From b96814b6bc97b32a590521ae8f401c40dac4cc7c Mon Sep 17 00:00:00 2001 From: Gary Scavone Date: Sat, 22 Nov 2008 03:32:52 +0000 Subject: Various updates to configure script, rtaudio files, and test files in preparation for upcoming release (gps). --- Makefile.in | 6 +- RtAudio.cpp | 302 +++++++--- RtAudio.h | 41 +- config.guess | 1371 ---------------------------------------------- config.sub | 1362 --------------------------------------------- config/config.guess | 1371 ++++++++++++++++++++++++++++++++++++++++++++++ config/config.sub | 1362 +++++++++++++++++++++++++++++++++++++++++++++ config/install.sh | 0 configure.ac | 119 ++-- doc/doxygen/tutorial.txt | 2 +- install.sh | 0 readme | 2 +- rtaudio-config.in | 13 +- tests/Makefile.in | 13 +- tests/testall.cpp | 30 +- 15 files changed, 3072 insertions(+), 2922 deletions(-) delete mode 100644 config.guess delete mode 100755 config.sub create mode 100644 config/config.guess create mode 100755 config/config.sub create mode 100755 config/install.sh delete mode 100755 install.sh diff --git a/Makefile.in b/Makefile.in index 23a444f..3230002 100644 --- a/Makefile.in +++ b/Makefile.in @@ -11,10 +11,8 @@ CC = @CXX@ AR = @AR@ RANLIB = @RANLIB@ -DEFS = @debug@ -DEFS += @audio_apis@ -CFLAGS = @CFLAGS@ -Iinclude -CFLAGS += @warn@ +DEFS = @CPPFLAGS@ +CFLAGS = @CXXFLAGS@ -Iinclude all : $(LIBRARY) diff --git a/RtAudio.cpp b/RtAudio.cpp index 8d15a34..05e3afa 100644 --- a/RtAudio.cpp +++ b/RtAudio.cpp @@ -359,6 +359,13 @@ double RtApi :: getStreamTime( void ) #endif } +unsigned int RtApi :: getStreamSampleRate( void ) +{ + verifyStream(); + + return stream_.sampleRate; +} + // *************************************************** // // @@ -393,6 +400,7 @@ double RtApi :: getStreamTime( void ) // implementation. struct CoreHandle { AudioDeviceID id[2]; // device ids + AudioDeviceIOProcID procId[2]; UInt32 iStream[2]; // device stream index (first for mono mode) bool xrun[2]; char *deviceBuffer; @@ -1105,7 +1113,8 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne handle->id[mode] = id; // Allocate necessary internal buffers. - unsigned long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); if ( stream_.userBuffer[mode] == NULL ) { errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory."; @@ -1172,7 +1181,12 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne // Only one callback procedure per device. stream_.mode = DUPLEX; else { +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] ); +#else + // deprecated in favor of AudioDeviceCreateIOProcID() result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo ); +#endif if ( result != noErr ) { errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ")."; errorText_ = errorStream_.str(); @@ -1225,13 +1239,23 @@ void RtApiCore :: closeStream( void ) if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { if ( stream_.state == STREAM_RUNNING ) AudioDeviceStop( handle->id[0], callbackHandler ); +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] ); +#else + // deprecated in favor of AudioDeviceDestroyIOProcID() AudioDeviceRemoveIOProc( handle->id[0], callbackHandler ); +#endif } if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { if ( stream_.state == STREAM_RUNNING ) AudioDeviceStop( handle->id[1], callbackHandler ); +#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) + AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] ); +#else + // deprecated in favor of AudioDeviceDestroyIOProcID() AudioDeviceRemoveIOProc( handle->id[1], callbackHandler ); +#endif } for ( int i=0; i<2; i++ ) { @@ -1470,7 +1494,8 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, } } - AudioDeviceID inputDevice = handle->id[1]; + AudioDeviceID inputDevice; + inputDevice = handle->id[1]; if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) { if ( stream_.doConvertBuffer[1] ) { @@ -1610,7 +1635,9 @@ RtApiJack :: ~RtApiJack() unsigned int RtApiJack :: getDeviceCount( void ) { // See if we can become a jack client. - jack_client_t *client = jack_client_new( "RtApiJackCount" ); + jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption; + jack_status_t *status = NULL; + jack_client_t *client = jack_client_open( "RtApiJackCount", options, status ); if ( client == 0 ) return 0; const char **ports; @@ -1619,7 +1646,7 @@ unsigned int RtApiJack :: getDeviceCount( void ) ports = jack_get_ports( client, NULL, NULL, 0 ); if ( ports ) { // Parse the port names up to the first colon (:). - unsigned int iColon = 0; + size_t iColon = 0; do { port = (char *) ports[ nChannels ]; iColon = port.find(":"); @@ -1643,7 +1670,9 @@ RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) RtAudio::DeviceInfo info; info.probed = false; - jack_client_t *client = jack_client_new( "RtApiJackInfo" ); + jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption + jack_status_t *status = NULL; + jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status ); if ( client == 0 ) { errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!"; error( RtError::WARNING ); @@ -1656,7 +1685,7 @@ RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) ports = jack_get_ports( client, NULL, NULL, 0 ); if ( ports ) { // Parse the port names up to the first colon (:). - unsigned int iColon = 0; + size_t iColon = 0; do { port = (char *) ports[ nPorts ]; iColon = port.find(":"); @@ -1771,10 +1800,12 @@ bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne // Look for jack server and try to become a client (only do once per stream). jack_client_t *client = 0; if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) { + jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption; + jack_status_t *status = NULL; if ( options && !options->streamName.empty() ) - client = jack_client_new( options->streamName.c_str() ); + client = jack_client_open( options->streamName.c_str(), jackoptions, status ); else - client = jack_client_new( "RtApiJack" ); + client = jack_client_open( "RtApiJack", jackoptions, status ); if ( client == 0 ) { errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!"; error( RtError::WARNING ); @@ -1792,7 +1823,7 @@ bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne ports = jack_get_ports( client, NULL, NULL, 0 ); if ( ports ) { // Parse the port names up to the first colon (:). - unsigned int iColon = 0; + size_t iColon = 0; do { port = (char *) ports[ nPorts ]; iColon = port.find(":"); @@ -2625,11 +2656,28 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; else if ( granularity == -1 ) { // Make sure bufferSize is a power of two. - double power = std::log10( (double) *bufferSize ) / log10( 2.0 ); - *bufferSize = (int) pow( 2.0, floor(power+0.5) ); - if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; - else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; - else *bufferSize = preferSize; + int log2_of_min_size = 0; + int log2_of_max_size = 0; + + for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) { + if ( minSize & ((long)1 << i) ) log2_of_min_size = i; + if ( maxSize & ((long)1 << i) ) log2_of_max_size = i; + } + + long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) ); + int min_delta_num = log2_of_min_size; + + for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) { + long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) ); + if (current_delta < min_delta) { + min_delta = current_delta; + min_delta_num = i; + } + } + + *bufferSize = ( (unsigned int)1 << min_delta_num ); + if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; + else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; } else if ( granularity != 0 ) { // Set to an even multiple of granularity, rounding up. @@ -3235,7 +3283,7 @@ static const char* getAsioErrorString( ASIOError result ) static inline DWORD dsPointerDifference( DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) { - if (laterPointer > earlierPointer) + if ( laterPointer > earlierPointer ) return laterPointer - earlierPointer; else return laterPointer - earlierPointer + bufferSize; @@ -4828,6 +4876,7 @@ struct AlsaHandle { snd_pcm_t *handles[2]; bool synchronized; bool xrun[2]; + pthread_cond_t runnable; AlsaHandle() :synchronized(false) { xrun[0] = false; xrun[1] = false; } @@ -5414,19 +5463,10 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne } // Set the buffer number, which in ALSA is referred to as the "period". - int dir; + int totalSize, dir; unsigned int periods = 0; if ( options ) periods = options->numberOfBuffers; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; - // Even though the hardware might allow 1 buffer, it won't work reliably. - if ( periods < 2 ) periods = 2; - result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } + totalSize = *bufferSize * periods; // Set the buffer (or period) size. snd_pcm_uframes_t periodSize = *bufferSize; @@ -5439,6 +5479,18 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne } *bufferSize = periodSize; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; + else periods = totalSize / *bufferSize; + // Even though the hardware might allow 1 buffer, it won't work reliably. + if ( periods < 2 ) periods = 2; + result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + // If attempting to setup a duplex stream, the bufferSize parameter // MUST be the same in both directions! if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { @@ -5468,9 +5520,19 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne snd_pcm_sw_params_alloca( &sw_params ); snd_pcm_sw_params_current( phandle, sw_params ); snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize ); - snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, 0x7fffffff ); + snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX ); snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 ); - snd_pcm_sw_params_set_silence_size( phandle, sw_params, INT_MAX ); + + // The following two settings were suggested by Theo Veenker + //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize ); + //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 ); + + // here are two options for a fix + //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX ); + snd_pcm_uframes_t val; + snd_pcm_sw_params_get_boundary( sw_params, &val ); + snd_pcm_sw_params_set_silence_size( phandle, sw_params, val ); + result = snd_pcm_sw_params( phandle, sw_params ); if ( result < 0 ) { snd_pcm_close( phandle ); @@ -5504,6 +5566,12 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory."; goto error; } + + if ( pthread_cond_init( &apiInfo->runnable, NULL ) ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + stream_.apiHandle = (void *) apiInfo; apiInfo->handles[0] = 0; apiInfo->handles[1] = 0; @@ -5572,13 +5640,28 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne stream_.callbackInfo.object = (void *) this; // Set the thread attributes for joinable and realtime scheduling - // priority. The higher priority will only take affect if the - // program is run as root or suid. + // priority (optional). The higher priority will only take affect + // if the program is run as root or suid. Note, under Linux + // processes with CAP_SYS_NICE privilege, a user can change + // scheduling policy and priority (thus need not be root). See + // POSIX "capabilities". pthread_attr_t attr; pthread_attr_init( &attr ); pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) - pthread_attr_setschedpolicy( &attr, SCHED_RR ); + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { + struct sched_param param; + int priority = options->priority; + int min = sched_get_priority_min( SCHED_RR ); + int max = sched_get_priority_max( SCHED_RR ); + if ( priority < min ) priority = min; + else if ( priority > max ) priority = max; + param.sched_priority = priority; + pthread_attr_setschedparam( &attr, ¶m ); + pthread_attr_setschedpolicy( &attr, SCHED_RR ); + } + else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); #else pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); #endif @@ -5597,6 +5680,7 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne error: if ( apiInfo ) { + pthread_cond_destroy( &apiInfo->runnable ); if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); delete apiInfo; @@ -5626,10 +5710,14 @@ void RtApiAlsa :: closeStream() return; } + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; stream_.callbackInfo.isRunning = false; + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) + pthread_cond_signal( &apiInfo->runnable ); + MUTEX_UNLOCK( &stream_.mutex ); pthread_join( stream_.callbackInfo.thread, NULL ); - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; if ( stream_.state == STREAM_RUNNING ) { stream_.state = STREAM_STOPPED; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) @@ -5639,6 +5727,7 @@ void RtApiAlsa :: closeStream() } if ( apiInfo ) { + pthread_cond_destroy( &apiInfo->runnable ); if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); delete apiInfo; @@ -5707,6 +5796,8 @@ void RtApiAlsa :: startStream() unlock: MUTEX_UNLOCK( &stream_.mutex ); + pthread_cond_signal( &apiInfo->runnable ); + if ( result >= 0 ) return; error( RtError::SYSTEM_ERROR ); } @@ -5720,8 +5811,7 @@ void RtApiAlsa :: stopStream() return; } - // Change the state before the lock to improve shutdown response - // when using a callback. + // Change the state before the lock to improve shutdown response. stream_.state = STREAM_STOPPED; MUTEX_LOCK( &stream_.mutex ); @@ -5765,8 +5855,7 @@ void RtApiAlsa :: abortStream() return; } - // Change the state before the lock to improve shutdown response - // when using a callback. + // Change the state before the lock to improve shutdown response. stream_.state = STREAM_STOPPED; MUTEX_LOCK( &stream_.mutex ); @@ -5794,16 +5883,21 @@ void RtApiAlsa :: abortStream() unlock: MUTEX_UNLOCK( &stream_.mutex ); - stream_.state = STREAM_STOPPED; if ( result >= 0 ) return; error( RtError::SYSTEM_ERROR ); } void RtApiAlsa :: callbackEvent() { + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; if ( stream_.state == STREAM_STOPPED ) { - if ( stream_.callbackInfo.isRunning ) usleep( 50000 ); // sleep 50 milliseconds - return; + MUTEX_LOCK( &stream_.mutex ); + pthread_cond_wait( &apiInfo->runnable, &stream_.mutex ); + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); } if ( stream_.state == STREAM_CLOSED ) { @@ -5813,7 +5907,6 @@ void RtApiAlsa :: callbackEvent() } int doStopStream = 0; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; double streamTime = getStreamTime(); RtAudioStreamStatus status = 0; @@ -5828,6 +5921,11 @@ void RtApiAlsa :: callbackEvent() doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); + if ( doStopStream == 2 ) { + abortStream(); + return; + } + MUTEX_LOCK( &stream_.mutex ); // The state might change while waiting on a mutex. @@ -5867,7 +5965,7 @@ void RtApiAlsa :: callbackEvent() } if ( result < (int) stream_.bufferSize ) { - // Either an error or underrun occured. + // Either an error or overrun occured. if ( result == -EPIPE ) { snd_pcm_state_t state = snd_pcm_state( handle[1] ); if ( state == SND_PCM_STATE_XRUN ) { @@ -5888,7 +5986,7 @@ void RtApiAlsa :: callbackEvent() errorText_ = errorStream_.str(); } error( RtError::WARNING ); - goto unlock; + goto tryOutput; } // Do byte swapping if necessary. @@ -5904,6 +6002,8 @@ void RtApiAlsa :: callbackEvent() if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; } + tryOutput: + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { // Setup parameters and do buffer conversion if necessary. @@ -5969,7 +6069,6 @@ void RtApiAlsa :: callbackEvent() RtApi::tickStreamTime(); if ( doStopStream == 1 ) this->stopStream(); - else if ( doStopStream == 2 ) this->abortStream(); } extern "C" void *alsaCallbackHandler( void *ptr ) @@ -5978,15 +6077,6 @@ extern "C" void *alsaCallbackHandler( void *ptr ) RtApiAlsa *object = (RtApiAlsa *) info->object; bool *isRunning = &info->isRunning; -#ifdef SCHED_RR - // Set a higher scheduler priority (P.J. Leonard) - struct sched_param param; - int min = sched_get_priority_min( SCHED_RR ); - int max = sched_get_priority_max( SCHED_RR ); - param.sched_priority = min + ( max - min ) / 2; // Is this the best number? - sched_setscheduler( 0, SCHED_RR, ¶m ); -#endif - while ( *isRunning == true ) { pthread_testcancel(); object->callbackEvent(); @@ -6017,6 +6107,7 @@ struct OssHandle { int id[2]; // device ids bool xrun[2]; bool triggered; + pthread_cond_t runnable; OssHandle() :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } @@ -6474,6 +6565,11 @@ bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned goto error; } + if ( pthread_cond_init( &handle->runnable, NULL ) ) { + errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + stream_.apiHandle = (void *) handle; } else { @@ -6537,7 +6633,19 @@ bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned pthread_attr_init( &attr ); pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) - pthread_attr_setschedpolicy( &attr, SCHED_RR ); + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { + struct sched_param param; + int priority = options->priority; + int min = sched_get_priority_min( SCHED_RR ); + int max = sched_get_priority_max( SCHED_RR ); + if ( priority < min ) priority = min; + else if ( priority > max ) priority = max; + param.sched_priority = priority; + pthread_attr_setschedparam( &attr, ¶m ); + pthread_attr_setschedpolicy( &attr, SCHED_RR ); + } + else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); #else pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); #endif @@ -6556,6 +6664,7 @@ bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned error: if ( handle ) { + pthread_cond_destroy( &handle->runnable ); if ( handle->id[0] ) close( handle->id[0] ); if ( handle->id[1] ) close( handle->id[1] ); delete handle; @@ -6585,10 +6694,14 @@ void RtApiOss :: closeStream() return; } + OssHandle *handle = (OssHandle *) stream_.apiHandle; stream_.callbackInfo.isRunning = false; + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) + pthread_cond_signal( &handle->runnable ); + MUTEX_UNLOCK( &stream_.mutex ); pthread_join( stream_.callbackInfo.thread, NULL ); - OssHandle *handle = (OssHandle *) stream_.apiHandle; if ( stream_.state == STREAM_RUNNING ) { if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); @@ -6598,6 +6711,7 @@ void RtApiOss :: closeStream() } if ( handle ) { + pthread_cond_destroy( &handle->runnable ); if ( handle->id[0] ) close( handle->id[0] ); if ( handle->id[1] ) close( handle->id[1] ); delete handle; @@ -6637,6 +6751,9 @@ void RtApiOss :: startStream() // when fed samples. MUTEX_UNLOCK( &stream_.mutex ); + + OssHandle *handle = (OssHandle *) stream_.apiHandle; + pthread_cond_signal( &handle->runnable ); } void RtApiOss :: stopStream() @@ -6753,9 +6870,15 @@ void RtApiOss :: abortStream() void RtApiOss :: callbackEvent() { + OssHandle *handle = (OssHandle *) stream_.apiHandle; if ( stream_.state == STREAM_STOPPED ) { - if ( stream_.callbackInfo.isRunning ) usleep( 50000 ); // sleep 50 milliseconds - return; + MUTEX_LOCK( &stream_.mutex ); + pthread_cond_wait( &handle->runnable, &stream_.mutex ); + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); } if ( stream_.state == STREAM_CLOSED ) { @@ -6769,7 +6892,6 @@ void RtApiOss :: callbackEvent() RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; double streamTime = getStreamTime(); RtAudioStreamStatus status = 0; - OssHandle *handle = (OssHandle *) stream_.apiHandle; if ( stream_.mode != INPUT && handle->xrun[0] == true ) { status |= RTAUDIO_OUTPUT_UNDERFLOW; handle->xrun[0] = false; @@ -6780,6 +6902,10 @@ void RtApiOss :: callbackEvent() } doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); + if ( doStopStream == 2 ) { + this->abortStream(); + return; + } MUTEX_LOCK( &stream_.mutex ); @@ -6828,7 +6954,7 @@ void RtApiOss :: callbackEvent() handle->xrun[0] = true; errorText_ = "RtApiOss::callbackEvent: audio write error."; error( RtError::WARNING ); - goto unlock; + // Continue on to input section. } } @@ -6872,7 +6998,6 @@ void RtApiOss :: callbackEvent() RtApi::tickStreamTime(); if ( doStopStream == 1 ) this->stopStream(); - else if ( doStopStream == 2 ) this->abortStream(); } extern "C" void *ossCallbackHandler( void *ptr ) @@ -6881,13 +7006,6 @@ extern "C" void *ossCallbackHandler( void *ptr ) RtApiOss *object = (RtApiOss *) info->object; bool *isRunning = &info->isRunning; -#ifdef SCHED_RR - // Set a higher scheduler priority (P.J. Leonard) - struct sched_param param; - param.sched_priority = 39; // Is this the best number? - sched_setscheduler( 0, SCHED_RR, ¶m ); -#endif - while ( *isRunning == true ) { pthread_testcancel(); object->callbackEvent(); @@ -7077,10 +7195,11 @@ void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info if (info.inFormat == RTAUDIO_SINT8) { signed char *in = (signed char *)inBuffer; - scale = 1.0 / 128.0; + scale = 1.0 / 127.5; for (unsigned int i=0; i>8) | (x<<8); } +//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); } +//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); } + void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ) { register char val; @@ -7516,8 +7646,8 @@ void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat *(ptr) = *(ptr+1); *(ptr+1) = val; - // Increment 4 bytes. - ptr += 4; + // Increment 3 more bytes. + ptr += 3; } } else if ( format == RTAUDIO_FLOAT64 ) { @@ -7545,8 +7675,8 @@ void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat *(ptr) = *(ptr+1); *(ptr+1) = val; - // Increment 8 bytes. - ptr += 8; + // Increment 5 more bytes. + ptr += 5; } } } diff --git a/RtAudio.h b/RtAudio.h index 8696262..370ba26 100644 --- a/RtAudio.h +++ b/RtAudio.h @@ -108,11 +108,15 @@ static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/mi If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to open the input and/or output stream device(s) for exclusive use. Note that this is not possible with all supported audio APIs. + + If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt + to select realtime scheduling (round-robin) for the callback thread. */ typedef unsigned int RtAudioStreamFlags; static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1; // Use non-interleaved buffers (default = interleaved). static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2; // Attempt to set stream parameters for lowest possible latency. static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4; // Attempt grab device and prevent use by others. +static const RtAudioStreamFlags RTAUDIO_SCHEDULE_REALTIME = 0x8; // Try to select realtime scheduling for callback thread. /*! \typedef typedef unsigned long RtAudioStreamStatus; \brief RtAudio stream status (over- or underflow) flags. @@ -240,9 +244,10 @@ class RtAudio The following flags can be OR'ed together to allow a client to make changes to the default stream behavior: - - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved). - - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency. - - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use. + - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved). + - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency. + - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use. + - \e RTAUDIO_SCHEDULE_REALTIME: Attempt to select realtime scheduling for callback thread. By default, RtAudio streams pass and receive audio data from the client in an interleaved format. By passing the @@ -268,6 +273,11 @@ class RtAudio open the input and/or output stream device(s) for exclusive use. Note that this is not possible with all supported audio APIs. + If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt + to select realtime scheduling (round-robin) for the callback thread. + The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME + flag is set. It defines the thread's realtime priority. + The \c numberOfBuffers parameter can be used to control stream latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs only. A value of two is usually the smallest allowed. Larger @@ -285,10 +295,11 @@ class RtAudio RtAudioStreamFlags flags; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE). */ unsigned int numberOfBuffers; /*!< Number of stream buffers. */ std::string streamName; /*!< A stream name (currently used only in Jack). */ + int priority; /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */ // Default constructor. StreamOptions() - : flags(0), numberOfBuffers(0) {} + : flags(0), numberOfBuffers(0), priority(0) {} }; //! A static function to determine the available compiled audio APIs. @@ -440,10 +451,10 @@ class RtAudio void abortStream( void ); //! Returns true if a stream is open and false if not. - bool isStreamOpen( void ) throw(); + bool isStreamOpen( void ) const throw(); //! Returns true if the stream is running and false if it is stopped or not open. - bool isStreamRunning( void ) throw(); + bool isStreamRunning( void ) const throw(); //! Returns the number of elapsed seconds since the stream was started. /*! @@ -462,6 +473,14 @@ class RtAudio */ long getStreamLatency( void ); + //! Returns actual sample rate in use by the stream. + /*! + On some systems, the sample rate used may be slightly different + than that specified in the stream parameters. If a stream is not + open, an RtError (type = INVALID_USE) will be thrown. + */ + unsigned int getStreamSampleRate( void ); + //! Specify whether warning messages should be printed to stderr. void showWarnings( bool value = true ) throw(); @@ -551,9 +570,10 @@ public: virtual void stopStream( void ) = 0; virtual void abortStream( void ) = 0; long getStreamLatency( void ); + unsigned int getStreamSampleRate( void ); virtual double getStreamTime( void ); - bool isStreamOpen( void ) { return stream_.state != STREAM_CLOSED; }; - bool isStreamRunning( void ) { return stream_.state == STREAM_RUNNING; }; + bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; }; + bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; }; void showWarnings( bool value ) { showWarnings_ = value; }; @@ -688,9 +708,10 @@ inline void RtAudio :: closeStream( void ) throw() { return rtapi_->closeStream( inline void RtAudio :: startStream( void ) { return rtapi_->startStream(); } inline void RtAudio :: stopStream( void ) { return rtapi_->stopStream(); } inline void RtAudio :: abortStream( void ) { return rtapi_->abortStream(); } -inline bool RtAudio :: isStreamOpen( void ) throw() { return rtapi_->isStreamOpen(); } -inline bool RtAudio :: isStreamRunning( void ) throw() { return rtapi_->isStreamRunning(); } +inline bool RtAudio :: isStreamOpen( void ) const throw() { return rtapi_->isStreamOpen(); } +inline bool RtAudio :: isStreamRunning( void ) const throw() { return rtapi_->isStreamRunning(); } inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); } +inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); }; inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); } inline void RtAudio :: showWarnings( bool value ) throw() { rtapi_->showWarnings( value ); } diff --git a/config.guess b/config.guess deleted file mode 100644 index 313be34..0000000 --- a/config.guess +++ /dev/null @@ -1,1371 +0,0 @@ -#! /bin/sh -# Attempt to guess a canonical system name. -# Copyright (C) 1992, 1993, 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001 -# Free Software Foundation, Inc. - -timestamp='2004-02-26' - -# This file is free software; you can redistribute it and/or modify it -# under the terms of the GNU General Public License as published by -# the Free Software Foundation; either version 2 of the License, or -# (at your option) any later version. -# -# This program is distributed in the hope that it will be useful, but -# WITHOUT ANY WARRANTY; without even the implied warranty of -# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU -# General Public License for more details. -# -# You should have received a copy of the GNU General Public License -# along with this program; if not, write to the Free Software -# Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. -# -# As a special exception to the GNU General Public License, if you -# distribute this file as part of a program that contains a -# configuration script generated by Autoconf, you may include it under -# the same distribution terms that you use for the rest of that program. - -# Written by Per Bothner . -# Please send patches to . -# -# This script attempts to guess a canonical system name similar to -# config.sub. If it succeeds, it prints the system name on stdout, and -# exits with 0. Otherwise, it exits with 1. -# -# The plan is that this can be called by configure scripts if you -# don't specify an explicit build system type. - -me=`echo "$0" | sed -e 's,.*/,,'` - -usage="\ -Usage: $0 [OPTION] - -Output the configuration name of the system \`$me' is run on. - -Operation modes: - -h, --help print this help, then exit - -t, --time-stamp print date of last modification, then exit - -v, --version print version number, then exit - -Report bugs and patches to ." - -version="\ -GNU config.guess ($timestamp) - -Originally written by Per Bothner. -Copyright (C) 1992, 93, 94, 95, 96, 97, 98, 99, 2000 -Free Software Foundation, Inc. - -This is free software; see the source for copying conditions. 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The machine name - # can be virtually everything (everything which is not - # "atarist" or "atariste" at least should have a processor - # > m68000). The system name ranges from "MiNT" over "FreeMiNT" - # to the lowercase version "mint" (or "freemint"). Finally - # the system name "TOS" denotes a system which is actually not - # MiNT. But MiNT is downward compatible to TOS, so this should - # be no problem. - atarist[e]:*MiNT:*:* | atarist[e]:*mint:*:* | atarist[e]:*TOS:*:*) - echo m68k-atari-mint${UNAME_RELEASE} - exit 0 ;; - atari*:*MiNT:*:* | atari*:*mint:*:* | atarist[e]:*TOS:*:*) - echo m68k-atari-mint${UNAME_RELEASE} - exit 0 ;; - *falcon*:*MiNT:*:* | *falcon*:*mint:*:* | *falcon*:*TOS:*:*) - echo m68k-atari-mint${UNAME_RELEASE} - exit 0 ;; - milan*:*MiNT:*:* | milan*:*mint:*:* | *milan*:*TOS:*:*) - echo m68k-milan-mint${UNAME_RELEASE} - exit 0 ;; - hades*:*MiNT:*:* | hades*:*mint:*:* | *hades*:*TOS:*:*) - echo m68k-hades-mint${UNAME_RELEASE} - exit 0 ;; - *:*MiNT:*:* | *:*mint:*:* | *:*TOS:*:*) - echo m68k-unknown-mint${UNAME_RELEASE} - exit 0 ;; - sun3*:OpenBSD:*:*) - echo m68k-unknown-openbsd${UNAME_RELEASE} - exit 0 ;; - mac68k:OpenBSD:*:*) - echo m68k-unknown-openbsd${UNAME_RELEASE} - exit 0 ;; - mvme68k:OpenBSD:*:*) - echo m68k-unknown-openbsd${UNAME_RELEASE} - exit 0 ;; - mvme88k:OpenBSD:*:*) - echo m88k-unknown-openbsd${UNAME_RELEASE} - exit 0 ;; - powerpc:machten:*:*) - echo powerpc-apple-machten${UNAME_RELEASE} - exit 0 ;; - RISC*:Mach:*:*) - echo mips-dec-mach_bsd4.3 - exit 0 ;; - RISC*:ULTRIX:*:*) - echo mips-dec-ultrix${UNAME_RELEASE} - exit 0 ;; - VAX*:ULTRIX*:*:*) - echo vax-dec-ultrix${UNAME_RELEASE} - exit 0 ;; - 2020:CLIX:*:* | 2430:CLIX:*:*) - echo clipper-intergraph-clix${UNAME_RELEASE} - exit 0 ;; - mips:*:*:UMIPS | mips:*:*:RISCos) - sed 's/^ //' << EOF >$dummy.c -#ifdef __cplusplus -#include /* for printf() prototype */ - int main (int argc, char *argv[]) { -#else - int main (argc, argv) int argc; char *argv[]; { -#endif - #if defined (host_mips) && defined (MIPSEB) - #if defined (SYSTYPE_SYSV) - printf ("mips-mips-riscos%ssysv\n", argv[1]); exit (0); - #endif - #if defined (SYSTYPE_SVR4) - printf ("mips-mips-riscos%ssvr4\n", argv[1]); exit (0); - #endif - #if defined (SYSTYPE_BSD43) || defined(SYSTYPE_BSD) - printf ("mips-mips-riscos%sbsd\n", argv[1]); exit (0); - #endif - #endif - exit (-1); - } -EOF - $CC_FOR_BUILD $dummy.c -o $dummy \ - && ./$dummy `echo "${UNAME_RELEASE}" | sed -n 's/\([0-9]*\).*/\1/p'` \ - && rm -f $dummy.c $dummy && exit 0 - rm -f $dummy.c $dummy - echo mips-mips-riscos${UNAME_RELEASE} - exit 0 ;; - Motorola:PowerMAX_OS:*:*) - echo powerpc-motorola-powermax - exit 0 ;; - Night_Hawk:Power_UNIX:*:*) - echo powerpc-harris-powerunix - exit 0 ;; - m88k:CX/UX:7*:*) - echo m88k-harris-cxux7 - exit 0 ;; - m88k:*:4*:R4*) - echo m88k-motorola-sysv4 - exit 0 ;; - m88k:*:3*:R3*) - echo m88k-motorola-sysv3 - exit 0 ;; - AViiON:dgux:*:*) - # DG/UX returns AViiON for all architectures - UNAME_PROCESSOR=`/usr/bin/uname -p` - if [ $UNAME_PROCESSOR = mc88100 ] || [ $UNAME_PROCESSOR = mc88110 ] - then - if [ ${TARGET_BINARY_INTERFACE}x = m88kdguxelfx ] || \ - [ ${TARGET_BINARY_INTERFACE}x = x ] - then - echo m88k-dg-dgux${UNAME_RELEASE} - else - echo m88k-dg-dguxbcs${UNAME_RELEASE} - fi - else - echo i586-dg-dgux${UNAME_RELEASE} - fi - exit 0 ;; - M88*:DolphinOS:*:*) # DolphinOS (SVR3) - echo m88k-dolphin-sysv3 - exit 0 ;; - M88*:*:R3*:*) - # Delta 88k system running SVR3 - echo m88k-motorola-sysv3 - exit 0 ;; - XD88*:*:*:*) # Tektronix XD88 system running UTekV (SVR3) - echo m88k-tektronix-sysv3 - exit 0 ;; - Tek43[0-9][0-9]:UTek:*:*) # Tektronix 4300 system running UTek (BSD) - echo m68k-tektronix-bsd - exit 0 ;; - *:IRIX*:*:*) - echo mips-sgi-irix`echo ${UNAME_RELEASE}|sed -e 's/-/_/g'` - exit 0 ;; - ????????:AIX?:[12].1:2) # AIX 2.2.1 or AIX 2.1.1 is RT/PC AIX. - echo romp-ibm-aix # uname -m gives an 8 hex-code CPU id - exit 0 ;; # Note that: echo "'`uname -s`'" gives 'AIX ' - i*86:AIX:*:*) - echo i386-ibm-aix - exit 0 ;; - ia64:AIX:*:*) - if [ -x /usr/bin/oslevel ] ; then - IBM_REV=`/usr/bin/oslevel` - else - IBM_REV=${UNAME_VERSION}.${UNAME_RELEASE} - fi - echo ${UNAME_MACHINE}-ibm-aix${IBM_REV} - exit 0 ;; - *:AIX:2:3) - if grep bos325 /usr/include/stdio.h >/dev/null 2>&1; then - sed 's/^ //' << EOF >$dummy.c - #include - - main() - { - if (!__power_pc()) - exit(1); - puts("powerpc-ibm-aix3.2.5"); - exit(0); - } -EOF - $CC_FOR_BUILD $dummy.c -o $dummy && ./$dummy && rm -f $dummy.c $dummy && exit 0 - rm -f $dummy.c $dummy - echo rs6000-ibm-aix3.2.5 - elif grep bos324 /usr/include/stdio.h >/dev/null 2>&1; then - echo rs6000-ibm-aix3.2.4 - else - echo rs6000-ibm-aix3.2 - fi - exit 0 ;; - *:AIX:*:[45]) - IBM_CPU_ID=`/usr/sbin/lsdev -C -c processor -S available | head -1 | awk '{ print $1 }'` - if /usr/sbin/lsattr -El ${IBM_CPU_ID} | grep ' POWER' >/dev/null 2>&1; then - IBM_ARCH=rs6000 - else - IBM_ARCH=powerpc - fi - if [ -x /usr/bin/oslevel ] ; then - IBM_REV=`/usr/bin/oslevel` - else - IBM_REV=${UNAME_VERSION}.${UNAME_RELEASE} - fi - echo ${IBM_ARCH}-ibm-aix${IBM_REV} - exit 0 ;; - *:AIX:*:*) - echo rs6000-ibm-aix - exit 0 ;; - ibmrt:4.4BSD:*|romp-ibm:BSD:*) - echo romp-ibm-bsd4.4 - exit 0 ;; - ibmrt:*BSD:*|romp-ibm:BSD:*) # covers RT/PC BSD and - echo romp-ibm-bsd${UNAME_RELEASE} # 4.3 with uname added to - exit 0 ;; # report: romp-ibm BSD 4.3 - *:BOSX:*:*) - echo rs6000-bull-bosx - exit 0 ;; - DPX/2?00:B.O.S.:*:*) - echo m68k-bull-sysv3 - exit 0 ;; - 9000/[34]??:4.3bsd:1.*:*) - echo m68k-hp-bsd - exit 0 ;; - hp300:4.4BSD:*:* | 9000/[34]??:4.3bsd:2.*:*) - echo m68k-hp-bsd4.4 - exit 0 ;; - 9000/[34678]??:HP-UX:*:*) - HPUX_REV=`echo ${UNAME_RELEASE}|sed -e 's/[^.]*.[0B]*//'` - case "${UNAME_MACHINE}" in - 9000/31? ) HP_ARCH=m68000 ;; - 9000/[34]?? ) HP_ARCH=m68k ;; - 9000/[678][0-9][0-9]) - case "${HPUX_REV}" in - 11.[0-9][0-9]) - if [ -x /usr/bin/getconf ]; then - sc_cpu_version=`/usr/bin/getconf SC_CPU_VERSION 2>/dev/null` - sc_kernel_bits=`/usr/bin/getconf SC_KERNEL_BITS 2>/dev/null` - case "${sc_cpu_version}" in - 523) HP_ARCH="hppa1.0" ;; # CPU_PA_RISC1_0 - 528) HP_ARCH="hppa1.1" ;; # CPU_PA_RISC1_1 - 532) # CPU_PA_RISC2_0 - case "${sc_kernel_bits}" in - 32) HP_ARCH="hppa2.0n" ;; - 64) HP_ARCH="hppa2.0w" ;; - esac ;; - esac - fi ;; - esac - if [ "${HP_ARCH}" = "" ]; then - sed 's/^ //' << EOF >$dummy.c - - #define _HPUX_SOURCE - #include - #include - - int main () - { - #if defined(_SC_KERNEL_BITS) - long bits = sysconf(_SC_KERNEL_BITS); - #endif - long cpu = sysconf (_SC_CPU_VERSION); - - switch (cpu) - { - case CPU_PA_RISC1_0: puts ("hppa1.0"); break; - case CPU_PA_RISC1_1: puts ("hppa1.1"); break; - case CPU_PA_RISC2_0: - #if defined(_SC_KERNEL_BITS) - switch (bits) - { - case 64: puts ("hppa2.0w"); break; - case 32: puts ("hppa2.0n"); break; - default: puts ("hppa2.0"); break; - } break; - #else /* !defined(_SC_KERNEL_BITS) */ - puts ("hppa2.0"); break; - #endif - default: puts ("hppa1.0"); break; - } - exit (0); - } -EOF - (CCOPTS= $CC_FOR_BUILD $dummy.c -o $dummy 2>/dev/null ) && HP_ARCH=`./$dummy` - if test -z "$HP_ARCH"; then HP_ARCH=hppa; fi - rm -f $dummy.c $dummy - fi ;; - esac - echo ${HP_ARCH}-hp-hpux${HPUX_REV} - exit 0 ;; - ia64:HP-UX:*:*) - HPUX_REV=`echo ${UNAME_RELEASE}|sed -e 's/[^.]*.[0B]*//'` - echo ia64-hp-hpux${HPUX_REV} - exit 0 ;; - 3050*:HI-UX:*:*) - sed 's/^ //' << EOF >$dummy.c - #include - int - main () - { - long cpu = sysconf (_SC_CPU_VERSION); - /* The order matters, because CPU_IS_HP_MC68K erroneously returns - true for CPU_PA_RISC1_0. CPU_IS_PA_RISC returns correct - results, however. */ - if (CPU_IS_PA_RISC (cpu)) - { - switch (cpu) - { - case CPU_PA_RISC1_0: puts ("hppa1.0-hitachi-hiuxwe2"); break; - case CPU_PA_RISC1_1: puts ("hppa1.1-hitachi-hiuxwe2"); break; - case CPU_PA_RISC2_0: puts ("hppa2.0-hitachi-hiuxwe2"); break; - default: puts ("hppa-hitachi-hiuxwe2"); break; - } - } - else if (CPU_IS_HP_MC68K (cpu)) - puts ("m68k-hitachi-hiuxwe2"); - else puts ("unknown-hitachi-hiuxwe2"); - exit (0); - } -EOF - $CC_FOR_BUILD $dummy.c -o $dummy && ./$dummy && rm -f $dummy.c $dummy && exit 0 - rm -f $dummy.c $dummy - echo unknown-hitachi-hiuxwe2 - exit 0 ;; - 9000/7??:4.3bsd:*:* | 9000/8?[79]:4.3bsd:*:* ) - echo hppa1.1-hp-bsd - exit 0 ;; - 9000/8??:4.3bsd:*:*) - echo hppa1.0-hp-bsd - exit 0 ;; - *9??*:MPE/iX:*:*) - echo hppa1.0-hp-mpeix - exit 0 ;; - hp7??:OSF1:*:* | hp8?[79]:OSF1:*:* ) - echo hppa1.1-hp-osf - exit 0 ;; - hp8??:OSF1:*:*) - echo hppa1.0-hp-osf - exit 0 ;; - i*86:OSF1:*:*) - if [ -x /usr/sbin/sysversion ] ; then - echo ${UNAME_MACHINE}-unknown-osf1mk - else - echo ${UNAME_MACHINE}-unknown-osf1 - fi - exit 0 ;; - parisc*:Lites*:*:*) - echo hppa1.1-hp-lites - exit 0 ;; - hppa*:OpenBSD:*:*) - echo hppa-unknown-openbsd - exit 0 ;; - C1*:ConvexOS:*:* | convex:ConvexOS:C1*:*) - echo c1-convex-bsd - exit 0 ;; - C2*:ConvexOS:*:* | convex:ConvexOS:C2*:*) - if getsysinfo -f scalar_acc - then echo c32-convex-bsd - else echo c2-convex-bsd - fi - exit 0 ;; - C34*:ConvexOS:*:* | convex:ConvexOS:C34*:*) - echo c34-convex-bsd - exit 0 ;; - C38*:ConvexOS:*:* | convex:ConvexOS:C38*:*) - echo c38-convex-bsd - exit 0 ;; - C4*:ConvexOS:*:* | convex:ConvexOS:C4*:*) - echo c4-convex-bsd - exit 0 ;; - CRAY*X-MP:*:*:*) - echo xmp-cray-unicos - exit 0 ;; - CRAY*Y-MP:*:*:*) - echo ymp-cray-unicos${UNAME_RELEASE} - exit 0 ;; - CRAY*[A-Z]90:*:*:*) - echo ${UNAME_MACHINE}-cray-unicos${UNAME_RELEASE} \ - | sed -e 's/CRAY.*\([A-Z]90\)/\1/' \ - -e y/ABCDEFGHIJKLMNOPQRSTUVWXYZ/abcdefghijklmnopqrstuvwxyz/ - exit 0 ;; - CRAY*TS:*:*:*) - echo t90-cray-unicos${UNAME_RELEASE} | sed -e 's/\.[^.]*$/.X/' - exit 0 ;; - CRAY*T3D:*:*:*) - echo alpha-cray-unicosmk${UNAME_RELEASE} | sed -e 's/\.[^.]*$/.X/' - exit 0 ;; - CRAY*T3E:*:*:*) - echo alphaev5-cray-unicosmk${UNAME_RELEASE} | sed -e 's/\.[^.]*$/.X/' - exit 0 ;; - CRAY*SV1:*:*:*) - echo sv1-cray-unicos${UNAME_RELEASE} | sed -e 's/\.[^.]*$/.X/' - exit 0 ;; - CRAY-2:*:*:*) - echo cray2-cray-unicos - exit 0 ;; - F30[01]:UNIX_System_V:*:* | F700:UNIX_System_V:*:*) - FUJITSU_PROC=`uname -m | tr 'ABCDEFGHIJKLMNOPQRSTUVWXYZ' 'abcdefghijklmnopqrstuvwxyz'` - FUJITSU_SYS=`uname -p | tr 'ABCDEFGHIJKLMNOPQRSTUVWXYZ' 'abcdefghijklmnopqrstuvwxyz' | sed -e 's/\///'` - FUJITSU_REL=`echo ${UNAME_RELEASE} | sed -e 's/ /_/'` - echo "${FUJITSU_PROC}-fujitsu-${FUJITSU_SYS}${FUJITSU_REL}" - exit 0 ;; - hp300:OpenBSD:*:*) - echo m68k-unknown-openbsd${UNAME_RELEASE} - exit 0 ;; - i*86:BSD/386:*:* | i*86:BSD/OS:*:* | *:Ascend\ Embedded/OS:*:*) - echo ${UNAME_MACHINE}-pc-bsdi${UNAME_RELEASE} - exit 0 ;; - sparc*:BSD/OS:*:*) - echo sparc-unknown-bsdi${UNAME_RELEASE} - exit 0 ;; - *:BSD/OS:*:*) - echo ${UNAME_MACHINE}-unknown-bsdi${UNAME_RELEASE} - exit 0 ;; - *:FreeBSD:*:*) - echo ${UNAME_MACHINE}-unknown-freebsd`echo ${UNAME_RELEASE}|sed -e 's/[-(].*//'` - exit 0 ;; - *:OpenBSD:*:*) - echo ${UNAME_MACHINE}-unknown-openbsd`echo ${UNAME_RELEASE}|sed -e 's/[-_].*/\./'` - exit 0 ;; - i*:CYGWIN*:*) - echo ${UNAME_MACHINE}-pc-cygwin - exit 0 ;; - i*:MINGW*:*) - echo ${UNAME_MACHINE}-pc-mingw32 - exit 0 ;; - i*:PW*:*) - echo ${UNAME_MACHINE}-pc-pw32 - exit 0 ;; - i*:Windows_NT*:* | Pentium*:Windows_NT*:*) - # How do we know it's Interix rather than the generic POSIX subsystem? - # It also conflicts with pre-2.0 versions of AT&T UWIN. Should we - # UNAME_MACHINE based on the output of uname instead of i386? - echo i386-pc-interix - exit 0 ;; - i*:UWIN*:*) - echo ${UNAME_MACHINE}-pc-uwin - exit 0 ;; - p*:CYGWIN*:*) - echo powerpcle-unknown-cygwin - exit 0 ;; - prep*:SunOS:5.*:*) - echo powerpcle-unknown-solaris2`echo ${UNAME_RELEASE}|sed -e 's/[^.]*//'` - exit 0 ;; - *:GNU:*:*) - echo `echo ${UNAME_MACHINE}|sed -e 's,[-/].*$,,'`-unknown-gnu`echo ${UNAME_RELEASE}|sed -e 's,/.*$,,'` - exit 0 ;; - i*86:Minix:*:*) - echo ${UNAME_MACHINE}-pc-minix - exit 0 ;; - arm*:Linux:*:*) - echo ${UNAME_MACHINE}-unknown-linux-gnu - exit 0 ;; - ia64:Linux:*:*) - echo ${UNAME_MACHINE}-unknown-linux - exit 0 ;; - m68*:Linux:*:*) - echo ${UNAME_MACHINE}-unknown-linux-gnu - exit 0 ;; - mips:Linux:*:*) - cat >$dummy.c < /* for printf() prototype */ -int main (int argc, char *argv[]) { -#else -int main (argc, argv) int argc; char *argv[]; { -#endif -#ifdef __MIPSEB__ - printf ("%s-unknown-linux-gnu\n", argv[1]); -#endif -#ifdef __MIPSEL__ - printf ("%sel-unknown-linux-gnu\n", argv[1]); -#endif - return 0; -} -EOF - $CC_FOR_BUILD $dummy.c -o $dummy 2>/dev/null && ./$dummy "${UNAME_MACHINE}" && rm -f $dummy.c $dummy && exit 0 - rm -f $dummy.c $dummy - ;; - ppc:Linux:*:*) - # Determine Lib Version - cat >$dummy.c < -#if defined(__GLIBC__) -extern char __libc_version[]; -extern char __libc_release[]; -#endif -main(argc, argv) - int argc; - char *argv[]; -{ -#if defined(__GLIBC__) - printf("%s %s\n", __libc_version, __libc_release); -#else - printf("unknown\n"); -#endif - return 0; -} -EOF - LIBC="" - $CC_FOR_BUILD $dummy.c -o $dummy 2>/dev/null - if test "$?" = 0 ; then - ./$dummy | grep 1\.99 > /dev/null - if test "$?" = 0 ; then LIBC="libc1" ; fi - fi - rm -f $dummy.c $dummy - echo powerpc-unknown-linux-gnu${LIBC} - exit 0 ;; - alpha:Linux:*:*) - cat <$dummy.s - .data - \$Lformat: - .byte 37,100,45,37,120,10,0 # "%d-%x\n" - .text - .globl main - .align 4 - .ent main - main: - .frame \$30,16,\$26,0 - ldgp \$29,0(\$27) - .prologue 1 - .long 0x47e03d80 # implver \$0 - lda \$2,-1 - .long 0x47e20c21 # amask \$2,\$1 - lda \$16,\$Lformat - mov \$0,\$17 - not \$1,\$18 - jsr \$26,printf - ldgp \$29,0(\$26) - mov 0,\$16 - jsr \$26,exit - .end main -EOF - LIBC="" - $CC_FOR_BUILD $dummy.s -o $dummy 2>/dev/null - if test "$?" = 0 ; then - case `./$dummy` in - 0-0) UNAME_MACHINE="alpha" ;; - 1-0) UNAME_MACHINE="alphaev5" ;; - 1-1) UNAME_MACHINE="alphaev56" ;; - 1-101) UNAME_MACHINE="alphapca56" ;; - 2-303) UNAME_MACHINE="alphaev6" ;; - 2-307) UNAME_MACHINE="alphaev67" ;; - esac - objdump --private-headers $dummy | \ - grep ld.so.1 > /dev/null - if test "$?" = 0 ; then - LIBC="libc1" - fi - fi - rm -f $dummy.s $dummy - echo ${UNAME_MACHINE}-unknown-linux-gnu${LIBC} - exit 0 ;; - parisc:Linux:*:* | hppa:Linux:*:*) - # Look for CPU level - case `grep '^cpu[^a-z]*:' /proc/cpuinfo 2>/dev/null | cut -d' ' -f2` in - PA7*) echo hppa1.1-unknown-linux-gnu ;; - PA8*) echo hppa2.0-unknown-linux-gnu ;; - *) echo hppa-unknown-linux-gnu ;; - esac - exit 0 ;; - parisc64:Linux:*:* | hppa64:Linux:*:*) - echo hppa64-unknown-linux-gnu - exit 0 ;; - s390:Linux:*:* | s390x:Linux:*:*) - echo ${UNAME_MACHINE}-ibm-linux - exit 0 ;; - sh*:Linux:*:*) - echo ${UNAME_MACHINE}-unknown-linux-gnu - exit 0 ;; - sparc:Linux:*:* | sparc64:Linux:*:*) - echo ${UNAME_MACHINE}-unknown-linux-gnu - exit 0 ;; - x86_64:Linux:*:*) - echo x86_64-unknown-linux-gnu - exit 0 ;; - i*86:Linux:*:*) - # The BFD linker knows what the default object file format is, so - # first see if it will tell us. cd to the root directory to prevent - # problems with other programs or directories called `ld' in the path. - ld_supported_emulations=`cd /; ld --help 2>&1 \ - | sed -ne '/supported emulations:/!d - s/[ ][ ]*/ /g - s/.*supported emulations: *// - s/ .*// - p'` - case "$ld_supported_emulations" in - i*86linux) - echo "${UNAME_MACHINE}-pc-linux-gnuaout" - exit 0 - ;; - elf_i*86) - TENTATIVE="${UNAME_MACHINE}-pc-linux-gnu" - ;; - i*86coff) - echo "${UNAME_MACHINE}-pc-linux-gnucoff" - exit 0 - ;; - esac - # Either a pre-BFD a.out linker (linux-gnuoldld) - # or one that does not give us useful --help. - # GCC wants to distinguish between linux-gnuoldld and linux-gnuaout. - # If ld does not provide *any* "supported emulations:" - # that means it is gnuoldld. - test -z "$ld_supported_emulations" && echo "${UNAME_MACHINE}-pc-linux-gnuoldld" && exit 0 - case "${UNAME_MACHINE}" in - i*86) - VENDOR=pc; - ;; - *) - VENDOR=unknown; - ;; - esac - # Determine whether the default compiler is a.out or elf - cat >$dummy.c < -#ifdef __cplusplus -#include /* for printf() prototype */ - int main (int argc, char *argv[]) { -#else - int main (argc, argv) int argc; char *argv[]; { -#endif -#ifdef __ELF__ -# ifdef __GLIBC__ -# if __GLIBC__ >= 2 - printf ("%s-${VENDOR}-linux-gnu\n", argv[1]); -# else - printf ("%s-${VENDOR}-linux-gnulibc1\n", argv[1]); -# endif -# else - printf ("%s-${VENDOR}-linux-gnulibc1\n", argv[1]); -# endif -#else - printf ("%s-${VENDOR}-linux-gnuaout\n", argv[1]); -#endif - return 0; -} -EOF - $CC_FOR_BUILD $dummy.c -o $dummy 2>/dev/null && ./$dummy "${UNAME_MACHINE}" && rm -f $dummy.c $dummy && exit 0 - rm -f $dummy.c $dummy - test x"${TENTATIVE}" != x && echo "${TENTATIVE}" && exit 0 - ;; -# ptx 4.0 does uname -s correctly, with DYNIX/ptx in there. earlier versions -# are messed up and put the nodename in both sysname and nodename. - i*86:DYNIX/ptx:4*:*) - echo i386-sequent-sysv4 - exit 0 ;; - i*86:UNIX_SV:4.2MP:2.*) - # Unixware is an offshoot of SVR4, but it has its own version - # number series starting with 2... - # I am not positive that other SVR4 systems won't match this, - # I just have to hope. -- rms. - # Use sysv4.2uw... so that sysv4* matches it. - echo ${UNAME_MACHINE}-pc-sysv4.2uw${UNAME_VERSION} - exit 0 ;; - i*86:*:4.*:* | i*86:SYSTEM_V:4.*:*) - UNAME_REL=`echo ${UNAME_RELEASE} | sed 's/\/MP$//'` - if grep Novell /usr/include/link.h >/dev/null 2>/dev/null; then - echo ${UNAME_MACHINE}-univel-sysv${UNAME_REL} - else - echo ${UNAME_MACHINE}-pc-sysv${UNAME_REL} - fi - exit 0 ;; - i*86:*:5:7*) - # Fixed at (any) Pentium or better - UNAME_MACHINE=i586 - if [ ${UNAME_SYSTEM} = "UnixWare" ] ; then - echo ${UNAME_MACHINE}-sco-sysv${UNAME_RELEASE}uw${UNAME_VERSION} - else - echo ${UNAME_MACHINE}-pc-sysv${UNAME_RELEASE} - fi - exit 0 ;; - i*86:*:3.2:*) - if test -f /usr/options/cb.name; then - UNAME_REL=`sed -n 's/.*Version //p' /dev/null >/dev/null ; then - UNAME_REL=`(/bin/uname -X|egrep Release|sed -e 's/.*= //')` - (/bin/uname -X|egrep i80486 >/dev/null) && UNAME_MACHINE=i486 - (/bin/uname -X|egrep '^Machine.*Pentium' >/dev/null) \ - && UNAME_MACHINE=i586 - (/bin/uname -X|egrep '^Machine.*Pent ?II' >/dev/null) \ - && UNAME_MACHINE=i686 - (/bin/uname -X|egrep '^Machine.*Pentium Pro' >/dev/null) \ - && UNAME_MACHINE=i686 - echo ${UNAME_MACHINE}-pc-sco$UNAME_REL - else - echo ${UNAME_MACHINE}-pc-sysv32 - fi - exit 0 ;; - i*86:*DOS:*:*) - echo ${UNAME_MACHINE}-pc-msdosdjgpp - exit 0 ;; - pc:*:*:*) - # Left here for compatibility: - # uname -m prints for DJGPP always 'pc', but it prints nothing about - # the processor, so we play safe by assuming i386. - echo i386-pc-msdosdjgpp - exit 0 ;; - Intel:Mach:3*:*) - echo i386-pc-mach3 - exit 0 ;; - paragon:*:*:*) - echo i860-intel-osf1 - exit 0 ;; - i860:*:4.*:*) # i860-SVR4 - if grep Stardent /usr/include/sys/uadmin.h >/dev/null 2>&1 ; then - echo i860-stardent-sysv${UNAME_RELEASE} # Stardent Vistra i860-SVR4 - else # Add other i860-SVR4 vendors below as they are discovered. - echo i860-unknown-sysv${UNAME_RELEASE} # Unknown i860-SVR4 - fi - exit 0 ;; - mini*:CTIX:SYS*5:*) - # "miniframe" - echo m68010-convergent-sysv - exit 0 ;; - M68*:*:R3V[567]*:*) - test -r /sysV68 && echo 'm68k-motorola-sysv' && exit 0 ;; - 3[34]??:*:4.0:3.0 | 3[34]??A:*:4.0:3.0 | 3[34]??,*:*:4.0:3.0 | 4850:*:4.0:3.0) - OS_REL='' - test -r /etc/.relid \ - && OS_REL=.`sed -n 's/[^ ]* [^ ]* \([0-9][0-9]\).*/\1/p' < /etc/.relid` - /bin/uname -p 2>/dev/null | grep 86 >/dev/null \ - && echo i486-ncr-sysv4.3${OS_REL} && exit 0 - /bin/uname -p 2>/dev/null | /bin/grep entium >/dev/null \ - && echo i586-ncr-sysv4.3${OS_REL} && exit 0 ;; - 3[34]??:*:4.0:* | 3[34]??,*:*:4.0:*) - /bin/uname -p 2>/dev/null | grep 86 >/dev/null \ - && echo i486-ncr-sysv4 && exit 0 ;; - m68*:LynxOS:2.*:* | m68*:LynxOS:3.0*:*) - echo m68k-unknown-lynxos${UNAME_RELEASE} - exit 0 ;; - mc68030:UNIX_System_V:4.*:*) - echo m68k-atari-sysv4 - exit 0 ;; - i*86:LynxOS:2.*:* | i*86:LynxOS:3.[01]*:* | i*86:LynxOS:4.0*:*) - echo i386-unknown-lynxos${UNAME_RELEASE} - exit 0 ;; - TSUNAMI:LynxOS:2.*:*) - echo sparc-unknown-lynxos${UNAME_RELEASE} - exit 0 ;; - rs6000:LynxOS:2.*:*) - echo rs6000-unknown-lynxos${UNAME_RELEASE} - exit 0 ;; - PowerPC:LynxOS:2.*:* | PowerPC:LynxOS:3.[01]*:* | PowerPC:LynxOS:4.0*:*) - echo powerpc-unknown-lynxos${UNAME_RELEASE} - exit 0 ;; - SM[BE]S:UNIX_SV:*:*) - echo mips-dde-sysv${UNAME_RELEASE} - exit 0 ;; - RM*:ReliantUNIX-*:*:*) - echo mips-sni-sysv4 - exit 0 ;; - RM*:SINIX-*:*:*) - echo mips-sni-sysv4 - exit 0 ;; - *:SINIX-*:*:*) - if uname -p 2>/dev/null >/dev/null ; then - UNAME_MACHINE=`(uname -p) 2>/dev/null` - echo ${UNAME_MACHINE}-sni-sysv4 - else - echo ns32k-sni-sysv - fi - exit 0 ;; - PENTIUM:CPunix:4.0*:*) # Unisys `ClearPath HMP IX 4000' SVR4/MP effort - # says - echo i586-unisys-sysv4 - exit 0 ;; - *:UNIX_System_V:4*:FTX*) - # From Gerald Hewes . - # How about differentiating between stratus architectures? -djm - echo hppa1.1-stratus-sysv4 - exit 0 ;; - *:*:*:FTX*) - # From seanf@swdc.stratus.com. - echo i860-stratus-sysv4 - exit 0 ;; - mc68*:A/UX:*:*) - echo m68k-apple-aux${UNAME_RELEASE} - exit 0 ;; - news*:NEWS-OS:6*:*) - echo mips-sony-newsos6 - exit 0 ;; - R[34]000:*System_V*:*:* | R4000:UNIX_SYSV:*:* | R*000:UNIX_SV:*:*) - if [ -d /usr/nec ]; then - echo mips-nec-sysv${UNAME_RELEASE} - else - echo mips-unknown-sysv${UNAME_RELEASE} - fi - exit 0 ;; - BeBox:BeOS:*:*) # BeOS running on hardware made by Be, PPC only. - echo powerpc-be-beos - exit 0 ;; - BeMac:BeOS:*:*) # BeOS running on Mac or Mac clone, PPC only. - echo powerpc-apple-beos - exit 0 ;; - BePC:BeOS:*:*) # BeOS running on Intel PC compatible. - echo i586-pc-beos - exit 0 ;; - SX-4:SUPER-UX:*:*) - echo sx4-nec-superux${UNAME_RELEASE} - exit 0 ;; - SX-5:SUPER-UX:*:*) - echo sx5-nec-superux${UNAME_RELEASE} - exit 0 ;; - Power*:Rhapsody:*:*) - echo powerpc-apple-rhapsody${UNAME_RELEASE} - exit 0 ;; - *:Rhapsody:*:*) - echo ${UNAME_MACHINE}-apple-rhapsody${UNAME_RELEASE} - exit 0 ;; - *:Darwin:*:*) - echo `uname -p`-apple-darwin${UNAME_RELEASE} - exit 0 ;; - *:procnto*:*:* | *:QNX:[0123456789]*:*) - if test "${UNAME_MACHINE}" = "x86pc"; then - UNAME_MACHINE=pc - fi - echo `uname -p`-${UNAME_MACHINE}-nto-qnx - exit 0 ;; - *:QNX:*:4*) - echo i386-pc-qnx - exit 0 ;; - NSR-[KW]:NONSTOP_KERNEL:*:*) - echo nsr-tandem-nsk${UNAME_RELEASE} - exit 0 ;; - *:NonStop-UX:*:*) - echo mips-compaq-nonstopux - exit 0 ;; - BS2000:POSIX*:*:*) - echo bs2000-siemens-sysv - exit 0 ;; - DS/*:UNIX_System_V:*:*) - echo ${UNAME_MACHINE}-${UNAME_SYSTEM}-${UNAME_RELEASE} - exit 0 ;; - *:Plan9:*:*) - # "uname -m" is not consistent, so use $cputype instead. 386 - # is converted to i386 for consistency with other x86 - # operating systems. - if test "$cputype" = "386"; then - UNAME_MACHINE=i386 - else - UNAME_MACHINE="$cputype" - fi - echo ${UNAME_MACHINE}-unknown-plan9 - exit 0 ;; - i*86:OS/2:*:*) - # If we were able to find `uname', then EMX Unix compatibility - # is probably installed. - echo ${UNAME_MACHINE}-pc-os2-emx - exit 0 ;; - *:TOPS-10:*:*) - echo pdp10-unknown-tops10 - exit 0 ;; - *:TENEX:*:*) - echo pdp10-unknown-tenex - exit 0 ;; - KS10:TOPS-20:*:* | KL10:TOPS-20:*:* | TYPE4:TOPS-20:*:*) - echo pdp10-dec-tops20 - exit 0 ;; - XKL-1:TOPS-20:*:* | TYPE5:TOPS-20:*:*) - echo pdp10-xkl-tops20 - exit 0 ;; - *:TOPS-20:*:*) - echo pdp10-unknown-tops20 - exit 0 ;; - *:ITS:*:*) - echo pdp10-unknown-its - exit 0 ;; -esac - -#echo '(No uname command or uname output not recognized.)' 1>&2 -#echo "${UNAME_MACHINE}:${UNAME_SYSTEM}:${UNAME_RELEASE}:${UNAME_VERSION}" 1>&2 - -cat >$dummy.c < -# include -#endif -main () -{ -#if defined (sony) -#if defined (MIPSEB) - /* BFD wants "bsd" instead of "newsos". Perhaps BFD should be changed, - I don't know.... */ - printf ("mips-sony-bsd\n"); exit (0); -#else -#include - printf ("m68k-sony-newsos%s\n", -#ifdef NEWSOS4 - "4" -#else - "" -#endif - ); exit (0); -#endif -#endif - -#if defined (__arm) && defined (__acorn) && defined (__unix) - printf ("arm-acorn-riscix"); exit (0); -#endif - -#if defined (hp300) && !defined (hpux) - printf ("m68k-hp-bsd\n"); exit (0); -#endif - -#if defined (NeXT) -#if !defined (__ARCHITECTURE__) -#define __ARCHITECTURE__ "m68k" -#endif - int version; - version=`(hostinfo | sed -n 's/.*NeXT Mach \([0-9]*\).*/\1/p') 2>/dev/null`; - if (version < 4) - printf ("%s-next-nextstep%d\n", __ARCHITECTURE__, version); - else - printf ("%s-next-openstep%d\n", __ARCHITECTURE__, version); - exit (0); -#endif - -#if defined (MULTIMAX) || defined (n16) -#if defined (UMAXV) - printf ("ns32k-encore-sysv\n"); exit (0); -#else -#if defined (CMU) - printf ("ns32k-encore-mach\n"); exit (0); -#else - printf ("ns32k-encore-bsd\n"); exit (0); -#endif -#endif -#endif - -#if defined (__386BSD__) - printf ("i386-pc-bsd\n"); exit (0); -#endif - -#if defined (sequent) -#if defined (i386) - printf ("i386-sequent-dynix\n"); exit (0); -#endif -#if defined (ns32000) - printf ("ns32k-sequent-dynix\n"); exit (0); -#endif -#endif - -#if defined (_SEQUENT_) - struct utsname un; - - uname(&un); - - if (strncmp(un.version, "V2", 2) == 0) { - printf ("i386-sequent-ptx2\n"); exit (0); - } - if (strncmp(un.version, "V1", 2) == 0) { /* XXX is V1 correct? */ - printf ("i386-sequent-ptx1\n"); exit (0); - } - printf ("i386-sequent-ptx\n"); exit (0); - -#endif - -#if defined (vax) -# if !defined (ultrix) -# include -# if defined (BSD) -# if BSD == 43 - printf ("vax-dec-bsd4.3\n"); exit (0); -# else -# if BSD == 199006 - printf ("vax-dec-bsd4.3reno\n"); exit (0); -# else - printf ("vax-dec-bsd\n"); exit (0); -# endif -# endif -# else - printf ("vax-dec-bsd\n"); exit (0); -# endif -# else - printf ("vax-dec-ultrix\n"); exit (0); -# endif -#endif - -#if defined (alliant) && defined (i860) - printf ("i860-alliant-bsd\n"); exit (0); -#endif - - exit (1); -} -EOF - -$CC_FOR_BUILD $dummy.c -o $dummy 2>/dev/null && ./$dummy && rm -f $dummy.c $dummy && exit 0 -rm -f $dummy.c $dummy - -# Apollos put the system type in the environment. - -test -d /usr/apollo && { echo ${ISP}-apollo-${SYSTYPE}; exit 0; } - -# Convex versions that predate uname can use getsysinfo(1) - -if [ -x /usr/convex/getsysinfo ] -then - case `getsysinfo -f cpu_type` in - c1*) - echo c1-convex-bsd - exit 0 ;; - c2*) - if getsysinfo -f scalar_acc - then echo c32-convex-bsd - else echo c2-convex-bsd - fi - exit 0 ;; - c34*) - echo c34-convex-bsd - exit 0 ;; - c38*) - echo c38-convex-bsd - exit 0 ;; - c4*) - echo c4-convex-bsd - exit 0 ;; - esac -fi - -cat >&2 < in order to provide the needed -information to handle your system. - -config.guess timestamp = $timestamp - -uname -m = `(uname -m) 2>/dev/null || echo unknown` -uname -r = `(uname -r) 2>/dev/null || echo unknown` -uname -s = `(uname -s) 2>/dev/null || echo unknown` -uname -v = `(uname -v) 2>/dev/null || echo unknown` - -/usr/bin/uname -p = `(/usr/bin/uname -p) 2>/dev/null` -/bin/uname -X = `(/bin/uname -X) 2>/dev/null` - -hostinfo = `(hostinfo) 2>/dev/null` -/bin/universe = `(/bin/universe) 2>/dev/null` -/usr/bin/arch -k = `(/usr/bin/arch -k) 2>/dev/null` -/bin/arch = `(/bin/arch) 2>/dev/null` -/usr/bin/oslevel = `(/usr/bin/oslevel) 2>/dev/null` -/usr/convex/getsysinfo = `(/usr/convex/getsysinfo) 2>/dev/null` - -UNAME_MACHINE = ${UNAME_MACHINE} -UNAME_RELEASE = ${UNAME_RELEASE} -UNAME_SYSTEM = ${UNAME_SYSTEM} -UNAME_VERSION = ${UNAME_VERSION} -EOF - -exit 1 - -# Local variables: -# eval: (add-hook 'write-file-hooks 'time-stamp) -# time-stamp-start: "timestamp='" -# time-stamp-format: "%:y-%02m-%02d" -# time-stamp-end: "'" -# End: diff --git a/config.sub b/config.sub deleted file mode 100755 index 7d3f49f..0000000 --- a/config.sub +++ /dev/null @@ -1,1362 +0,0 @@ -#! /bin/sh -# Configuration validation subroutine script. -# Copyright (C) 1992, 1993, 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001 -# Free Software Foundation, Inc. - -timestamp='2004-02-26' - -# This file is (in principle) common to ALL GNU software. -# The presence of a machine in this file suggests that SOME GNU software -# can handle that machine. It does not imply ALL GNU software can. -# -# This file is free software; you can redistribute it and/or modify -# it under the terms of the GNU General Public License as published by -# the Free Software Foundation; either version 2 of the License, or -# (at your option) any later version. -# -# This program is distributed in the hope that it will be useful, -# but WITHOUT ANY WARRANTY; without even the implied warranty of -# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -# GNU General Public License for more details. -# -# You should have received a copy of the GNU General Public License -# along with this program; if not, write to the Free Software -# Foundation, Inc., 59 Temple Place - Suite 330, -# Boston, MA 02111-1307, USA. - -# As a special exception to the GNU General Public License, if you -# distribute this file as part of a program that contains a -# configuration script generated by Autoconf, you may include it under -# the same distribution terms that you use for the rest of that program. - -# Please send patches to . -# -# Configuration subroutine to validate and canonicalize a configuration type. -# Supply the specified configuration type as an argument. -# If it is invalid, we print an error message on stderr and exit with code 1. -# Otherwise, we print the canonical config type on stdout and succeed. - -# This file is supposed to be the same for all GNU packages -# and recognize all the CPU types, system types and aliases -# that are meaningful with *any* GNU software. -# Each package is responsible for reporting which valid configurations -# it does not support. The user should be able to distinguish -# a failure to support a valid configuration from a meaningless -# configuration. - -# The goal of this file is to map all the various variations of a given -# machine specification into a single specification in the form: -# CPU_TYPE-MANUFACTURER-OPERATING_SYSTEM -# or in some cases, the newer four-part form: -# CPU_TYPE-MANUFACTURER-KERNEL-OPERATING_SYSTEM -# It is wrong to echo any other type of specification. - -me=`echo "$0" | sed -e 's,.*/,,'` - -usage="\ -Usage: $0 [OPTION] CPU-MFR-OPSYS - $0 [OPTION] ALIAS - -Canonicalize a configuration name. - -Operation modes: - -h, --help print this help, then exit - -t, --time-stamp print date of last modification, then exit - -v, --version print version number, then exit - -Report bugs and patches to ." - -version="\ -GNU config.sub ($timestamp) - -Copyright (C) 1992, 1993, 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001 -Free Software Foundation, Inc. - -This is free software; see the source for copying conditions. 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Otherwise, code above -# will signal an error saying that MANUFACTURER isn't an operating -# system, and we'll never get to this point. - -case $basic_machine in - *-acorn) - os=-riscix1.2 - ;; - arm*-rebel) - os=-linux - ;; - arm*-semi) - os=-aout - ;; - pdp10-*) - os=-tops20 - ;; - pdp11-*) - os=-none - ;; - *-dec | vax-*) - os=-ultrix4.2 - ;; - m68*-apollo) - os=-domain - ;; - i386-sun) - os=-sunos4.0.2 - ;; - m68000-sun) - os=-sunos3 - # This also exists in the configure program, but was not the - # default. - # os=-sunos4 - ;; - m68*-cisco) - os=-aout - ;; - mips*-cisco) - os=-elf - ;; - mips*-*) - os=-elf - ;; - *-tti) # must be before sparc entry or we get the wrong os. - os=-sysv3 - ;; - sparc-* | *-sun) - os=-sunos4.1.1 - ;; - *-be) - os=-beos - ;; - *-ibm) - os=-aix - ;; - *-wec) - os=-proelf - ;; - *-winbond) - os=-proelf - ;; - *-oki) - os=-proelf - ;; - *-hp) - os=-hpux - ;; - *-hitachi) - os=-hiux - ;; - i860-* | *-att | *-ncr | *-altos | *-motorola | *-convergent) - os=-sysv - ;; - *-cbm) - os=-amigaos - ;; - *-dg) - os=-dgux - ;; - *-dolphin) - os=-sysv3 - ;; - m68k-ccur) - os=-rtu - ;; - m88k-omron*) - os=-luna - ;; - *-next ) - os=-nextstep - ;; - *-sequent) - os=-ptx - ;; - *-crds) - os=-unos - ;; - *-ns) - os=-genix - ;; - i370-*) - os=-mvs - ;; - *-next) - os=-nextstep3 - ;; - *-gould) - os=-sysv - ;; - *-highlevel) - os=-bsd - ;; - *-encore) - os=-bsd - ;; - *-sgi) - os=-irix - ;; - *-siemens) - os=-sysv4 - ;; - *-masscomp) - os=-rtu - ;; - f30[01]-fujitsu | f700-fujitsu) - os=-uxpv - ;; - *-rom68k) - os=-coff - ;; - *-*bug) - os=-coff - ;; - *-apple) - os=-macos - ;; - *-atari*) - os=-mint - ;; - *) - os=-none - ;; -esac -fi - -# Here we handle the case where we know the os, and the CPU type, but not the -# manufacturer. We pick the logical manufacturer. -vendor=unknown -case $basic_machine in - *-unknown) - case $os in - -riscix*) - vendor=acorn - ;; - -sunos*) - vendor=sun - ;; - -aix*) - vendor=ibm - ;; - -beos*) - vendor=be - ;; - -hpux*) - vendor=hp - ;; - -mpeix*) - vendor=hp - ;; - -hiux*) - vendor=hitachi - ;; - -unos*) - vendor=crds - ;; - -dgux*) - vendor=dg - ;; - -luna*) - vendor=omron - ;; - -genix*) - vendor=ns - ;; - -mvs* | -opened*) - vendor=ibm - ;; - -ptx*) - vendor=sequent - ;; - -vxsim* | -vxworks*) - vendor=wrs - ;; - -aux*) - vendor=apple - ;; - -hms*) - vendor=hitachi - ;; - -mpw* | -macos*) - vendor=apple - ;; - -*mint | -mint[0-9]* | -*MiNT | -MiNT[0-9]*) - vendor=atari - ;; - esac - basic_machine=`echo $basic_machine | sed "s/unknown/$vendor/"` - ;; -esac - -echo $basic_machine$os -exit 0 - -# Local variables: -# eval: (add-hook 'write-file-hooks 'time-stamp) -# time-stamp-start: "timestamp='" -# time-stamp-format: "%:y-%02m-%02d" -# time-stamp-end: "'" -# End: diff --git a/config/config.guess b/config/config.guess new file mode 100644 index 0000000..313be34 --- /dev/null +++ b/config/config.guess @@ -0,0 +1,1371 @@ +#! /bin/sh +# Attempt to guess a canonical system name. +# Copyright (C) 1992, 1993, 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001 +# Free Software Foundation, Inc. + +timestamp='2004-02-26' + +# This file is free software; you can redistribute it and/or modify it +# under the terms of the GNU General Public License as published by +# the Free Software Foundation; either version 2 of the License, or +# (at your option) any later version. +# +# This program is distributed in the hope that it will be useful, but +# WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +# General Public License for more details. +# +# You should have received a copy of the GNU General Public License +# along with this program; if not, write to the Free Software +# Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. +# +# As a special exception to the GNU General Public License, if you +# distribute this file as part of a program that contains a +# configuration script generated by Autoconf, you may include it under +# the same distribution terms that you use for the rest of that program. + +# Written by Per Bothner . +# Please send patches to . +# +# This script attempts to guess a canonical system name similar to +# config.sub. If it succeeds, it prints the system name on stdout, and +# exits with 0. Otherwise, it exits with 1. +# +# The plan is that this can be called by configure scripts if you +# don't specify an explicit build system type. + +me=`echo "$0" | sed -e 's,.*/,,'` + +usage="\ +Usage: $0 [OPTION] + +Output the configuration name of the system \`$me' is run on. + +Operation modes: + -h, --help print this help, then exit + -t, --time-stamp print date of last modification, then exit + -v, --version print version number, then exit + +Report bugs and patches to ." + +version="\ +GNU config.guess ($timestamp) + +Originally written by Per Bothner. +Copyright (C) 1992, 93, 94, 95, 96, 97, 98, 99, 2000 +Free Software Foundation, Inc. + +This is free software; see the source for copying conditions. There is NO +warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE." + +help=" +Try \`$me --help' for more information." + +# Parse command line +while test $# -gt 0 ; do + case $1 in + --time-stamp | --time* | -t ) + echo "$timestamp" ; exit 0 ;; + --version | -v ) + echo "$version" ; exit 0 ;; + --help | --h* | -h ) + echo "$usage"; exit 0 ;; + -- ) # Stop option processing + shift; break ;; + - ) # Use stdin as input. + break ;; + -* ) + echo "$me: invalid option $1$help" >&2 + exit 1 ;; + * ) + break ;; + esac +done + +if test $# != 0; then + echo "$me: too many arguments$help" >&2 + exit 1 +fi + + +dummy=dummy-$$ +trap 'rm -f $dummy.c $dummy.o $dummy.rel $dummy; exit 1' 1 2 15 + +# CC_FOR_BUILD -- compiler used by this script. +# Historically, `CC_FOR_BUILD' used to be named `HOST_CC'. We still +# use `HOST_CC' if defined, but it is deprecated. + +case $CC_FOR_BUILD,$HOST_CC,$CC in + ,,) echo "int dummy(){}" > $dummy.c + for c in cc gcc c89 ; do + ($c $dummy.c -c -o $dummy.o) >/dev/null 2>&1 + if test $? = 0 ; then + CC_FOR_BUILD="$c"; break + fi + done + rm -f $dummy.c $dummy.o $dummy.rel + if test x"$CC_FOR_BUILD" = x ; then + CC_FOR_BUILD=no_compiler_found + fi + ;; + ,,*) CC_FOR_BUILD=$CC ;; + ,*,*) CC_FOR_BUILD=$HOST_CC ;; +esac + +# This is needed to find uname on a Pyramid OSx when run in the BSD universe. +# (ghazi@noc.rutgers.edu 8/24/94.) +if (test -f /.attbin/uname) >/dev/null 2>&1 ; then + PATH=$PATH:/.attbin ; export PATH +fi + +UNAME_MACHINE=`(uname -m) 2>/dev/null` || UNAME_MACHINE=unknown +UNAME_RELEASE=`(uname -r) 2>/dev/null` || UNAME_RELEASE=unknown +UNAME_SYSTEM=`(uname -s) 2>/dev/null` || UNAME_SYSTEM=unknown +UNAME_VERSION=`(uname -v) 2>/dev/null` || UNAME_VERSION=unknown + +# Note: order is significant - the case branches are not exclusive. + +case "${UNAME_MACHINE}:${UNAME_SYSTEM}:${UNAME_RELEASE}:${UNAME_VERSION}" in + *:NetBSD:*:*) + # Netbsd (nbsd) targets should (where applicable) match one or + # more of the tupples: *-*-netbsdelf*, *-*-netbsdaout*, + # *-*-netbsdecoff* and *-*-netbsd*. For targets that recently + # switched to ELF, *-*-netbsd* would select the old + # object file format. This provides both forward + # compatibility and a consistent mechanism for selecting the + # object file format. + # Determine the machine/vendor (is the vendor relevant). + case "${UNAME_MACHINE}" in + amiga) machine=m68k-unknown ;; + arm32) machine=arm-unknown ;; + atari*) machine=m68k-atari ;; + sun3*) machine=m68k-sun ;; + mac68k) machine=m68k-apple ;; + macppc) machine=powerpc-apple ;; + hp3[0-9][05]) machine=m68k-hp ;; + ibmrt|romp-ibm) machine=romp-ibm ;; + *) machine=${UNAME_MACHINE}-unknown ;; + esac + # The Operating System including object format, if it has switched + # to ELF recently, or will in the future. + case "${UNAME_MACHINE}" in + i386|sparc|amiga|arm*|hp300|mvme68k|vax|atari|luna68k|mac68k|news68k|next68k|pc532|sun3*|x68k) + if echo __ELF__ | $CC_FOR_BUILD -E - 2>/dev/null \ + | grep __ELF__ >/dev/null + then + # Once all utilities can be ECOFF (netbsdecoff) or a.out (netbsdaout). + # Return netbsd for either. FIX? + os=netbsd + else + os=netbsdelf + fi + ;; + *) + os=netbsd + ;; + esac + # The OS release + release=`echo ${UNAME_RELEASE}|sed -e 's/[-_].*/\./'` + # Since CPU_TYPE-MANUFACTURER-KERNEL-OPERATING_SYSTEM: + # contains redundant information, the shorter form: + # CPU_TYPE-MANUFACTURER-OPERATING_SYSTEM is used. + echo "${machine}-${os}${release}" + exit 0 ;; + alpha:OSF1:*:*) + if test $UNAME_RELEASE = "V4.0"; then + UNAME_RELEASE=`/usr/sbin/sizer -v | awk '{print $3}'` + fi + # A Vn.n version is a released version. + # A Tn.n version is a released field test version. + # A Xn.n version is an unreleased experimental baselevel. + # 1.2 uses "1.2" for uname -r. + cat <$dummy.s + .data +\$Lformat: + .byte 37,100,45,37,120,10,0 # "%d-%x\n" + + .text + .globl main + .align 4 + .ent main +main: + .frame \$30,16,\$26,0 + ldgp \$29,0(\$27) + .prologue 1 + .long 0x47e03d80 # implver \$0 + lda \$2,-1 + .long 0x47e20c21 # amask \$2,\$1 + lda \$16,\$Lformat + mov \$0,\$17 + not \$1,\$18 + jsr \$26,printf + ldgp \$29,0(\$26) + mov 0,\$16 + jsr \$26,exit + .end main +EOF + $CC_FOR_BUILD $dummy.s -o $dummy 2>/dev/null + if test "$?" = 0 ; then + case `./$dummy` in + 0-0) + UNAME_MACHINE="alpha" + ;; + 1-0) + UNAME_MACHINE="alphaev5" + ;; + 1-1) + UNAME_MACHINE="alphaev56" + ;; + 1-101) + UNAME_MACHINE="alphapca56" + ;; + 2-303) + UNAME_MACHINE="alphaev6" + ;; + 2-307) + UNAME_MACHINE="alphaev67" + ;; + esac + fi + rm -f $dummy.s $dummy + echo ${UNAME_MACHINE}-dec-osf`echo ${UNAME_RELEASE} | sed -e 's/^[VTX]//' | tr 'ABCDEFGHIJKLMNOPQRSTUVWXYZ' 'abcdefghijklmnopqrstuvwxyz'` + exit 0 ;; + Alpha\ *:Windows_NT*:*) + # How do we know it's Interix rather than the generic POSIX subsystem? + # Should we change UNAME_MACHINE based on the output of uname instead + # of the specific Alpha model? + echo alpha-pc-interix + exit 0 ;; + 21064:Windows_NT:50:3) + echo alpha-dec-winnt3.5 + exit 0 ;; + Amiga*:UNIX_System_V:4.0:*) + echo m68k-unknown-sysv4 + exit 0;; + amiga:OpenBSD:*:*) + echo m68k-unknown-openbsd${UNAME_RELEASE} + exit 0 ;; + *:[Aa]miga[Oo][Ss]:*:*) + echo ${UNAME_MACHINE}-unknown-amigaos + exit 0 ;; + arc64:OpenBSD:*:*) + echo mips64el-unknown-openbsd${UNAME_RELEASE} + exit 0 ;; + arc:OpenBSD:*:*) + echo mipsel-unknown-openbsd${UNAME_RELEASE} + exit 0 ;; + hkmips:OpenBSD:*:*) + echo mips-unknown-openbsd${UNAME_RELEASE} + exit 0 ;; + pmax:OpenBSD:*:*) + echo mipsel-unknown-openbsd${UNAME_RELEASE} + exit 0 ;; + sgi:OpenBSD:*:*) + echo mips-unknown-openbsd${UNAME_RELEASE} + exit 0 ;; + wgrisc:OpenBSD:*:*) + echo mipsel-unknown-openbsd${UNAME_RELEASE} + exit 0 ;; + *:OS/390:*:*) + echo i370-ibm-openedition + exit 0 ;; + arm:RISC*:1.[012]*:*|arm:riscix:1.[012]*:*) + echo arm-acorn-riscix${UNAME_RELEASE} + exit 0;; + SR2?01:HI-UX/MPP:*:* | SR8000:HI-UX/MPP:*:*) + echo hppa1.1-hitachi-hiuxmpp + exit 0;; + Pyramid*:OSx*:*:* | MIS*:OSx*:*:* | MIS*:SMP_DC-OSx*:*:*) + # akee@wpdis03.wpafb.af.mil (Earle F. Ake) contributed MIS and NILE. + if test "`(/bin/universe) 2>/dev/null`" = att ; then + echo pyramid-pyramid-sysv3 + else + echo pyramid-pyramid-bsd + fi + exit 0 ;; + NILE*:*:*:dcosx) + echo pyramid-pyramid-svr4 + exit 0 ;; + sun4H:SunOS:5.*:*) + echo sparc-hal-solaris2`echo ${UNAME_RELEASE}|sed -e 's/[^.]*//'` + exit 0 ;; + sun4*:SunOS:5.*:* | tadpole*:SunOS:5.*:*) + echo sparc-sun-solaris2`echo ${UNAME_RELEASE}|sed -e 's/[^.]*//'` + exit 0 ;; + i86pc:SunOS:5.*:*) + echo i386-pc-solaris2`echo ${UNAME_RELEASE}|sed -e 's/[^.]*//'` + exit 0 ;; + sun4*:SunOS:6*:*) + # According to config.sub, this is the proper way to canonicalize + # SunOS6. Hard to guess exactly what SunOS6 will be like, but + # it's likely to be more like Solaris than SunOS4. + echo sparc-sun-solaris3`echo ${UNAME_RELEASE}|sed -e 's/[^.]*//'` + exit 0 ;; + sun4*:SunOS:*:*) + case "`/usr/bin/arch -k`" in + Series*|S4*) + UNAME_RELEASE=`uname -v` + ;; + esac + # Japanese Language versions have a version number like `4.1.3-JL'. + echo sparc-sun-sunos`echo ${UNAME_RELEASE}|sed -e 's/-/_/'` + exit 0 ;; + sun3*:SunOS:*:*) + echo m68k-sun-sunos${UNAME_RELEASE} + exit 0 ;; + sun*:*:4.2BSD:*) + UNAME_RELEASE=`(head -1 /etc/motd | awk '{print substr($5,1,3)}') 2>/dev/null` + test "x${UNAME_RELEASE}" = "x" && UNAME_RELEASE=3 + case "`/bin/arch`" in + sun3) + echo m68k-sun-sunos${UNAME_RELEASE} + ;; + sun4) + echo sparc-sun-sunos${UNAME_RELEASE} + ;; + esac + exit 0 ;; + aushp:SunOS:*:*) + echo sparc-auspex-sunos${UNAME_RELEASE} + exit 0 ;; + atari*:OpenBSD:*:*) + echo m68k-unknown-openbsd${UNAME_RELEASE} + exit 0 ;; + # The situation for MiNT is a little confusing. The machine name + # can be virtually everything (everything which is not + # "atarist" or "atariste" at least should have a processor + # > m68000). The system name ranges from "MiNT" over "FreeMiNT" + # to the lowercase version "mint" (or "freemint"). Finally + # the system name "TOS" denotes a system which is actually not + # MiNT. But MiNT is downward compatible to TOS, so this should + # be no problem. + atarist[e]:*MiNT:*:* | atarist[e]:*mint:*:* | atarist[e]:*TOS:*:*) + echo m68k-atari-mint${UNAME_RELEASE} + exit 0 ;; + atari*:*MiNT:*:* | atari*:*mint:*:* | atarist[e]:*TOS:*:*) + echo m68k-atari-mint${UNAME_RELEASE} + exit 0 ;; + *falcon*:*MiNT:*:* | *falcon*:*mint:*:* | *falcon*:*TOS:*:*) + echo m68k-atari-mint${UNAME_RELEASE} + exit 0 ;; + milan*:*MiNT:*:* | milan*:*mint:*:* | *milan*:*TOS:*:*) + echo m68k-milan-mint${UNAME_RELEASE} + exit 0 ;; + hades*:*MiNT:*:* | hades*:*mint:*:* | *hades*:*TOS:*:*) + echo m68k-hades-mint${UNAME_RELEASE} + exit 0 ;; + *:*MiNT:*:* | *:*mint:*:* | *:*TOS:*:*) + echo m68k-unknown-mint${UNAME_RELEASE} + exit 0 ;; + sun3*:OpenBSD:*:*) + echo m68k-unknown-openbsd${UNAME_RELEASE} + exit 0 ;; + mac68k:OpenBSD:*:*) + echo m68k-unknown-openbsd${UNAME_RELEASE} + exit 0 ;; + mvme68k:OpenBSD:*:*) + echo m68k-unknown-openbsd${UNAME_RELEASE} + exit 0 ;; + mvme88k:OpenBSD:*:*) + echo m88k-unknown-openbsd${UNAME_RELEASE} + exit 0 ;; + powerpc:machten:*:*) + echo powerpc-apple-machten${UNAME_RELEASE} + exit 0 ;; + RISC*:Mach:*:*) + echo mips-dec-mach_bsd4.3 + exit 0 ;; + RISC*:ULTRIX:*:*) + echo mips-dec-ultrix${UNAME_RELEASE} + exit 0 ;; + VAX*:ULTRIX*:*:*) + echo vax-dec-ultrix${UNAME_RELEASE} + exit 0 ;; + 2020:CLIX:*:* | 2430:CLIX:*:*) + echo clipper-intergraph-clix${UNAME_RELEASE} + exit 0 ;; + mips:*:*:UMIPS | mips:*:*:RISCos) + sed 's/^ //' << EOF >$dummy.c +#ifdef __cplusplus +#include /* for printf() prototype */ + int main (int argc, char *argv[]) { +#else + int main (argc, argv) int argc; char *argv[]; { +#endif + #if defined (host_mips) && defined (MIPSEB) + #if defined (SYSTYPE_SYSV) + printf ("mips-mips-riscos%ssysv\n", argv[1]); exit (0); + #endif + #if defined (SYSTYPE_SVR4) + printf ("mips-mips-riscos%ssvr4\n", argv[1]); exit (0); + #endif + #if defined (SYSTYPE_BSD43) || defined(SYSTYPE_BSD) + printf ("mips-mips-riscos%sbsd\n", argv[1]); exit (0); + #endif + #endif + exit (-1); + } +EOF + $CC_FOR_BUILD $dummy.c -o $dummy \ + && ./$dummy `echo "${UNAME_RELEASE}" | sed -n 's/\([0-9]*\).*/\1/p'` \ + && rm -f $dummy.c $dummy && exit 0 + rm -f $dummy.c $dummy + echo mips-mips-riscos${UNAME_RELEASE} + exit 0 ;; + Motorola:PowerMAX_OS:*:*) + echo powerpc-motorola-powermax + exit 0 ;; + Night_Hawk:Power_UNIX:*:*) + echo powerpc-harris-powerunix + exit 0 ;; + m88k:CX/UX:7*:*) + echo m88k-harris-cxux7 + exit 0 ;; + m88k:*:4*:R4*) + echo m88k-motorola-sysv4 + exit 0 ;; + m88k:*:3*:R3*) + echo m88k-motorola-sysv3 + exit 0 ;; + AViiON:dgux:*:*) + # DG/UX returns AViiON for all architectures + UNAME_PROCESSOR=`/usr/bin/uname -p` + if [ $UNAME_PROCESSOR = mc88100 ] || [ $UNAME_PROCESSOR = mc88110 ] + then + if [ ${TARGET_BINARY_INTERFACE}x = m88kdguxelfx ] || \ + [ ${TARGET_BINARY_INTERFACE}x = x ] + then + echo m88k-dg-dgux${UNAME_RELEASE} + else + echo m88k-dg-dguxbcs${UNAME_RELEASE} + fi + else + echo i586-dg-dgux${UNAME_RELEASE} + fi + exit 0 ;; + M88*:DolphinOS:*:*) # DolphinOS (SVR3) + echo m88k-dolphin-sysv3 + exit 0 ;; + M88*:*:R3*:*) + # Delta 88k system running SVR3 + echo m88k-motorola-sysv3 + exit 0 ;; + XD88*:*:*:*) # Tektronix XD88 system running UTekV (SVR3) + echo m88k-tektronix-sysv3 + exit 0 ;; + Tek43[0-9][0-9]:UTek:*:*) # Tektronix 4300 system running UTek (BSD) + echo m68k-tektronix-bsd + exit 0 ;; + *:IRIX*:*:*) + echo mips-sgi-irix`echo ${UNAME_RELEASE}|sed -e 's/-/_/g'` + exit 0 ;; + ????????:AIX?:[12].1:2) # AIX 2.2.1 or AIX 2.1.1 is RT/PC AIX. + echo romp-ibm-aix # uname -m gives an 8 hex-code CPU id + exit 0 ;; # Note that: echo "'`uname -s`'" gives 'AIX ' + i*86:AIX:*:*) + echo i386-ibm-aix + exit 0 ;; + ia64:AIX:*:*) + if [ -x /usr/bin/oslevel ] ; then + IBM_REV=`/usr/bin/oslevel` + else + IBM_REV=${UNAME_VERSION}.${UNAME_RELEASE} + fi + echo ${UNAME_MACHINE}-ibm-aix${IBM_REV} + exit 0 ;; + *:AIX:2:3) + if grep bos325 /usr/include/stdio.h >/dev/null 2>&1; then + sed 's/^ //' << EOF >$dummy.c + #include + + main() + { + if (!__power_pc()) + exit(1); + puts("powerpc-ibm-aix3.2.5"); + exit(0); + } +EOF + $CC_FOR_BUILD $dummy.c -o $dummy && ./$dummy && rm -f $dummy.c $dummy && exit 0 + rm -f $dummy.c $dummy + echo rs6000-ibm-aix3.2.5 + elif grep bos324 /usr/include/stdio.h >/dev/null 2>&1; then + echo rs6000-ibm-aix3.2.4 + else + echo rs6000-ibm-aix3.2 + fi + exit 0 ;; + *:AIX:*:[45]) + IBM_CPU_ID=`/usr/sbin/lsdev -C -c processor -S available | head -1 | awk '{ print $1 }'` + if /usr/sbin/lsattr -El ${IBM_CPU_ID} | grep ' POWER' >/dev/null 2>&1; then + IBM_ARCH=rs6000 + else + IBM_ARCH=powerpc + fi + if [ -x /usr/bin/oslevel ] ; then + IBM_REV=`/usr/bin/oslevel` + else + IBM_REV=${UNAME_VERSION}.${UNAME_RELEASE} + fi + echo ${IBM_ARCH}-ibm-aix${IBM_REV} + exit 0 ;; + *:AIX:*:*) + echo rs6000-ibm-aix + exit 0 ;; + ibmrt:4.4BSD:*|romp-ibm:BSD:*) + echo romp-ibm-bsd4.4 + exit 0 ;; + ibmrt:*BSD:*|romp-ibm:BSD:*) # covers RT/PC BSD and + echo romp-ibm-bsd${UNAME_RELEASE} # 4.3 with uname added to + exit 0 ;; # report: romp-ibm BSD 4.3 + *:BOSX:*:*) + echo rs6000-bull-bosx + exit 0 ;; + DPX/2?00:B.O.S.:*:*) + echo m68k-bull-sysv3 + exit 0 ;; + 9000/[34]??:4.3bsd:1.*:*) + echo m68k-hp-bsd + exit 0 ;; + hp300:4.4BSD:*:* | 9000/[34]??:4.3bsd:2.*:*) + echo m68k-hp-bsd4.4 + exit 0 ;; + 9000/[34678]??:HP-UX:*:*) + HPUX_REV=`echo ${UNAME_RELEASE}|sed -e 's/[^.]*.[0B]*//'` + case "${UNAME_MACHINE}" in + 9000/31? ) HP_ARCH=m68000 ;; + 9000/[34]?? ) HP_ARCH=m68k ;; + 9000/[678][0-9][0-9]) + case "${HPUX_REV}" in + 11.[0-9][0-9]) + if [ -x /usr/bin/getconf ]; then + sc_cpu_version=`/usr/bin/getconf SC_CPU_VERSION 2>/dev/null` + sc_kernel_bits=`/usr/bin/getconf SC_KERNEL_BITS 2>/dev/null` + case "${sc_cpu_version}" in + 523) HP_ARCH="hppa1.0" ;; # CPU_PA_RISC1_0 + 528) HP_ARCH="hppa1.1" ;; # CPU_PA_RISC1_1 + 532) # CPU_PA_RISC2_0 + case "${sc_kernel_bits}" in + 32) HP_ARCH="hppa2.0n" ;; + 64) HP_ARCH="hppa2.0w" ;; + esac ;; + esac + fi ;; + esac + if [ "${HP_ARCH}" = "" ]; then + sed 's/^ //' << EOF >$dummy.c + + #define _HPUX_SOURCE + #include + #include + + int main () + { + #if defined(_SC_KERNEL_BITS) + long bits = sysconf(_SC_KERNEL_BITS); + #endif + long cpu = sysconf (_SC_CPU_VERSION); + + switch (cpu) + { + case CPU_PA_RISC1_0: puts ("hppa1.0"); break; + case CPU_PA_RISC1_1: puts ("hppa1.1"); break; + case CPU_PA_RISC2_0: + #if defined(_SC_KERNEL_BITS) + switch (bits) + { + case 64: puts ("hppa2.0w"); break; + case 32: puts ("hppa2.0n"); break; + default: puts ("hppa2.0"); break; + } break; + #else /* !defined(_SC_KERNEL_BITS) */ + puts ("hppa2.0"); break; + #endif + default: puts ("hppa1.0"); break; + } + exit (0); + } +EOF + (CCOPTS= $CC_FOR_BUILD $dummy.c -o $dummy 2>/dev/null ) && HP_ARCH=`./$dummy` + if test -z "$HP_ARCH"; then HP_ARCH=hppa; fi + rm -f $dummy.c $dummy + fi ;; + esac + echo ${HP_ARCH}-hp-hpux${HPUX_REV} + exit 0 ;; + ia64:HP-UX:*:*) + HPUX_REV=`echo ${UNAME_RELEASE}|sed -e 's/[^.]*.[0B]*//'` + echo ia64-hp-hpux${HPUX_REV} + exit 0 ;; + 3050*:HI-UX:*:*) + sed 's/^ //' << EOF >$dummy.c + #include + int + main () + { + long cpu = sysconf (_SC_CPU_VERSION); + /* The order matters, because CPU_IS_HP_MC68K erroneously returns + true for CPU_PA_RISC1_0. CPU_IS_PA_RISC returns correct + results, however. */ + if (CPU_IS_PA_RISC (cpu)) + { + switch (cpu) + { + case CPU_PA_RISC1_0: puts ("hppa1.0-hitachi-hiuxwe2"); break; + case CPU_PA_RISC1_1: puts ("hppa1.1-hitachi-hiuxwe2"); break; + case CPU_PA_RISC2_0: puts ("hppa2.0-hitachi-hiuxwe2"); break; + default: puts ("hppa-hitachi-hiuxwe2"); break; + } + } + else if (CPU_IS_HP_MC68K (cpu)) + puts ("m68k-hitachi-hiuxwe2"); + else puts ("unknown-hitachi-hiuxwe2"); + exit (0); + } +EOF + $CC_FOR_BUILD $dummy.c -o $dummy && ./$dummy && rm -f $dummy.c $dummy && exit 0 + rm -f $dummy.c $dummy + echo unknown-hitachi-hiuxwe2 + exit 0 ;; + 9000/7??:4.3bsd:*:* | 9000/8?[79]:4.3bsd:*:* ) + echo hppa1.1-hp-bsd + exit 0 ;; + 9000/8??:4.3bsd:*:*) + echo hppa1.0-hp-bsd + exit 0 ;; + *9??*:MPE/iX:*:*) + echo hppa1.0-hp-mpeix + exit 0 ;; + hp7??:OSF1:*:* | hp8?[79]:OSF1:*:* ) + echo hppa1.1-hp-osf + exit 0 ;; + hp8??:OSF1:*:*) + echo hppa1.0-hp-osf + exit 0 ;; + i*86:OSF1:*:*) + if [ -x /usr/sbin/sysversion ] ; then + echo ${UNAME_MACHINE}-unknown-osf1mk + else + echo ${UNAME_MACHINE}-unknown-osf1 + fi + exit 0 ;; + parisc*:Lites*:*:*) + echo hppa1.1-hp-lites + exit 0 ;; + hppa*:OpenBSD:*:*) + echo hppa-unknown-openbsd + exit 0 ;; + C1*:ConvexOS:*:* | convex:ConvexOS:C1*:*) + echo c1-convex-bsd + exit 0 ;; + C2*:ConvexOS:*:* | convex:ConvexOS:C2*:*) + if getsysinfo -f scalar_acc + then echo c32-convex-bsd + else echo c2-convex-bsd + fi + exit 0 ;; + C34*:ConvexOS:*:* | convex:ConvexOS:C34*:*) + echo c34-convex-bsd + exit 0 ;; + C38*:ConvexOS:*:* | convex:ConvexOS:C38*:*) + echo c38-convex-bsd + exit 0 ;; + C4*:ConvexOS:*:* | convex:ConvexOS:C4*:*) + echo c4-convex-bsd + exit 0 ;; + CRAY*X-MP:*:*:*) + echo xmp-cray-unicos + exit 0 ;; + CRAY*Y-MP:*:*:*) + echo ymp-cray-unicos${UNAME_RELEASE} + exit 0 ;; + CRAY*[A-Z]90:*:*:*) + echo ${UNAME_MACHINE}-cray-unicos${UNAME_RELEASE} \ + | sed -e 's/CRAY.*\([A-Z]90\)/\1/' \ + -e y/ABCDEFGHIJKLMNOPQRSTUVWXYZ/abcdefghijklmnopqrstuvwxyz/ + exit 0 ;; + CRAY*TS:*:*:*) + echo t90-cray-unicos${UNAME_RELEASE} | sed -e 's/\.[^.]*$/.X/' + exit 0 ;; + CRAY*T3D:*:*:*) + echo alpha-cray-unicosmk${UNAME_RELEASE} | sed -e 's/\.[^.]*$/.X/' + exit 0 ;; + CRAY*T3E:*:*:*) + echo alphaev5-cray-unicosmk${UNAME_RELEASE} | sed -e 's/\.[^.]*$/.X/' + exit 0 ;; + CRAY*SV1:*:*:*) + echo sv1-cray-unicos${UNAME_RELEASE} | sed -e 's/\.[^.]*$/.X/' + exit 0 ;; + CRAY-2:*:*:*) + echo cray2-cray-unicos + exit 0 ;; + F30[01]:UNIX_System_V:*:* | F700:UNIX_System_V:*:*) + FUJITSU_PROC=`uname -m | tr 'ABCDEFGHIJKLMNOPQRSTUVWXYZ' 'abcdefghijklmnopqrstuvwxyz'` + FUJITSU_SYS=`uname -p | tr 'ABCDEFGHIJKLMNOPQRSTUVWXYZ' 'abcdefghijklmnopqrstuvwxyz' | sed -e 's/\///'` + FUJITSU_REL=`echo ${UNAME_RELEASE} | sed -e 's/ /_/'` + echo "${FUJITSU_PROC}-fujitsu-${FUJITSU_SYS}${FUJITSU_REL}" + exit 0 ;; + hp300:OpenBSD:*:*) + echo m68k-unknown-openbsd${UNAME_RELEASE} + exit 0 ;; + i*86:BSD/386:*:* | i*86:BSD/OS:*:* | *:Ascend\ Embedded/OS:*:*) + echo ${UNAME_MACHINE}-pc-bsdi${UNAME_RELEASE} + exit 0 ;; + sparc*:BSD/OS:*:*) + echo sparc-unknown-bsdi${UNAME_RELEASE} + exit 0 ;; + *:BSD/OS:*:*) + echo ${UNAME_MACHINE}-unknown-bsdi${UNAME_RELEASE} + exit 0 ;; + *:FreeBSD:*:*) + echo ${UNAME_MACHINE}-unknown-freebsd`echo ${UNAME_RELEASE}|sed -e 's/[-(].*//'` + exit 0 ;; + *:OpenBSD:*:*) + echo ${UNAME_MACHINE}-unknown-openbsd`echo ${UNAME_RELEASE}|sed -e 's/[-_].*/\./'` + exit 0 ;; + i*:CYGWIN*:*) + echo ${UNAME_MACHINE}-pc-cygwin + exit 0 ;; + i*:MINGW*:*) + echo ${UNAME_MACHINE}-pc-mingw32 + exit 0 ;; + i*:PW*:*) + echo ${UNAME_MACHINE}-pc-pw32 + exit 0 ;; + i*:Windows_NT*:* | Pentium*:Windows_NT*:*) + # How do we know it's Interix rather than the generic POSIX subsystem? + # It also conflicts with pre-2.0 versions of AT&T UWIN. Should we + # UNAME_MACHINE based on the output of uname instead of i386? + echo i386-pc-interix + exit 0 ;; + i*:UWIN*:*) + echo ${UNAME_MACHINE}-pc-uwin + exit 0 ;; + p*:CYGWIN*:*) + echo powerpcle-unknown-cygwin + exit 0 ;; + prep*:SunOS:5.*:*) + echo powerpcle-unknown-solaris2`echo ${UNAME_RELEASE}|sed -e 's/[^.]*//'` + exit 0 ;; + *:GNU:*:*) + echo `echo ${UNAME_MACHINE}|sed -e 's,[-/].*$,,'`-unknown-gnu`echo ${UNAME_RELEASE}|sed -e 's,/.*$,,'` + exit 0 ;; + i*86:Minix:*:*) + echo ${UNAME_MACHINE}-pc-minix + exit 0 ;; + arm*:Linux:*:*) + echo ${UNAME_MACHINE}-unknown-linux-gnu + exit 0 ;; + ia64:Linux:*:*) + echo ${UNAME_MACHINE}-unknown-linux + exit 0 ;; + m68*:Linux:*:*) + echo ${UNAME_MACHINE}-unknown-linux-gnu + exit 0 ;; + mips:Linux:*:*) + cat >$dummy.c < /* for printf() prototype */ +int main (int argc, char *argv[]) { +#else +int main (argc, argv) int argc; char *argv[]; { +#endif +#ifdef __MIPSEB__ + printf ("%s-unknown-linux-gnu\n", argv[1]); +#endif +#ifdef __MIPSEL__ + printf ("%sel-unknown-linux-gnu\n", argv[1]); +#endif + return 0; +} +EOF + $CC_FOR_BUILD $dummy.c -o $dummy 2>/dev/null && ./$dummy "${UNAME_MACHINE}" && rm -f $dummy.c $dummy && exit 0 + rm -f $dummy.c $dummy + ;; + ppc:Linux:*:*) + # Determine Lib Version + cat >$dummy.c < +#if defined(__GLIBC__) +extern char __libc_version[]; +extern char __libc_release[]; +#endif +main(argc, argv) + int argc; + char *argv[]; +{ +#if defined(__GLIBC__) + printf("%s %s\n", __libc_version, __libc_release); +#else + printf("unknown\n"); +#endif + return 0; +} +EOF + LIBC="" + $CC_FOR_BUILD $dummy.c -o $dummy 2>/dev/null + if test "$?" = 0 ; then + ./$dummy | grep 1\.99 > /dev/null + if test "$?" = 0 ; then LIBC="libc1" ; fi + fi + rm -f $dummy.c $dummy + echo powerpc-unknown-linux-gnu${LIBC} + exit 0 ;; + alpha:Linux:*:*) + cat <$dummy.s + .data + \$Lformat: + .byte 37,100,45,37,120,10,0 # "%d-%x\n" + .text + .globl main + .align 4 + .ent main + main: + .frame \$30,16,\$26,0 + ldgp \$29,0(\$27) + .prologue 1 + .long 0x47e03d80 # implver \$0 + lda \$2,-1 + .long 0x47e20c21 # amask \$2,\$1 + lda \$16,\$Lformat + mov \$0,\$17 + not \$1,\$18 + jsr \$26,printf + ldgp \$29,0(\$26) + mov 0,\$16 + jsr \$26,exit + .end main +EOF + LIBC="" + $CC_FOR_BUILD $dummy.s -o $dummy 2>/dev/null + if test "$?" = 0 ; then + case `./$dummy` in + 0-0) UNAME_MACHINE="alpha" ;; + 1-0) UNAME_MACHINE="alphaev5" ;; + 1-1) UNAME_MACHINE="alphaev56" ;; + 1-101) UNAME_MACHINE="alphapca56" ;; + 2-303) UNAME_MACHINE="alphaev6" ;; + 2-307) UNAME_MACHINE="alphaev67" ;; + esac + objdump --private-headers $dummy | \ + grep ld.so.1 > /dev/null + if test "$?" = 0 ; then + LIBC="libc1" + fi + fi + rm -f $dummy.s $dummy + echo ${UNAME_MACHINE}-unknown-linux-gnu${LIBC} + exit 0 ;; + parisc:Linux:*:* | hppa:Linux:*:*) + # Look for CPU level + case `grep '^cpu[^a-z]*:' /proc/cpuinfo 2>/dev/null | cut -d' ' -f2` in + PA7*) echo hppa1.1-unknown-linux-gnu ;; + PA8*) echo hppa2.0-unknown-linux-gnu ;; + *) echo hppa-unknown-linux-gnu ;; + esac + exit 0 ;; + parisc64:Linux:*:* | hppa64:Linux:*:*) + echo hppa64-unknown-linux-gnu + exit 0 ;; + s390:Linux:*:* | s390x:Linux:*:*) + echo ${UNAME_MACHINE}-ibm-linux + exit 0 ;; + sh*:Linux:*:*) + echo ${UNAME_MACHINE}-unknown-linux-gnu + exit 0 ;; + sparc:Linux:*:* | sparc64:Linux:*:*) + echo ${UNAME_MACHINE}-unknown-linux-gnu + exit 0 ;; + x86_64:Linux:*:*) + echo x86_64-unknown-linux-gnu + exit 0 ;; + i*86:Linux:*:*) + # The BFD linker knows what the default object file format is, so + # first see if it will tell us. cd to the root directory to prevent + # problems with other programs or directories called `ld' in the path. + ld_supported_emulations=`cd /; ld --help 2>&1 \ + | sed -ne '/supported emulations:/!d + s/[ ][ ]*/ /g + s/.*supported emulations: *// + s/ .*// + p'` + case "$ld_supported_emulations" in + i*86linux) + echo "${UNAME_MACHINE}-pc-linux-gnuaout" + exit 0 + ;; + elf_i*86) + TENTATIVE="${UNAME_MACHINE}-pc-linux-gnu" + ;; + i*86coff) + echo "${UNAME_MACHINE}-pc-linux-gnucoff" + exit 0 + ;; + esac + # Either a pre-BFD a.out linker (linux-gnuoldld) + # or one that does not give us useful --help. + # GCC wants to distinguish between linux-gnuoldld and linux-gnuaout. + # If ld does not provide *any* "supported emulations:" + # that means it is gnuoldld. + test -z "$ld_supported_emulations" && echo "${UNAME_MACHINE}-pc-linux-gnuoldld" && exit 0 + case "${UNAME_MACHINE}" in + i*86) + VENDOR=pc; + ;; + *) + VENDOR=unknown; + ;; + esac + # Determine whether the default compiler is a.out or elf + cat >$dummy.c < +#ifdef __cplusplus +#include /* for printf() prototype */ + int main (int argc, char *argv[]) { +#else + int main (argc, argv) int argc; char *argv[]; { +#endif +#ifdef __ELF__ +# ifdef __GLIBC__ +# if __GLIBC__ >= 2 + printf ("%s-${VENDOR}-linux-gnu\n", argv[1]); +# else + printf ("%s-${VENDOR}-linux-gnulibc1\n", argv[1]); +# endif +# else + printf ("%s-${VENDOR}-linux-gnulibc1\n", argv[1]); +# endif +#else + printf ("%s-${VENDOR}-linux-gnuaout\n", argv[1]); +#endif + return 0; +} +EOF + $CC_FOR_BUILD $dummy.c -o $dummy 2>/dev/null && ./$dummy "${UNAME_MACHINE}" && rm -f $dummy.c $dummy && exit 0 + rm -f $dummy.c $dummy + test x"${TENTATIVE}" != x && echo "${TENTATIVE}" && exit 0 + ;; +# ptx 4.0 does uname -s correctly, with DYNIX/ptx in there. earlier versions +# are messed up and put the nodename in both sysname and nodename. + i*86:DYNIX/ptx:4*:*) + echo i386-sequent-sysv4 + exit 0 ;; + i*86:UNIX_SV:4.2MP:2.*) + # Unixware is an offshoot of SVR4, but it has its own version + # number series starting with 2... + # I am not positive that other SVR4 systems won't match this, + # I just have to hope. -- rms. + # Use sysv4.2uw... so that sysv4* matches it. + echo ${UNAME_MACHINE}-pc-sysv4.2uw${UNAME_VERSION} + exit 0 ;; + i*86:*:4.*:* | i*86:SYSTEM_V:4.*:*) + UNAME_REL=`echo ${UNAME_RELEASE} | sed 's/\/MP$//'` + if grep Novell /usr/include/link.h >/dev/null 2>/dev/null; then + echo ${UNAME_MACHINE}-univel-sysv${UNAME_REL} + else + echo ${UNAME_MACHINE}-pc-sysv${UNAME_REL} + fi + exit 0 ;; + i*86:*:5:7*) + # Fixed at (any) Pentium or better + UNAME_MACHINE=i586 + if [ ${UNAME_SYSTEM} = "UnixWare" ] ; then + echo ${UNAME_MACHINE}-sco-sysv${UNAME_RELEASE}uw${UNAME_VERSION} + else + echo ${UNAME_MACHINE}-pc-sysv${UNAME_RELEASE} + fi + exit 0 ;; + i*86:*:3.2:*) + if test -f /usr/options/cb.name; then + UNAME_REL=`sed -n 's/.*Version //p' /dev/null >/dev/null ; then + UNAME_REL=`(/bin/uname -X|egrep Release|sed -e 's/.*= //')` + (/bin/uname -X|egrep i80486 >/dev/null) && UNAME_MACHINE=i486 + (/bin/uname -X|egrep '^Machine.*Pentium' >/dev/null) \ + && UNAME_MACHINE=i586 + (/bin/uname -X|egrep '^Machine.*Pent ?II' >/dev/null) \ + && UNAME_MACHINE=i686 + (/bin/uname -X|egrep '^Machine.*Pentium Pro' >/dev/null) \ + && UNAME_MACHINE=i686 + echo ${UNAME_MACHINE}-pc-sco$UNAME_REL + else + echo ${UNAME_MACHINE}-pc-sysv32 + fi + exit 0 ;; + i*86:*DOS:*:*) + echo ${UNAME_MACHINE}-pc-msdosdjgpp + exit 0 ;; + pc:*:*:*) + # Left here for compatibility: + # uname -m prints for DJGPP always 'pc', but it prints nothing about + # the processor, so we play safe by assuming i386. + echo i386-pc-msdosdjgpp + exit 0 ;; + Intel:Mach:3*:*) + echo i386-pc-mach3 + exit 0 ;; + paragon:*:*:*) + echo i860-intel-osf1 + exit 0 ;; + i860:*:4.*:*) # i860-SVR4 + if grep Stardent /usr/include/sys/uadmin.h >/dev/null 2>&1 ; then + echo i860-stardent-sysv${UNAME_RELEASE} # Stardent Vistra i860-SVR4 + else # Add other i860-SVR4 vendors below as they are discovered. + echo i860-unknown-sysv${UNAME_RELEASE} # Unknown i860-SVR4 + fi + exit 0 ;; + mini*:CTIX:SYS*5:*) + # "miniframe" + echo m68010-convergent-sysv + exit 0 ;; + M68*:*:R3V[567]*:*) + test -r /sysV68 && echo 'm68k-motorola-sysv' && exit 0 ;; + 3[34]??:*:4.0:3.0 | 3[34]??A:*:4.0:3.0 | 3[34]??,*:*:4.0:3.0 | 4850:*:4.0:3.0) + OS_REL='' + test -r /etc/.relid \ + && OS_REL=.`sed -n 's/[^ ]* [^ ]* \([0-9][0-9]\).*/\1/p' < /etc/.relid` + /bin/uname -p 2>/dev/null | grep 86 >/dev/null \ + && echo i486-ncr-sysv4.3${OS_REL} && exit 0 + /bin/uname -p 2>/dev/null | /bin/grep entium >/dev/null \ + && echo i586-ncr-sysv4.3${OS_REL} && exit 0 ;; + 3[34]??:*:4.0:* | 3[34]??,*:*:4.0:*) + /bin/uname -p 2>/dev/null | grep 86 >/dev/null \ + && echo i486-ncr-sysv4 && exit 0 ;; + m68*:LynxOS:2.*:* | m68*:LynxOS:3.0*:*) + echo m68k-unknown-lynxos${UNAME_RELEASE} + exit 0 ;; + mc68030:UNIX_System_V:4.*:*) + echo m68k-atari-sysv4 + exit 0 ;; + i*86:LynxOS:2.*:* | i*86:LynxOS:3.[01]*:* | i*86:LynxOS:4.0*:*) + echo i386-unknown-lynxos${UNAME_RELEASE} + exit 0 ;; + TSUNAMI:LynxOS:2.*:*) + echo sparc-unknown-lynxos${UNAME_RELEASE} + exit 0 ;; + rs6000:LynxOS:2.*:*) + echo rs6000-unknown-lynxos${UNAME_RELEASE} + exit 0 ;; + PowerPC:LynxOS:2.*:* | PowerPC:LynxOS:3.[01]*:* | PowerPC:LynxOS:4.0*:*) + echo powerpc-unknown-lynxos${UNAME_RELEASE} + exit 0 ;; + SM[BE]S:UNIX_SV:*:*) + echo mips-dde-sysv${UNAME_RELEASE} + exit 0 ;; + RM*:ReliantUNIX-*:*:*) + echo mips-sni-sysv4 + exit 0 ;; + RM*:SINIX-*:*:*) + echo mips-sni-sysv4 + exit 0 ;; + *:SINIX-*:*:*) + if uname -p 2>/dev/null >/dev/null ; then + UNAME_MACHINE=`(uname -p) 2>/dev/null` + echo ${UNAME_MACHINE}-sni-sysv4 + else + echo ns32k-sni-sysv + fi + exit 0 ;; + PENTIUM:CPunix:4.0*:*) # Unisys `ClearPath HMP IX 4000' SVR4/MP effort + # says + echo i586-unisys-sysv4 + exit 0 ;; + *:UNIX_System_V:4*:FTX*) + # From Gerald Hewes . + # How about differentiating between stratus architectures? -djm + echo hppa1.1-stratus-sysv4 + exit 0 ;; + *:*:*:FTX*) + # From seanf@swdc.stratus.com. + echo i860-stratus-sysv4 + exit 0 ;; + mc68*:A/UX:*:*) + echo m68k-apple-aux${UNAME_RELEASE} + exit 0 ;; + news*:NEWS-OS:6*:*) + echo mips-sony-newsos6 + exit 0 ;; + R[34]000:*System_V*:*:* | R4000:UNIX_SYSV:*:* | R*000:UNIX_SV:*:*) + if [ -d /usr/nec ]; then + echo mips-nec-sysv${UNAME_RELEASE} + else + echo mips-unknown-sysv${UNAME_RELEASE} + fi + exit 0 ;; + BeBox:BeOS:*:*) # BeOS running on hardware made by Be, PPC only. + echo powerpc-be-beos + exit 0 ;; + BeMac:BeOS:*:*) # BeOS running on Mac or Mac clone, PPC only. + echo powerpc-apple-beos + exit 0 ;; + BePC:BeOS:*:*) # BeOS running on Intel PC compatible. + echo i586-pc-beos + exit 0 ;; + SX-4:SUPER-UX:*:*) + echo sx4-nec-superux${UNAME_RELEASE} + exit 0 ;; + SX-5:SUPER-UX:*:*) + echo sx5-nec-superux${UNAME_RELEASE} + exit 0 ;; + Power*:Rhapsody:*:*) + echo powerpc-apple-rhapsody${UNAME_RELEASE} + exit 0 ;; + *:Rhapsody:*:*) + echo ${UNAME_MACHINE}-apple-rhapsody${UNAME_RELEASE} + exit 0 ;; + *:Darwin:*:*) + echo `uname -p`-apple-darwin${UNAME_RELEASE} + exit 0 ;; + *:procnto*:*:* | *:QNX:[0123456789]*:*) + if test "${UNAME_MACHINE}" = "x86pc"; then + UNAME_MACHINE=pc + fi + echo `uname -p`-${UNAME_MACHINE}-nto-qnx + exit 0 ;; + *:QNX:*:4*) + echo i386-pc-qnx + exit 0 ;; + NSR-[KW]:NONSTOP_KERNEL:*:*) + echo nsr-tandem-nsk${UNAME_RELEASE} + exit 0 ;; + *:NonStop-UX:*:*) + echo mips-compaq-nonstopux + exit 0 ;; + BS2000:POSIX*:*:*) + echo bs2000-siemens-sysv + exit 0 ;; + DS/*:UNIX_System_V:*:*) + echo ${UNAME_MACHINE}-${UNAME_SYSTEM}-${UNAME_RELEASE} + exit 0 ;; + *:Plan9:*:*) + # "uname -m" is not consistent, so use $cputype instead. 386 + # is converted to i386 for consistency with other x86 + # operating systems. + if test "$cputype" = "386"; then + UNAME_MACHINE=i386 + else + UNAME_MACHINE="$cputype" + fi + echo ${UNAME_MACHINE}-unknown-plan9 + exit 0 ;; + i*86:OS/2:*:*) + # If we were able to find `uname', then EMX Unix compatibility + # is probably installed. + echo ${UNAME_MACHINE}-pc-os2-emx + exit 0 ;; + *:TOPS-10:*:*) + echo pdp10-unknown-tops10 + exit 0 ;; + *:TENEX:*:*) + echo pdp10-unknown-tenex + exit 0 ;; + KS10:TOPS-20:*:* | KL10:TOPS-20:*:* | TYPE4:TOPS-20:*:*) + echo pdp10-dec-tops20 + exit 0 ;; + XKL-1:TOPS-20:*:* | TYPE5:TOPS-20:*:*) + echo pdp10-xkl-tops20 + exit 0 ;; + *:TOPS-20:*:*) + echo pdp10-unknown-tops20 + exit 0 ;; + *:ITS:*:*) + echo pdp10-unknown-its + exit 0 ;; +esac + +#echo '(No uname command or uname output not recognized.)' 1>&2 +#echo "${UNAME_MACHINE}:${UNAME_SYSTEM}:${UNAME_RELEASE}:${UNAME_VERSION}" 1>&2 + +cat >$dummy.c < +# include +#endif +main () +{ +#if defined (sony) +#if defined (MIPSEB) + /* BFD wants "bsd" instead of "newsos". Perhaps BFD should be changed, + I don't know.... */ + printf ("mips-sony-bsd\n"); exit (0); +#else +#include + printf ("m68k-sony-newsos%s\n", +#ifdef NEWSOS4 + "4" +#else + "" +#endif + ); exit (0); +#endif +#endif + +#if defined (__arm) && defined (__acorn) && defined (__unix) + printf ("arm-acorn-riscix"); exit (0); +#endif + +#if defined (hp300) && !defined (hpux) + printf ("m68k-hp-bsd\n"); exit (0); +#endif + +#if defined (NeXT) +#if !defined (__ARCHITECTURE__) +#define __ARCHITECTURE__ "m68k" +#endif + int version; + version=`(hostinfo | sed -n 's/.*NeXT Mach \([0-9]*\).*/\1/p') 2>/dev/null`; + if (version < 4) + printf ("%s-next-nextstep%d\n", __ARCHITECTURE__, version); + else + printf ("%s-next-openstep%d\n", __ARCHITECTURE__, version); + exit (0); +#endif + +#if defined (MULTIMAX) || defined (n16) +#if defined (UMAXV) + printf ("ns32k-encore-sysv\n"); exit (0); +#else +#if defined (CMU) + printf ("ns32k-encore-mach\n"); exit (0); +#else + printf ("ns32k-encore-bsd\n"); exit (0); +#endif +#endif +#endif + +#if defined (__386BSD__) + printf ("i386-pc-bsd\n"); exit (0); +#endif + +#if defined (sequent) +#if defined (i386) + printf ("i386-sequent-dynix\n"); exit (0); +#endif +#if defined (ns32000) + printf ("ns32k-sequent-dynix\n"); exit (0); +#endif +#endif + +#if defined (_SEQUENT_) + struct utsname un; + + uname(&un); + + if (strncmp(un.version, "V2", 2) == 0) { + printf ("i386-sequent-ptx2\n"); exit (0); + } + if (strncmp(un.version, "V1", 2) == 0) { /* XXX is V1 correct? */ + printf ("i386-sequent-ptx1\n"); exit (0); + } + printf ("i386-sequent-ptx\n"); exit (0); + +#endif + +#if defined (vax) +# if !defined (ultrix) +# include +# if defined (BSD) +# if BSD == 43 + printf ("vax-dec-bsd4.3\n"); exit (0); +# else +# if BSD == 199006 + printf ("vax-dec-bsd4.3reno\n"); exit (0); +# else + printf ("vax-dec-bsd\n"); exit (0); +# endif +# endif +# else + printf ("vax-dec-bsd\n"); exit (0); +# endif +# else + printf ("vax-dec-ultrix\n"); exit (0); +# endif +#endif + +#if defined (alliant) && defined (i860) + printf ("i860-alliant-bsd\n"); exit (0); +#endif + + exit (1); +} +EOF + +$CC_FOR_BUILD $dummy.c -o $dummy 2>/dev/null && ./$dummy && rm -f $dummy.c $dummy && exit 0 +rm -f $dummy.c $dummy + +# Apollos put the system type in the environment. + +test -d /usr/apollo && { echo ${ISP}-apollo-${SYSTYPE}; exit 0; } + +# Convex versions that predate uname can use getsysinfo(1) + +if [ -x /usr/convex/getsysinfo ] +then + case `getsysinfo -f cpu_type` in + c1*) + echo c1-convex-bsd + exit 0 ;; + c2*) + if getsysinfo -f scalar_acc + then echo c32-convex-bsd + else echo c2-convex-bsd + fi + exit 0 ;; + c34*) + echo c34-convex-bsd + exit 0 ;; + c38*) + echo c38-convex-bsd + exit 0 ;; + c4*) + echo c4-convex-bsd + exit 0 ;; + esac +fi + +cat >&2 < in order to provide the needed +information to handle your system. + +config.guess timestamp = $timestamp + +uname -m = `(uname -m) 2>/dev/null || echo unknown` +uname -r = `(uname -r) 2>/dev/null || echo unknown` +uname -s = `(uname -s) 2>/dev/null || echo unknown` +uname -v = `(uname -v) 2>/dev/null || echo unknown` + +/usr/bin/uname -p = `(/usr/bin/uname -p) 2>/dev/null` +/bin/uname -X = `(/bin/uname -X) 2>/dev/null` + +hostinfo = `(hostinfo) 2>/dev/null` +/bin/universe = `(/bin/universe) 2>/dev/null` +/usr/bin/arch -k = `(/usr/bin/arch -k) 2>/dev/null` +/bin/arch = `(/bin/arch) 2>/dev/null` +/usr/bin/oslevel = `(/usr/bin/oslevel) 2>/dev/null` +/usr/convex/getsysinfo = `(/usr/convex/getsysinfo) 2>/dev/null` + +UNAME_MACHINE = ${UNAME_MACHINE} +UNAME_RELEASE = ${UNAME_RELEASE} +UNAME_SYSTEM = ${UNAME_SYSTEM} +UNAME_VERSION = ${UNAME_VERSION} +EOF + +exit 1 + +# Local variables: +# eval: (add-hook 'write-file-hooks 'time-stamp) +# time-stamp-start: "timestamp='" +# time-stamp-format: "%:y-%02m-%02d" +# time-stamp-end: "'" +# End: diff --git a/config/config.sub b/config/config.sub new file mode 100755 index 0000000..7d3f49f --- /dev/null +++ b/config/config.sub @@ -0,0 +1,1362 @@ +#! /bin/sh +# Configuration validation subroutine script. +# Copyright (C) 1992, 1993, 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001 +# Free Software Foundation, Inc. + +timestamp='2004-02-26' + +# This file is (in principle) common to ALL GNU software. +# The presence of a machine in this file suggests that SOME GNU software +# can handle that machine. It does not imply ALL GNU software can. +# +# This file is free software; you can redistribute it and/or modify +# it under the terms of the GNU General Public License as published by +# the Free Software Foundation; either version 2 of the License, or +# (at your option) any later version. +# +# This program is distributed in the hope that it will be useful, +# but WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +# GNU General Public License for more details. +# +# You should have received a copy of the GNU General Public License +# along with this program; if not, write to the Free Software +# Foundation, Inc., 59 Temple Place - Suite 330, +# Boston, MA 02111-1307, USA. + +# As a special exception to the GNU General Public License, if you +# distribute this file as part of a program that contains a +# configuration script generated by Autoconf, you may include it under +# the same distribution terms that you use for the rest of that program. + +# Please send patches to . +# +# Configuration subroutine to validate and canonicalize a configuration type. +# Supply the specified configuration type as an argument. +# If it is invalid, we print an error message on stderr and exit with code 1. +# Otherwise, we print the canonical config type on stdout and succeed. + +# This file is supposed to be the same for all GNU packages +# and recognize all the CPU types, system types and aliases +# that are meaningful with *any* GNU software. +# Each package is responsible for reporting which valid configurations +# it does not support. The user should be able to distinguish +# a failure to support a valid configuration from a meaningless +# configuration. + +# The goal of this file is to map all the various variations of a given +# machine specification into a single specification in the form: +# CPU_TYPE-MANUFACTURER-OPERATING_SYSTEM +# or in some cases, the newer four-part form: +# CPU_TYPE-MANUFACTURER-KERNEL-OPERATING_SYSTEM +# It is wrong to echo any other type of specification. + +me=`echo "$0" | sed -e 's,.*/,,'` + +usage="\ +Usage: $0 [OPTION] CPU-MFR-OPSYS + $0 [OPTION] ALIAS + +Canonicalize a configuration name. + +Operation modes: + -h, --help print this help, then exit + -t, --time-stamp print date of last modification, then exit + -v, --version print version number, then exit + +Report bugs and patches to ." + +version="\ +GNU config.sub ($timestamp) + +Copyright (C) 1992, 1993, 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001 +Free Software Foundation, Inc. + +This is free software; see the source for copying conditions. There is NO +warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE." + +help=" +Try \`$me --help' for more information." + +# Parse command line +while test $# -gt 0 ; do + case $1 in + --time-stamp | --time* | -t ) + echo "$timestamp" ; exit 0 ;; + --version | -v ) + echo "$version" ; exit 0 ;; + --help | --h* | -h ) + echo "$usage"; exit 0 ;; + -- ) # Stop option processing + shift; break ;; + - ) # Use stdin as input. + break ;; + -* ) + echo "$me: invalid option $1$help" + exit 1 ;; + + *local*) + # First pass through any local machine types. + echo $1 + exit 0;; + + * ) + break ;; + esac +done + +case $# in + 0) echo "$me: missing argument$help" >&2 + exit 1;; + 1) ;; + *) echo "$me: too many arguments$help" >&2 + exit 1;; +esac + +# Separate what the user gave into CPU-COMPANY and OS or KERNEL-OS (if any). +# Here we must recognize all the valid KERNEL-OS combinations. +maybe_os=`echo $1 | sed 's/^\(.*\)-\([^-]*-[^-]*\)$/\2/'` +case $maybe_os in + nto-qnx* | linux-gnu* | storm-chaos* | os2-emx*) + os=-$maybe_os + basic_machine=`echo $1 | sed 's/^\(.*\)-\([^-]*-[^-]*\)$/\1/'` + ;; + *) + basic_machine=`echo $1 | sed 's/-[^-]*$//'` + if [ $basic_machine != $1 ] + then os=`echo $1 | sed 's/.*-/-/'` + else os=; fi + ;; +esac + +### Let's recognize common machines as not being operating systems so +### that things like config.sub decstation-3100 work. We also +### recognize some manufacturers as not being operating systems, so we +### can provide default operating systems below. +case $os in + -sun*os*) + # Prevent following clause from handling this invalid input. + ;; + -dec* | -mips* | -sequent* | -encore* | -pc532* | -sgi* | -sony* | \ + -att* | -7300* | -3300* | -delta* | -motorola* | -sun[234]* | \ + -unicom* | -ibm* | -next | -hp | -isi* | -apollo | -altos* | \ + -convergent* | -ncr* | -news | -32* | -3600* | -3100* | -hitachi* |\ + -c[123]* | -convex* | -sun | -crds | -omron* | -dg | -ultra | -tti* | \ + -harris | -dolphin | -highlevel | -gould | -cbm | -ns | -masscomp | \ + -apple | -axis) + os= + basic_machine=$1 + ;; + -sim | -cisco | -oki | -wec | -winbond) + os= + basic_machine=$1 + ;; + -scout) + ;; + -wrs) + os=-vxworks + basic_machine=$1 + ;; + -hiux*) + os=-hiuxwe2 + ;; + -sco5) + os=-sco3.2v5 + basic_machine=`echo $1 | sed -e 's/86-.*/86-pc/'` + ;; + -sco4) + os=-sco3.2v4 + basic_machine=`echo $1 | sed -e 's/86-.*/86-pc/'` + ;; + -sco3.2.[4-9]*) + os=`echo $os | sed -e 's/sco3.2./sco3.2v/'` + basic_machine=`echo $1 | sed -e 's/86-.*/86-pc/'` + ;; + -sco3.2v[4-9]*) + # Don't forget version if it is 3.2v4 or newer. + basic_machine=`echo $1 | sed -e 's/86-.*/86-pc/'` + ;; + -sco*) + os=-sco3.2v2 + basic_machine=`echo $1 | sed -e 's/86-.*/86-pc/'` + ;; + -udk*) + basic_machine=`echo $1 | sed -e 's/86-.*/86-pc/'` + ;; + -isc) + os=-isc2.2 + basic_machine=`echo $1 | sed -e 's/86-.*/86-pc/'` + ;; + -clix*) + basic_machine=clipper-intergraph + ;; + -isc*) + basic_machine=`echo $1 | sed -e 's/86-.*/86-pc/'` + ;; + -lynx*) + os=-lynxos + ;; + -ptx*) + basic_machine=`echo $1 | sed -e 's/86-.*/86-sequent/'` + ;; + -windowsnt*) + os=`echo $os | sed -e 's/windowsnt/winnt/'` + ;; + -psos*) + os=-psos + ;; + -mint | -mint[0-9]*) + basic_machine=m68k-atari + os=-mint + ;; +esac + +# Decode aliases for certain CPU-COMPANY combinations. +case $basic_machine in + # Recognize the basic CPU types without company name. + # Some are omitted here because they have special meanings below. + tahoe | i860 | ia64 | m32r | m68k | m68000 | m88k | ns32k | arc \ + | arm | arme[lb] | arm[bl]e | armv[2345] | armv[345][lb] | strongarm | xscale \ + | pyramid | mn10200 | mn10300 | tron | a29k \ + | 580 | i960 | h8300 \ + | x86 | ppcbe | mipsbe | mipsle | shbe | shle \ + | hppa | hppa1.0 | hppa1.1 | hppa2.0 | hppa2.0w | hppa2.0n \ + | hppa64 \ + | alpha | alphaev[4-8] | alphaev56 | alphapca5[67] \ + | alphaev6[78] \ + | we32k | ns16k | clipper | i370 | sh | sh[34] \ + | powerpc | powerpcle \ + | 1750a | dsp16xx | pdp10 | pdp11 \ + | mips16 | mips64 | mipsel | mips64el \ + | mips64orion | mips64orionel | mipstx39 | mipstx39el \ + | mips64vr4300 | mips64vr4300el | mips64vr4100 | mips64vr4100el \ + | mips64vr5000 | miprs64vr5000el | mcore | s390 | s390x \ + | sparc | sparclet | sparclite | sparc64 | sparcv9 | sparcv9b \ + | v850 | c4x \ + | thumb | d10v | d30v | fr30 | avr | openrisc | tic80 \ + | pj | pjl | h8500) + basic_machine=$basic_machine-unknown + ;; + m6811 | m68hc11 | m6812 | m68hc12) + # Motorola 68HC11/12. + basic_machine=$basic_machine-unknown + os=-none + ;; + m88110 | m680[12346]0 | m683?2 | m68360 | m5200 | z8k | v70 | w65) + ;; + + # We use `pc' rather than `unknown' + # because (1) that's what they normally are, and + # (2) the word "unknown" tends to confuse beginning users. + i*86 | x86_64) + basic_machine=$basic_machine-pc + ;; + # Object if more than one company name word. + *-*-*) + echo Invalid configuration \`$1\': machine \`$basic_machine\' not recognized 1>&2 + exit 1 + ;; + # Recognize the basic CPU types with company name. + # FIXME: clean up the formatting here. + vax-* | tahoe-* | i*86-* | i860-* | ia64-* | m32r-* | m68k-* | m68000-* \ + | m88k-* | sparc-* | ns32k-* | fx80-* | arc-* | c[123]* \ + | arm-* | armbe-* | armle-* | armv*-* | strongarm-* | xscale-* \ + | mips-* | pyramid-* | tron-* | a29k-* | romp-* | rs6000-* \ + | power-* | none-* | 580-* | cray2-* | h8300-* | h8500-* | i960-* \ + | xmp-* | ymp-* \ + | x86-* | ppcbe-* | mipsbe-* | mipsle-* | shbe-* | shle-* \ + | hppa-* | hppa1.0-* | hppa1.1-* | hppa2.0-* | hppa2.0w-* \ + | hppa2.0n-* | hppa64-* \ + | alpha-* | alphaev[4-8]-* | alphaev56-* | alphapca5[67]-* \ + | alphaev6[78]-* \ + | we32k-* | cydra-* | ns16k-* | pn-* | np1-* | xps100-* \ + | clipper-* | orion-* \ + | sparclite-* | pdp10-* | pdp11-* | sh-* | powerpc-* | powerpcle-* \ + | sparc64-* | sparcv9-* | sparcv9b-* | sparc86x-* \ + | mips16-* | mips64-* | mipsel-* \ + | mips64el-* | mips64orion-* | mips64orionel-* \ + | mips64vr4100-* | mips64vr4100el-* | mips64vr4300-* | mips64vr4300el-* \ + | mipstx39-* | mipstx39el-* | mcore-* \ + | f30[01]-* | f700-* | s390-* | s390x-* | sv1-* | t3e-* \ + | [cjt]90-* \ + | m88110-* | m680[01234]0-* | m683?2-* | m68360-* | z8k-* | d10v-* \ + | thumb-* | v850-* | d30v-* | tic30-* | tic80-* | c30-* | fr30-* \ + | bs2000-* | tic54x-* | c54x-* | x86_64-* | pj-* | pjl-*) + ;; + # Recognize the various machine names and aliases which stand + # for a CPU type and a company and sometimes even an OS. + 386bsd) + basic_machine=i386-unknown + os=-bsd + ;; + 3b1 | 7300 | 7300-att | att-7300 | pc7300 | safari | unixpc) + basic_machine=m68000-att + ;; + 3b*) + basic_machine=we32k-att + ;; + a29khif) + basic_machine=a29k-amd + os=-udi + ;; + adobe68k) + basic_machine=m68010-adobe + os=-scout + ;; + alliant | fx80) + basic_machine=fx80-alliant + ;; + altos | altos3068) + basic_machine=m68k-altos + ;; + am29k) + basic_machine=a29k-none + os=-bsd + ;; + amdahl) + basic_machine=580-amdahl + os=-sysv + ;; + amiga | amiga-*) + basic_machine=m68k-unknown + ;; + amigaos | amigados) + basic_machine=m68k-unknown + os=-amigaos + ;; + amigaunix | amix) + basic_machine=m68k-unknown + os=-sysv4 + ;; + apollo68) + basic_machine=m68k-apollo + os=-sysv + ;; + apollo68bsd) + basic_machine=m68k-apollo + os=-bsd + ;; + aux) + basic_machine=m68k-apple + os=-aux + ;; + balance) + basic_machine=ns32k-sequent + os=-dynix + ;; + convex-c1) + basic_machine=c1-convex + os=-bsd + ;; + convex-c2) + basic_machine=c2-convex + os=-bsd + ;; + convex-c32) + basic_machine=c32-convex + os=-bsd + ;; + convex-c34) + basic_machine=c34-convex + os=-bsd + ;; + convex-c38) + basic_machine=c38-convex + os=-bsd + ;; + cray | ymp) + basic_machine=ymp-cray + os=-unicos + ;; + cray2) + basic_machine=cray2-cray + os=-unicos + ;; + [cjt]90) + basic_machine=${basic_machine}-cray + os=-unicos + ;; + crds | unos) + basic_machine=m68k-crds + ;; + cris | cris-* | etrax*) + basic_machine=cris-axis + ;; + da30 | da30-*) + basic_machine=m68k-da30 + ;; + decstation | decstation-3100 | pmax | pmax-* | pmin | dec3100 | decstatn) + basic_machine=mips-dec + ;; + delta | 3300 | motorola-3300 | motorola-delta \ + | 3300-motorola | delta-motorola) + basic_machine=m68k-motorola + ;; + delta88) + basic_machine=m88k-motorola + os=-sysv3 + ;; + dpx20 | dpx20-*) + basic_machine=rs6000-bull + os=-bosx + ;; + dpx2* | dpx2*-bull) + basic_machine=m68k-bull + os=-sysv3 + ;; + ebmon29k) + basic_machine=a29k-amd + os=-ebmon + ;; + elxsi) + basic_machine=elxsi-elxsi + os=-bsd + ;; + encore | umax | mmax) + basic_machine=ns32k-encore + ;; + es1800 | OSE68k | ose68k | ose | OSE) + basic_machine=m68k-ericsson + os=-ose + ;; + fx2800) + basic_machine=i860-alliant + ;; + genix) + basic_machine=ns32k-ns + ;; + gmicro) + basic_machine=tron-gmicro + os=-sysv + ;; + go32) + basic_machine=i386-pc + os=-go32 + ;; + h3050r* | hiux*) + basic_machine=hppa1.1-hitachi + os=-hiuxwe2 + ;; + h8300hms) + basic_machine=h8300-hitachi + os=-hms + ;; + h8300xray) + basic_machine=h8300-hitachi + os=-xray + ;; + h8500hms) + basic_machine=h8500-hitachi + os=-hms + ;; + harris) + basic_machine=m88k-harris + os=-sysv3 + ;; + hp300-*) + basic_machine=m68k-hp + ;; + hp300bsd) + basic_machine=m68k-hp + os=-bsd + ;; + hp300hpux) + basic_machine=m68k-hp + os=-hpux + ;; + hp3k9[0-9][0-9] | hp9[0-9][0-9]) + basic_machine=hppa1.0-hp + ;; + hp9k2[0-9][0-9] | hp9k31[0-9]) + basic_machine=m68000-hp + ;; + hp9k3[2-9][0-9]) + basic_machine=m68k-hp + ;; + hp9k6[0-9][0-9] | hp6[0-9][0-9]) + basic_machine=hppa1.0-hp + ;; + hp9k7[0-79][0-9] | hp7[0-79][0-9]) + basic_machine=hppa1.1-hp + ;; + hp9k78[0-9] | hp78[0-9]) + # FIXME: really hppa2.0-hp + basic_machine=hppa1.1-hp + ;; + hp9k8[67]1 | hp8[67]1 | hp9k80[24] | hp80[24] | hp9k8[78]9 | hp8[78]9 | hp9k893 | hp893) + # FIXME: really hppa2.0-hp + basic_machine=hppa1.1-hp + ;; + hp9k8[0-9][13679] | hp8[0-9][13679]) + basic_machine=hppa1.1-hp + ;; + hp9k8[0-9][0-9] | hp8[0-9][0-9]) + basic_machine=hppa1.0-hp + ;; + hppa-next) + os=-nextstep3 + ;; + hppaosf) + basic_machine=hppa1.1-hp + os=-osf + ;; + hppro) + basic_machine=hppa1.1-hp + os=-proelf + ;; + i370-ibm* | ibm*) + basic_machine=i370-ibm + ;; +# I'm not sure what "Sysv32" means. Should this be sysv3.2? + i*86v32) + basic_machine=`echo $1 | sed -e 's/86.*/86-pc/'` + os=-sysv32 + ;; + i*86v4*) + basic_machine=`echo $1 | sed -e 's/86.*/86-pc/'` + os=-sysv4 + ;; + i*86v) + basic_machine=`echo $1 | sed -e 's/86.*/86-pc/'` + os=-sysv + ;; + i*86sol2) + basic_machine=`echo $1 | sed -e 's/86.*/86-pc/'` + os=-solaris2 + ;; + i386mach) + basic_machine=i386-mach + os=-mach + ;; + i386-vsta | vsta) + basic_machine=i386-unknown + os=-vsta + ;; + iris | iris4d) + basic_machine=mips-sgi + case $os in + -irix*) + ;; + *) + os=-irix4 + ;; + esac + ;; + isi68 | isi) + basic_machine=m68k-isi + os=-sysv + ;; + m88k-omron*) + basic_machine=m88k-omron + ;; + magnum | m3230) + basic_machine=mips-mips + os=-sysv + ;; + merlin) + basic_machine=ns32k-utek + os=-sysv + ;; + mingw32) + basic_machine=i386-pc + os=-mingw32 + ;; + miniframe) + basic_machine=m68000-convergent + ;; + *mint | -mint[0-9]* | *MiNT | *MiNT[0-9]*) + basic_machine=m68k-atari + os=-mint + ;; + mipsel*-linux*) + basic_machine=mipsel-unknown + os=-linux-gnu + ;; + mips*-linux*) + basic_machine=mips-unknown + os=-linux-gnu + ;; + mips3*-*) + basic_machine=`echo $basic_machine | sed -e 's/mips3/mips64/'` + ;; + mips3*) + basic_machine=`echo $basic_machine | sed -e 's/mips3/mips64/'`-unknown + ;; + mmix*) + basic_machine=mmix-knuth + os=-mmixware + ;; + monitor) + basic_machine=m68k-rom68k + os=-coff + ;; + msdos) + basic_machine=i386-pc + os=-msdos + ;; + mvs) + basic_machine=i370-ibm + os=-mvs + ;; + ncr3000) + basic_machine=i486-ncr + os=-sysv4 + ;; + netbsd386) + basic_machine=i386-unknown + os=-netbsd + ;; + netwinder) + basic_machine=armv4l-rebel + os=-linux + ;; + news | news700 | news800 | news900) + basic_machine=m68k-sony + os=-newsos + ;; + news1000) + basic_machine=m68030-sony + os=-newsos + ;; + news-3600 | risc-news) + basic_machine=mips-sony + os=-newsos + ;; + necv70) + basic_machine=v70-nec + os=-sysv + ;; + next | m*-next ) + basic_machine=m68k-next + case $os in + -nextstep* ) + ;; + -ns2*) + os=-nextstep2 + ;; + *) + os=-nextstep3 + ;; + esac + ;; + nh3000) + basic_machine=m68k-harris + os=-cxux + ;; + nh[45]000) + basic_machine=m88k-harris + os=-cxux + ;; + nindy960) + basic_machine=i960-intel + os=-nindy + ;; + mon960) + basic_machine=i960-intel + os=-mon960 + ;; + nonstopux) + basic_machine=mips-compaq + os=-nonstopux + ;; + np1) + basic_machine=np1-gould + ;; + nsr-tandem) + basic_machine=nsr-tandem + ;; + op50n-* | op60c-*) + basic_machine=hppa1.1-oki + os=-proelf + ;; + OSE68000 | ose68000) + basic_machine=m68000-ericsson + os=-ose + ;; + os68k) + basic_machine=m68k-none + os=-os68k + ;; + pa-hitachi) + basic_machine=hppa1.1-hitachi + os=-hiuxwe2 + ;; + paragon) + basic_machine=i860-intel + os=-osf + ;; + pbd) + basic_machine=sparc-tti + ;; + pbb) + basic_machine=m68k-tti + ;; + pc532 | pc532-*) + basic_machine=ns32k-pc532 + ;; + pentium | p5 | k5 | k6 | nexgen) + basic_machine=i586-pc + ;; + pentiumpro | p6 | 6x86 | athlon) + basic_machine=i686-pc + ;; + pentiumii | pentium2) + basic_machine=i686-pc + ;; + pentium-* | p5-* | k5-* | k6-* | nexgen-*) + basic_machine=i586-`echo $basic_machine | sed 's/^[^-]*-//'` + ;; + pentiumpro-* | p6-* | 6x86-* | athlon-*) + basic_machine=i686-`echo $basic_machine | sed 's/^[^-]*-//'` + ;; + pentiumii-* | pentium2-*) + basic_machine=i686-`echo $basic_machine | sed 's/^[^-]*-//'` + ;; + pn) + basic_machine=pn-gould + ;; + power) basic_machine=power-ibm + ;; + ppc) basic_machine=powerpc-unknown + ;; + ppc-*) basic_machine=powerpc-`echo $basic_machine | sed 's/^[^-]*-//'` + ;; + ppcle | powerpclittle | ppc-le | powerpc-little) + basic_machine=powerpcle-unknown + ;; + ppcle-* | powerpclittle-*) + basic_machine=powerpcle-`echo $basic_machine | sed 's/^[^-]*-//'` + ;; + ps2) + basic_machine=i386-ibm + ;; + pw32) + basic_machine=i586-unknown + os=-pw32 + ;; + rom68k) + basic_machine=m68k-rom68k + os=-coff + ;; + rm[46]00) + basic_machine=mips-siemens + ;; + rtpc | rtpc-*) + basic_machine=romp-ibm + ;; + sa29200) + basic_machine=a29k-amd + os=-udi + ;; + sequent) + basic_machine=i386-sequent + ;; + sh) + basic_machine=sh-hitachi + os=-hms + ;; + sparclite-wrs) + basic_machine=sparclite-wrs + os=-vxworks + ;; + sps7) + basic_machine=m68k-bull + os=-sysv2 + ;; + spur) + basic_machine=spur-unknown + ;; + st2000) + basic_machine=m68k-tandem + ;; + stratus) + basic_machine=i860-stratus + os=-sysv4 + ;; + sun2) + basic_machine=m68000-sun + ;; + sun2os3) + basic_machine=m68000-sun + os=-sunos3 + ;; + sun2os4) + basic_machine=m68000-sun + os=-sunos4 + ;; + sun3os3) + basic_machine=m68k-sun + os=-sunos3 + ;; + sun3os4) + basic_machine=m68k-sun + os=-sunos4 + ;; + sun4os3) + basic_machine=sparc-sun + os=-sunos3 + ;; + sun4os4) + basic_machine=sparc-sun + os=-sunos4 + ;; + sun4sol2) + basic_machine=sparc-sun + os=-solaris2 + ;; + sun3 | sun3-*) + basic_machine=m68k-sun + ;; + sun4) + basic_machine=sparc-sun + ;; + sun386 | sun386i | roadrunner) + basic_machine=i386-sun + ;; + sv1) + basic_machine=sv1-cray + os=-unicos + ;; + symmetry) + basic_machine=i386-sequent + os=-dynix + ;; + t3e) + basic_machine=t3e-cray + os=-unicos + ;; + tic54x | c54x*) + basic_machine=tic54x-unknown + os=-coff + ;; + tx39) + basic_machine=mipstx39-unknown + ;; + tx39el) + basic_machine=mipstx39el-unknown + ;; + tower | tower-32) + basic_machine=m68k-ncr + ;; + udi29k) + basic_machine=a29k-amd + os=-udi + ;; + ultra3) + basic_machine=a29k-nyu + os=-sym1 + ;; + v810 | necv810) + basic_machine=v810-nec + os=-none + ;; + vaxv) + basic_machine=vax-dec + os=-sysv + ;; + vms) + basic_machine=vax-dec + os=-vms + ;; + vpp*|vx|vx-*) + basic_machine=f301-fujitsu + ;; + vxworks960) + basic_machine=i960-wrs + os=-vxworks + ;; + vxworks68) + basic_machine=m68k-wrs + os=-vxworks + ;; + vxworks29k) + basic_machine=a29k-wrs + os=-vxworks + ;; + w65*) + basic_machine=w65-wdc + os=-none + ;; + w89k-*) + basic_machine=hppa1.1-winbond + os=-proelf + ;; + xmp) + basic_machine=xmp-cray + os=-unicos + ;; + xps | xps100) + basic_machine=xps100-honeywell + ;; + z8k-*-coff) + basic_machine=z8k-unknown + os=-sim + ;; + none) + basic_machine=none-none + os=-none + ;; + +# Here we handle the default manufacturer of certain CPU types. It is in +# some cases the only manufacturer, in others, it is the most popular. + w89k) + basic_machine=hppa1.1-winbond + ;; + op50n) + basic_machine=hppa1.1-oki + ;; + op60c) + basic_machine=hppa1.1-oki + ;; + mips) + if [ x$os = x-linux-gnu ]; then + basic_machine=mips-unknown + else + basic_machine=mips-mips + fi + ;; + romp) + basic_machine=romp-ibm + ;; + rs6000) + basic_machine=rs6000-ibm + ;; + vax) + basic_machine=vax-dec + ;; + pdp10) + # there are many clones, so DEC is not a safe bet + basic_machine=pdp10-unknown + ;; + pdp11) + basic_machine=pdp11-dec + ;; + we32k) + basic_machine=we32k-att + ;; + sh3 | sh4) + basic_machine=sh-unknown + ;; + sparc | sparcv9 | sparcv9b) + basic_machine=sparc-sun + ;; + cydra) + basic_machine=cydra-cydrome + ;; + orion) + basic_machine=orion-highlevel + ;; + orion105) + basic_machine=clipper-highlevel + ;; + mac | mpw | mac-mpw) + basic_machine=m68k-apple + ;; + pmac | pmac-mpw) + basic_machine=powerpc-apple + ;; + c4x*) + basic_machine=c4x-none + os=-coff + ;; + *-unknown) + # Make sure to match an already-canonicalized machine name. + ;; + *) + echo Invalid configuration \`$1\': machine \`$basic_machine\' not recognized 1>&2 + exit 1 + ;; +esac + +# Here we canonicalize certain aliases for manufacturers. +case $basic_machine in + *-digital*) + basic_machine=`echo $basic_machine | sed 's/digital.*/dec/'` + ;; + *-commodore*) + basic_machine=`echo $basic_machine | sed 's/commodore.*/cbm/'` + ;; + *) + ;; +esac + +# Decode manufacturer-specific aliases for certain operating systems. + +if [ x"$os" != x"" ] +then +case $os in + # First match some system type aliases + # that might get confused with valid system types. + # -solaris* is a basic system type, with this one exception. + -solaris1 | -solaris1.*) + os=`echo $os | sed -e 's|solaris1|sunos4|'` + ;; + -solaris) + os=-solaris2 + ;; + -svr4*) + os=-sysv4 + ;; + -unixware*) + os=-sysv4.2uw + ;; + -gnu/linux*) + os=`echo $os | sed -e 's|gnu/linux|linux-gnu|'` + ;; + # First accept the basic system types. + # The portable systems comes first. + # Each alternative MUST END IN A *, to match a version number. + # -sysv* is not here because it comes later, after sysvr4. + -gnu* | -bsd* | -mach* | -minix* | -genix* | -ultrix* | -irix* \ + | -*vms* | -sco* | -esix* | -isc* | -aix* | -sunos | -sunos[34]*\ + | -hpux* | -unos* | -osf* | -luna* | -dgux* | -solaris* | -sym* \ + | -amigaos* | -amigados* | -msdos* | -newsos* | -unicos* | -aof* \ + | -aos* \ + | -nindy* | -vxsim* | -vxworks* | -ebmon* | -hms* | -mvs* \ + | -clix* | -riscos* | -uniplus* | -iris* | -rtu* | -xenix* \ + | -hiux* | -386bsd* | -netbsd* | -openbsd* | -freebsd* | -riscix* \ + | -lynxos* | -bosx* | -nextstep* | -cxux* | -aout* | -elf* | -oabi* \ + | -ptx* | -coff* | -ecoff* | -winnt* | -domain* | -vsta* \ + | -udi* | -eabi* | -lites* | -ieee* | -go32* | -aux* \ + | -cygwin* | -pe* | -psos* | -moss* | -proelf* | -rtems* \ + | -mingw32* | -linux-gnu* | -uxpv* | -beos* | -mpeix* | -udk* \ + | -interix* | -uwin* | -rhapsody* | -darwin* | -opened* \ + | -openstep* | -oskit* | -conix* | -pw32* | -nonstopux* \ + | -storm-chaos* | -tops10* | -tenex* | -tops20* | -its* | -os2*) + # Remember, each alternative MUST END IN *, to match a version number. + ;; + -qnx*) + case $basic_machine in + x86-* | i*86-*) + ;; + *) + os=-nto$os + ;; + esac + ;; + -nto*) + os=-nto-qnx + ;; + -sim | -es1800* | -hms* | -xray | -os68k* | -none* | -v88r* \ + | -windows* | -osx | -abug | -netware* | -os9* | -beos* \ + | -macos* | -mpw* | -magic* | -mmixware* | -mon960* | -lnews*) + ;; + -mac*) + os=`echo $os | sed -e 's|mac|macos|'` + ;; + -linux*) + os=`echo $os | sed -e 's|linux|linux-gnu|'` + ;; + -sunos5*) + os=`echo $os | sed -e 's|sunos5|solaris2|'` + ;; + -sunos6*) + os=`echo $os | sed -e 's|sunos6|solaris3|'` + ;; + -opened*) + os=-openedition + ;; + -wince*) + os=-wince + ;; + -osfrose*) + os=-osfrose + ;; + -osf*) + os=-osf + ;; + -utek*) + os=-bsd + ;; + -dynix*) + os=-bsd + ;; + -acis*) + os=-aos + ;; + -386bsd) + os=-bsd + ;; + -ctix* | -uts*) + os=-sysv + ;; + -ns2 ) + os=-nextstep2 + ;; + -nsk*) + os=-nsk + ;; + # Preserve the version number of sinix5. + -sinix5.*) + os=`echo $os | sed -e 's|sinix|sysv|'` + ;; + -sinix*) + os=-sysv4 + ;; + -triton*) + os=-sysv3 + ;; + -oss*) + os=-sysv3 + ;; + -svr4) + os=-sysv4 + ;; + -svr3) + os=-sysv3 + ;; + -sysvr4) + os=-sysv4 + ;; + # This must come after -sysvr4. + -sysv*) + ;; + -ose*) + os=-ose + ;; + -es1800*) + os=-ose + ;; + -xenix) + os=-xenix + ;; + -*mint | -mint[0-9]* | -*MiNT | -MiNT[0-9]*) + os=-mint + ;; + -none) + ;; + *) + # Get rid of the `-' at the beginning of $os. + os=`echo $os | sed 's/[^-]*-//'` + echo Invalid configuration \`$1\': system \`$os\' not recognized 1>&2 + exit 1 + ;; +esac +else + +# Here we handle the default operating systems that come with various machines. +# The value should be what the vendor currently ships out the door with their +# machine or put another way, the most popular os provided with the machine. + +# Note that if you're going to try to match "-MANUFACTURER" here (say, +# "-sun"), then you have to tell the case statement up towards the top +# that MANUFACTURER isn't an operating system. Otherwise, code above +# will signal an error saying that MANUFACTURER isn't an operating +# system, and we'll never get to this point. + +case $basic_machine in + *-acorn) + os=-riscix1.2 + ;; + arm*-rebel) + os=-linux + ;; + arm*-semi) + os=-aout + ;; + pdp10-*) + os=-tops20 + ;; + pdp11-*) + os=-none + ;; + *-dec | vax-*) + os=-ultrix4.2 + ;; + m68*-apollo) + os=-domain + ;; + i386-sun) + os=-sunos4.0.2 + ;; + m68000-sun) + os=-sunos3 + # This also exists in the configure program, but was not the + # default. + # os=-sunos4 + ;; + m68*-cisco) + os=-aout + ;; + mips*-cisco) + os=-elf + ;; + mips*-*) + os=-elf + ;; + *-tti) # must be before sparc entry or we get the wrong os. + os=-sysv3 + ;; + sparc-* | *-sun) + os=-sunos4.1.1 + ;; + *-be) + os=-beos + ;; + *-ibm) + os=-aix + ;; + *-wec) + os=-proelf + ;; + *-winbond) + os=-proelf + ;; + *-oki) + os=-proelf + ;; + *-hp) + os=-hpux + ;; + *-hitachi) + os=-hiux + ;; + i860-* | *-att | *-ncr | *-altos | *-motorola | *-convergent) + os=-sysv + ;; + *-cbm) + os=-amigaos + ;; + *-dg) + os=-dgux + ;; + *-dolphin) + os=-sysv3 + ;; + m68k-ccur) + os=-rtu + ;; + m88k-omron*) + os=-luna + ;; + *-next ) + os=-nextstep + ;; + *-sequent) + os=-ptx + ;; + *-crds) + os=-unos + ;; + *-ns) + os=-genix + ;; + i370-*) + os=-mvs + ;; + *-next) + os=-nextstep3 + ;; + *-gould) + os=-sysv + ;; + *-highlevel) + os=-bsd + ;; + *-encore) + os=-bsd + ;; + *-sgi) + os=-irix + ;; + *-siemens) + os=-sysv4 + ;; + *-masscomp) + os=-rtu + ;; + f30[01]-fujitsu | f700-fujitsu) + os=-uxpv + ;; + *-rom68k) + os=-coff + ;; + *-*bug) + os=-coff + ;; + *-apple) + os=-macos + ;; + *-atari*) + os=-mint + ;; + *) + os=-none + ;; +esac +fi + +# Here we handle the case where we know the os, and the CPU type, but not the +# manufacturer. We pick the logical manufacturer. +vendor=unknown +case $basic_machine in + *-unknown) + case $os in + -riscix*) + vendor=acorn + ;; + -sunos*) + vendor=sun + ;; + -aix*) + vendor=ibm + ;; + -beos*) + vendor=be + ;; + -hpux*) + vendor=hp + ;; + -mpeix*) + vendor=hp + ;; + -hiux*) + vendor=hitachi + ;; + -unos*) + vendor=crds + ;; + -dgux*) + vendor=dg + ;; + -luna*) + vendor=omron + ;; + -genix*) + vendor=ns + ;; + -mvs* | -opened*) + vendor=ibm + ;; + -ptx*) + vendor=sequent + ;; + -vxsim* | -vxworks*) + vendor=wrs + ;; + -aux*) + vendor=apple + ;; + -hms*) + vendor=hitachi + ;; + -mpw* | -macos*) + vendor=apple + ;; + -*mint | -mint[0-9]* | -*MiNT | -MiNT[0-9]*) + vendor=atari + ;; + esac + basic_machine=`echo $basic_machine | sed "s/unknown/$vendor/"` + ;; +esac + +echo $basic_machine$os +exit 0 + +# Local variables: +# eval: (add-hook 'write-file-hooks 'time-stamp) +# time-stamp-start: "timestamp='" +# time-stamp-format: "%:y-%02m-%02d" +# time-stamp-end: "'" +# End: diff --git a/config/install.sh b/config/install.sh new file mode 100755 index 0000000..e69de29 diff --git a/configure.ac b/configure.ac index 027dae0..97989d2 100644 --- a/configure.ac +++ b/configure.ac @@ -1,5 +1,6 @@ # Process this file with autoconf to produce a configure script. AC_INIT(RtAudio, 4.0, gary@music.mcgill.ca, rtaudio) +AC_CONFIG_AUX_DIR(config) AC_CONFIG_SRCDIR(RtAudio.cpp) AC_CONFIG_FILES([rtaudio-config Makefile tests/Makefile]) @@ -7,8 +8,7 @@ AC_CONFIG_FILES([rtaudio-config Makefile tests/Makefile]) AC_SUBST( GXX, ["no"] ) # Checks for programs. -AC_PROG_CC -AC_PROG_CXX(g++ CC c++ cxx) +AC_PROG_CXX AC_PROG_RANLIB AC_PATH_PROG(AR, ar, no) if [[ $AR = "no" ]] ; then @@ -19,66 +19,65 @@ fi AC_HEADER_STDC AC_CHECK_HEADERS(sys/ioctl.h unistd.h) -# Checks for typedefs, structures, and compiler characteristics. -AC_C_CONST - # Check for debug AC_MSG_CHECKING(whether to compile debug version) AC_ARG_ENABLE(debug, [ --enable-debug = enable various debug output], - [AC_SUBST( debug, [-D__RTAUDIO_DEBUG__] ) AC_SUBST( cflags, [-g] ) AC_SUBST( object_path, [Debug] ) AC_MSG_RESULT(yes)], - [AC_SUBST( debug, [] ) AC_SUBST( cflags, [-O2] ) AC_SUBST( object_path, [Release] ) AC_MSG_RESULT(no)]) + [AC_SUBST( cppflag, [-D__RTAUDIO_DEBUG__] ) AC_SUBST( cxxflag, [-g] ) AC_SUBST( object_path, [Debug] ) AC_MSG_RESULT(yes)], + [AC_SUBST( cppflag, [] ) AC_SUBST( cxxflag, [-O2] ) AC_SUBST( object_path, [Release] ) AC_MSG_RESULT(no)]) + # Checks for functions -AC_CHECK_FUNC(gettimeofday, [CFLAGS=$CFLAGS" -DHAVE_GETTIMEOFDAY"], ) +AC_CHECK_FUNC(gettimeofday, [cppflag="$cppflag -DHAVE_GETTIMEOFDAY"], ) + +# For -I and -D flags +CPPFLAGS="$CPPFLAGS $cppflag" + +# For debugging and optimization ... overwrite default because it has both -g and -O2 +#CXXFLAGS="$CXXFLAGS $cxxflag" +CXXFLAGS="$cxxflag" # Check compiler and use -Wall if gnu. if [test $GXX = "yes" ;] then - AC_SUBST( warn, [-Wall] ) + AC_SUBST( cxxflag, [-Wall] ) fi -CFLAGS="$CFLAGS $cflags" +CXXFLAGS="$CXXFLAGS $cxxflag" # Checks for package options and external software +AC_SUBST( api, [""] ) AC_CANONICAL_HOST AC_MSG_CHECKING(for audio API) case $host in *-*-netbsd*) - AC_SUBST( sound_api, [-D__LINUX_OSS__] ) AC_MSG_RESULT(using OSS) - AC_SUBST( audio_apis, [-D__LINUX_OSS__] ) - CFLAGS=$CFLAGS" -lossaudio" + api="$api -D__LINUX_OSS__" + LIBS="$LIBS -lossaudio" AC_CHECK_LIB(pthread, pthread_create, , AC_MSG_ERROR(RtAudio requires the pthread library!)) ;; *-*-linux*) - AC_SUBST( sound_api, [_NO_API_] ) - AC_ARG_WITH(jack, [ --with-jack = choose JACK server support (mac and linux only)], [AC_SUBST( sound_api, [-D__UNIX_JACK__] ) AC_MSG_RESULT(using JACK)], ) - if [test $sound_api = -D__UNIX_JACK__;] then - TEMP_LIBS=$LIBS - AC_CHECK_LIB(jack, jack_client_new, , AC_MSG_ERROR(JACK support requires the jack library!)) - AC_CHECK_LIB(asound, snd_pcm_open, , AC_MSG_ERROR(Jack support also requires the asound library!)) - LIBS="`pkg-config --CFLAGS --libs jack` $TEMP_LIBS -lasound" - audio_apis="-D__UNIX_JACK__" - fi + AC_ARG_WITH(jack, [ --with-jack = choose JACK server support (mac and linux only)], [ + api="$api -D__UNIX_JACK__" + AC_MSG_RESULT(using JACK) + AC_CHECK_LIB(jack, jack_client_open, , AC_MSG_ERROR(JACK support requires the jack library!)) + AC_CHECK_LIB(asound, snd_pcm_open, , AC_MSG_ERROR(Jack support also requires the asound library!))], ) # Look for ALSA flag - AC_ARG_WITH(alsa, [ --with-alsa = choose native ALSA API support (linux only)], [AC_SUBST( sound_api, [-D__LINUX_ALSA__] ) AC_MSG_RESULT(using ALSA)], ) - if [test $sound_api = -D__LINUX_ALSA__;] then - AC_CHECK_LIB(asound, snd_pcm_open, , AC_MSG_ERROR(ALSA support requires the asound library!)) - audio_apis="-D__LINUX_ALSA__ $audio_apis" - fi + AC_ARG_WITH(alsa, [ --with-alsa = choose native ALSA API support (linux only)], [ + api="$api -D__LINUX_ALSA__" + AC_MSG_RESULT(using ALSA) + AC_CHECK_LIB(asound, snd_pcm_open, , AC_MSG_ERROR(ALSA support requires the asound library!))], ) # Look for OSS flag - AC_ARG_WITH(oss, [ --with-oss = choose OSS API support (linux only)], [AC_SUBST( sound_api, [-D__LINUX_OSS__] ) AC_MSG_RESULT(using OSS)], ) - if test $sound_api = -D__LINUX_OSS__; then - audio_apis="-D__LINUX_OSS__ $audio_apis" - fi + AC_ARG_WITH(oss, [ --with-oss = choose OSS API support (linux only)], [ + api="$api -D__LINUX_OSS__" + AC_MSG_RESULT(using OSS)], ) # If no audio api flags specified, use ALSA - if [test $sound_api = _NO_API_;] then + if [test "$api" == "";] then AC_MSG_RESULT(using ALSA) - AC_SUBST( audio_apis, [-D__LINUX_ALSA__] ) + AC_SUBST( api, [-D__LINUX_ALSA__] ) AC_CHECK_LIB(asound, snd_pcm_open, , AC_MSG_ERROR(ALSA support requires the asound library!)) fi @@ -86,54 +85,47 @@ case $host in ;; *-apple*) - AC_SUBST( sound_api, [_NO_API_] ) - AC_ARG_WITH(jack, [ --with-jack = choose JACK server support (unix only)], [AC_SUBST( sound_api, [-D__UNIX_JACK__] ) AC_MSG_RESULT(using JACK)], ) - if [test $sound_api = -D__UNIX_JACK__;] then - AC_CHECK_LIB(jack, jack_client_new, , AC_MSG_ERROR(JACK support requires the jack library!)) - audio_apis="-D__UNIX_JACK__" - fi + AC_ARG_WITH(jack, [ --with-jack = choose JACK server support (unix only)], [ + api="$api -D__UNIX_JACK__" + AC_MSG_RESULT(using JACK) + AC_CHECK_LIB(jack, jack_client_new, , AC_MSG_ERROR(JACK support requires the jack library!))], ) # Look for Core flag - AC_ARG_WITH(core, [ --with-core = choose CoreAudio API support (mac only)], [AC_SUBST( sound_api, [-D__MACOSX_CORE__] ) AC_MSG_RESULT(using CoreAudio)], ) - if test $sound_api = -D__MACOSX_CORE__; then + AC_ARG_WITH(core, [ --with-core = choose CoreAudio API support (mac only)], [ + api="$api -D__MACOSX_CORE__" + AC_MSG_RESULT(using CoreAudio) AC_CHECK_HEADER(CoreAudio/CoreAudio.h, [], [AC_MSG_ERROR(CoreAudio header files not found!)] ) - AC_SUBST( frameworks, ["-framework CoreAudio -framework CoreFoundation"] ) - audio_apis="-D__MACOSX_CORE__ $audio_apis" - fi + LIBS="$LIBS -framework CoreAudio -framework CoreFoundation" ], ) # If no audio api flags specified, use CoreAudio - if [test $sound_api = _NO_API_;] then - AC_SUBST( sound_api, [-D__MACOSX_CORE__] ) + if [test "$api" == ""; ] then + AC_SUBST( api, [-D__MACOSX_CORE__] ) AC_MSG_RESULT(using CoreAudio) AC_CHECK_HEADER(CoreAudio/CoreAudio.h, - [AC_SUBST( audio_apis, [-D__MACOSX_CORE__] )], + [], [AC_MSG_ERROR(CoreAudio header files not found!)] ) - AC_SUBST( frameworks, ["-framework CoreAudio -framework CoreFoundation"] ) + AC_SUBST( LIBS, ["-framework CoreAudio -framework CoreFoundation"] ) fi AC_CHECK_LIB(pthread, pthread_create, , AC_MSG_ERROR(RtAudio requires the pthread library!)) ;; *-mingw32*) - AC_SUBST( sound_api, [_NO_API_] ) - AC_ARG_WITH(asio, [ --with-asio = choose ASIO API support (windoze only)], [AC_SUBST( sound_api, [-D__WINDOWS_ASIO__] ) AC_MSG_RESULT(using ASIO)], ) - if [test $sound_api = -D__WINDOWS_ASIO__;] then - audio_apis="-D__WINDOWS_ASIO__" - AC_SUBST( objects, ["asio.o asiodrivers.o asiolist.o iasiothiscallresolver.o"] ) - fi + AC_ARG_WITH(asio, [ --with-asio = choose ASIO API support (windoze only)], [ + api="$api -D__WINDOWS_ASIO__" + AC_MSG_RESULT(using ASIO) + AC_SUBST( objects, ["asio.o asiodrivers.o asiolist.o iasiothiscallresolver.o"] ) ], ) # Look for DirectSound flag - AC_ARG_WITH(ds, [ --with-ds = choose DirectSound API support (windoze only)], [AC_SUBST( sound_api, [-D__WINDOWS_DS__] ) AC_MSG_RESULT(using DirectSound)], ) - if test $sound_api = -D__WINDOWS_DS__; then - audio_apis="-D__WINDOWS_DS__ $audio_apis" - LIBS="-ldsound -lwinmm $LIBS" - fi + AC_ARG_WITH(ds, [ --with-ds = choose DirectSound API support (windoze only)], [ + api="$api -D__WINDOWS_DS__" + AC_MSG_RESULT(using DirectSound) + LIBS="-ldsound -lwinmm $LIBS" ], ) # If no audio api flags specified, use DirectSound - if [test $sound_api = _NO_API_;] then - AC_SUBST( sound_api, [-D__WINDOWS_DS__] ) + if [test "$api" == "";] then + AC_SUBST( api, [-D__WINDOWS_DS__] ) AC_MSG_RESULT(using DirectSound) - audio_apis="-D__WINDOWS_DS__" LIBS="-ldsound -lwinmm $LIBS" fi @@ -146,8 +138,7 @@ case $host in ;; esac -# Checks for library functions. -AC_PROG_GCC_TRADITIONAL +CPPFLAGS="$CPPFLAGS $api" AC_OUTPUT diff --git a/doc/doxygen/tutorial.txt b/doc/doxygen/tutorial.txt index 906afe3..c8fd9cf 100644 --- a/doc/doxygen/tutorial.txt +++ b/doc/doxygen/tutorial.txt @@ -32,7 +32,7 @@ Devices are now re-enumerated every time the RtAudio::getDeviceCount(), RtAudio: \section download Download -Latest Release (24 January 2008): Version 4.0.4 +Latest Release (24 January 2008): Version 4.0.4 \section documentation Documentation Links diff --git a/install.sh b/install.sh deleted file mode 100755 index e69de29..0000000 diff --git a/readme b/readme index 8228cd3..7c2c7b9 100644 --- a/readme +++ b/readme @@ -11,7 +11,7 @@ tests/Windows: Visual C++ .net test program workspace and projects OVERVIEW: -RtAudio is a set of C++ classes that provide a common API (Application Programming Interface) for realtime audio input/output across Linux (native ALSA, JACK, and OSS), Macintosh OS X, SGI, and Windows (DirectSound and ASIO) operating systems. RtAudio significantly simplifies the process of interacting with computer audio hardware. It was designed with the following objectives: +RtAudio is a set of C++ classes that provides a common API (Application Programming Interface) for realtime audio input/output across Linux (native ALSA, JACK, and OSS), Macintosh OS X, SGI, and Windows (DirectSound and ASIO) operating systems. RtAudio significantly simplifies the process of interacting with computer audio hardware. It was designed with the following objectives: - object-oriented C++ design - simple, common API across all supported platforms diff --git a/rtaudio-config.in b/rtaudio-config.in index c689a98..2422b60 100644 --- a/rtaudio-config.in +++ b/rtaudio-config.in @@ -1,16 +1,19 @@ #! /bin/sh if (test "x$#" != "x1") ; then - echo "Usage: $0 [--libs | --cxxflags]" + echo "Usage: $0 [--libs | --cxxflags | --cppflags]" exit; fi -LIBRARY="@LIBS@ @frameworks@" -CFLAGS="@audio_apis@" +LIBRARY="@LIBS@" +CXXFLAGS="@CXXFLAGS@" +CPPFLAGS="@CPPFLAGS@" if (test "x$1" == "x--libs") ; then echo "$LIBRARY" -elif (test "x$1" == "x--CFLAGS") ; then - echo "$CFLAGS" +elif (test "x$1" == "x--cxxflags") ; then + echo "$CXXFLAGS" +elif (test "x$1" == "x--cppflags") ; then + echo "$CPPFLAGS" else echo "Unknown option: $1" fi diff --git a/tests/Makefile.in b/tests/Makefile.in index 059b5ce..4e93823 100644 --- a/tests/Makefile.in +++ b/tests/Makefile.in @@ -11,12 +11,10 @@ vpath %.o $(OBJECT_PATH) OBJECTS = RtAudio.o @objects@ CC = @CXX@ -DEFS = @debug@ -DEFS += @audio_apis@ -CFLAGS = @CFLAGS@ -CFLAGS += @warn@ -I$(INCLUDE) -I../include +DEFS = @CPPFLAGS@ +CFLAGS = @CXXFLAGS@ +CFLAGS += -I$(INCLUDE) -I../include LIBRARY = @LIBS@ -LIBRARY += @frameworks@ %.o : $(SRC_PATH)/%.cpp $(CC) $(CFLAGS) $(DEFS) -c $(<) -o $(OBJECT_PATH)/$@ @@ -44,11 +42,14 @@ duplex : duplex.cpp $(OBJECTS) testall : testall.cpp $(OBJECTS) $(CC) $(CFLAGS) $(DEFS) -o testall testall.cpp $(OBJECT_PATH)/*.o $(LIBRARY) - clean : -rm $(OBJECT_PATH)/*.o -rm $(PROGRAMS) -rm -f *.raw *~ *.exe + -rm -fR *.dSYM + +distclean: clean + -rm Makefile strip : strip $(PROGRAMS) diff --git a/tests/testall.cpp b/tests/testall.cpp index 2eeb330..1f3ea26 100644 --- a/tests/testall.cpp +++ b/tests/testall.cpp @@ -1,7 +1,7 @@ /******************************************/ /* testall.cpp - by Gary P. Scavone, 2007 + by Gary P. Scavone, 2007-2008 This program will make a variety of calls to extensively test RtAudio functionality. @@ -17,11 +17,13 @@ void usage( void ) { // Error function in case of incorrect command-line // argument specifications - std::cout << "\nuseage: testall N fs \n"; + std::cout << "\nuseage: testall N fs \n"; std::cout << " where N = number of channels,\n"; std::cout << " fs = the sample rate,\n"; - std::cout << " device = optional device to use (default = 0),\n"; - std::cout << " and channelOffset = an optional channel offset on the device (default = 0).\n\n"; + std::cout << " iDevice = optional input device to use (default = 0),\n"; + std::cout << " oDevice = optional output device to use (default = 0),\n"; + std::cout << " iChannelOffset = an optional input channel offset (default = 0),\n"; + std::cout << " and oChannelOffset = optional output channel offset (default = 0).\n\n"; exit( 0 ); } @@ -89,11 +91,11 @@ int inout( void *outputBuffer, void *inputBuffer, unsigned int nBufferFrames, int main( int argc, char *argv[] ) { - unsigned int bufferFrames, fs, device = 0, offset = 0; + unsigned int bufferFrames, fs, oDevice = 0, iDevice = 0, iOffset = 0, oOffset = 0; char input; // minimal command-line checking - if (argc < 3 || argc > 5 ) usage(); + if (argc < 3 || argc > 7 ) usage(); RtAudio dac; if ( dac.getDeviceCount() < 1 ) { @@ -104,9 +106,13 @@ int main( int argc, char *argv[] ) channels = (unsigned int) atoi( argv[1] ); fs = (unsigned int) atoi( argv[2] ); if ( argc > 3 ) - device = (unsigned int) atoi( argv[3] ); + iDevice = (unsigned int) atoi( argv[3] ); if ( argc > 4 ) - offset = (unsigned int) atoi( argv[4] ); + oDevice = (unsigned int) atoi(argv[4]); + if ( argc > 5 ) + iOffset = (unsigned int) atoi(argv[5]); + if ( argc > 6 ) + oOffset = (unsigned int) atoi(argv[6]); double *data = (double *) calloc( channels, sizeof( double ) ); @@ -116,9 +122,9 @@ int main( int argc, char *argv[] ) // Set our stream parameters for output only. bufferFrames = 256; RtAudio::StreamParameters oParams, iParams; - oParams.deviceId = device; + oParams.deviceId = oDevice; oParams.nChannels = channels; - oParams.firstChannel = offset; + oParams.firstChannel = oOffset; RtAudio::StreamOptions options; options.flags = RTAUDIO_HOG_DEVICE; @@ -181,9 +187,9 @@ int main( int argc, char *argv[] ) // Now open a duplex stream. unsigned int bufferBytes; - iParams.deviceId = device; + iParams.deviceId = iDevice; iParams.nChannels = channels; - iParams.firstChannel = offset; + iParams.firstChannel = iOffset; options.flags = RTAUDIO_NONINTERLEAVED; try { dac.openStream( &oParams, &iParams, RTAUDIO_SINT32, fs, &bufferFrames, &inout, (void *)&bufferBytes, &options ); -- cgit v1.2.3 From 287e68ea212610c225613876da4e643d43fc2aba Mon Sep 17 00:00:00 2001 From: Gary Scavone Date: Fri, 2 Jan 2009 15:59:43 +0000 Subject: Updates to OS-X for multi-stream support (GS). --- RtAudio.cpp | 10505 ++++++++++++++++++++++++++-------------------------- RtAudio.h | 2 +- tests/playraw.cpp | 4 +- tests/playsaw.cpp | 6 +- tests/record.cpp | 4 +- 5 files changed, 5311 insertions(+), 5210 deletions(-) diff --git a/RtAudio.cpp b/RtAudio.cpp index 05e3afa..9b186cc 100644 --- a/RtAudio.cpp +++ b/RtAudio.cpp @@ -4,7 +4,7 @@ RtAudio provides a common API (Application Programming Interface) for realtime audio input/output across Linux (native ALSA, Jack, - and OSS), SGI, Macintosh OS X (CoreAudio and Jack), and Windows + and OSS), Macintosh OS X (CoreAudio and Jack), and Windows (DirectSound and ASIO) operating systems. RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/ @@ -401,7 +401,8 @@ unsigned int RtApi :: getStreamSampleRate( void ) struct CoreHandle { AudioDeviceID id[2]; // device ids AudioDeviceIOProcID procId[2]; - UInt32 iStream[2]; // device stream index (first for mono mode) + UInt32 iStream[2]; // device stream index (or first if using multiple) + UInt32 nStreams[2]; // number of streams to use bool xrun[2]; char *deviceBuffer; pthread_cond_t condition; @@ -409,7 +410,7 @@ struct CoreHandle { bool internalDrain; // Indicates if stop is initiated from callback or not. CoreHandle() - :deviceBuffer(0), drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } + :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } }; RtApiCore :: RtApiCore() @@ -813,69 +814,73 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne return FAILURE; } - // Search for a stream that contains the desired number of + // Search for one or more streams that contain the desired number of // channels. CoreAudio devices can have an arbitrary number of // streams and each stream can have an arbitrary number of channels. // For each stream, a single buffer of interleaved samples is - // provided. RtAudio currently only supports the use of one stream - // of interleaved data or multiple consecutive single-channel - // streams. Thus, our search below is limited to these two - // contexts. - unsigned int streamChannels = 0, nStreams = 0; - UInt32 iChannel = 0, iStream = 0; - unsigned int offsetCounter = firstChannel; - stream_.deviceInterleaved[mode] = true; - nStreams = bufferList->mNumberBuffers; + // provided. RtAudio prefers the use of one stream of interleaved + // data or multiple consecutive single-channel streams. However, we + // now support multiple consecutive multi-channel streams of + // interleaved data as well. + UInt32 iStream, offsetCounter = firstChannel; + UInt32 nStreams = bufferList->mNumberBuffers; + bool monoMode = false; bool foundStream = false; + // First check that the device supports the requested number of + // channels. + UInt32 deviceChannels = 0; + for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; + + if ( deviceChannels < ( channels + firstChannel ) ) { + free( bufferList ); + errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Look for a single stream meeting our needs. + UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0; for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; if ( streamChannels >= channels + offsetCounter ) { - iChannel += offsetCounter; + firstStream = iStream; + channelOffset = offsetCounter; foundStream = true; break; } if ( streamChannels > offsetCounter ) break; offsetCounter -= streamChannels; - iChannel += streamChannels; } - // If we didn't find a single stream above, see if we can meet - // the channel specification in mono mode (i.e. using separate - // non-interleaved buffers). This can only work if there are N - // consecutive one-channel streams, where N is the number of - // desired channels (+ channel offset). + // If we didn't find a single stream above, then we should be able + // to meet the channel specification with multiple streams. if ( foundStream == false ) { - unsigned int counter = 0; + monoMode = true; offsetCounter = firstChannel; - iChannel = 0; for ( iStream=0; iStreammBuffers[iStream].mNumberChannels; - if ( offsetCounter ) { - if ( streamChannels > offsetCounter ) break; - offsetCounter -= streamChannels; - } - else if ( streamChannels == 1 ) - counter++; - else - counter = 0; - if ( counter == channels ) { - iStream -= channels - 1; - iChannel -= channels - 1; - stream_.deviceInterleaved[mode] = false; - foundStream = true; - break; - } - iChannel += streamChannels; + if ( streamChannels > offsetCounter ) break; + offsetCounter -= streamChannels; + } + + firstStream = iStream; + channelOffset = offsetCounter; + Int32 channelCounter = channels + offsetCounter - streamChannels; + + if ( streamChannels > 1 ) monoMode = false; + while ( channelCounter > 0 ) { + streamChannels = bufferList->mBuffers[++iStream].mNumberChannels; + if ( streamChannels > 1 ) monoMode = false; + channelCounter -= streamChannels; + streamCount++; } } + free( bufferList ); - if ( foundStream == false ) { - errorStream_ << "RtApiCore::probeDeviceOpen: unable to find OS-X stream on device (" << device << ") for requested channels."; - errorText_ = errorStream_.str(); - return FAILURE; - } + std::cout << "deviceStreams = " << nStreams << ", firstStream = " << firstStream << ", streamCount = " << streamCount << ", channelOffset = " << channelOffset << std::endl; // Determine the buffer size. AudioValueRange bufferRange; @@ -893,8 +898,8 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum; if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum; - // Set the buffer size. For mono mode, I'm assuming we only need to - // make this setting for the master channel. + // Set the buffer size. For multiple streams, I'm assuming we only + // need to make this setting for the master channel. UInt32 theSize = (UInt32) *bufferSize; dataSize = sizeof( UInt32 ); result = AudioDeviceSetProperty( id, NULL, 0, isInput, @@ -919,8 +924,8 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne stream_.bufferSize = *bufferSize; stream_.nBuffers = 1; - // Get the stream ID(s) so we can set the stream format. In mono - // mode, we'll have to do this for each stream (channel). + // Get the stream ID(s) so we can set the stream format. We'll have + // to do this for each stream. AudioStreamID streamIDs[ nStreams ]; dataSize = nStreams * sizeof( AudioStreamID ); result = AudioDeviceGetProperty( id, 0, isInput, @@ -936,13 +941,11 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne // device and change that if necessary. AudioStreamBasicDescription description; dataSize = sizeof( AudioStreamBasicDescription ); - if ( stream_.deviceInterleaved[mode] ) nStreams = 1; - else nStreams = channels; bool updateFormat; - for ( unsigned int i=0; iflags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; else stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + if ( monoMode == true ) stream_.deviceInterleaved[mode] = false; // Set flags for buffer conversion. stream_.doConvertBuffer[mode] = false; @@ -1086,10 +1089,16 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne stream_.doConvertBuffer[mode] = true; if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) + if ( streamCount == 1 ) { + if ( stream_.nUserChannels[mode] > 1 && + stream_.userInterleaved != stream_.deviceInterleaved[mode] ) + stream_.doConvertBuffer[mode] = true; + } + else if ( monoMode && stream_.userInterleaved ) stream_.doConvertBuffer[mode] = true; + std::cout << "doConvert = " << stream_.doConvertBuffer[mode] << ", userInterleaved = " << stream_.userInterleaved << ", deviceInterleaved = " << stream_.deviceInterleaved[mode] << std::endl; + // Allocate our CoreHandle structure for the stream. CoreHandle *handle = 0; if ( stream_.apiHandle == 0 ) { @@ -1109,7 +1118,8 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne } else handle = (CoreHandle *) stream_.apiHandle; - handle->iStream[mode] = iStream; + handle->iStream[mode] = firstStream; + handle->nStreams[mode] = streamCount; handle->id[mode] = id; // Allocate necessary internal buffers. @@ -1122,9 +1132,9 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne } // If possible, we will make use of the CoreAudio stream buffers as - // "device buffers". However, we can't do this if the device - // buffers are non-interleaved ("mono" mode). - if ( !stream_.deviceInterleaved[mode] && stream_.doConvertBuffer[mode] ) { + // "device buffers". However, we can't do this if using multiple + // streams. + if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) { bool makeBuffer = true; bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); @@ -1143,13 +1153,6 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory."; goto error; } - - // Save a pointer to our own device buffer in the CoreHandle - // structure because we may need to use the stream_.deviceBuffer - // variable to point to the CoreAudio buffer before buffer - // conversion (if we have a duplex stream with two different - // conversion schemes). - handle->deviceBuffer = stream_.deviceBuffer; } } @@ -1158,23 +1161,10 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne stream_.state = STREAM_STOPPED; stream_.callbackInfo.object = (void *) this; - // Setup the buffer conversion information structure. We override - // the channel offset value and perform our own setting for that - // here. + // Setup the buffer conversion information structure. if ( stream_.doConvertBuffer[mode] ) { - setConvertInfo( mode, 0 ); - - // Add channel offset for interleaved channels. - if ( firstChannel > 0 && stream_.deviceInterleaved[mode] ) { - if ( mode == OUTPUT ) { - for ( int k=0; k 1 ) setConvertInfo( mode, 0 ); + else setConvertInfo( mode, channelOffset ); } if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device ) @@ -1265,8 +1255,8 @@ void RtApiCore :: closeStream( void ) } } - if ( handle->deviceBuffer ) { - free( handle->deviceBuffer ); + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); stream_.deviceBuffer = 0; } @@ -1442,48 +1432,96 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, if ( handle->drainCounter > 1 ) { // write zeros to the output stream - if ( stream_.deviceInterleaved[0] ) { + if ( handle->nStreams[0] == 1 ) { memset( outBufferList->mBuffers[handle->iStream[0]].mData, 0, outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); } - else { - for ( unsigned int i=0; inStreams[0]; i++ ) { memset( outBufferList->mBuffers[handle->iStream[0]+i].mData, 0, outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize ); } } } - else if ( stream_.doConvertBuffer[0] ) { - - if ( stream_.deviceInterleaved[0] ) - stream_.deviceBuffer = (char *) outBufferList->mBuffers[handle->iStream[0]].mData; - else - stream_.deviceBuffer = handle->deviceBuffer; - - convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - - if ( !stream_.deviceInterleaved[0] ) { - UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize; - for ( unsigned int i=0; imBuffers[handle->iStream[0]+i].mData, - &stream_.deviceBuffer[i*bufferBytes], bufferBytes ); - } + else if ( handle->nStreams[0] == 1 ) { + if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer + convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData, + stream_.userBuffer[0], stream_.convertInfo[0] ); } - - } - else { - if ( stream_.deviceInterleaved[0] ) { + else { // copy from user buffer memcpy( outBufferList->mBuffers[handle->iStream[0]].mData, stream_.userBuffer[0], outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); } - else { + } + else { // fill multiple streams + Float32 *inBuffer = (Float32 *) stream_.userBuffer[0]; + if ( stream_.doConvertBuffer[0] ) { + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + inBuffer = (Float32 *) stream_.deviceBuffer; + } + + if ( stream_.deviceInterleaved[0] == false ) { // mono mode UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize; - for ( unsigned int i=0; imBuffers[handle->iStream[0]+i].mData, - &stream_.userBuffer[0][i*bufferBytes], bufferBytes ); + &inBuffer[i*bufferBytes], bufferBytes ); + } + } + else { // fill multiple multi-channel streams with interleaved data + UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset; + Float32 *out, *in; + + bool inInterleaved = ( stream_.userInterleaved ) ? true : false; + UInt32 inChannels = stream_.nUserChannels[0]; + if ( stream_.doConvertBuffer[0] ) { + inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 + inChannels = stream_.nDeviceChannels[0]; + } + + if ( inInterleaved ) inOffset = 1; + else inOffset = stream_.bufferSize; + + channelsLeft = inChannels; + for ( unsigned int i=0; inStreams[0]; i++ ) { + in = inBuffer; + out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData; + streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels; + + outJump = 0; + // Account for possible channel offset in first stream + if ( i == 0 && stream_.channelOffset[0] > 0 ) { + streamChannels -= stream_.channelOffset[0]; + outJump = stream_.channelOffset[0]; + out += outJump; + } + + // Account for possible unfilled channels at end of the last stream + if ( streamChannels > channelsLeft ) { + outJump = streamChannels - channelsLeft; + streamChannels = channelsLeft; + } + + // Determine input buffer offsets and skips + if ( inInterleaved ) { + inJump = inChannels; + in += inChannels - channelsLeft; + } + else { + inJump = 1; + in += (inChannels - channelsLeft) * inOffset; + } + + for ( unsigned int i=0; iid[1]; if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) { - if ( stream_.doConvertBuffer[1] ) { + if ( handle->nStreams[1] == 1 ) { + if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer + convertBuffer( stream_.userBuffer[1], + (char *) inBufferList->mBuffers[handle->iStream[1]].mData, + stream_.convertInfo[1] ); + } + else { // copy to user buffer + memcpy( stream_.userBuffer[1], + inBufferList->mBuffers[handle->iStream[1]].mData, + inBufferList->mBuffers[handle->iStream[1]].mDataByteSize ); + } + } + else { // read from multiple streams + Float32 *outBuffer = (Float32 *) stream_.userBuffer[1]; + if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer; - if ( stream_.deviceInterleaved[1] ) - stream_.deviceBuffer = (char *) inBufferList->mBuffers[handle->iStream[1]].mData; - else { - stream_.deviceBuffer = (char *) handle->deviceBuffer; + if ( stream_.deviceInterleaved[1] == false ) { // mono mode UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize; - for ( unsigned int i=0; imBuffers[handle->iStream[1]+i].mData, bufferBytes ); } } + else { // read from multiple multi-channel streams + UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset; + Float32 *out, *in; + + bool outInterleaved = ( stream_.userInterleaved ) ? true : false; + UInt32 outChannels = stream_.nUserChannels[1]; + if ( stream_.doConvertBuffer[1] ) { + outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 + outChannels = stream_.nDeviceChannels[1]; + } - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + if ( outInterleaved ) outOffset = 1; + else outOffset = stream_.bufferSize; + + channelsLeft = outChannels; + for ( unsigned int i=0; inStreams[1]; i++ ) { + out = outBuffer; + in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData; + streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels; + + inJump = 0; + // Account for possible channel offset in first stream + if ( i == 0 && stream_.channelOffset[1] > 0 ) { + streamChannels -= stream_.channelOffset[1]; + inJump = stream_.channelOffset[1]; + in += inJump; + } - } - else { - memcpy( stream_.userBuffer[1], - inBufferList->mBuffers[handle->iStream[1]].mData, - inBufferList->mBuffers[handle->iStream[1]].mDataByteSize ); + // Account for possible unread channels at end of the last stream + if ( streamChannels > channelsLeft ) { + inJump = streamChannels - channelsLeft; + streamChannels = channelsLeft; + } + + // Determine output buffer offsets and skips + if ( outInterleaved ) { + outJump = outChannels; + out += outChannels - channelsLeft; + } + else { + outJump = 1; + out += (outChannels - channelsLeft) * outOffset; + } + + for ( unsigned int i=0; i #include -// A structure to hold various information related to the Jack API -// implementation. -struct JackHandle { - jack_client_t *client; - jack_port_t **ports[2]; - std::string deviceName[2]; - bool xrun[2]; - pthread_cond_t condition; - int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - - JackHandle() - :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; } -}; + // A structure to hold various information related to the Jack API + // implementation. + struct JackHandle { + jack_client_t *client; + jack_port_t **ports[2]; + std::string deviceName[2]; + bool xrun[2]; + pthread_cond_t condition; + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + + JackHandle() + :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; } + }; -RtApiJack :: RtApiJack() -{ - // Nothing to do here. -} + RtApiJack :: RtApiJack() + { + // Nothing to do here. + } -RtApiJack :: ~RtApiJack() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} + RtApiJack :: ~RtApiJack() + { + if ( stream_.state != STREAM_CLOSED ) closeStream(); + } -unsigned int RtApiJack :: getDeviceCount( void ) -{ - // See if we can become a jack client. - jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption; - jack_status_t *status = NULL; - jack_client_t *client = jack_client_open( "RtApiJackCount", options, status ); - if ( client == 0 ) return 0; - - const char **ports; - std::string port, previousPort; - unsigned int nChannels = 0, nDevices = 0; - ports = jack_get_ports( client, NULL, NULL, 0 ); - if ( ports ) { - // Parse the port names up to the first colon (:). - size_t iColon = 0; - do { - port = (char *) ports[ nChannels ]; - iColon = port.find(":"); - if ( iColon != std::string::npos ) { - port = port.substr( 0, iColon + 1 ); - if ( port != previousPort ) { - nDevices++; - previousPort = port; + unsigned int RtApiJack :: getDeviceCount( void ) + { + // See if we can become a jack client. + jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption; + jack_status_t *status = NULL; + jack_client_t *client = jack_client_open( "RtApiJackCount", options, status ); + if ( client == 0 ) return 0; + + const char **ports; + std::string port, previousPort; + unsigned int nChannels = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, NULL, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nChannels ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon + 1 ); + if ( port != previousPort ) { + nDevices++; + previousPort = port; + } } - } - } while ( ports[++nChannels] ); - free( ports ); - } + } while ( ports[++nChannels] ); + free( ports ); + } - jack_client_close( client ); - return nDevices; -} + jack_client_close( client ); + return nDevices; + } -RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; + RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) + { + RtAudio::DeviceInfo info; + info.probed = false; - jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption - jack_status_t *status = NULL; - jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status ); - if ( client == 0 ) { - errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!"; - error( RtError::WARNING ); - return info; - } + jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption + jack_status_t *status = NULL; + jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status ); + if ( client == 0 ) { + errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!"; + error( RtError::WARNING ); + return info; + } - const char **ports; - std::string port, previousPort; - unsigned int nPorts = 0, nDevices = 0; - ports = jack_get_ports( client, NULL, NULL, 0 ); - if ( ports ) { - // Parse the port names up to the first colon (:). - size_t iColon = 0; - do { - port = (char *) ports[ nPorts ]; - iColon = port.find(":"); - if ( iColon != std::string::npos ) { - port = port.substr( 0, iColon ); - if ( port != previousPort ) { - if ( nDevices == device ) info.name = port; - nDevices++; - previousPort = port; + const char **ports; + std::string port, previousPort; + unsigned int nPorts = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, NULL, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nPorts ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon ); + if ( port != previousPort ) { + if ( nDevices == device ) info.name = port; + nDevices++; + previousPort = port; + } } - } - } while ( ports[++nPorts] ); - free( ports ); - } + } while ( ports[++nPorts] ); + free( ports ); + } - if ( device >= nDevices ) { - errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - } + if ( device >= nDevices ) { + errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } - // Get the current jack server sample rate. - info.sampleRates.clear(); - info.sampleRates.push_back( jack_get_sample_rate( client ) ); - - // Count the available ports containing the client name as device - // channels. Jack "input ports" equal RtAudio output channels. - unsigned int nChannels = 0; - ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput ); - if ( ports ) { - while ( ports[ nChannels ] ) nChannels++; - free( ports ); - info.outputChannels = nChannels; - } + // Get the current jack server sample rate. + info.sampleRates.clear(); + info.sampleRates.push_back( jack_get_sample_rate( client ) ); - // Jack "output ports" equal RtAudio input channels. - nChannels = 0; - ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput ); - if ( ports ) { - while ( ports[ nChannels ] ) nChannels++; - free( ports ); - info.inputChannels = nChannels; - } + // Count the available ports containing the client name as device + // channels. Jack "input ports" equal RtAudio output channels. + unsigned int nChannels = 0; + ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + info.outputChannels = nChannels; + } + + // Jack "output ports" equal RtAudio input channels. + nChannels = 0; + ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + info.inputChannels = nChannels; + } + + if ( info.outputChannels == 0 && info.inputChannels == 0 ) { + jack_client_close(client); + errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!"; + error( RtError::WARNING ); + return info; + } + + // If device opens for both playback and capture, we determine the channels. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + + // Jack always uses 32-bit floats. + info.nativeFormats = RTAUDIO_FLOAT32; + + // Jack doesn't provide default devices so we'll use the first available one. + if ( device == 0 && info.outputChannels > 0 ) + info.isDefaultOutput = true; + if ( device == 0 && info.inputChannels > 0 ) + info.isDefaultInput = true; - if ( info.outputChannels == 0 && info.inputChannels == 0 ) { jack_client_close(client); - errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!"; - error( RtError::WARNING ); + info.probed = true; return info; } - // If device opens for both playback and capture, we determine the channels. - if ( info.outputChannels > 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - - // Jack always uses 32-bit floats. - info.nativeFormats = RTAUDIO_FLOAT32; - - // Jack doesn't provide default devices so we'll use the first available one. - if ( device == 0 && info.outputChannels > 0 ) - info.isDefaultOutput = true; - if ( device == 0 && info.inputChannels > 0 ) - info.isDefaultInput = true; + int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer ) + { + CallbackInfo *info = (CallbackInfo *) infoPointer; - jack_client_close(client); - info.probed = true; - return info; -} + RtApiJack *object = (RtApiJack *) info->object; + if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1; -int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer ) -{ - CallbackInfo *info = (CallbackInfo *) infoPointer; + return 0; + } - RtApiJack *object = (RtApiJack *) info->object; - if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1; + void jackShutdown( void *infoPointer ) + { + CallbackInfo *info = (CallbackInfo *) infoPointer; + RtApiJack *object = (RtApiJack *) info->object; - return 0; -} + // Check current stream state. If stopped, then we'll assume this + // was called as a result of a call to RtApiJack::stopStream (the + // deactivation of a client handle causes this function to be called). + // If not, we'll assume the Jack server is shutting down or some + // other problem occurred and we should close the stream. + if ( object->isStreamRunning() == false ) return; -void jackShutdown( void *infoPointer ) -{ - CallbackInfo *info = (CallbackInfo *) infoPointer; - RtApiJack *object = (RtApiJack *) info->object; + object->closeStream(); + std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl; + } - // Check current stream state. If stopped, then we'll assume this - // was called as a result of a call to RtApiJack::stopStream (the - // deactivation of a client handle causes this function to be called). - // If not, we'll assume the Jack server is shutting down or some - // other problem occurred and we should close the stream. - if ( object->isStreamRunning() == false ) return; + int jackXrun( void *infoPointer ) + { + JackHandle *handle = (JackHandle *) infoPointer; - object->closeStream(); - std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl; -} + if ( handle->ports[0] ) handle->xrun[0] = true; + if ( handle->ports[1] ) handle->xrun[1] = true; -int jackXrun( void *infoPointer ) -{ - JackHandle *handle = (JackHandle *) infoPointer; + return 0; + } - if ( handle->ports[0] ) handle->xrun[0] = true; - if ( handle->ports[1] ) handle->xrun[1] = true; + bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) + { + JackHandle *handle = (JackHandle *) stream_.apiHandle; + + // Look for jack server and try to become a client (only do once per stream). + jack_client_t *client = 0; + if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) { + jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption; + jack_status_t *status = NULL; + if ( options && !options->streamName.empty() ) + client = jack_client_open( options->streamName.c_str(), jackoptions, status ); + else + client = jack_client_open( "RtApiJack", jackoptions, status ); + if ( client == 0 ) { + errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!"; + error( RtError::WARNING ); + return FAILURE; + } + } + else { + // The handle must have been created on an earlier pass. + client = handle->client; + } + + const char **ports; + std::string port, previousPort, deviceName; + unsigned int nPorts = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, NULL, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nPorts ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon ); + if ( port != previousPort ) { + if ( nDevices == device ) deviceName = port; + nDevices++; + previousPort = port; + } + } + } while ( ports[++nPorts] ); + free( ports ); + } - return 0; -} + if ( device >= nDevices ) { + errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } -bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - JackHandle *handle = (JackHandle *) stream_.apiHandle; + // Count the available ports containing the client name as device + // channels. Jack "input ports" equal RtAudio output channels. + unsigned int nChannels = 0; + unsigned long flag = JackPortIsInput; + if ( mode == INPUT ) flag = JackPortIsOutput; + ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + } - // Look for jack server and try to become a client (only do once per stream). - jack_client_t *client = 0; - if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) { - jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption; - jack_status_t *status = NULL; - if ( options && !options->streamName.empty() ) - client = jack_client_open( options->streamName.c_str(), jackoptions, status ); - else - client = jack_client_open( "RtApiJack", jackoptions, status ); - if ( client == 0 ) { - errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!"; - error( RtError::WARNING ); + // Compare the jack ports for specified client to the requested number of channels. + if ( nChannels < (channels + firstChannel) ) { + errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ")."; + errorText_ = errorStream_.str(); return FAILURE; } - } - else { - // The handle must have been created on an earlier pass. - client = handle->client; - } - - const char **ports; - std::string port, previousPort, deviceName; - unsigned int nPorts = 0, nDevices = 0; - ports = jack_get_ports( client, NULL, NULL, 0 ); - if ( ports ) { - // Parse the port names up to the first colon (:). - size_t iColon = 0; - do { - port = (char *) ports[ nPorts ]; - iColon = port.find(":"); - if ( iColon != std::string::npos ) { - port = port.substr( 0, iColon ); - if ( port != previousPort ) { - if ( nDevices == device ) deviceName = port; - nDevices++; - previousPort = port; - } - } - } while ( ports[++nPorts] ); - free( ports ); - } - if ( device >= nDevices ) { - errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } + // Check the jack server sample rate. + unsigned int jackRate = jack_get_sample_rate( client ); + if ( sampleRate != jackRate ) { + jack_client_close( client ); + errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.sampleRate = jackRate; - // Count the available ports containing the client name as device - // channels. Jack "input ports" equal RtAudio output channels. - unsigned int nChannels = 0; - unsigned long flag = JackPortIsInput; - if ( mode == INPUT ) flag = JackPortIsOutput; - ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); - if ( ports ) { - while ( ports[ nChannels ] ) nChannels++; + // Get the latency of the JACK port. + ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); + if ( ports[ firstChannel ] ) + stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) ); free( ports ); - } - // Compare the jack ports for specified client to the requested number of channels. - if ( nChannels < (channels + firstChannel) ) { - errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // The jack server always uses 32-bit floating-point data. + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + stream_.userFormat = format; - // Check the jack server sample rate. - unsigned int jackRate = jack_get_sample_rate( client ); - if ( sampleRate != jackRate ) { - jack_client_close( client ); - errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.sampleRate = jackRate; + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; - // Get the latency of the JACK port. - ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); - if ( ports[ firstChannel ] ) - stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) ); - free( ports ); + // Jack always uses non-interleaved buffers. + stream_.deviceInterleaved[mode] = false; - // The jack server always uses 32-bit floating-point data. - stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - stream_.userFormat = format; + // Jack always provides host byte-ordered data. + stream_.doByteSwap[mode] = false; - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; + // Get the buffer size. The buffer size and number of buffers + // (periods) is set when the jack server is started. + stream_.bufferSize = (int) jack_get_buffer_size( client ); + *bufferSize = stream_.bufferSize; - // Jack always uses non-interleaved buffers. - stream_.deviceInterleaved[mode] = false; + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate our JackHandle structure for the stream. + if ( handle == 0 ) { + try { + handle = new JackHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory."; + goto error; + } - // Jack always provides host byte-ordered data. - stream_.doByteSwap[mode] = false; + if ( pthread_cond_init(&handle->condition, NULL) ) { + errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + stream_.apiHandle = (void *) handle; + handle->client = client; + } + handle->deviceName[mode] = deviceName; - // Get the buffer size. The buffer size and number of buffers - // (periods) is set when the jack server is started. - stream_.bufferSize = (int) jack_get_buffer_size( client ); - *bufferSize = stream_.bufferSize; + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } - stream_.nDeviceChannels[mode] = channels; - stream_.nUserChannels[mode] = channels; + if ( stream_.doConvertBuffer[mode] ) { - // Set flags for buffer conversion. - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; + bool makeBuffer = true; + if ( mode == OUTPUT ) + bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + else { // mode == INPUT + bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] ); + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]); + if ( bufferBytes < bytesOut ) makeBuffer = false; + } + } - // Allocate our JackHandle structure for the stream. - if ( handle == 0 ) { - try { - handle = new JackHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory."; - goto error; + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } } - if ( pthread_cond_init(&handle->condition, NULL) ) { - errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable."; + // Allocate memory for the Jack ports (channels) identifiers. + handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels ); + if ( handle->ports[mode] == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory."; goto error; } - stream_.apiHandle = (void *) handle; - handle->client = client; - } - handle->deviceName[mode] = deviceName; - - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - if ( stream_.doConvertBuffer[mode] ) { + stream_.device[mode] = device; + stream_.channelOffset[mode] = firstChannel; + stream_.state = STREAM_STOPPED; + stream_.callbackInfo.object = (void *) this; - bool makeBuffer = true; - if ( mode == OUTPUT ) - bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - else { // mode == INPUT - bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] ); - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]); - if ( bufferBytes < bytesOut ) makeBuffer = false; - } + if ( stream_.mode == OUTPUT && mode == INPUT ) + // We had already set up the stream for output. + stream_.mode = DUPLEX; + else { + stream_.mode = mode; + jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo ); + jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle ); + jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo ); } - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory."; - goto error; + // Register our ports. + char label[64]; + if ( mode == OUTPUT ) { + for ( unsigned int i=0; iports[0][i] = jack_port_register( handle->client, (const char *)label, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 ); } } - } - - // Allocate memory for the Jack ports (channels) identifiers. - handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels ); - if ( handle->ports[mode] == NULL ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory."; - goto error; - } - - stream_.device[mode] = device; - stream_.channelOffset[mode] = firstChannel; - stream_.state = STREAM_STOPPED; - stream_.callbackInfo.object = (void *) this; - - if ( stream_.mode == OUTPUT && mode == INPUT ) - // We had already set up the stream for output. - stream_.mode = DUPLEX; - else { - stream_.mode = mode; - jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo ); - jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle ); - jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo ); - } - - // Register our ports. - char label[64]; - if ( mode == OUTPUT ) { - for ( unsigned int i=0; iports[0][i] = jack_port_register( handle->client, (const char *)label, - JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 ); - } - } - else { - for ( unsigned int i=0; iports[1][i] = jack_port_register( handle->client, (const char *)label, - JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 ); + else { + for ( unsigned int i=0; iports[1][i] = jack_port_register( handle->client, (const char *)label, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 ); + } } - } - // Setup the buffer conversion information structure. We don't use - // buffers to do channel offsets, so we override that parameter - // here. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); + // Setup the buffer conversion information structure. We don't use + // buffers to do channel offsets, so we override that parameter + // here. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); - return SUCCESS; - - error: - if ( handle ) { - pthread_cond_destroy( &handle->condition ); - jack_client_close( handle->client ); + return SUCCESS; - if ( handle->ports[0] ) free( handle->ports[0] ); - if ( handle->ports[1] ) free( handle->ports[1] ); + error: + if ( handle ) { + pthread_cond_destroy( &handle->condition ); + jack_client_close( handle->client ); - delete handle; - stream_.apiHandle = 0; - } + if ( handle->ports[0] ) free( handle->ports[0] ); + if ( handle->ports[1] ) free( handle->ports[1] ); - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; + delete handle; + stream_.apiHandle = 0; } - } - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } - return FAILURE; -} + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } -void RtApiJack :: closeStream( void ) -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiJack::closeStream(): no open stream to close!"; - error( RtError::WARNING ); - return; + return FAILURE; } - JackHandle *handle = (JackHandle *) stream_.apiHandle; - if ( handle ) { + void RtApiJack :: closeStream( void ) + { + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiJack::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } - if ( stream_.state == STREAM_RUNNING ) - jack_deactivate( handle->client ); + JackHandle *handle = (JackHandle *) stream_.apiHandle; + if ( handle ) { - jack_client_close( handle->client ); - } + if ( stream_.state == STREAM_RUNNING ) + jack_deactivate( handle->client ); - if ( handle ) { - if ( handle->ports[0] ) free( handle->ports[0] ); - if ( handle->ports[1] ) free( handle->ports[1] ); - pthread_cond_destroy( &handle->condition ); - delete handle; - stream_.apiHandle = 0; - } + jack_client_close( handle->client ); + } - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; + if ( handle ) { + if ( handle->ports[0] ) free( handle->ports[0] ); + if ( handle->ports[1] ) free( handle->ports[1] ); + pthread_cond_destroy( &handle->condition ); + delete handle; + stream_.apiHandle = 0; } - } - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } -void RtApiJack :: startStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiJack::startStream(): the stream is already running!"; - error( RtError::WARNING ); - return; + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; } - MUTEX_LOCK(&stream_.mutex); - - JackHandle *handle = (JackHandle *) stream_.apiHandle; - int result = jack_activate( handle->client ); - if ( result ) { - errorText_ = "RtApiJack::startStream(): unable to activate JACK client!"; - goto unlock; - } + void RtApiJack :: startStream( void ) + { + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiJack::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } - const char **ports; + MUTEX_LOCK(&stream_.mutex); - // Get the list of available ports. - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - result = 1; - ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput); - if ( ports == NULL) { - errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!"; + JackHandle *handle = (JackHandle *) stream_.apiHandle; + int result = jack_activate( handle->client ); + if ( result ) { + errorText_ = "RtApiJack::startStream(): unable to activate JACK client!"; goto unlock; } - // Now make the port connections. Since RtAudio wasn't designed to - // allow the user to select particular channels of a device, we'll - // just open the first "nChannels" ports with offset. - for ( unsigned int i=0; iclient, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] ); - if ( result ) { - free( ports ); - errorText_ = "RtApiJack::startStream(): error connecting output ports!"; + ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput); + if ( ports == NULL) { + errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!"; goto unlock; } - } - free(ports); - } - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - result = 1; - ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput ); - if ( ports == NULL) { - errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!"; - goto unlock; + // Now make the port connections. Since RtAudio wasn't designed to + // allow the user to select particular channels of a device, we'll + // just open the first "nChannels" ports with offset. + for ( unsigned int i=0; iclient, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] ); + if ( result ) { + free( ports ); + errorText_ = "RtApiJack::startStream(): error connecting output ports!"; + goto unlock; + } + } + free(ports); } - // Now make the port connections. See note above. - for ( unsigned int i=0; iclient, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) ); - if ( result ) { - free( ports ); - errorText_ = "RtApiJack::startStream(): error connecting input ports!"; + ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput ); + if ( ports == NULL) { + errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!"; goto unlock; } - } - free(ports); - } - handle->drainCounter = 0; - handle->internalDrain = false; - stream_.state = STREAM_RUNNING; + // Now make the port connections. See note above. + for ( unsigned int i=0; iclient, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) ); + if ( result ) { + free( ports ); + errorText_ = "RtApiJack::startStream(): error connecting input ports!"; + goto unlock; + } + } + free(ports); + } - unlock: - MUTEX_UNLOCK(&stream_.mutex); + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; - if ( result == 0 ) return; - error( RtError::SYSTEM_ERROR ); -} + unlock: + MUTEX_UNLOCK(&stream_.mutex); -void RtApiJack :: stopStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiJack::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; + if ( result == 0 ) return; + error( RtError::SYSTEM_ERROR ); } - MUTEX_LOCK( &stream_.mutex ); - - JackHandle *handle = (JackHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 1; - pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled + void RtApiJack :: stopStream( void ) + { + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiJack::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; } - } - - jack_deactivate( handle->client ); - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); -} + MUTEX_LOCK( &stream_.mutex ); -void RtApiJack :: abortStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiJack::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } + JackHandle *handle = (JackHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - JackHandle *handle = (JackHandle *) stream_.apiHandle; - handle->drainCounter = 1; + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 1; + pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled + } + } - stopStream(); -} + jack_deactivate( handle->client ); + stream_.state = STREAM_STOPPED; -bool RtApiJack :: callbackEvent( unsigned long nframes ) -{ - if ( stream_.state == STREAM_STOPPED ) return SUCCESS; - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); - return FAILURE; - } - if ( stream_.bufferSize != nframes ) { - errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!"; - error( RtError::WARNING ); - return FAILURE; + MUTEX_UNLOCK( &stream_.mutex ); } - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - JackHandle *handle = (JackHandle *) stream_.apiHandle; + void RtApiJack :: abortStream( void ) + { + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiJack::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } - // Check if we were draining the stream and signal is finished. - if ( handle->drainCounter > 3 ) { - if ( handle->internalDrain == false ) - pthread_cond_signal( &handle->condition ); - else - stopStream(); - return SUCCESS; - } + JackHandle *handle = (JackHandle *) stream_.apiHandle; + handle->drainCounter = 1; - MUTEX_LOCK( &stream_.mutex ); + stopStream(); + } - // Invoke user callback first, to get fresh output data. - if ( handle->drainCounter == 0 ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; + bool RtApiJack :: callbackEvent( unsigned long nframes ) + { + if ( stream_.state == STREAM_STOPPED ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return FAILURE; } - handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( handle->drainCounter == 2 ) { - MUTEX_UNLOCK( &stream_.mutex ); - abortStream(); - return SUCCESS; + if ( stream_.bufferSize != nframes ) { + errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!"; + error( RtError::WARNING ); + return FAILURE; } - else if ( handle->drainCounter == 1 ) - handle->internalDrain = true; - } - jack_default_audio_sample_t *jackbuffer; - unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t ); - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - if ( handle->drainCounter > 0 ) { // write zeros to the output stream - - for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); - memset( jackbuffer, 0, bufferBytes ); - } + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + JackHandle *handle = (JackHandle *) stream_.apiHandle; + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + if ( handle->internalDrain == false ) + pthread_cond_signal( &handle->condition ); + else + stopStream(); + return SUCCESS; } - else if ( stream_.doConvertBuffer[0] ) { - convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + MUTEX_LOCK( &stream_.mutex ); - for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); - memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes ); + // Invoke user callback first, to get fresh output data. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; } - } - else { // no buffer conversion - for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); - memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes ); + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; } + handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( handle->drainCounter == 2 ) { + MUTEX_UNLOCK( &stream_.mutex ); + abortStream(); + return SUCCESS; + } + else if ( handle->drainCounter == 1 ) + handle->internalDrain = true; } - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } - } + jack_default_audio_sample_t *jackbuffer; + unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t ); + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + if ( handle->drainCounter > 0 ) { // write zeros to the output stream + + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memset( jackbuffer, 0, bufferBytes ); + } - if ( stream_.doConvertBuffer[1] ) { - for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); - memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes ); - } - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - } - else { // no buffer conversion - for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); - memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes ); } - } - } + else if ( stream_.doConvertBuffer[0] ) { - unlock: - MUTEX_UNLOCK(&stream_.mutex); + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - RtApi::tickStreamTime(); - return SUCCESS; -} -//******************** End of __UNIX_JACK__ *********************// + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes ); + } + } + else { // no buffer conversion + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes ); + } + } + + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + } + + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + if ( stream_.doConvertBuffer[1] ) { + for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); + memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes ); + } + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + } + else { // no buffer conversion + for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); + memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes ); + } + } + } + + unlock: + MUTEX_UNLOCK(&stream_.mutex); + + RtApi::tickStreamTime(); + return SUCCESS; + } + //******************** End of __UNIX_JACK__ *********************// #endif #if defined(__WINDOWS_ASIO__) // ASIO API on Windows -// The ASIO API is designed around a callback scheme, so this -// implementation is similar to that used for OS-X CoreAudio and Linux -// Jack. The primary constraint with ASIO is that it only allows -// access to a single driver at a time. Thus, it is not possible to -// have more than one simultaneous RtAudio stream. -// -// This implementation also requires a number of external ASIO files -// and a few global variables. The ASIO callback scheme does not -// allow for the passing of user data, so we must create a global -// pointer to our callbackInfo structure. -// -// On unix systems, we make use of a pthread condition variable. -// Since there is no equivalent in Windows, I hacked something based -// on information found in -// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html. + // The ASIO API is designed around a callback scheme, so this + // implementation is similar to that used for OS-X CoreAudio and Linux + // Jack. The primary constraint with ASIO is that it only allows + // access to a single driver at a time. Thus, it is not possible to + // have more than one simultaneous RtAudio stream. + // + // This implementation also requires a number of external ASIO files + // and a few global variables. The ASIO callback scheme does not + // allow for the passing of user data, so we must create a global + // pointer to our callbackInfo structure. + // + // On unix systems, we make use of a pthread condition variable. + // Since there is no equivalent in Windows, I hacked something based + // on information found in + // http://www.cs.wustl.edu/~schmidt/win32-cv-1.html. #include "asiosys.h" #include "asio.h" @@ -2328,947 +2429,947 @@ bool RtApiJack :: callbackEvent( unsigned long nframes ) #include "asiodrivers.h" #include -AsioDrivers drivers; -ASIOCallbacks asioCallbacks; -ASIODriverInfo driverInfo; -CallbackInfo *asioCallbackInfo; -bool asioXRun; + AsioDrivers drivers; + ASIOCallbacks asioCallbacks; + ASIODriverInfo driverInfo; + CallbackInfo *asioCallbackInfo; + bool asioXRun; -struct AsioHandle { - int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - ASIOBufferInfo *bufferInfos; - HANDLE condition; + struct AsioHandle { + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + ASIOBufferInfo *bufferInfos; + HANDLE condition; - AsioHandle() - :drainCounter(0), internalDrain(false), bufferInfos(0) {} -}; + AsioHandle() + :drainCounter(0), internalDrain(false), bufferInfos(0) {} + }; -// Function declarations (definitions at end of section) -static const char* getAsioErrorString( ASIOError result ); -void sampleRateChanged( ASIOSampleRate sRate ); -long asioMessages( long selector, long value, void* message, double* opt ); + // Function declarations (definitions at end of section) + static const char* getAsioErrorString( ASIOError result ); + void sampleRateChanged( ASIOSampleRate sRate ); + long asioMessages( long selector, long value, void* message, double* opt ); -RtApiAsio :: RtApiAsio() -{ - // ASIO cannot run on a multi-threaded appartment. You can call - // CoInitialize beforehand, but it must be for appartment threading - // (in which case, CoInitilialize will return S_FALSE here). - coInitialized_ = false; - HRESULT hr = CoInitialize( NULL ); - if ( FAILED(hr) ) { - errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"; - error( RtError::WARNING ); + RtApiAsio :: RtApiAsio() + { + // ASIO cannot run on a multi-threaded appartment. You can call + // CoInitialize beforehand, but it must be for appartment threading + // (in which case, CoInitilialize will return S_FALSE here). + coInitialized_ = false; + HRESULT hr = CoInitialize( NULL ); + if ( FAILED(hr) ) { + errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"; + error( RtError::WARNING ); + } + coInitialized_ = true; + + drivers.removeCurrentDriver(); + driverInfo.asioVersion = 2; + + // See note in DirectSound implementation about GetDesktopWindow(). + driverInfo.sysRef = GetForegroundWindow(); } - coInitialized_ = true; - drivers.removeCurrentDriver(); - driverInfo.asioVersion = 2; + RtApiAsio :: ~RtApiAsio() + { + if ( stream_.state != STREAM_CLOSED ) closeStream(); + if ( coInitialized_ ) CoUninitialize(); + } - // See note in DirectSound implementation about GetDesktopWindow(). - driverInfo.sysRef = GetForegroundWindow(); -} + unsigned int RtApiAsio :: getDeviceCount( void ) + { + return (unsigned int) drivers.asioGetNumDev(); + } -RtApiAsio :: ~RtApiAsio() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); - if ( coInitialized_ ) CoUninitialize(); -} + RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) + { + RtAudio::DeviceInfo info; + info.probed = false; -unsigned int RtApiAsio :: getDeviceCount( void ) -{ - return (unsigned int) drivers.asioGetNumDev(); -} + // Get device ID + unsigned int nDevices = getDeviceCount(); + if ( nDevices == 0 ) { + errorText_ = "RtApiAsio::getDeviceInfo: no devices found!"; + error( RtError::INVALID_USE ); + } -RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; + if ( device >= nDevices ) { + errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } - // Get device ID - unsigned int nDevices = getDeviceCount(); - if ( nDevices == 0 ) { - errorText_ = "RtApiAsio::getDeviceInfo: no devices found!"; - error( RtError::INVALID_USE ); - } + // If a stream is already open, we cannot probe other devices. Thus, use the saved results. + if ( stream_.state != STREAM_CLOSED ) { + if ( device >= devices_.size() ) { + errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened."; + error( RtError::WARNING ); + return info; + } + return devices_[ device ]; + } - if ( device >= nDevices ) { - errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - } + char driverName[32]; + ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } - // If a stream is already open, we cannot probe other devices. Thus, use the saved results. - if ( stream_.state != STREAM_CLOSED ) { - if ( device >= devices_.size() ) { - errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened."; + info.name = driverName; + + if ( !drivers.loadDriver( driverName ) ) { + errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ")."; + errorText_ = errorStream_.str(); error( RtError::WARNING ); return info; } - return devices_[ device ]; - } - char driverName[32]; - ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } + result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } - info.name = driverName; + // Determine the device channel information. + long inputChannels, outputChannels; + result = ASIOGetChannels( &inputChannels, &outputChannels ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } - if ( !drivers.loadDriver( driverName ) ) { - errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } + info.outputChannels = outputChannels; + info.inputChannels = inputChannels; + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - result = ASIOInit( &driverInfo ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } + // Determine the supported sample rates. + info.sampleRates.clear(); + for ( unsigned int i=0; i 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + info.nativeFormats = 0; + if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) + info.nativeFormats |= RTAUDIO_SINT16; + else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) + info.nativeFormats |= RTAUDIO_SINT32; + else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) + info.nativeFormats |= RTAUDIO_FLOAT32; + else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) + info.nativeFormats |= RTAUDIO_FLOAT64; - // Determine the supported sample rates. - info.sampleRates.clear(); - for ( unsigned int i=0; iobject; + object->callbackEvent( index ); + } - if ( getDefaultOutputDevice() == device ) - info.isDefaultOutput = true; - if ( getDefaultInputDevice() == device ) - info.isDefaultInput = true; + void RtApiAsio :: saveDeviceInfo( void ) + { + devices_.clear(); - info.probed = true; - drivers.removeCurrentDriver(); - return info; -} + unsigned int nDevices = getDeviceCount(); + devices_.resize( nDevices ); + for ( unsigned int i=0; iobject; - object->callbackEvent( index ); -} + bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) + { + // For ASIO, a duplex stream MUST use the same driver. + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) { + errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!"; + return FAILURE; + } -void RtApiAsio :: saveDeviceInfo( void ) -{ - devices_.clear(); + char driverName[32]; + ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } - unsigned int nDevices = getDeviceCount(); - devices_.resize( nDevices ); - for ( unsigned int i=0; isaveDeviceInfo(); -bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - // For ASIO, a duplex stream MUST use the same driver. - if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) { - errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!"; - return FAILURE; - } + // Only load the driver once for duplex stream. + if ( mode != INPUT || stream_.mode != OUTPUT ) { + if ( !drivers.loadDriver( driverName ) ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } - char driverName[32]; - ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } + result = ASIOInit( &driverInfo ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } - // The getDeviceInfo() function will not work when a stream is open - // because ASIO does not allow multiple devices to run at the same - // time. Thus, we'll probe the system before opening a stream and - // save the results for use by getDeviceInfo(). - this->saveDeviceInfo(); + // Check the device channel count. + long inputChannels, outputChannels; + result = ASIOGetChannels( &inputChannels, &outputChannels ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Only load the driver once for duplex stream. - if ( mode != INPUT || stream_.mode != OUTPUT ) { - if ( !drivers.loadDriver( driverName ) ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ")."; + if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) || + ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ")."; errorText_ = errorStream_.str(); return FAILURE; } + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; + stream_.channelOffset[mode] = firstChannel; - result = ASIOInit( &driverInfo ); + // Verify the sample rate is supported. + result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate ); if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ")."; errorText_ = errorStream_.str(); return FAILURE; } - } - - // Check the device channel count. - long inputChannels, outputChannels; - result = ASIOGetChannels( &inputChannels, &outputChannels ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) || - ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.nDeviceChannels[mode] = channels; - stream_.nUserChannels[mode] = channels; - stream_.channelOffset[mode] = firstChannel; - // Verify the sample rate is supported. - result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Get the current sample rate + ASIOSampleRate currentRate; + result = ASIOGetSampleRate( ¤tRate ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Get the current sample rate - ASIOSampleRate currentRate; - result = ASIOGetSampleRate( ¤tRate ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Set the sample rate only if necessary + if ( currentRate != sampleRate ) { + result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } - // Set the sample rate only if necessary - if ( currentRate != sampleRate ) { - result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); + // Determine the driver data type. + ASIOChannelInfo channelInfo; + channelInfo.channel = 0; + if ( mode == OUTPUT ) channelInfo.isInput = false; + else channelInfo.isInput = true; + result = ASIOGetChannelInfo( &channelInfo ); if ( result != ASE_OK ) { drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ")."; + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format."; errorText_ = errorStream_.str(); return FAILURE; } - } - - // Determine the driver data type. - ASIOChannelInfo channelInfo; - channelInfo.channel = 0; - if ( mode == OUTPUT ) channelInfo.isInput = false; - else channelInfo.isInput = true; - result = ASIOGetChannelInfo( &channelInfo ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format."; - errorText_ = errorStream_.str(); - return FAILURE; - } - // Assuming WINDOWS host is always little-endian. - stream_.doByteSwap[mode] = false; - stream_.userFormat = format; - stream_.deviceFormat[mode] = 0; - if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true; - } - else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true; - } - else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true; - } - else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; - if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true; - } + // Assuming WINDOWS host is always little-endian. + stream_.doByteSwap[mode] = false; + stream_.userFormat = format; + stream_.deviceFormat[mode] = 0; + if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; + if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true; + } - if ( stream_.deviceFormat[mode] == 0 ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - return FAILURE; - } + if ( stream_.deviceFormat[mode] == 0 ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Set the buffer size. For a duplex stream, this will end up - // setting the buffer size based on the input constraints, which - // should be ok. - long minSize, maxSize, preferSize, granularity; - result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Set the buffer size. For a duplex stream, this will end up + // setting the buffer size based on the input constraints, which + // should be ok. + long minSize, maxSize, preferSize, granularity; + result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size."; + errorText_ = errorStream_.str(); + return FAILURE; + } - if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; - else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; - else if ( granularity == -1 ) { - // Make sure bufferSize is a power of two. - int log2_of_min_size = 0; - int log2_of_max_size = 0; + if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; + else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; + else if ( granularity == -1 ) { + // Make sure bufferSize is a power of two. + int log2_of_min_size = 0; + int log2_of_max_size = 0; - for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) { - if ( minSize & ((long)1 << i) ) log2_of_min_size = i; - if ( maxSize & ((long)1 << i) ) log2_of_max_size = i; - } + for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) { + if ( minSize & ((long)1 << i) ) log2_of_min_size = i; + if ( maxSize & ((long)1 << i) ) log2_of_max_size = i; + } - long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) ); - int min_delta_num = log2_of_min_size; + long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) ); + int min_delta_num = log2_of_min_size; - for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) { - long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) ); - if (current_delta < min_delta) { - min_delta = current_delta; - min_delta_num = i; - } - } + for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) { + long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) ); + if (current_delta < min_delta) { + min_delta = current_delta; + min_delta_num = i; + } + } - *bufferSize = ( (unsigned int)1 << min_delta_num ); - if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; - else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; - } - else if ( granularity != 0 ) { - // Set to an even multiple of granularity, rounding up. - *bufferSize = (*bufferSize + granularity-1) / granularity * granularity; - } + *bufferSize = ( (unsigned int)1 << min_delta_num ); + if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; + else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; + } + else if ( granularity != 0 ) { + // Set to an even multiple of granularity, rounding up. + *bufferSize = (*bufferSize + granularity-1) / granularity * granularity; + } - if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) { - drivers.removeCurrentDriver(); - errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!"; - return FAILURE; - } + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) { + drivers.removeCurrentDriver(); + errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!"; + return FAILURE; + } - stream_.bufferSize = *bufferSize; - stream_.nBuffers = 2; + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 2; - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; - // ASIO always uses non-interleaved buffers. - stream_.deviceInterleaved[mode] = false; + // ASIO always uses non-interleaved buffers. + stream_.deviceInterleaved[mode] = false; - // Allocate, if necessary, our AsioHandle structure for the stream. - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - if ( handle == 0 ) { - try { - handle = new AsioHandle; + // Allocate, if necessary, our AsioHandle structure for the stream. + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( handle == 0 ) { + try { + handle = new AsioHandle; + } + catch ( std::bad_alloc& ) { + //if ( handle == NULL ) { + drivers.removeCurrentDriver(); + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory."; + return FAILURE; + } + handle->bufferInfos = 0; + + // Create a manual-reset event. + handle->condition = CreateEvent( NULL, // no security + TRUE, // manual-reset + FALSE, // non-signaled initially + NULL ); // unnamed + stream_.apiHandle = (void *) handle; + } + + // Create the ASIO internal buffers. Since RtAudio sets up input + // and output separately, we'll have to dispose of previously + // created output buffers for a duplex stream. + long inputLatency, outputLatency; + if ( mode == INPUT && stream_.mode == OUTPUT ) { + ASIODisposeBuffers(); + if ( handle->bufferInfos ) free( handle->bufferInfos ); + } + + // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. + bool buffersAllocated = false; + unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; + handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) ); + if ( handle->bufferInfos == NULL ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + goto error; } - catch ( std::bad_alloc& ) { - //if ( handle == NULL ) { - drivers.removeCurrentDriver(); - errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory."; - return FAILURE; + + ASIOBufferInfo *infos; + infos = handle->bufferInfos; + for ( i=0; iisInput = ASIOFalse; + infos->channelNum = i + stream_.channelOffset[0]; + infos->buffers[0] = infos->buffers[1] = 0; + } + for ( i=0; iisInput = ASIOTrue; + infos->channelNum = i + stream_.channelOffset[1]; + infos->buffers[0] = infos->buffers[1] = 0; } - handle->bufferInfos = 0; - // Create a manual-reset event. - handle->condition = CreateEvent( NULL, // no security - TRUE, // manual-reset - FALSE, // non-signaled initially - NULL ); // unnamed - stream_.apiHandle = (void *) handle; - } + // Set up the ASIO callback structure and create the ASIO data buffers. + asioCallbacks.bufferSwitch = &bufferSwitch; + asioCallbacks.sampleRateDidChange = &sampleRateChanged; + asioCallbacks.asioMessage = &asioMessages; + asioCallbacks.bufferSwitchTimeInfo = NULL; + result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers."; + errorText_ = errorStream_.str(); + goto error; + } + buffersAllocated = true; + + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } - // Create the ASIO internal buffers. Since RtAudio sets up input - // and output separately, we'll have to dispose of previously - // created output buffers for a duplex stream. - long inputLatency, outputLatency; - if ( mode == INPUT && stream_.mode == OUTPUT ) { - ASIODisposeBuffers(); - if ( handle->bufferInfos ) free( handle->bufferInfos ); - } + if ( stream_.doConvertBuffer[mode] ) { - // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. - bool buffersAllocated = false; - unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; - handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) ); - if ( handle->bufferInfos == NULL ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - goto error; - } + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + } - ASIOBufferInfo *infos; - infos = handle->bufferInfos; - for ( i=0; iisInput = ASIOFalse; - infos->channelNum = i + stream_.channelOffset[0]; - infos->buffers[0] = infos->buffers[1] = 0; - } - for ( i=0; iisInput = ASIOTrue; - infos->channelNum = i + stream_.channelOffset[1]; - infos->buffers[0] = infos->buffers[1] = 0; - } + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } - // Set up the ASIO callback structure and create the ASIO data buffers. - asioCallbacks.bufferSwitch = &bufferSwitch; - asioCallbacks.sampleRateDidChange = &sampleRateChanged; - asioCallbacks.asioMessage = &asioMessages; - asioCallbacks.bufferSwitchTimeInfo = NULL; - result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers."; - errorText_ = errorStream_.str(); - goto error; - } - buffersAllocated = true; + stream_.sampleRate = sampleRate; + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + asioCallbackInfo = &stream_.callbackInfo; + stream_.callbackInfo.object = (void *) this; + if ( stream_.mode == OUTPUT && mode == INPUT ) + // We had already set up an output stream. + stream_.mode = DUPLEX; + else + stream_.mode = mode; - // Set flags for buffer conversion. - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; + // Determine device latencies + result = ASIOGetLatencies( &inputLatency, &outputLatency ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency."; + errorText_ = errorStream_.str(); + error( RtError::WARNING); // warn but don't fail + } + else { + stream_.latency[0] = outputLatency; + stream_.latency[1] = inputLatency; + } - // Allocate necessary internal buffers - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } + // Setup the buffer conversion information structure. We don't use + // buffers to do channel offsets, so we override that parameter + // here. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); - if ( stream_.doConvertBuffer[mode] ) { + return SUCCESS; - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } + error: + if ( buffersAllocated ) + ASIODisposeBuffers(); + drivers.removeCurrentDriver(); + + if ( handle ) { + CloseHandle( handle->condition ); + if ( handle->bufferInfos ) + free( handle->bufferInfos ); + delete handle; + stream_.apiHandle = 0; } - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory."; - goto error; + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; } } - } - stream_.sampleRate = sampleRate; - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - asioCallbackInfo = &stream_.callbackInfo; - stream_.callbackInfo.object = (void *) this; - if ( stream_.mode == OUTPUT && mode == INPUT ) - // We had already set up an output stream. - stream_.mode = DUPLEX; - else - stream_.mode = mode; + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } - // Determine device latencies - result = ASIOGetLatencies( &inputLatency, &outputLatency ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency."; - errorText_ = errorStream_.str(); - error( RtError::WARNING); // warn but don't fail - } - else { - stream_.latency[0] = outputLatency; - stream_.latency[1] = inputLatency; + return FAILURE; } - // Setup the buffer conversion information structure. We don't use - // buffers to do channel offsets, so we override that parameter - // here. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); - - return SUCCESS; + void RtApiAsio :: closeStream() + { + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAsio::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } - error: - if ( buffersAllocated ) + if ( stream_.state == STREAM_RUNNING ) { + stream_.state = STREAM_STOPPED; + ASIOStop(); + } ASIODisposeBuffers(); - drivers.removeCurrentDriver(); - - if ( handle ) { - CloseHandle( handle->condition ); - if ( handle->bufferInfos ) - free( handle->bufferInfos ); - delete handle; - stream_.apiHandle = 0; - } + drivers.removeCurrentDriver(); - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( handle ) { + CloseHandle( handle->condition ); + if ( handle->bufferInfos ) + free( handle->bufferInfos ); + delete handle; + stream_.apiHandle = 0; } - } - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } - return FAILURE; -} + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } -void RtApiAsio :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAsio::closeStream(): no open stream to close!"; - error( RtError::WARNING ); - return; + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; } - if ( stream_.state == STREAM_RUNNING ) { - stream_.state = STREAM_STOPPED; - ASIOStop(); - } - ASIODisposeBuffers(); - drivers.removeCurrentDriver(); + void RtApiAsio :: startStream() + { + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiAsio::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - if ( handle ) { - CloseHandle( handle->condition ); - if ( handle->bufferInfos ) - free( handle->bufferInfos ); - delete handle; - stream_.apiHandle = 0; - } + MUTEX_LOCK( &stream_.mutex ); - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + ASIOError result = ASIOStart(); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device."; + errorText_ = errorStream_.str(); + goto unlock; } - } - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; + asioXRun = false; - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} + unlock: + MUTEX_UNLOCK( &stream_.mutex ); -void RtApiAsio :: startStream() -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiAsio::startStream(): the stream is already running!"; - error( RtError::WARNING ); - return; + if ( result == ASE_OK ) return; + error( RtError::SYSTEM_ERROR ); } - MUTEX_LOCK( &stream_.mutex ); + void RtApiAsio :: stopStream() + { + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - ASIOError result = ASIOStart(); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device."; - errorText_ = errorStream_.str(); - goto unlock; - } + MUTEX_LOCK( &stream_.mutex ); - handle->drainCounter = 0; - handle->internalDrain = false; - stream_.state = STREAM_RUNNING; - asioXRun = false; + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 1; + MUTEX_UNLOCK( &stream_.mutex ); + WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled + ResetEvent( handle->condition ); + MUTEX_LOCK( &stream_.mutex ); + } + } - unlock: - MUTEX_UNLOCK( &stream_.mutex ); + ASIOError result = ASIOStop(); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device."; + errorText_ = errorStream_.str(); + } - if ( result == ASE_OK ) return; - error( RtError::SYSTEM_ERROR ); -} + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); -void RtApiAsio :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; + if ( result == ASE_OK ) return; + error( RtError::SYSTEM_ERROR ); } - MUTEX_LOCK( &stream_.mutex ); - - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 1; - MUTEX_UNLOCK( &stream_.mutex ); - WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled - ResetEvent( handle->condition ); - MUTEX_LOCK( &stream_.mutex ); + void RtApiAsio :: abortStream() + { + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; } - } - ASIOError result = ASIOStop(); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device."; - errorText_ = errorStream_.str(); + // The following lines were commented-out because some behavior was + // noted where the device buffers need to be zeroed to avoid + // continuing sound, even when the device buffers are completed + // disposed. So now, calling abort is the same as calling stop. + //AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + //handle->drainCounter = 1; + stopStream(); } - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); + bool RtApiAsio :: callbackEvent( long bufferIndex ) + { + if ( stream_.state == STREAM_STOPPED ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return FAILURE; + } - if ( result == ASE_OK ) return; - error( RtError::SYSTEM_ERROR ); -} + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; -void RtApiAsio :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + if ( handle->internalDrain == false ) + SetEvent( handle->condition ); + else + stopStream(); + return SUCCESS; + } - // The following lines were commented-out because some behavior was - // noted where the device buffers need to be zeroed to avoid - // continuing sound, even when the device buffers are completed - // disposed. So now, calling abort is the same as calling stop. - //AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - //handle->drainCounter = 1; - stopStream(); -} + MUTEX_LOCK( &stream_.mutex ); -bool RtApiAsio :: callbackEvent( long bufferIndex ) -{ - if ( stream_.state == STREAM_STOPPED ) return SUCCESS; - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); - return FAILURE; - } + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + // Invoke user callback to get fresh output data UNLESS we are + // draining stream. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && asioXRun == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + asioXRun = false; + } + if ( stream_.mode != OUTPUT && asioXRun == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + asioXRun = false; + } + handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( handle->drainCounter == 2 ) { + MUTEX_UNLOCK( &stream_.mutex ); + abortStream(); + return SUCCESS; + } + else if ( handle->drainCounter == 1 ) + handle->internalDrain = true; + } - // Check if we were draining the stream and signal is finished. - if ( handle->drainCounter > 3 ) { - if ( handle->internalDrain == false ) - SetEvent( handle->condition ); - else - stopStream(); - return SUCCESS; - } + unsigned int nChannels, bufferBytes, i, j; + nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - MUTEX_LOCK( &stream_.mutex ); + bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] ); - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) goto unlock; + if ( handle->drainCounter > 1 ) { // write zeros to the output stream - // Invoke user callback to get fresh output data UNLESS we are - // draining stream. - if ( handle->drainCounter == 0 ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && asioXRun == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - asioXRun = false; - } - if ( stream_.mode != OUTPUT && asioXRun == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - asioXRun = false; - } - handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( handle->drainCounter == 2 ) { - MUTEX_UNLOCK( &stream_.mutex ); - abortStream(); - return SUCCESS; - } - else if ( handle->drainCounter == 1 ) - handle->internalDrain = true; - } + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes ); + } - unsigned int nChannels, bufferBytes, i, j; - nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + } + else if ( stream_.doConvertBuffer[0] ) { + + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( stream_.deviceBuffer, + stream_.bufferSize * stream_.nDeviceChannels[0], + stream_.deviceFormat[0] ); + + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memcpy( handle->bufferInfos[i].buffers[bufferIndex], + &stream_.deviceBuffer[j++*bufferBytes], bufferBytes ); + } + + } + else { - bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] ); + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( stream_.userBuffer[0], + stream_.bufferSize * stream_.nUserChannels[0], + stream_.userFormat ); - if ( handle->drainCounter > 1 ) { // write zeros to the output stream + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memcpy( handle->bufferInfos[i].buffers[bufferIndex], + &stream_.userBuffer[0][bufferBytes*j++], bufferBytes ); + } - for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) - memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes ); } - } - else if ( stream_.doConvertBuffer[0] ) { - - convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - if ( stream_.doByteSwap[0] ) - byteSwapBuffer( stream_.deviceBuffer, - stream_.bufferSize * stream_.nDeviceChannels[0], - stream_.deviceFormat[0] ); - - for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) - memcpy( handle->bufferInfos[i].buffers[bufferIndex], - &stream_.deviceBuffer[j++*bufferBytes], bufferBytes ); + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; } - } - else { - - if ( stream_.doByteSwap[0] ) - byteSwapBuffer( stream_.userBuffer[0], - stream_.bufferSize * stream_.nUserChannels[0], - stream_.userFormat ); - - for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) - memcpy( handle->bufferInfos[i].buffers[bufferIndex], - &stream_.userBuffer[0][bufferBytes*j++], bufferBytes ); - } - } + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } - } + bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]); - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + if (stream_.doConvertBuffer[1]) { - bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]); + // Always interleave ASIO input data. + for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) + memcpy( &stream_.deviceBuffer[j++*bufferBytes], + handle->bufferInfos[i].buffers[bufferIndex], + bufferBytes ); + } - if (stream_.doConvertBuffer[1]) { + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( stream_.deviceBuffer, + stream_.bufferSize * stream_.nDeviceChannels[1], + stream_.deviceFormat[1] ); + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - // Always interleave ASIO input data. - for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) - memcpy( &stream_.deviceBuffer[j++*bufferBytes], - handle->bufferInfos[i].buffers[bufferIndex], - bufferBytes ); } - - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( stream_.deviceBuffer, - stream_.bufferSize * stream_.nDeviceChannels[1], - stream_.deviceFormat[1] ); - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - - } - else { - for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) { - memcpy( &stream_.userBuffer[1][bufferBytes*j++], - handle->bufferInfos[i].buffers[bufferIndex], - bufferBytes ); + else { + for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) { + memcpy( &stream_.userBuffer[1][bufferBytes*j++], + handle->bufferInfos[i].buffers[bufferIndex], + bufferBytes ); + } } - } - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( stream_.userBuffer[1], - stream_.bufferSize * stream_.nUserChannels[1], - stream_.userFormat ); + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( stream_.userBuffer[1], + stream_.bufferSize * stream_.nUserChannels[1], + stream_.userFormat ); + } } - } - - unlock: - // The following call was suggested by Malte Clasen. While the API - // documentation indicates it should not be required, some device - // drivers apparently do not function correctly without it. - ASIOOutputReady(); - MUTEX_UNLOCK( &stream_.mutex ); + unlock: + // The following call was suggested by Malte Clasen. While the API + // documentation indicates it should not be required, some device + // drivers apparently do not function correctly without it. + ASIOOutputReady(); - RtApi::tickStreamTime(); - return SUCCESS; -} + MUTEX_UNLOCK( &stream_.mutex ); -void sampleRateChanged( ASIOSampleRate sRate ) -{ - // The ASIO documentation says that this usually only happens during - // external sync. Audio processing is not stopped by the driver, - // actual sample rate might not have even changed, maybe only the - // sample rate status of an AES/EBU or S/PDIF digital input at the - // audio device. - - RtApi *object = (RtApi *) asioCallbackInfo->object; - try { - object->stopStream(); - } - catch ( RtError &exception ) { - std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl; - return; + RtApi::tickStreamTime(); + return SUCCESS; } - std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl; -} + void sampleRateChanged( ASIOSampleRate sRate ) + { + // The ASIO documentation says that this usually only happens during + // external sync. Audio processing is not stopped by the driver, + // actual sample rate might not have even changed, maybe only the + // sample rate status of an AES/EBU or S/PDIF digital input at the + // audio device. -long asioMessages( long selector, long value, void* message, double* opt ) -{ - long ret = 0; - - switch( selector ) { - case kAsioSelectorSupported: - if ( value == kAsioResetRequest - || value == kAsioEngineVersion - || value == kAsioResyncRequest - || value == kAsioLatenciesChanged - // The following three were added for ASIO 2.0, you don't - // necessarily have to support them. - || value == kAsioSupportsTimeInfo - || value == kAsioSupportsTimeCode - || value == kAsioSupportsInputMonitor) - ret = 1L; - break; - case kAsioResetRequest: - // Defer the task and perform the reset of the driver during the - // next "safe" situation. You cannot reset the driver right now, - // as this code is called from the driver. Reset the driver is - // done by completely destruct is. I.e. ASIOStop(), - // ASIODisposeBuffers(), Destruction Afterwards you initialize the - // driver again. - std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl; - ret = 1L; - break; - case kAsioResyncRequest: - // This informs the application that the driver encountered some - // non-fatal data loss. It is used for synchronization purposes - // of different media. Added mainly to work around the Win16Mutex - // problems in Windows 95/98 with the Windows Multimedia system, - // which could lose data because the Mutex was held too long by - // another thread. However a driver can issue it in other - // situations, too. - // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl; - asioXRun = true; - ret = 1L; - break; - case kAsioLatenciesChanged: - // This will inform the host application that the drivers were - // latencies changed. Beware, it this does not mean that the - // buffer sizes have changed! You might need to update internal - // delay data. - std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl; - ret = 1L; - break; - case kAsioEngineVersion: - // Return the supported ASIO version of the host application. If - // a host application does not implement this selector, ASIO 1.0 - // is assumed by the driver. - ret = 2L; - break; - case kAsioSupportsTimeInfo: - // Informs the driver whether the - // asioCallbacks.bufferSwitchTimeInfo() callback is supported. - // For compatibility with ASIO 1.0 drivers the host application - // should always support the "old" bufferSwitch method, too. - ret = 0; - break; - case kAsioSupportsTimeCode: - // Informs the driver whether application is interested in time - // code info. If an application does not need to know about time - // code, the driver has less work to do. - ret = 0; - break; - } - return ret; -} + RtApi *object = (RtApi *) asioCallbackInfo->object; + try { + object->stopStream(); + } + catch ( RtError &exception ) { + std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl; + return; + } -static const char* getAsioErrorString( ASIOError result ) -{ - struct Messages - { - ASIOError value; - const char*message; - }; + std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl; + } - static Messages m[] = + long asioMessages( long selector, long value, void* message, double* opt ) { - { ASE_NotPresent, "Hardware input or output is not present or available." }, - { ASE_HWMalfunction, "Hardware is malfunctioning." }, - { ASE_InvalidParameter, "Invalid input parameter." }, - { ASE_InvalidMode, "Invalid mode." }, - { ASE_SPNotAdvancing, "Sample position not advancing." }, - { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." }, - { ASE_NoMemory, "Not enough memory to complete the request." } - }; - - for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i ) - if ( m[i].value == result ) return m[i].message; + long ret = 0; + + switch( selector ) { + case kAsioSelectorSupported: + if ( value == kAsioResetRequest + || value == kAsioEngineVersion + || value == kAsioResyncRequest + || value == kAsioLatenciesChanged + // The following three were added for ASIO 2.0, you don't + // necessarily have to support them. + || value == kAsioSupportsTimeInfo + || value == kAsioSupportsTimeCode + || value == kAsioSupportsInputMonitor) + ret = 1L; + break; + case kAsioResetRequest: + // Defer the task and perform the reset of the driver during the + // next "safe" situation. You cannot reset the driver right now, + // as this code is called from the driver. Reset the driver is + // done by completely destruct is. I.e. ASIOStop(), + // ASIODisposeBuffers(), Destruction Afterwards you initialize the + // driver again. + std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl; + ret = 1L; + break; + case kAsioResyncRequest: + // This informs the application that the driver encountered some + // non-fatal data loss. It is used for synchronization purposes + // of different media. Added mainly to work around the Win16Mutex + // problems in Windows 95/98 with the Windows Multimedia system, + // which could lose data because the Mutex was held too long by + // another thread. However a driver can issue it in other + // situations, too. + // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl; + asioXRun = true; + ret = 1L; + break; + case kAsioLatenciesChanged: + // This will inform the host application that the drivers were + // latencies changed. Beware, it this does not mean that the + // buffer sizes have changed! You might need to update internal + // delay data. + std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl; + ret = 1L; + break; + case kAsioEngineVersion: + // Return the supported ASIO version of the host application. If + // a host application does not implement this selector, ASIO 1.0 + // is assumed by the driver. + ret = 2L; + break; + case kAsioSupportsTimeInfo: + // Informs the driver whether the + // asioCallbacks.bufferSwitchTimeInfo() callback is supported. + // For compatibility with ASIO 1.0 drivers the host application + // should always support the "old" bufferSwitch method, too. + ret = 0; + break; + case kAsioSupportsTimeCode: + // Informs the driver whether application is interested in time + // code info. If an application does not need to know about time + // code, the driver has less work to do. + ret = 0; + break; + } + return ret; + } - return "Unknown error."; -} -//******************** End of __WINDOWS_ASIO__ *********************// + static const char* getAsioErrorString( ASIOError result ) + { + struct Messages + { + ASIOError value; + const char*message; + }; + + static Messages m[] = + { + { ASE_NotPresent, "Hardware input or output is not present or available." }, + { ASE_HWMalfunction, "Hardware is malfunctioning." }, + { ASE_InvalidParameter, "Invalid input parameter." }, + { ASE_InvalidMode, "Invalid mode." }, + { ASE_SPNotAdvancing, "Sample position not advancing." }, + { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." }, + { ASE_NoMemory, "Not enough memory to complete the request." } + }; + + for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i ) + if ( m[i].value == result ) return m[i].message; + + return "Unknown error."; + } + //******************** End of __WINDOWS_ASIO__ *********************// #endif #if defined(__WINDOWS_DS__) // Windows DirectSound API -// Modified by Robin Davies, October 2005 -// - Improvements to DirectX pointer chasing. -// - Backdoor RtDsStatistics hook provides DirectX performance information. -// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30. -// - Auto-call CoInitialize for DSOUND and ASIO platforms. -// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007 + // Modified by Robin Davies, October 2005 + // - Improvements to DirectX pointer chasing. + // - Backdoor RtDsStatistics hook provides DirectX performance information. + // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30. + // - Auto-call CoInitialize for DSOUND and ASIO platforms. + // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007 #include #include #if defined(__MINGW32__) -// missing from latest mingw winapi + // missing from latest mingw winapi #define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */ #define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */ #define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */ @@ -3281,1587 +3382,1587 @@ static const char* getAsioErrorString( ASIOError result ) #pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually. #endif -static inline DWORD dsPointerDifference( DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) -{ - if ( laterPointer > earlierPointer ) - return laterPointer - earlierPointer; - else - return laterPointer - earlierPointer + bufferSize; -} - -static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) -{ - if ( pointer > bufferSize ) pointer -= bufferSize; - if ( laterPointer < earlierPointer ) laterPointer += bufferSize; - if ( pointer < earlierPointer ) pointer += bufferSize; - return pointer >= earlierPointer && pointer < laterPointer; -} - -// A structure to hold various information related to the DirectSound -// API implementation. -struct DsHandle { - unsigned int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - void *id[2]; - void *buffer[2]; - bool xrun[2]; - UINT bufferPointer[2]; - DWORD dsBufferSize[2]; - DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by. - HANDLE condition; + static inline DWORD dsPointerDifference( DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) + { + if ( laterPointer > earlierPointer ) + return laterPointer - earlierPointer; + else + return laterPointer - earlierPointer + bufferSize; + } - DsHandle() - :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; } -}; + static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) + { + if ( pointer > bufferSize ) pointer -= bufferSize; + if ( laterPointer < earlierPointer ) laterPointer += bufferSize; + if ( pointer < earlierPointer ) pointer += bufferSize; + return pointer >= earlierPointer && pointer < laterPointer; + } + + // A structure to hold various information related to the DirectSound + // API implementation. + struct DsHandle { + unsigned int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + void *id[2]; + void *buffer[2]; + bool xrun[2]; + UINT bufferPointer[2]; + DWORD dsBufferSize[2]; + DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by. + HANDLE condition; + + DsHandle() + :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; } + }; -/* -RtApiDs::RtDsStatistics RtApiDs::statistics; + /* + RtApiDs::RtDsStatistics RtApiDs::statistics; -// Provides a backdoor hook to monitor for DirectSound read overruns and write underruns. -RtApiDs::RtDsStatistics RtApiDs::getDsStatistics() -{ - RtDsStatistics s = statistics; + // Provides a backdoor hook to monitor for DirectSound read overruns and write underruns. + RtApiDs::RtDsStatistics RtApiDs::getDsStatistics() + { + RtDsStatistics s = statistics; - // update the calculated fields. - if ( s.inputFrameSize != 0 ) + // update the calculated fields. + if ( s.inputFrameSize != 0 ) s.latency += s.readDeviceSafeLeadBytes * 1.0 / s.inputFrameSize / s.sampleRate; - if ( s.outputFrameSize != 0 ) + if ( s.outputFrameSize != 0 ) s.latency += (s.writeDeviceSafeLeadBytes + s.writeDeviceBufferLeadBytes) * 1.0 / s.outputFrameSize / s.sampleRate; - return s; -} -*/ - -// Declarations for utility functions, callbacks, and structures -// specific to the DirectSound implementation. -static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, - LPCTSTR description, - LPCTSTR module, - LPVOID lpContext ); - -static char* getErrorString( int code ); - -extern "C" unsigned __stdcall callbackHandler( void *ptr ); - -struct EnumInfo { - bool isInput; - bool getDefault; - bool findIndex; - unsigned int counter; - unsigned int index; - LPGUID id; - std::string name; - - EnumInfo() - : isInput(false), getDefault(false), findIndex(false), counter(0), index(0) {} -}; - -RtApiDs :: RtApiDs() -{ - // Dsound will run both-threaded. If CoInitialize fails, then just - // accept whatever the mainline chose for a threading model. - coInitialized_ = false; - HRESULT hr = CoInitialize( NULL ); - if ( !FAILED( hr ) ) coInitialized_ = true; -} - -RtApiDs :: ~RtApiDs() -{ - if ( coInitialized_ ) CoUninitialize(); // balanced call. - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} - -unsigned int RtApiDs :: getDefaultInputDevice( void ) -{ - // Count output devices. - EnumInfo info; - HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") counting output devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return 0; - } - - // Now enumerate input devices until we find the id = NULL. - info.isInput = true; - info.getDefault = true; - result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDefaultInputDevice: error (" << getErrorString( result ) << ") enumerating input devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return 0; - } - - if ( info.counter > 0 ) return info.counter - 1; - return 0; -} - -unsigned int RtApiDs :: getDefaultOutputDevice( void ) -{ - // Enumerate output devices until we find the id = NULL. - EnumInfo info; - info.getDefault = true; - HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") enumerating output devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return 0; - } - - if ( info.counter > 0 ) return info.counter - 1; - return 0; -} - -unsigned int RtApiDs :: getDeviceCount( void ) -{ - // Count DirectSound devices. - EnumInfo info; - HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } - - // Count DirectSoundCapture devices. - info.isInput = true; - result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } - - return info.counter; -} - -RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device ) -{ - // Because DirectSound always enumerates input and output devices - // separately (and because we don't attempt to combine devices - // internally), none of our "devices" will ever be duplex. - - RtAudio::DeviceInfo info; - info.probed = false; + return s; + } + */ - // Enumerate through devices to find the id (if it exists). Note - // that we have to do the output enumeration first, even if this is - // an input device, in order for the device counter to be correct. - EnumInfo dsinfo; - dsinfo.findIndex = true; - dsinfo.index = device; - HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating output devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } + // Declarations for utility functions, callbacks, and structures + // specific to the DirectSound implementation. + static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, + LPCTSTR description, + LPCTSTR module, + LPVOID lpContext ); - if ( dsinfo.name.empty() ) goto probeInput; + static char* getErrorString( int code ); - LPDIRECTSOUND output; - DSCAPS outCaps; - result = DirectSoundCreate( dsinfo.id, &output, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } + extern "C" unsigned __stdcall callbackHandler( void *ptr ); - outCaps.dwSize = sizeof( outCaps ); - result = output->GetCaps( &outCaps ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } + struct EnumInfo { + bool isInput; + bool getDefault; + bool findIndex; + unsigned int counter; + unsigned int index; + LPGUID id; + std::string name; - // Get output channel information. - info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; + EnumInfo() + : isInput(false), getDefault(false), findIndex(false), counter(0), index(0) {} + }; - // Get sample rate information. - info.sampleRates.clear(); - for ( unsigned int k=0; k= (unsigned int) outCaps.dwMinSecondarySampleRate && - SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) - info.sampleRates.push_back( SAMPLE_RATES[k] ); + RtApiDs :: RtApiDs() + { + // Dsound will run both-threaded. If CoInitialize fails, then just + // accept whatever the mainline chose for a threading model. + coInitialized_ = false; + HRESULT hr = CoInitialize( NULL ); + if ( !FAILED( hr ) ) coInitialized_ = true; } - // Get format information. - if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16; - if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8; - - output->Release(); - - if ( getDefaultOutputDevice() == device ) - info.isDefaultOutput = true; - - // Copy name and return. - info.name = dsinfo.name; - - info.probed = true; - return info; - - probeInput: - - dsinfo.isInput = true; - result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating input devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); + RtApiDs :: ~RtApiDs() + { + if ( coInitialized_ ) CoUninitialize(); // balanced call. + if ( stream_.state != STREAM_CLOSED ) closeStream(); } - if ( dsinfo.name.empty() ) return info; + unsigned int RtApiDs :: getDefaultInputDevice( void ) + { + // Count output devices. + EnumInfo info; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") counting output devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return 0; + } - LPDIRECTSOUNDCAPTURE input; - result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } + // Now enumerate input devices until we find the id = NULL. + info.isInput = true; + info.getDefault = true; + result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDefaultInputDevice: error (" << getErrorString( result ) << ") enumerating input devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return 0; + } - DSCCAPS inCaps; - inCaps.dwSize = sizeof( inCaps ); - result = input->GetCaps( &inCaps ); - if ( FAILED( result ) ) { - input->Release(); - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; + if ( info.counter > 0 ) return info.counter - 1; + return 0; } - // Get input channel information. - info.inputChannels = inCaps.dwChannels; - - // Get sample rate and format information. - if ( inCaps.dwChannels == 2 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8; - - if ( info.nativeFormats & RTAUDIO_SINT16 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.sampleRates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.sampleRates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.sampleRates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.sampleRates.push_back( 96000 ); - } - else if ( info.nativeFormats & RTAUDIO_SINT8 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.sampleRates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.sampleRates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.sampleRates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.sampleRates.push_back( 44100 ); - } - } - else if ( inCaps.dwChannels == 1 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8; - - if ( info.nativeFormats & RTAUDIO_SINT16 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.sampleRates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.sampleRates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.sampleRates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.sampleRates.push_back( 96000 ); - } - else if ( info.nativeFormats & RTAUDIO_SINT8 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.sampleRates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.sampleRates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.sampleRates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.sampleRates.push_back( 96000 ); - } - } - else info.inputChannels = 0; // technically, this would be an error - - input->Release(); - - if ( info.inputChannels == 0 ) return info; - - if ( getDefaultInputDevice() == device ) - info.isDefaultInput = true; - - // Copy name and return. - info.name = dsinfo.name; - info.probed = true; - return info; -} - -bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - if ( channels + firstChannel > 2 ) { - errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device."; - return FAILURE; - } + unsigned int RtApiDs :: getDefaultOutputDevice( void ) + { + // Enumerate output devices until we find the id = NULL. + EnumInfo info; + info.getDefault = true; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") enumerating output devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return 0; + } - // Enumerate through devices to find the id (if it exists). Note - // that we have to do the output enumeration first, even if this is - // an input device, in order for the device counter to be correct. - EnumInfo dsinfo; - dsinfo.findIndex = true; - dsinfo.index = device; - HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating output devices!"; - errorText_ = errorStream_.str(); - return FAILURE; + if ( info.counter > 0 ) return info.counter - 1; + return 0; } - if ( mode == OUTPUT ) { - if ( dsinfo.name.empty() ) { - errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!"; + unsigned int RtApiDs :: getDeviceCount( void ) + { + // Count DirectSound devices. + EnumInfo info; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!"; errorText_ = errorStream_.str(); - return FAILURE; + error( RtError::WARNING ); } - } - else { // mode == INPUT - dsinfo.isInput = true; - HRESULT result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); + + // Count DirectSoundCapture devices. + info.isInput = true; + result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating input devices!"; + errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!"; errorText_ = errorStream_.str(); - return FAILURE; + error( RtError::WARNING ); } - if ( dsinfo.name.empty() ) { - errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!"; + + return info.counter; + } + + RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device ) + { + // Because DirectSound always enumerates input and output devices + // separately (and because we don't attempt to combine devices + // internally), none of our "devices" will ever be duplex. + + RtAudio::DeviceInfo info; + info.probed = false; + + // Enumerate through devices to find the id (if it exists). Note + // that we have to do the output enumeration first, even if this is + // an input device, in order for the device counter to be correct. + EnumInfo dsinfo; + dsinfo.findIndex = true; + dsinfo.index = device; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating output devices!"; errorText_ = errorStream_.str(); - return FAILURE; + error( RtError::WARNING ); } - } - // According to a note in PortAudio, using GetDesktopWindow() - // instead of GetForegroundWindow() is supposed to avoid problems - // that occur when the application's window is not the foreground - // window. Also, if the application window closes before the - // DirectSound buffer, DirectSound can crash. However, for console - // applications, no sound was produced when using GetDesktopWindow(). - HWND hWnd = GetForegroundWindow(); - - // Check the numberOfBuffers parameter and limit the lowest value to - // two. This is a judgement call and a value of two is probably too - // low for capture, but it should work for playback. - int nBuffers = 0; - if ( options ) nBuffers = options->numberOfBuffers; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2; - if ( nBuffers < 2 ) nBuffers = 3; - - // Create the wave format structure. The data format setting will - // be determined later. - WAVEFORMATEX waveFormat; - ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) ); - waveFormat.wFormatTag = WAVE_FORMAT_PCM; - waveFormat.nChannels = channels + firstChannel; - waveFormat.nSamplesPerSec = (unsigned long) sampleRate; - - // Determine the device buffer size. By default, 32k, but we will - // grow it to make allowances for very large software buffer sizes. - DWORD dsBufferSize = 0; - DWORD dsPointerLeadTime = 0; - long bufferBytes = MINIMUM_DEVICE_BUFFER_SIZE; // sound cards will always *knock wood* support this - - void *ohandle = 0, *bhandle = 0; - if ( mode == OUTPUT ) { + if ( dsinfo.name.empty() ) goto probeInput; LPDIRECTSOUND output; + DSCAPS outCaps; result = DirectSoundCreate( dsinfo.id, &output, NULL ); if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!"; + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!"; errorText_ = errorStream_.str(); - return FAILURE; + error( RtError::WARNING ); + return info; } - DSCAPS outCaps; outCaps.dwSize = sizeof( outCaps ); result = output->GetCaps( &outCaps ); if ( FAILED( result ) ) { output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsinfo.name << ")!"; + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!"; errorText_ = errorStream_.str(); - return FAILURE; + error( RtError::WARNING ); + return info; } - // Check channel information. - if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsinfo.name << ") does not support stereo playback."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Get output channel information. + info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; - // Check format information. Use 16-bit format unless not - // supported or user requests 8-bit. - if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT && - !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) { - waveFormat.wBitsPerSample = 16; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - else { - waveFormat.wBitsPerSample = 8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; + // Get sample rate information. + info.sampleRates.clear(); + for ( unsigned int k=0; k= (unsigned int) outCaps.dwMinSecondarySampleRate && + SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) + info.sampleRates.push_back( SAMPLE_RATES[k] ); } - stream_.userFormat = format; - // Update wave format structure and buffer information. - waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; - waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; - dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; + // Get format information. + if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16; + if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8; + + output->Release(); + + if ( getDefaultOutputDevice() == device ) + info.isDefaultOutput = true; + + // Copy name and return. + info.name = dsinfo.name; - // If the user wants an even bigger buffer, increase the device buffer size accordingly. - while ( dsPointerLeadTime * 2U > (DWORD) bufferBytes ) - bufferBytes *= 2; + info.probed = true; + return info; - // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes. - //result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE ); - // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes. - result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY ); + probeInput: + + dsinfo.isInput = true; + result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsinfo.name << ")!"; + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating input devices!"; errorText_ = errorStream_.str(); - return FAILURE; + error( RtError::WARNING ); } - // Even though we will write to the secondary buffer, we need to - // access the primary buffer to set the correct output format - // (since the default is 8-bit, 22 kHz!). Setup the DS primary - // buffer description. - DSBUFFERDESC bufferDescription; - ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); - bufferDescription.dwSize = sizeof( DSBUFFERDESC ); - bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER; + if ( dsinfo.name.empty() ) return info; - // Obtain the primary buffer - LPDIRECTSOUNDBUFFER buffer; - result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsinfo.name << ")!"; + LPDIRECTSOUNDCAPTURE input; + result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!"; errorText_ = errorStream_.str(); - return FAILURE; + error( RtError::WARNING ); + return info; } - // Set the primary DS buffer sound format. - result = buffer->SetFormat( &waveFormat ); + DSCCAPS inCaps; + inCaps.dwSize = sizeof( inCaps ); + result = input->GetCaps( &inCaps ); if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsinfo.name << ")!"; + input->Release(); + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsinfo.name << ")!"; errorText_ = errorStream_.str(); - return FAILURE; + error( RtError::WARNING ); + return info; } - // Setup the secondary DS buffer description. - dsBufferSize = (DWORD) bufferBytes; - ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); - bufferDescription.dwSize = sizeof( DSBUFFERDESC ); - bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | - DSBCAPS_GLOBALFOCUS | - DSBCAPS_GETCURRENTPOSITION2 | - DSBCAPS_LOCHARDWARE ); // Force hardware mixing - bufferDescription.dwBufferBytes = bufferBytes; - bufferDescription.lpwfxFormat = &waveFormat; - - // Try to create the secondary DS buffer. If that doesn't work, - // try to use software mixing. Otherwise, there's a problem. - result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | - DSBCAPS_GLOBALFOCUS | - DSBCAPS_GETCURRENTPOSITION2 | - DSBCAPS_LOCSOFTWARE ); // Force software mixing - result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; + // Get input channel information. + info.inputChannels = inCaps.dwChannels; + + // Get sample rate and format information. + if ( inCaps.dwChannels == 2 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8; + + if ( info.nativeFormats & RTAUDIO_SINT16 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.sampleRates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.sampleRates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.sampleRates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.sampleRates.push_back( 96000 ); + } + else if ( info.nativeFormats & RTAUDIO_SINT8 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.sampleRates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.sampleRates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.sampleRates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.sampleRates.push_back( 44100 ); } } - - // Get the buffer size ... might be different from what we specified. - DSBCAPS dsbcaps; - dsbcaps.dwSize = sizeof( DSBCAPS ); - result = buffer->GetCaps( &dsbcaps ); - if ( FAILED( result ) ) { - output->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; + else if ( inCaps.dwChannels == 1 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8; + + if ( info.nativeFormats & RTAUDIO_SINT16 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.sampleRates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.sampleRates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.sampleRates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.sampleRates.push_back( 96000 ); + } + else if ( info.nativeFormats & RTAUDIO_SINT8 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.sampleRates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.sampleRates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.sampleRates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.sampleRates.push_back( 96000 ); + } } + else info.inputChannels = 0; // technically, this would be an error - bufferBytes = dsbcaps.dwBufferBytes; - - // Lock the DS buffer - LPVOID audioPtr; - DWORD dataLen; - result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - output->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + input->Release(); - // Zero the DS buffer - ZeroMemory( audioPtr, dataLen ); + if ( info.inputChannels == 0 ) return info; - // Unlock the DS buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - output->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + if ( getDefaultInputDevice() == device ) + info.isDefaultInput = true; - dsBufferSize = bufferBytes; - ohandle = (void *) output; - bhandle = (void *) buffer; + // Copy name and return. + info.name = dsinfo.name; + info.probed = true; + return info; } - if ( mode == INPUT ) { - - LPDIRECTSOUNDCAPTURE input; - result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); + bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) + { + if ( channels + firstChannel > 2 ) { + errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device."; return FAILURE; } - DSCCAPS inCaps; - inCaps.dwSize = sizeof( inCaps ); - result = input->GetCaps( &inCaps ); + // Enumerate through devices to find the id (if it exists). Note + // that we have to do the output enumeration first, even if this is + // an input device, in order for the device counter to be correct. + EnumInfo dsinfo; + dsinfo.findIndex = true; + dsinfo.index = device; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); if ( FAILED( result ) ) { - input->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsinfo.name << ")!"; + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating output devices!"; errorText_ = errorStream_.str(); return FAILURE; } - // Check channel information. - if ( inCaps.dwChannels < channels + firstChannel ) { - errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels."; - return FAILURE; + if ( mode == OUTPUT ) { + if ( dsinfo.name.empty() ) { + errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + else { // mode == INPUT + dsinfo.isInput = true; + HRESULT result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating input devices!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + if ( dsinfo.name.empty() ) { + errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!"; + errorText_ = errorStream_.str(); + return FAILURE; + } } - // Check format information. Use 16-bit format unless user - // requests 8-bit. - DWORD deviceFormats; - if ( channels + firstChannel == 2 ) { - deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08; - if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { - waveFormat.wBitsPerSample = 8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; + // According to a note in PortAudio, using GetDesktopWindow() + // instead of GetForegroundWindow() is supposed to avoid problems + // that occur when the application's window is not the foreground + // window. Also, if the application window closes before the + // DirectSound buffer, DirectSound can crash. However, for console + // applications, no sound was produced when using GetDesktopWindow(). + HWND hWnd = GetForegroundWindow(); + + // Check the numberOfBuffers parameter and limit the lowest value to + // two. This is a judgement call and a value of two is probably too + // low for capture, but it should work for playback. + int nBuffers = 0; + if ( options ) nBuffers = options->numberOfBuffers; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2; + if ( nBuffers < 2 ) nBuffers = 3; + + // Create the wave format structure. The data format setting will + // be determined later. + WAVEFORMATEX waveFormat; + ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) ); + waveFormat.wFormatTag = WAVE_FORMAT_PCM; + waveFormat.nChannels = channels + firstChannel; + waveFormat.nSamplesPerSec = (unsigned long) sampleRate; + + // Determine the device buffer size. By default, 32k, but we will + // grow it to make allowances for very large software buffer sizes. + DWORD dsBufferSize = 0; + DWORD dsPointerLeadTime = 0; + long bufferBytes = MINIMUM_DEVICE_BUFFER_SIZE; // sound cards will always *knock wood* support this + + void *ohandle = 0, *bhandle = 0; + if ( mode == OUTPUT ) { + + LPDIRECTSOUND output; + result = DirectSoundCreate( dsinfo.id, &output, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + + DSCAPS outCaps; + outCaps.dwSize = sizeof( outCaps ); + result = output->GetCaps( &outCaps ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; } - else { // assume 16-bit is supported + + // Check channel information. + if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsinfo.name << ") does not support stereo playback."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Check format information. Use 16-bit format unless not + // supported or user requests 8-bit. + if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT && + !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) { waveFormat.wBitsPerSample = 16; stream_.deviceFormat[mode] = RTAUDIO_SINT16; } - } - else { // channel == 1 - deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08; - if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { + else { waveFormat.wBitsPerSample = 8; stream_.deviceFormat[mode] = RTAUDIO_SINT8; } - else { // assume 16-bit is supported - waveFormat.wBitsPerSample = 16; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - } - stream_.userFormat = format; + stream_.userFormat = format; - // Update wave format structure and buffer information. - waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; - waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; - - // Setup the secondary DS buffer description. - dsBufferSize = bufferBytes; - DSCBUFFERDESC bufferDescription; - ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) ); - bufferDescription.dwSize = sizeof( DSCBUFFERDESC ); - bufferDescription.dwFlags = 0; - bufferDescription.dwReserved = 0; - bufferDescription.dwBufferBytes = bufferBytes; - bufferDescription.lpwfxFormat = &waveFormat; - - // Create the capture buffer. - LPDIRECTSOUNDCAPTUREBUFFER buffer; - result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - input->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Update wave format structure and buffer information. + waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; + waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; - // Lock the capture buffer - LPVOID audioPtr; - DWORD dataLen; - result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - input->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + // If the user wants an even bigger buffer, increase the device buffer size accordingly. + while ( dsPointerLeadTime * 2U > (DWORD) bufferBytes ) + bufferBytes *= 2; - // Zero the buffer - ZeroMemory( audioPtr, dataLen ); + // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes. + //result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE ); + // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes. + result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Unlock the buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - input->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Even though we will write to the secondary buffer, we need to + // access the primary buffer to set the correct output format + // (since the default is 8-bit, 22 kHz!). Setup the DS primary + // buffer description. + DSBUFFERDESC bufferDescription; + ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSBUFFERDESC ); + bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER; + + // Obtain the primary buffer + LPDIRECTSOUNDBUFFER buffer; + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - dsBufferSize = bufferBytes; - ohandle = (void *) input; - bhandle = (void *) buffer; - } + // Set the primary DS buffer sound format. + result = buffer->SetFormat( &waveFormat ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Set various stream parameters - DsHandle *handle = 0; - stream_.nDeviceChannels[mode] = channels + firstChannel; - stream_.nUserChannels[mode] = channels; - stream_.bufferSize = *bufferSize; - stream_.channelOffset[mode] = firstChannel; - stream_.deviceInterleaved[mode] = true; - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; + // Setup the secondary DS buffer description. + dsBufferSize = (DWORD) bufferBytes; + ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSBUFFERDESC ); + bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | + DSBCAPS_GLOBALFOCUS | + DSBCAPS_GETCURRENTPOSITION2 | + DSBCAPS_LOCHARDWARE ); // Force hardware mixing + bufferDescription.dwBufferBytes = bufferBytes; + bufferDescription.lpwfxFormat = &waveFormat; - // Set flag for buffer conversion - stream_.doConvertBuffer[mode] = false; - if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode]) - stream_.doConvertBuffer[mode] = true; - if (stream_.userFormat != stream_.deviceFormat[mode]) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; + // Try to create the secondary DS buffer. If that doesn't work, + // try to use software mixing. Otherwise, there's a problem. + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | + DSBCAPS_GLOBALFOCUS | + DSBCAPS_GETCURRENTPOSITION2 | + DSBCAPS_LOCSOFTWARE ); // Force software mixing + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } - // Allocate necessary internal buffers - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } + // Get the buffer size ... might be different from what we specified. + DSBCAPS dsbcaps; + dsbcaps.dwSize = sizeof( DSBCAPS ); + result = buffer->GetCaps( &dsbcaps ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - if ( stream_.doConvertBuffer[mode] ) { + bufferBytes = dsbcaps.dwBufferBytes; - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= (long) bytesOut ) makeBuffer = false; + // Lock the DS buffer + LPVOID audioPtr; + DWORD dataLen; + result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; } - } - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory."; - goto error; + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; } - } - } - // Allocate our DsHandle structures for the stream. - if ( stream_.apiHandle == 0 ) { - try { - handle = new DsHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory."; - goto error; + dsBufferSize = bufferBytes; + ohandle = (void *) output; + bhandle = (void *) buffer; } - // Create a manual-reset event. - handle->condition = CreateEvent( NULL, // no security - TRUE, // manual-reset - FALSE, // non-signaled initially - NULL ); // unnamed - stream_.apiHandle = (void *) handle; - } - else - handle = (DsHandle *) stream_.apiHandle; - handle->id[mode] = ohandle; - handle->buffer[mode] = bhandle; - handle->dsBufferSize[mode] = dsBufferSize; - handle->dsPointerLeadTime[mode] = dsPointerLeadTime; + if ( mode == INPUT ) { - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - if ( stream_.mode == OUTPUT && mode == INPUT ) - // We had already set up an output stream. - stream_.mode = DUPLEX; - else - stream_.mode = mode; - stream_.nBuffers = nBuffers; - stream_.sampleRate = sampleRate; + LPDIRECTSOUNDCAPTURE input; + result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + DSCCAPS inCaps; + inCaps.dwSize = sizeof( inCaps ); + result = input->GetCaps( &inCaps ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Setup the callback thread. - unsigned threadId; - stream_.callbackInfo.object = (void *) this; - stream_.callbackInfo.isRunning = true; - stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler, - &stream_.callbackInfo, 0, &threadId ); - if ( stream_.callbackInfo.thread == 0 ) { - errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!"; - goto error; - } + // Check channel information. + if ( inCaps.dwChannels < channels + firstChannel ) { + errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels."; + return FAILURE; + } - // Boost DS thread priority - SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST ); - return SUCCESS; + // Check format information. Use 16-bit format unless user + // requests 8-bit. + DWORD deviceFormats; + if ( channels + firstChannel == 2 ) { + deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08; + if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + else { // assume 16-bit is supported + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + } + else { // channel == 1 + deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08; + if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } + else { // assume 16-bit is supported + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + } + stream_.userFormat = format; + + // Update wave format structure and buffer information. + waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; + waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + + // Setup the secondary DS buffer description. + dsBufferSize = bufferBytes; + DSCBUFFERDESC bufferDescription; + ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSCBUFFERDESC ); + bufferDescription.dwFlags = 0; + bufferDescription.dwReserved = 0; + bufferDescription.dwBufferBytes = bufferBytes; + bufferDescription.lpwfxFormat = &waveFormat; + + // Create the capture buffer. + LPDIRECTSOUNDCAPTUREBUFFER buffer; + result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - error: - if ( handle ) { - if ( handle->buffer[0] ) { // the object pointer can be NULL and valid - LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - if ( buffer ) buffer->Release(); - object->Release(); - } - if ( handle->buffer[1] ) { - LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - if ( buffer ) buffer->Release(); - object->Release(); - } - CloseHandle( handle->condition ); - delete handle; - stream_.apiHandle = 0; - } + // Lock the capture buffer + LPVOID audioPtr; + DWORD dataLen; + result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } + // Zero the buffer + ZeroMemory( audioPtr, dataLen ); - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } + // Unlock the buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - return FAILURE; -} + dsBufferSize = bufferBytes; + ohandle = (void *) input; + bhandle = (void *) buffer; + } + + // Set various stream parameters + DsHandle *handle = 0; + stream_.nDeviceChannels[mode] = channels + firstChannel; + stream_.nUserChannels[mode] = channels; + stream_.bufferSize = *bufferSize; + stream_.channelOffset[mode] = firstChannel; + stream_.deviceInterleaved[mode] = true; + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; + + // Set flag for buffer conversion + stream_.doConvertBuffer[mode] = false; + if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode]) + stream_.doConvertBuffer[mode] = true; + if (stream_.userFormat != stream_.deviceFormat[mode]) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate necessary internal buffers + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } -void RtApiDs :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiDs::closeStream(): no open stream to close!"; - error( RtError::WARNING ); - return; - } + if ( stream_.doConvertBuffer[mode] ) { - // Stop the callback thread. - stream_.callbackInfo.isRunning = false; - WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE ); - CloseHandle( (HANDLE) stream_.callbackInfo.thread ); + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= (long) bytesOut ) makeBuffer = false; + } + } - DsHandle *handle = (DsHandle *) stream_.apiHandle; - if ( handle ) { - if ( handle->buffer[0] ) { // the object pointer can be NULL and valid - LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - if ( buffer ) { - buffer->Stop(); - buffer->Release(); + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } } - object->Release(); } - if ( handle->buffer[1] ) { - LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - if ( buffer ) { - buffer->Stop(); - buffer->Release(); + + // Allocate our DsHandle structures for the stream. + if ( stream_.apiHandle == 0 ) { + try { + handle = new DsHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory."; + goto error; } - object->Release(); - } - CloseHandle( handle->condition ); - delete handle; - stream_.apiHandle = 0; - } - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; + // Create a manual-reset event. + handle->condition = CreateEvent( NULL, // no security + TRUE, // manual-reset + FALSE, // non-signaled initially + NULL ); // unnamed + stream_.apiHandle = (void *) handle; } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } + else + handle = (DsHandle *) stream_.apiHandle; + handle->id[mode] = ohandle; + handle->buffer[mode] = bhandle; + handle->dsBufferSize[mode] = dsBufferSize; + handle->dsPointerLeadTime[mode] = dsPointerLeadTime; - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + if ( stream_.mode == OUTPUT && mode == INPUT ) + // We had already set up an output stream. + stream_.mode = DUPLEX; + else + stream_.mode = mode; + stream_.nBuffers = nBuffers; + stream_.sampleRate = sampleRate; -void RtApiDs :: startStream() -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiDs::startStream(): the stream is already running!"; - error( RtError::WARNING ); - return; - } + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - // Increase scheduler frequency on lesser windows (a side-effect of - // increasing timer accuracy). On greater windows (Win2K or later), - // this is already in effect. + // Setup the callback thread. + unsigned threadId; + stream_.callbackInfo.object = (void *) this; + stream_.callbackInfo.isRunning = true; + stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler, + &stream_.callbackInfo, 0, &threadId ); + if ( stream_.callbackInfo.thread == 0 ) { + errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!"; + goto error; + } - MUTEX_LOCK( &stream_.mutex ); - - DsHandle *handle = (DsHandle *) stream_.apiHandle; + // Boost DS thread priority + SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST ); + return SUCCESS; - timeBeginPeriod( 1 ); + error: + if ( handle ) { + if ( handle->buffer[0] ) { // the object pointer can be NULL and valid + LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + if ( buffer ) buffer->Release(); + object->Release(); + } + if ( handle->buffer[1] ) { + LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + if ( buffer ) buffer->Release(); + object->Release(); + } + CloseHandle( handle->condition ); + delete handle; + stream_.apiHandle = 0; + } - /* - memset( &statistics, 0, sizeof( statistics ) ); - statistics.sampleRate = stream_.sampleRate; - statistics.writeDeviceBufferLeadBytes = handle->dsPointerLeadTime[0]; - */ + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } - buffersRolling = false; - duplexPrerollBytes = 0; + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } - if ( stream_.mode == DUPLEX ) { - // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize. - duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] ); + return FAILURE; } - HRESULT result = 0; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - //statistics.outputFrameSize = formatBytes( stream_.deviceFormat[0] ) * stream_.nDeviceChannels[0]; + void RtApiDs :: closeStream() + { + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiDs::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - result = buffer->Play( 0, 0, DSBPLAY_LOOPING ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; + // Stop the callback thread. + stream_.callbackInfo.isRunning = false; + WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE ); + CloseHandle( (HANDLE) stream_.callbackInfo.thread ); + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + if ( handle ) { + if ( handle->buffer[0] ) { // the object pointer can be NULL and valid + LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + if ( buffer ) { + buffer->Stop(); + buffer->Release(); + } + object->Release(); + } + if ( handle->buffer[1] ) { + LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + if ( buffer ) { + buffer->Stop(); + buffer->Release(); + } + object->Release(); + } + CloseHandle( handle->condition ); + delete handle; + stream_.apiHandle = 0; } - } - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - //statistics.inputFrameSize = formatBytes( stream_.deviceFormat[1]) * stream_.nDeviceChannels[1]; + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - result = buffer->Start( DSCBSTART_LOOPING ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; } + + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; } - handle->drainCounter = 0; - handle->internalDrain = false; - stream_.state = STREAM_RUNNING; + void RtApiDs :: startStream() + { + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiDs::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } - unlock: - MUTEX_UNLOCK( &stream_.mutex ); + // Increase scheduler frequency on lesser windows (a side-effect of + // increasing timer accuracy). On greater windows (Win2K or later), + // this is already in effect. - if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); -} + MUTEX_LOCK( &stream_.mutex ); + + DsHandle *handle = (DsHandle *) stream_.apiHandle; -void RtApiDs :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } + timeBeginPeriod( 1 ); - MUTEX_LOCK( &stream_.mutex ); + /* + memset( &statistics, 0, sizeof( statistics ) ); + statistics.sampleRate = stream_.sampleRate; + statistics.writeDeviceBufferLeadBytes = handle->dsPointerLeadTime[0]; + */ - HRESULT result = 0; - LPVOID audioPtr; - DWORD dataLen; - DsHandle *handle = (DsHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 1; - MUTEX_UNLOCK( &stream_.mutex ); - WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled - ResetEvent( handle->condition ); - MUTEX_LOCK( &stream_.mutex ); - } + buffersRolling = false; + duplexPrerollBytes = 0; - // Stop the buffer and clear memory - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - result = buffer->Stop(); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; + if ( stream_.mode == DUPLEX ) { + // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize. + duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] ); } - // Lock the buffer and clear it so that if we start to play again, - // we won't have old data playing. - result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; + HRESULT result = 0; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + //statistics.outputFrameSize = formatBytes( stream_.deviceFormat[0] ) * stream_.nDeviceChannels[0]; + + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + result = buffer->Play( 0, 0, DSBPLAY_LOOPING ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } } - // Zero the DS buffer - ZeroMemory( audioPtr, dataLen ); + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + //statistics.inputFrameSize = formatBytes( stream_.deviceFormat[1]) * stream_.nDeviceChannels[1]; - // Unlock the DS buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + result = buffer->Start( DSCBSTART_LOOPING ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } } - // If we start playing again, we must begin at beginning of buffer. - handle->bufferPointer[0] = 0; - } + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - audioPtr = NULL; - dataLen = 0; + unlock: + MUTEX_UNLOCK( &stream_.mutex ); - result = buffer->Stop(); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } + if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); + } - // Lock the buffer and clear it so that if we start to play again, - // we won't have old data playing. - result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; + void RtApiDs :: stopStream() + { + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; } - // Zero the DS buffer - ZeroMemory( audioPtr, dataLen ); + MUTEX_LOCK( &stream_.mutex ); - // Unlock the DS buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; + HRESULT result = 0; + LPVOID audioPtr; + DWORD dataLen; + DsHandle *handle = (DsHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 1; + MUTEX_UNLOCK( &stream_.mutex ); + WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled + ResetEvent( handle->condition ); + MUTEX_LOCK( &stream_.mutex ); + } + + // Stop the buffer and clear memory + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + result = buffer->Stop(); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); + + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } + + // If we start playing again, we must begin at beginning of buffer. + handle->bufferPointer[0] = 0; } - // If we start recording again, we must begin at beginning of buffer. - handle->bufferPointer[1] = 0; - } + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + audioPtr = NULL; + dataLen = 0; - unlock: - timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); - if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); -} + result = buffer->Stop(); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } -void RtApiDs :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiDs::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } - DsHandle *handle = (DsHandle *) stream_.apiHandle; - handle->drainCounter = 1; + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); - stopStream(); -} + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } -void RtApiDs :: callbackEvent() -{ - if ( stream_.state == STREAM_STOPPED ) { - Sleep(50); // sleep 50 milliseconds - return; - } + // If we start recording again, we must begin at beginning of buffer. + handle->bufferPointer[1] = 0; + } - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); - return; + unlock: + timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); } - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - DsHandle *handle = (DsHandle *) stream_.apiHandle; + void RtApiDs :: abortStream() + { + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiDs::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } - // Check if we were draining the stream and signal is finished. - if ( handle->drainCounter > stream_.nBuffers + 2 ) { - if ( handle->internalDrain == false ) - SetEvent( handle->condition ); - else - stopStream(); - return; - } + DsHandle *handle = (DsHandle *) stream_.apiHandle; + handle->drainCounter = 1; - MUTEX_LOCK( &stream_.mutex ); + stopStream(); + } - // Invoke user callback to get fresh output data UNLESS we are - // draining stream. - if ( handle->drainCounter == 0 ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; + void RtApiDs :: callbackEvent() + { + if ( stream_.state == STREAM_STOPPED ) { + Sleep(50); // sleep 50 milliseconds + return; } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return; } - handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( handle->drainCounter == 2 ) { - MUTEX_UNLOCK( &stream_.mutex ); - abortStream(); + + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + DsHandle *handle = (DsHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > stream_.nBuffers + 2 ) { + if ( handle->internalDrain == false ) + SetEvent( handle->condition ); + else + stopStream(); return; } - else if ( handle->drainCounter == 1 ) - handle->internalDrain = true; - } - HRESULT result; - DWORD currentWritePos, safeWritePos; - DWORD currentReadPos, safeReadPos; - DWORD leadPos; - UINT nextWritePos; + MUTEX_LOCK( &stream_.mutex ); + + // Invoke user callback to get fresh output data UNLESS we are + // draining stream. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( handle->drainCounter == 2 ) { + MUTEX_UNLOCK( &stream_.mutex ); + abortStream(); + return; + } + else if ( handle->drainCounter == 1 ) + handle->internalDrain = true; + } + + HRESULT result; + DWORD currentWritePos, safeWritePos; + DWORD currentReadPos, safeReadPos; + DWORD leadPos; + UINT nextWritePos; #ifdef GENERATE_DEBUG_LOG - DWORD writeTime, readTime; + DWORD writeTime, readTime; #endif - LPVOID buffer1 = NULL; - LPVOID buffer2 = NULL; - DWORD bufferSize1 = 0; - DWORD bufferSize2 = 0; + LPVOID buffer1 = NULL; + LPVOID buffer2 = NULL; + DWORD bufferSize1 = 0; + DWORD bufferSize2 = 0; - char *buffer; - long bufferBytes; + char *buffer; + long bufferBytes; - if ( stream_.mode == DUPLEX && !buffersRolling ) { - assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); + if ( stream_.mode == DUPLEX && !buffersRolling ) { + assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); - // It takes a while for the devices to get rolling. As a result, - // there's no guarantee that the capture and write device pointers - // will move in lockstep. Wait here for both devices to start - // rolling, and then set our buffer pointers accordingly. - // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600 - // bytes later than the write buffer. + // It takes a while for the devices to get rolling. As a result, + // there's no guarantee that the capture and write device pointers + // will move in lockstep. Wait here for both devices to start + // rolling, and then set our buffer pointers accordingly. + // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600 + // bytes later than the write buffer. - // Stub: a serious risk of having a pre-emptive scheduling round - // take place between the two GetCurrentPosition calls... but I'm - // really not sure how to solve the problem. Temporarily boost to - // Realtime priority, maybe; but I'm not sure what priority the - // DirectSound service threads run at. We *should* be roughly - // within a ms or so of correct. + // Stub: a serious risk of having a pre-emptive scheduling round + // take place between the two GetCurrentPosition calls... but I'm + // really not sure how to solve the problem. Temporarily boost to + // Realtime priority, maybe; but I'm not sure what priority the + // DirectSound service threads run at. We *should* be roughly + // within a ms or so of correct. - LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - DWORD initialWritePos, initialSafeWritePos; - DWORD initialReadPos, initialSafeReadPos; + DWORD initialWritePos, initialSafeWritePos; + DWORD initialReadPos, initialSafeReadPos; - result = dsWriteBuffer->GetCurrentPosition( &initialWritePos, &initialSafeWritePos ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - result = dsCaptureBuffer->GetCurrentPosition( &initialReadPos, &initialSafeReadPos ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - while ( true ) { - result = dsWriteBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos ); + result = dsWriteBuffer->GetCurrentPosition( &initialWritePos, &initialSafeWritePos ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; errorText_ = errorStream_.str(); error( RtError::SYSTEM_ERROR ); } - result = dsCaptureBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); + result = dsCaptureBuffer->GetCurrentPosition( &initialReadPos, &initialSafeReadPos ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; errorText_ = errorStream_.str(); error( RtError::SYSTEM_ERROR ); } - if ( safeWritePos != initialSafeWritePos && safeReadPos != initialSafeReadPos ) break; - Sleep( 1 ); - } + while ( true ) { + result = dsWriteBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + result = dsCaptureBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + if ( safeWritePos != initialSafeWritePos && safeReadPos != initialSafeReadPos ) break; + Sleep( 1 ); + } - assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); + assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); - buffersRolling = true; - handle->bufferPointer[0] = ( safeWritePos + handle->dsPointerLeadTime[0] ); - handle->bufferPointer[1] = safeReadPos; - } + buffersRolling = true; + handle->bufferPointer[0] = ( safeWritePos + handle->dsPointerLeadTime[0] ); + handle->bufferPointer[1] = safeReadPos; + } - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - if ( handle->drainCounter > 1 ) { // write zeros to the output stream - bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; - bufferBytes *= formatBytes( stream_.userFormat ); - memset( stream_.userBuffer[0], 0, bufferBytes ); - } + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; + bufferBytes *= formatBytes( stream_.userFormat ); + memset( stream_.userBuffer[0], 0, bufferBytes ); + } - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0]; - bufferBytes *= formatBytes( stream_.deviceFormat[0] ); - } - else { - buffer = stream_.userBuffer[0]; - bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; - bufferBytes *= formatBytes( stream_.userFormat ); - } + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0]; + bufferBytes *= formatBytes( stream_.deviceFormat[0] ); + } + else { + buffer = stream_.userBuffer[0]; + bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; + bufferBytes *= formatBytes( stream_.userFormat ); + } + + // No byte swapping necessary in DirectSound implementation. + + // Ahhh ... windoze. 16-bit data is signed but 8-bit data is + // unsigned. So, we need to convert our signed 8-bit data here to + // unsigned. + if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 ) + for ( int i=0; idsBufferSize[0]; + nextWritePos = handle->bufferPointer[0]; - // Ahhh ... windoze. 16-bit data is signed but 8-bit data is - // unsigned. So, we need to convert our signed 8-bit data here to - // unsigned. - if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 ) - for ( int i=0; iGetCurrentPosition( ¤tWritePos, &safeWritePos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + + leadPos = safeWritePos + handle->dsPointerLeadTime[0]; + if ( leadPos > dsBufferSize ) leadPos -= dsBufferSize; + if ( leadPos < nextWritePos ) leadPos += dsBufferSize; // unwrap offset + endWrite = nextWritePos + bufferBytes; + + // Check whether the entire write region is behind the play pointer. + if ( leadPos >= endWrite ) break; + + // If we are here, then we must wait until the play pointer gets + // beyond the write region. The approach here is to use the + // Sleep() function to suspend operation until safePos catches + // up. Calculate number of milliseconds to wait as: + // time = distance * (milliseconds/second) * fudgefactor / + // ((bytes/sample) * (samples/second)) + // A "fudgefactor" less than 1 is used because it was found + // that sleeping too long was MUCH worse than sleeping for + // several shorter periods. + double millis = ( endWrite - leadPos ) * 900.0; + millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate); + if ( millis < 1.0 ) millis = 1.0; + if ( millis > 50.0 ) { + static int nOverruns = 0; + ++nOverruns; + } + Sleep( (DWORD) millis ); + } - DWORD dsBufferSize = handle->dsBufferSize[0]; - nextWritePos = handle->bufferPointer[0]; + //if ( statistics.writeDeviceSafeLeadBytes < dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ) ) { + // statistics.writeDeviceSafeLeadBytes = dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ); + //} + + if ( dsPointerBetween( nextWritePos, safeWritePos, currentWritePos, dsBufferSize ) + || dsPointerBetween( endWrite, safeWritePos, currentWritePos, dsBufferSize ) ) { + // We've strayed into the forbidden zone ... resync the read pointer. + //++statistics.numberOfWriteUnderruns; + handle->xrun[0] = true; + nextWritePos = safeWritePos + handle->dsPointerLeadTime[0] - bufferBytes + dsBufferSize; + while ( nextWritePos >= dsBufferSize ) nextWritePos -= dsBufferSize; + handle->bufferPointer[0] = nextWritePos; + endWrite = nextWritePos + bufferBytes; + } - DWORD endWrite; - while ( true ) { - // Find out where the read and "safe write" pointers are. - result = dsBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos ); + // Lock free space in the buffer + result = dsBuffer->Lock( nextWritePos, bufferBytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0 ); if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!"; errorText_ = errorStream_.str(); error( RtError::SYSTEM_ERROR ); } - leadPos = safeWritePos + handle->dsPointerLeadTime[0]; - if ( leadPos > dsBufferSize ) leadPos -= dsBufferSize; - if ( leadPos < nextWritePos ) leadPos += dsBufferSize; // unwrap offset - endWrite = nextWritePos + bufferBytes; - - // Check whether the entire write region is behind the play pointer. - if ( leadPos >= endWrite ) break; - - // If we are here, then we must wait until the play pointer gets - // beyond the write region. The approach here is to use the - // Sleep() function to suspend operation until safePos catches - // up. Calculate number of milliseconds to wait as: - // time = distance * (milliseconds/second) * fudgefactor / - // ((bytes/sample) * (samples/second)) - // A "fudgefactor" less than 1 is used because it was found - // that sleeping too long was MUCH worse than sleeping for - // several shorter periods. - double millis = ( endWrite - leadPos ) * 900.0; - millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate); - if ( millis < 1.0 ) millis = 1.0; - if ( millis > 50.0 ) { - static int nOverruns = 0; - ++nOverruns; - } - Sleep( (DWORD) millis ); - } - - //if ( statistics.writeDeviceSafeLeadBytes < dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ) ) { - // statistics.writeDeviceSafeLeadBytes = dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ); - //} - - if ( dsPointerBetween( nextWritePos, safeWritePos, currentWritePos, dsBufferSize ) - || dsPointerBetween( endWrite, safeWritePos, currentWritePos, dsBufferSize ) ) { - // We've strayed into the forbidden zone ... resync the read pointer. - //++statistics.numberOfWriteUnderruns; - handle->xrun[0] = true; - nextWritePos = safeWritePos + handle->dsPointerLeadTime[0] - bufferBytes + dsBufferSize; - while ( nextWritePos >= dsBufferSize ) nextWritePos -= dsBufferSize; - handle->bufferPointer[0] = nextWritePos; - endWrite = nextWritePos + bufferBytes; - } - - // Lock free space in the buffer - result = dsBuffer->Lock( nextWritePos, bufferBytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - - // Copy our buffer into the DS buffer - CopyMemory( buffer1, buffer, bufferSize1 ); - if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 ); - - // Update our buffer offset and unlock sound buffer - dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - nextWritePos = ( nextWritePos + bufferSize1 + bufferSize2 ) % dsBufferSize; - handle->bufferPointer[0] = nextWritePos; - - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } - } + // Copy our buffer into the DS buffer + CopyMemory( buffer1, buffer, bufferSize1 ); + if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 ); - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + // Update our buffer offset and unlock sound buffer + dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + nextWritePos = ( nextWritePos + bufferSize1 + bufferSize2 ) % dsBufferSize; + handle->bufferPointer[0] = nextWritePos; - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1]; - bufferBytes *= formatBytes( stream_.deviceFormat[1] ); - } - else { - buffer = stream_.userBuffer[1]; - bufferBytes = stream_.bufferSize * stream_.nUserChannels[1]; - bufferBytes *= formatBytes( stream_.userFormat ); + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } } - LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - long nextReadPos = handle->bufferPointer[1]; - DWORD dsBufferSize = handle->dsBufferSize[1]; + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - // Find out where the write and "safe read" pointers are. - result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1]; + bufferBytes *= formatBytes( stream_.deviceFormat[1] ); + } + else { + buffer = stream_.userBuffer[1]; + bufferBytes = stream_.bufferSize * stream_.nUserChannels[1]; + bufferBytes *= formatBytes( stream_.userFormat ); + } - if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset - DWORD endRead = nextReadPos + bufferBytes; + LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + long nextReadPos = handle->bufferPointer[1]; + DWORD dsBufferSize = handle->dsBufferSize[1]; - // Handling depends on whether we are INPUT or DUPLEX. - // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode, - // then a wait here will drag the write pointers into the forbidden zone. - // - // In DUPLEX mode, rather than wait, we will back off the read pointer until - // it's in a safe position. This causes dropouts, but it seems to be the only - // practical way to sync up the read and write pointers reliably, given the - // the very complex relationship between phase and increment of the read and write - // pointers. - // - // In order to minimize audible dropouts in DUPLEX mode, we will - // provide a pre-roll period of 0.5 seconds in which we return - // zeros from the read buffer while the pointers sync up. + // Find out where the write and "safe read" pointers are. + result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } - if ( stream_.mode == DUPLEX ) { - if ( safeReadPos < endRead ) { - if ( duplexPrerollBytes <= 0 ) { - // Pre-roll time over. Be more agressive. - int adjustment = endRead-safeReadPos; - - handle->xrun[1] = true; - //++statistics.numberOfReadOverruns; - // Two cases: - // - large adjustments: we've probably run out of CPU cycles, so just resync exactly, - // and perform fine adjustments later. - // - small adjustments: back off by twice as much. - if ( adjustment >= 2*bufferBytes ) - nextReadPos = safeReadPos-2*bufferBytes; - else - nextReadPos = safeReadPos-bufferBytes-adjustment; - - //statistics.readDeviceSafeLeadBytes = currentReadPos-nextReadPos; - //if ( statistics.readDeviceSafeLeadBytes < 0) statistics.readDeviceSafeLeadBytes += dsBufferSize; - if ( nextReadPos < 0 ) nextReadPos += dsBufferSize; + if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset + DWORD endRead = nextReadPos + bufferBytes; + + // Handling depends on whether we are INPUT or DUPLEX. + // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode, + // then a wait here will drag the write pointers into the forbidden zone. + // + // In DUPLEX mode, rather than wait, we will back off the read pointer until + // it's in a safe position. This causes dropouts, but it seems to be the only + // practical way to sync up the read and write pointers reliably, given the + // the very complex relationship between phase and increment of the read and write + // pointers. + // + // In order to minimize audible dropouts in DUPLEX mode, we will + // provide a pre-roll period of 0.5 seconds in which we return + // zeros from the read buffer while the pointers sync up. + + if ( stream_.mode == DUPLEX ) { + if ( safeReadPos < endRead ) { + if ( duplexPrerollBytes <= 0 ) { + // Pre-roll time over. Be more agressive. + int adjustment = endRead-safeReadPos; + + handle->xrun[1] = true; + //++statistics.numberOfReadOverruns; + // Two cases: + // - large adjustments: we've probably run out of CPU cycles, so just resync exactly, + // and perform fine adjustments later. + // - small adjustments: back off by twice as much. + if ( adjustment >= 2*bufferBytes ) + nextReadPos = safeReadPos-2*bufferBytes; + else + nextReadPos = safeReadPos-bufferBytes-adjustment; + + //statistics.readDeviceSafeLeadBytes = currentReadPos-nextReadPos; + //if ( statistics.readDeviceSafeLeadBytes < 0) statistics.readDeviceSafeLeadBytes += dsBufferSize; + if ( nextReadPos < 0 ) nextReadPos += dsBufferSize; + } + else { + // In pre=roll time. Just do it. + nextReadPos = safeReadPos-bufferBytes; + while ( nextReadPos < 0 ) nextReadPos += dsBufferSize; + } + endRead = nextReadPos + bufferBytes; } - else { - // In pre=roll time. Just do it. - nextReadPos = safeReadPos-bufferBytes; - while ( nextReadPos < 0 ) nextReadPos += dsBufferSize; - } - endRead = nextReadPos + bufferBytes; } - } - else { // mode == INPUT - while ( safeReadPos < endRead ) { - // See comments for playback. - double millis = (endRead - safeReadPos) * 900.0; - millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate); - if ( millis < 1.0 ) millis = 1.0; - Sleep( (DWORD) millis ); - - // Wake up, find out where we are now - result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } + else { // mode == INPUT + while ( safeReadPos < endRead ) { + // See comments for playback. + double millis = (endRead - safeReadPos) * 900.0; + millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate); + if ( millis < 1.0 ) millis = 1.0; + Sleep( (DWORD) millis ); + + // Wake up, find out where we are now + result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } - if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset + if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset + } } - } - //if (statistics.readDeviceSafeLeadBytes < dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ) ) - // statistics.readDeviceSafeLeadBytes = dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ); + //if (statistics.readDeviceSafeLeadBytes < dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ) ) + // statistics.readDeviceSafeLeadBytes = dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ); - // Lock free space in the buffer - result = dsBuffer->Lock( nextReadPos, bufferBytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } + // Lock free space in the buffer + result = dsBuffer->Lock( nextReadPos, bufferBytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } - if ( duplexPrerollBytes <= 0 ) { - // Copy our buffer into the DS buffer - CopyMemory( buffer, buffer1, bufferSize1 ); - if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 ); - } - else { - memset( buffer, 0, bufferSize1 ); - if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 ); - duplexPrerollBytes -= bufferSize1 + bufferSize2; - } + if ( duplexPrerollBytes <= 0 ) { + // Copy our buffer into the DS buffer + CopyMemory( buffer, buffer1, bufferSize1 ); + if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 ); + } + else { + memset( buffer, 0, bufferSize1 ); + if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 ); + duplexPrerollBytes -= bufferSize1 + bufferSize2; + } - // Update our buffer offset and unlock sound buffer - nextReadPos = ( nextReadPos + bufferSize1 + bufferSize2 ) % dsBufferSize; - dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - handle->bufferPointer[1] = nextReadPos; + // Update our buffer offset and unlock sound buffer + nextReadPos = ( nextReadPos + bufferSize1 + bufferSize2 ) % dsBufferSize; + dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + handle->bufferPointer[1] = nextReadPos; - // No byte swapping necessary in DirectSound implementation. + // No byte swapping necessary in DirectSound implementation. - // If necessary, convert 8-bit data from unsigned to signed. - if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 ) - for ( int j=0; jobject; - bool* isRunning = &info->isRunning; + extern "C" unsigned __stdcall callbackHandler( void *ptr ) + { + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiDs *object = (RtApiDs *) info->object; + bool* isRunning = &info->isRunning; - while ( *isRunning == true ) { - object->callbackEvent(); - } + while ( *isRunning == true ) { + object->callbackEvent(); + } - _endthreadex( 0 ); - return 0; -} + _endthreadex( 0 ); + return 0; + } #include "tchar.h" -std::string convertTChar( LPCTSTR name ) -{ - std::string s; + std::string convertTChar( LPCTSTR name ) + { + std::string s; #if defined( UNICODE ) || defined( _UNICODE ) - // Yes, this conversion doesn't make sense for two-byte characters - // but RtAudio is currently written to return an std::string of - // one-byte chars for the device name. - for ( unsigned int i=0; iisInput == true ) { - DSCCAPS caps; - LPDIRECTSOUNDCAPTURE object; + HRESULT hr; + if ( info->isInput == true ) { + DSCCAPS caps; + LPDIRECTSOUNDCAPTURE object; - hr = DirectSoundCaptureCreate( lpguid, &object, NULL ); - if ( hr != DS_OK ) return TRUE; + hr = DirectSoundCaptureCreate( lpguid, &object, NULL ); + if ( hr != DS_OK ) return TRUE; - caps.dwSize = sizeof(caps); - hr = object->GetCaps( &caps ); - if ( hr == DS_OK ) { - if ( caps.dwChannels > 0 && caps.dwFormats > 0 ) - info->counter++; + caps.dwSize = sizeof(caps); + hr = object->GetCaps( &caps ); + if ( hr == DS_OK ) { + if ( caps.dwChannels > 0 && caps.dwFormats > 0 ) + info->counter++; + } + object->Release(); } - object->Release(); - } - else { - DSCAPS caps; - LPDIRECTSOUND object; - hr = DirectSoundCreate( lpguid, &object, NULL ); - if ( hr != DS_OK ) return TRUE; - - caps.dwSize = sizeof(caps); - hr = object->GetCaps( &caps ); - if ( hr == DS_OK ) { - if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO ) - info->counter++; + else { + DSCAPS caps; + LPDIRECTSOUND object; + hr = DirectSoundCreate( lpguid, &object, NULL ); + if ( hr != DS_OK ) return TRUE; + + caps.dwSize = sizeof(caps); + hr = object->GetCaps( &caps ); + if ( hr == DS_OK ) { + if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO ) + info->counter++; + } + object->Release(); } - object->Release(); - } - if ( info->getDefault && lpguid == NULL ) return FALSE; + if ( info->getDefault && lpguid == NULL ) return FALSE; - if ( info->findIndex && info->counter > info->index ) { - info->id = lpguid; - info->name = convertTChar( description ); - return FALSE; - } + if ( info->findIndex && info->counter > info->index ) { + info->id = lpguid; + info->name = convertTChar( description ); + return FALSE; + } - return TRUE; -} + return TRUE; + } -static char* getErrorString( int code ) -{ - switch ( code ) { + static char* getErrorString( int code ) + { + switch ( code ) { - case DSERR_ALLOCATED: - return "Already allocated"; + case DSERR_ALLOCATED: + return "Already allocated"; - case DSERR_CONTROLUNAVAIL: - return "Control unavailable"; + case DSERR_CONTROLUNAVAIL: + return "Control unavailable"; - case DSERR_INVALIDPARAM: - return "Invalid parameter"; + case DSERR_INVALIDPARAM: + return "Invalid parameter"; - case DSERR_INVALIDCALL: - return "Invalid call"; + case DSERR_INVALIDCALL: + return "Invalid call"; - case DSERR_GENERIC: - return "Generic error"; + case DSERR_GENERIC: + return "Generic error"; - case DSERR_PRIOLEVELNEEDED: - return "Priority level needed"; + case DSERR_PRIOLEVELNEEDED: + return "Priority level needed"; - case DSERR_OUTOFMEMORY: - return "Out of memory"; + case DSERR_OUTOFMEMORY: + return "Out of memory"; - case DSERR_BADFORMAT: - return "The sample rate or the channel format is not supported"; + case DSERR_BADFORMAT: + return "The sample rate or the channel format is not supported"; - case DSERR_UNSUPPORTED: - return "Not supported"; + case DSERR_UNSUPPORTED: + return "Not supported"; - case DSERR_NODRIVER: - return "No driver"; + case DSERR_NODRIVER: + return "No driver"; - case DSERR_ALREADYINITIALIZED: - return "Already initialized"; + case DSERR_ALREADYINITIALIZED: + return "Already initialized"; - case DSERR_NOAGGREGATION: - return "No aggregation"; + case DSERR_NOAGGREGATION: + return "No aggregation"; - case DSERR_BUFFERLOST: - return "Buffer lost"; + case DSERR_BUFFERLOST: + return "Buffer lost"; - case DSERR_OTHERAPPHASPRIO: - return "Another application already has priority"; + case DSERR_OTHERAPPHASPRIO: + return "Another application already has priority"; - case DSERR_UNINITIALIZED: - return "Uninitialized"; + case DSERR_UNINITIALIZED: + return "Uninitialized"; - default: - return "DirectSound unknown error"; - } -} -//******************** End of __WINDOWS_DS__ *********************// + default: + return "DirectSound unknown error"; + } + } + //******************** End of __WINDOWS_DS__ *********************// #endif @@ -4870,1222 +4971,1222 @@ static char* getErrorString( int code ) #include #include -// A structure to hold various information related to the ALSA API -// implementation. -struct AlsaHandle { - snd_pcm_t *handles[2]; - bool synchronized; - bool xrun[2]; - pthread_cond_t runnable; + // A structure to hold various information related to the ALSA API + // implementation. + struct AlsaHandle { + snd_pcm_t *handles[2]; + bool synchronized; + bool xrun[2]; + pthread_cond_t runnable; - AlsaHandle() - :synchronized(false) { xrun[0] = false; xrun[1] = false; } -}; + AlsaHandle() + :synchronized(false) { xrun[0] = false; xrun[1] = false; } + }; -extern "C" void *alsaCallbackHandler( void * ptr ); + extern "C" void *alsaCallbackHandler( void * ptr ); -RtApiAlsa :: RtApiAlsa() -{ - // Nothing to do here. -} + RtApiAlsa :: RtApiAlsa() + { + // Nothing to do here. + } -RtApiAlsa :: ~RtApiAlsa() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} + RtApiAlsa :: ~RtApiAlsa() + { + if ( stream_.state != STREAM_CLOSED ) closeStream(); + } -unsigned int RtApiAlsa :: getDeviceCount( void ) -{ - unsigned nDevices = 0; - int result, subdevice, card; - char name[64]; - snd_ctl_t *handle; - - // Count cards and devices - card = -1; - snd_card_next( &card ); - while ( card >= 0 ) { - sprintf( name, "hw:%d", card ); - result = snd_ctl_open( &handle, name, 0 ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto nextcard; - } - subdevice = -1; - while( 1 ) { - result = snd_ctl_pcm_next_device( handle, &subdevice ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << "."; + unsigned int RtApiAlsa :: getDeviceCount( void ) + { + unsigned nDevices = 0; + int result, subdevice, card; + char name[64]; + snd_ctl_t *handle; + + // Count cards and devices + card = -1; + snd_card_next( &card ); + while ( card >= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &handle, name, 0 ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtError::WARNING ); - break; + goto nextcard; } - if ( subdevice < 0 ) - break; - nDevices++; + subdevice = -1; + while( 1 ) { + result = snd_ctl_pcm_next_device( handle, &subdevice ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + break; + } + if ( subdevice < 0 ) + break; + nDevices++; + } + nextcard: + snd_ctl_close( handle ); + snd_card_next( &card ); } - nextcard: - snd_ctl_close( handle ); - snd_card_next( &card ); + + return nDevices; } - return nDevices; -} + RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) + { + RtAudio::DeviceInfo info; + info.probed = false; -RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; + unsigned nDevices = 0; + int result, subdevice, card; + char name[64]; + snd_ctl_t *chandle; - unsigned nDevices = 0; - int result, subdevice, card; - char name[64]; - snd_ctl_t *chandle; - - // Count cards and devices - card = -1; - snd_card_next( &card ); - while ( card >= 0 ) { - sprintf( name, "hw:%d", card ); - result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto nextcard; - } - subdevice = -1; - while( 1 ) { - result = snd_ctl_pcm_next_device( chandle, &subdevice ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << "."; + // Count cards and devices + card = -1; + snd_card_next( &card ); + while ( card >= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtError::WARNING ); - break; + goto nextcard; } - if ( subdevice < 0 ) break; - if ( nDevices == device ) { - sprintf( name, "hw:%d,%d", card, subdevice ); - goto foundDevice; + subdevice = -1; + while( 1 ) { + result = snd_ctl_pcm_next_device( chandle, &subdevice ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + break; + } + if ( subdevice < 0 ) break; + if ( nDevices == device ) { + sprintf( name, "hw:%d,%d", card, subdevice ); + goto foundDevice; + } + nDevices++; } - nDevices++; + nextcard: + snd_ctl_close( chandle ); + snd_card_next( &card ); } - nextcard: - snd_ctl_close( chandle ); - snd_card_next( &card ); - } - if ( nDevices == 0 ) { - errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!"; - error( RtError::INVALID_USE ); - } + if ( nDevices == 0 ) { + errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!"; + error( RtError::INVALID_USE ); + } - if ( device >= nDevices ) { - errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - } + if ( device >= nDevices ) { + errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } - foundDevice: + foundDevice: - // If a stream is already open, we cannot probe the stream devices. - // Thus, use the saved results. - if ( stream_.state != STREAM_CLOSED && - ( stream_.device[0] == device || stream_.device[1] == device ) ) { - if ( device >= devices_.size() ) { - errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened."; - error( RtError::WARNING ); - return info; + // If a stream is already open, we cannot probe the stream devices. + // Thus, use the saved results. + if ( stream_.state != STREAM_CLOSED && + ( stream_.device[0] == device || stream_.device[1] == device ) ) { + if ( device >= devices_.size() ) { + errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened."; + error( RtError::WARNING ); + return info; + } + return devices_[ device ]; } - return devices_[ device ]; - } - int openMode = SND_PCM_ASYNC; - snd_pcm_stream_t stream; - snd_pcm_info_t *pcminfo; - snd_pcm_info_alloca( &pcminfo ); - snd_pcm_t *phandle; - snd_pcm_hw_params_t *params; - snd_pcm_hw_params_alloca( ¶ms ); + int openMode = SND_PCM_ASYNC; + snd_pcm_stream_t stream; + snd_pcm_info_t *pcminfo; + snd_pcm_info_alloca( &pcminfo ); + snd_pcm_t *phandle; + snd_pcm_hw_params_t *params; + snd_pcm_hw_params_alloca( ¶ms ); + + // First try for playback + stream = SND_PCM_STREAM_PLAYBACK; + snd_pcm_info_set_device( pcminfo, subdevice ); + snd_pcm_info_set_subdevice( pcminfo, 0 ); + snd_pcm_info_set_stream( pcminfo, stream ); - // First try for playback - stream = SND_PCM_STREAM_PLAYBACK; - snd_pcm_info_set_device( pcminfo, subdevice ); - snd_pcm_info_set_subdevice( pcminfo, 0 ); - snd_pcm_info_set_stream( pcminfo, stream ); + result = snd_ctl_pcm_info( chandle, pcminfo ); + if ( result < 0 ) { + // Device probably doesn't support playback. + goto captureProbe; + } - result = snd_ctl_pcm_info( chandle, pcminfo ); - if ( result < 0 ) { - // Device probably doesn't support playback. - goto captureProbe; - } + result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + goto captureProbe; + } - result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto captureProbe; - } + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + goto captureProbe; + } - // The device is open ... fill the parameter structure. - result = snd_pcm_hw_params_any( phandle, params ); - if ( result < 0 ) { + // Get output channel information. + unsigned int value; + result = snd_pcm_hw_params_get_channels_max( params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + goto captureProbe; + } + info.outputChannels = value; snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto captureProbe; - } - // Get output channel information. - unsigned int value; - result = snd_pcm_hw_params_get_channels_max( params, &value ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto captureProbe; - } - info.outputChannels = value; - snd_pcm_close( phandle ); + captureProbe: + // Now try for capture + stream = SND_PCM_STREAM_CAPTURE; + snd_pcm_info_set_stream( pcminfo, stream ); - captureProbe: - // Now try for capture - stream = SND_PCM_STREAM_CAPTURE; - snd_pcm_info_set_stream( pcminfo, stream ); + result = snd_ctl_pcm_info( chandle, pcminfo ); + snd_ctl_close( chandle ); + if ( result < 0 ) { + // Device probably doesn't support capture. + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } - result = snd_ctl_pcm_info( chandle, pcminfo ); - snd_ctl_close( chandle ); - if ( result < 0 ) { - // Device probably doesn't support capture. - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } + result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } - result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } - // The device is open ... fill the parameter structure. - result = snd_pcm_hw_params_any( phandle, params ); - if ( result < 0 ) { + result = snd_pcm_hw_params_get_channels_max( params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + info.inputChannels = value; snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } - result = snd_pcm_hw_params_get_channels_max( params, &value ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } - info.inputChannels = value; - snd_pcm_close( phandle ); + // If device opens for both playback and capture, we determine the channels. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - // If device opens for both playback and capture, we determine the channels. - if ( info.outputChannels > 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + // ALSA doesn't provide default devices so we'll use the first available one. + if ( device == 0 && info.outputChannels > 0 ) + info.isDefaultOutput = true; + if ( device == 0 && info.inputChannels > 0 ) + info.isDefaultInput = true; + + probeParameters: + // At this point, we just need to figure out the supported data + // formats and sample rates. We'll proceed by opening the device in + // the direction with the maximum number of channels, or playback if + // they are equal. This might limit our sample rate options, but so + // be it. + + if ( info.outputChannels >= info.inputChannels ) + stream = SND_PCM_STREAM_PLAYBACK; + else + stream = SND_PCM_STREAM_CAPTURE; + snd_pcm_info_set_stream( pcminfo, stream ); - // ALSA doesn't provide default devices so we'll use the first available one. - if ( device == 0 && info.outputChannels > 0 ) - info.isDefaultOutput = true; - if ( device == 0 && info.inputChannels > 0 ) - info.isDefaultInput = true; + result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } - probeParameters: - // At this point, we just need to figure out the supported data - // formats and sample rates. We'll proceed by opening the device in - // the direction with the maximum number of channels, or playback if - // they are equal. This might limit our sample rate options, but so - // be it. + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } - if ( info.outputChannels >= info.inputChannels ) - stream = SND_PCM_STREAM_PLAYBACK; - else - stream = SND_PCM_STREAM_CAPTURE; - snd_pcm_info_set_stream( pcminfo, stream ); + // Test our discrete set of sample rate values. + info.sampleRates.clear(); + for ( unsigned int i=0; i= 0 ) + sprintf( name, "hw:%s,%d", cardname, subdevice ); + info.name = name; - // Test our discrete set of sample rate values. - info.sampleRates.clear(); - for ( unsigned int i=0; i= 0 ) - sprintf( name, "hw:%s,%d", cardname, subdevice ); - info.name = name; - - // That's all ... close the device and return - snd_pcm_close( phandle ); - info.probed = true; - return info; -} - -void RtApiAlsa :: saveDeviceInfo( void ) -{ - devices_.clear(); + void RtApiAlsa :: saveDeviceInfo( void ) + { + devices_.clear(); - unsigned int nDevices = getDeviceCount(); - devices_.resize( nDevices ); - for ( unsigned int i=0; i= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + subdevice = -1; + while( 1 ) { + result = snd_ctl_pcm_next_device( chandle, &subdevice ); + if ( result < 0 ) break; + if ( subdevice < 0 ) break; + if ( nDevices == device ) { + sprintf( name, "hw:%d,%d", card, subdevice ); + snd_ctl_close( chandle ); + goto foundDevice; + } + nDevices++; + } + snd_ctl_close( chandle ); + snd_card_next( &card ); + } - // Count cards and devices - card = -1; - snd_card_next( &card ); - while ( card >= 0 ) { - sprintf( name, "hw:%d", card ); - result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!"; return FAILURE; - } - subdevice = -1; - while( 1 ) { - result = snd_ctl_pcm_next_device( chandle, &subdevice ); - if ( result < 0 ) break; - if ( subdevice < 0 ) break; - if ( nDevices == device ) { - sprintf( name, "hw:%d,%d", card, subdevice ); - snd_ctl_close( chandle ); - goto foundDevice; - } - nDevices++; } - snd_ctl_close( chandle ); - snd_card_next( &card ); - } - - if ( nDevices == 0 ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!"; - return FAILURE; - } - if ( device >= nDevices ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } - - foundDevice: + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } - // The getDeviceInfo() function will not work for a device that is - // already open. Thus, we'll probe the system before opening a - // stream and save the results for use by getDeviceInfo(). - if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once - this->saveDeviceInfo(); + foundDevice: - snd_pcm_stream_t stream; - if ( mode == OUTPUT ) - stream = SND_PCM_STREAM_PLAYBACK; - else - stream = SND_PCM_STREAM_CAPTURE; + // The getDeviceInfo() function will not work for a device that is + // already open. Thus, we'll probe the system before opening a + // stream and save the results for use by getDeviceInfo(). + if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once + this->saveDeviceInfo(); - snd_pcm_t *phandle; - int openMode = SND_PCM_ASYNC; - result = snd_pcm_open( &phandle, name, stream, openMode ); - if ( result < 0 ) { + snd_pcm_stream_t stream; if ( mode == OUTPUT ) - errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output."; + stream = SND_PCM_STREAM_PLAYBACK; else - errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input."; - errorText_ = errorStream_.str(); - return FAILURE; - } + stream = SND_PCM_STREAM_CAPTURE; - // Fill the parameter structure. - snd_pcm_hw_params_t *hw_params; - snd_pcm_hw_params_alloca( &hw_params ); - result = snd_pcm_hw_params_any( phandle, hw_params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } + snd_pcm_t *phandle; + int openMode = SND_PCM_ASYNC; + result = snd_pcm_open( &phandle, name, stream, openMode ); + if ( result < 0 ) { + if ( mode == OUTPUT ) + errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output."; + else + errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Fill the parameter structure. + snd_pcm_hw_params_t *hw_params; + snd_pcm_hw_params_alloca( &hw_params ); + result = snd_pcm_hw_params_any( phandle, hw_params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } #if defined(__RTAUDIO_DEBUG__) - fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" ); - snd_pcm_hw_params_dump( hw_params, out ); + fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" ); + snd_pcm_hw_params_dump( hw_params, out ); #endif - // Set access ... check user preference. - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) { - stream_.userInterleaved = false; - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); - if ( result < 0 ) { + // Set access ... check user preference. + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) { + stream_.userInterleaved = false; + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); + if ( result < 0 ) { + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); + stream_.deviceInterleaved[mode] = true; + } + else + stream_.deviceInterleaved[mode] = false; + } + else { + stream_.userInterleaved = true; result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); - stream_.deviceInterleaved[mode] = true; + if ( result < 0 ) { + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); + stream_.deviceInterleaved[mode] = false; + } + else + stream_.deviceInterleaved[mode] = true; } - else - stream_.deviceInterleaved[mode] = false; - } - else { - stream_.userInterleaved = true; - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); + if ( result < 0 ) { - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); - stream_.deviceInterleaved[mode] = false; + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; } - else - stream_.deviceInterleaved[mode] = true; - } - - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - // Determine how to set the device format. - stream_.userFormat = format; - snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN; + // Determine how to set the device format. + stream_.userFormat = format; + snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN; + + if ( format == RTAUDIO_SINT8 ) + deviceFormat = SND_PCM_FORMAT_S8; + else if ( format == RTAUDIO_SINT16 ) + deviceFormat = SND_PCM_FORMAT_S16; + else if ( format == RTAUDIO_SINT24 ) + deviceFormat = SND_PCM_FORMAT_S24; + else if ( format == RTAUDIO_SINT32 ) + deviceFormat = SND_PCM_FORMAT_S32; + else if ( format == RTAUDIO_FLOAT32 ) + deviceFormat = SND_PCM_FORMAT_FLOAT; + else if ( format == RTAUDIO_FLOAT64 ) + deviceFormat = SND_PCM_FORMAT_FLOAT64; + + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) { + stream_.deviceFormat[mode] = format; + goto setFormat; + } - if ( format == RTAUDIO_SINT8 ) - deviceFormat = SND_PCM_FORMAT_S8; - else if ( format == RTAUDIO_SINT16 ) - deviceFormat = SND_PCM_FORMAT_S16; - else if ( format == RTAUDIO_SINT24 ) - deviceFormat = SND_PCM_FORMAT_S24; - else if ( format == RTAUDIO_SINT32 ) - deviceFormat = SND_PCM_FORMAT_S32; - else if ( format == RTAUDIO_FLOAT32 ) - deviceFormat = SND_PCM_FORMAT_FLOAT; - else if ( format == RTAUDIO_FLOAT64 ) + // The user requested format is not natively supported by the device. deviceFormat = SND_PCM_FORMAT_FLOAT64; + if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; + goto setFormat; + } - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) { - stream_.deviceFormat[mode] = format; - goto setFormat; - } - - // The user requested format is not natively supported by the device. - deviceFormat = SND_PCM_FORMAT_FLOAT64; - if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; - goto setFormat; - } - - deviceFormat = SND_PCM_FORMAT_FLOAT; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - goto setFormat; - } - - deviceFormat = SND_PCM_FORMAT_S32; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - goto setFormat; - } + deviceFormat = SND_PCM_FORMAT_FLOAT; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + goto setFormat; + } - deviceFormat = SND_PCM_FORMAT_S24; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - goto setFormat; - } + deviceFormat = SND_PCM_FORMAT_S32; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + goto setFormat; + } - deviceFormat = SND_PCM_FORMAT_S16; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - goto setFormat; - } + deviceFormat = SND_PCM_FORMAT_S24; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + goto setFormat; + } - deviceFormat = SND_PCM_FORMAT_S8; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - goto setFormat; - } + deviceFormat = SND_PCM_FORMAT_S16; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + goto setFormat; + } - // If we get here, no supported format was found. - errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - return FAILURE; + deviceFormat = SND_PCM_FORMAT_S8; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + goto setFormat; + } - setFormat: - result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << "."; + // If we get here, no supported format was found. + errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio."; errorText_ = errorStream_.str(); return FAILURE; - } - // Determine whether byte-swaping is necessary. - stream_.doByteSwap[mode] = false; - if ( deviceFormat != SND_PCM_FORMAT_S8 ) { - result = snd_pcm_format_cpu_endian( deviceFormat ); - if ( result == 0 ) - stream_.doByteSwap[mode] = true; - else if (result < 0) { + setFormat: + result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat ); + if ( result < 0 ) { snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << "."; + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); return FAILURE; } - } - // Set the sample rate. - result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Determine whether byte-swaping is necessary. + stream_.doByteSwap[mode] = false; + if ( deviceFormat != SND_PCM_FORMAT_S8 ) { + result = snd_pcm_format_cpu_endian( deviceFormat ); + if ( result == 0 ) + stream_.doByteSwap[mode] = true; + else if (result < 0) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } - // Determine the number of channels for this device. We support a possible - // minimum device channel number > than the value requested by the user. - stream_.nUserChannels[mode] = channels; - unsigned int value; - result = snd_pcm_hw_params_get_channels_max( hw_params, &value ); - unsigned int deviceChannels = value; - if ( result < 0 || deviceChannels < channels + firstChannel ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Set the sample rate. + result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } - result = snd_pcm_hw_params_get_channels_min( hw_params, &value ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - deviceChannels = value; - if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel; - stream_.nDeviceChannels[mode] = deviceChannels; + // Determine the number of channels for this device. We support a possible + // minimum device channel number > than the value requested by the user. + stream_.nUserChannels[mode] = channels; + unsigned int value; + result = snd_pcm_hw_params_get_channels_max( hw_params, &value ); + unsigned int deviceChannels = value; + if ( result < 0 || deviceChannels < channels + firstChannel ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Set the device channels. - result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } + result = snd_pcm_hw_params_get_channels_min( hw_params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + deviceChannels = value; + if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel; + stream_.nDeviceChannels[mode] = deviceChannels; - // Set the buffer number, which in ALSA is referred to as the "period". - int totalSize, dir; - unsigned int periods = 0; - if ( options ) periods = options->numberOfBuffers; - totalSize = *bufferSize * periods; + // Set the device channels. + result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Set the buffer (or period) size. - snd_pcm_uframes_t periodSize = *bufferSize; - result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - *bufferSize = periodSize; + // Set the buffer number, which in ALSA is referred to as the "period". + int totalSize, dir; + unsigned int periods = 0; + if ( options ) periods = options->numberOfBuffers; + totalSize = *bufferSize * periods; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; - else periods = totalSize / *bufferSize; - // Even though the hardware might allow 1 buffer, it won't work reliably. - if ( periods < 2 ) periods = 2; - result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Set the buffer (or period) size. + snd_pcm_uframes_t periodSize = *bufferSize; + result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + *bufferSize = periodSize; - // If attempting to setup a duplex stream, the bufferSize parameter - // MUST be the same in both directions! - if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { - errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; + else periods = totalSize / *bufferSize; + // Even though the hardware might allow 1 buffer, it won't work reliably. + if ( periods < 2 ) periods = 2; + result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } - stream_.bufferSize = *bufferSize; + // If attempting to setup a duplex stream, the bufferSize parameter + // MUST be the same in both directions! + if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { + errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Install the hardware configuration - result = snd_pcm_hw_params( phandle, hw_params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } + stream_.bufferSize = *bufferSize; + + // Install the hardware configuration + result = snd_pcm_hw_params( phandle, hw_params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } #if defined(__RTAUDIO_DEBUG__) - fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n"); - snd_pcm_hw_params_dump( hw_params, out ); + fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n"); + snd_pcm_hw_params_dump( hw_params, out ); #endif - // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns. - snd_pcm_sw_params_t *sw_params = NULL; - snd_pcm_sw_params_alloca( &sw_params ); - snd_pcm_sw_params_current( phandle, sw_params ); - snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize ); - snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX ); - snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 ); - - // The following two settings were suggested by Theo Veenker - //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize ); - //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 ); - - // here are two options for a fix - //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX ); - snd_pcm_uframes_t val; - snd_pcm_sw_params_get_boundary( sw_params, &val ); - snd_pcm_sw_params_set_silence_size( phandle, sw_params, val ); - - result = snd_pcm_sw_params( phandle, sw_params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns. + snd_pcm_sw_params_t *sw_params = NULL; + snd_pcm_sw_params_alloca( &sw_params ); + snd_pcm_sw_params_current( phandle, sw_params ); + snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize ); + snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX ); + snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 ); + + // The following two settings were suggested by Theo Veenker + //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize ); + //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 ); + + // here are two options for a fix + //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX ); + snd_pcm_uframes_t val; + snd_pcm_sw_params_get_boundary( sw_params, &val ); + snd_pcm_sw_params_set_silence_size( phandle, sw_params, val ); + + result = snd_pcm_sw_params( phandle, sw_params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } #if defined(__RTAUDIO_DEBUG__) - fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n"); - snd_pcm_sw_params_dump( sw_params, out ); + fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n"); + snd_pcm_sw_params_dump( sw_params, out ); #endif - // Set flags for buffer conversion - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; + // Set flags for buffer conversion + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate the ApiHandle if necessary and then save. + AlsaHandle *apiInfo = 0; + if ( stream_.apiHandle == 0 ) { + try { + apiInfo = (AlsaHandle *) new AlsaHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory."; + goto error; + } - // Allocate the ApiHandle if necessary and then save. - AlsaHandle *apiInfo = 0; - if ( stream_.apiHandle == 0 ) { - try { - apiInfo = (AlsaHandle *) new AlsaHandle; + if ( pthread_cond_init( &apiInfo->runnable, NULL ) ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } + + stream_.apiHandle = (void *) apiInfo; + apiInfo->handles[0] = 0; + apiInfo->handles[1] = 0; } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory."; - goto error; + else { + apiInfo = (AlsaHandle *) stream_.apiHandle; } + apiInfo->handles[mode] = phandle; - if ( pthread_cond_init( &apiInfo->runnable, NULL ) ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable."; + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory."; goto error; } - stream_.apiHandle = (void *) apiInfo; - apiInfo->handles[0] = 0; - apiInfo->handles[1] = 0; - } - else { - apiInfo = (AlsaHandle *) stream_.apiHandle; - } - apiInfo->handles[mode] = phandle; + if ( stream_.doConvertBuffer[mode] ) { - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } + } - if ( stream_.doConvertBuffer[mode] ) { + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } + } + } - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; + stream_.sampleRate = sampleRate; + stream_.nBuffers = periods; + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + + // Setup thread if necessary. + if ( stream_.mode == OUTPUT && mode == INPUT ) { + // We had already set up an output stream. + stream_.mode = DUPLEX; + // Link the streams if possible. + apiInfo->synchronized = false; + if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 ) + apiInfo->synchronized = true; + else { + errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices."; + error( RtError::WARNING ); + } + } + else { + stream_.mode = mode; + + // Setup callback thread. + stream_.callbackInfo.object = (void *) this; + + // Set the thread attributes for joinable and realtime scheduling + // priority (optional). The higher priority will only take affect + // if the program is run as root or suid. Note, under Linux + // processes with CAP_SYS_NICE privilege, a user can change + // scheduling policy and priority (thus need not be root). See + // POSIX "capabilities". + pthread_attr_t attr; + pthread_attr_init( &attr ); + pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); +#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { + struct sched_param param; + int priority = options->priority; + int min = sched_get_priority_min( SCHED_RR ); + int max = sched_get_priority_max( SCHED_RR ); + if ( priority < min ) priority = min; + else if ( priority > max ) priority = max; + param.sched_priority = priority; + pthread_attr_setschedparam( &attr, ¶m ); + pthread_attr_setschedpolicy( &attr, SCHED_RR ); + } + else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#endif + + stream_.callbackInfo.isRunning = true; + result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo ); + pthread_attr_destroy( &attr ); + if ( result ) { + stream_.callbackInfo.isRunning = false; + errorText_ = "RtApiAlsa::error creating callback thread!"; + goto error; } } - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory."; - goto error; + return SUCCESS; + + error: + if ( apiInfo ) { + pthread_cond_destroy( &apiInfo->runnable ); + if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); + if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); + delete apiInfo; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; } } - } - stream_.sampleRate = sampleRate; - stream_.nBuffers = periods; - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + return FAILURE; + } - // Setup thread if necessary. - if ( stream_.mode == OUTPUT && mode == INPUT ) { - // We had already set up an output stream. - stream_.mode = DUPLEX; - // Link the streams if possible. - apiInfo->synchronized = false; - if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 ) - apiInfo->synchronized = true; - else { - errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices."; + void RtApiAlsa :: closeStream() + { + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAlsa::closeStream(): no open stream to close!"; error( RtError::WARNING ); + return; } - } - else { - stream_.mode = mode; - // Setup callback thread. - stream_.callbackInfo.object = (void *) this; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + stream_.callbackInfo.isRunning = false; + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) + pthread_cond_signal( &apiInfo->runnable ); + MUTEX_UNLOCK( &stream_.mutex ); + pthread_join( stream_.callbackInfo.thread, NULL ); - // Set the thread attributes for joinable and realtime scheduling - // priority (optional). The higher priority will only take affect - // if the program is run as root or suid. Note, under Linux - // processes with CAP_SYS_NICE privilege, a user can change - // scheduling policy and priority (thus need not be root). See - // POSIX "capabilities". - pthread_attr_t attr; - pthread_attr_init( &attr ); - pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); -#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) - if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { - struct sched_param param; - int priority = options->priority; - int min = sched_get_priority_min( SCHED_RR ); - int max = sched_get_priority_max( SCHED_RR ); - if ( priority < min ) priority = min; - else if ( priority > max ) priority = max; - param.sched_priority = priority; - pthread_attr_setschedparam( &attr, ¶m ); - pthread_attr_setschedpolicy( &attr, SCHED_RR ); + if ( stream_.state == STREAM_RUNNING ) { + stream_.state = STREAM_STOPPED; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) + snd_pcm_drop( apiInfo->handles[0] ); + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) + snd_pcm_drop( apiInfo->handles[1] ); } - else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#endif - stream_.callbackInfo.isRunning = true; - result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo ); - pthread_attr_destroy( &attr ); - if ( result ) { - stream_.callbackInfo.isRunning = false; - errorText_ = "RtApiAlsa::error creating callback thread!"; - goto error; + if ( apiInfo ) { + pthread_cond_destroy( &apiInfo->runnable ); + if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); + if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); + delete apiInfo; + stream_.apiHandle = 0; } - } - - return SUCCESS; - error: - if ( apiInfo ) { - pthread_cond_destroy( &apiInfo->runnable ); - if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); - if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); - delete apiInfo; - stream_.apiHandle = 0; - } + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; } - } - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; } - return FAILURE; -} - -void RtApiAlsa :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAlsa::closeStream(): no open stream to close!"; - error( RtError::WARNING ); - return; - } + void RtApiAlsa :: startStream() + { + // This method calls snd_pcm_prepare if the device isn't already in that state. - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - stream_.callbackInfo.isRunning = false; - MUTEX_LOCK( &stream_.mutex ); - if ( stream_.state == STREAM_STOPPED ) - pthread_cond_signal( &apiInfo->runnable ); - MUTEX_UNLOCK( &stream_.mutex ); - pthread_join( stream_.callbackInfo.thread, NULL ); + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiAlsa::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } - if ( stream_.state == STREAM_RUNNING ) { - stream_.state = STREAM_STOPPED; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) - snd_pcm_drop( apiInfo->handles[0] ); - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) - snd_pcm_drop( apiInfo->handles[1] ); - } + MUTEX_LOCK( &stream_.mutex ); - if ( apiInfo ) { - pthread_cond_destroy( &apiInfo->runnable ); - if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); - if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); - delete apiInfo; - stream_.apiHandle = 0; - } + int result = 0; + snd_pcm_state_t state; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + state = snd_pcm_state( handle[0] ); + if ( state != SND_PCM_STATE_PREPARED ) { + result = snd_pcm_prepare( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } + } - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + state = snd_pcm_state( handle[1] ); + if ( state != SND_PCM_STATE_PREPARED ) { + result = snd_pcm_prepare( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } + } } - } - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } + stream_.state = STREAM_RUNNING; - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} + unlock: + MUTEX_UNLOCK( &stream_.mutex ); -void RtApiAlsa :: startStream() -{ - // This method calls snd_pcm_prepare if the device isn't already in that state. + pthread_cond_signal( &apiInfo->runnable ); - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiAlsa::startStream(): the stream is already running!"; - error( RtError::WARNING ); - return; + if ( result >= 0 ) return; + error( RtError::SYSTEM_ERROR ); } - MUTEX_LOCK( &stream_.mutex ); + void RtApiAlsa :: stopStream() + { + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } - int result = 0; - snd_pcm_state_t state; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - state = snd_pcm_state( handle[0] ); - if ( state != SND_PCM_STATE_PREPARED ) { - result = snd_pcm_prepare( handle[0] ); + // Change the state before the lock to improve shutdown response. + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + int result = 0; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( apiInfo->synchronized ) + result = snd_pcm_drop( handle[0] ); + else + result = snd_pcm_drain( handle[0] ); if ( result < 0 ) { - errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << "."; + errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); goto unlock; } } - } - if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { - state = snd_pcm_state( handle[1] ); - if ( state != SND_PCM_STATE_PREPARED ) { - result = snd_pcm_prepare( handle[1] ); + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + result = snd_pcm_drop( handle[1] ); if ( result < 0 ) { - errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << "."; + errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); goto unlock; } } - } - - stream_.state = STREAM_RUNNING; - - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - pthread_cond_signal( &apiInfo->runnable ); - - if ( result >= 0 ) return; - error( RtError::SYSTEM_ERROR ); -} + unlock: + MUTEX_UNLOCK( &stream_.mutex ); -void RtApiAlsa :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; + if ( result >= 0 ) return; + error( RtError::SYSTEM_ERROR ); } - // Change the state before the lock to improve shutdown response. - stream_.state = STREAM_STOPPED; - MUTEX_LOCK( &stream_.mutex ); + void RtApiAlsa :: abortStream() + { + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } - int result = 0; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( apiInfo->synchronized ) + // Change the state before the lock to improve shutdown response. + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + int result = 0; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { result = snd_pcm_drop( handle[0] ); - else - result = snd_pcm_drain( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } } - } - if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { - result = snd_pcm_drop( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + result = snd_pcm_drop( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; + } } - } - - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - if ( result >= 0 ) return; - error( RtError::SYSTEM_ERROR ); -} + unlock: + MUTEX_UNLOCK( &stream_.mutex ); -void RtApiAlsa :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; + if ( result >= 0 ) return; + error( RtError::SYSTEM_ERROR ); } - // Change the state before the lock to improve shutdown response. - stream_.state = STREAM_STOPPED; - MUTEX_LOCK( &stream_.mutex ); - - int result = 0; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - result = snd_pcm_drop( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; + void RtApiAlsa :: callbackEvent() + { + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + pthread_cond_wait( &apiInfo->runnable, &stream_.mutex ); + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); } - } - if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { - result = snd_pcm_drop( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return; } - } - - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - if ( result >= 0 ) return; - error( RtError::SYSTEM_ERROR ); -} + int doStopStream = 0; + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + apiInfo->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + apiInfo->xrun[1] = false; + } + doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); -void RtApiAlsa :: callbackEvent() -{ - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_LOCK( &stream_.mutex ); - pthread_cond_wait( &apiInfo->runnable, &stream_.mutex ); - if ( stream_.state != STREAM_RUNNING ) { - MUTEX_UNLOCK( &stream_.mutex ); + if ( doStopStream == 2 ) { + abortStream(); return; } - MUTEX_UNLOCK( &stream_.mutex ); - } - - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); - return; - } - - int doStopStream = 0; - RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - apiInfo->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - apiInfo->xrun[1] = false; - } - doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); - if ( doStopStream == 2 ) { - abortStream(); - return; - } + MUTEX_LOCK( &stream_.mutex ); - MUTEX_LOCK( &stream_.mutex ); + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) goto unlock; + int result; + char *buffer; + int channels; + snd_pcm_t **handle; + snd_pcm_sframes_t frames; + RtAudioFormat format; + handle = (snd_pcm_t **) apiInfo->handles; - int result; - char *buffer; - int channels; - snd_pcm_t **handle; - snd_pcm_sframes_t frames; - RtAudioFormat format; - handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + channels = stream_.nDeviceChannels[1]; + format = stream_.deviceFormat[1]; + } + else { + buffer = stream_.userBuffer[1]; + channels = stream_.nUserChannels[1]; + format = stream_.userFormat; + } - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - channels = stream_.nDeviceChannels[1]; - format = stream_.deviceFormat[1]; - } - else { - buffer = stream_.userBuffer[1]; - channels = stream_.nUserChannels[1]; - format = stream_.userFormat; - } + // Read samples from device in interleaved/non-interleaved format. + if ( stream_.deviceInterleaved[1] ) + result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize ); + else { + void *bufs[channels]; + size_t offset = stream_.bufferSize * formatBytes( format ); + for ( int i=0; ixrun[1] = true; - result = snd_pcm_prepare( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << "."; + if ( result < (int) stream_.bufferSize ) { + // Either an error or overrun occured. + if ( result == -EPIPE ) { + snd_pcm_state_t state = snd_pcm_state( handle[1] ); + if ( state == SND_PCM_STATE_XRUN ) { + apiInfo->xrun[1] = true; + result = snd_pcm_prepare( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); } } else { - errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; + errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); } + error( RtError::WARNING ); + goto tryOutput; } - else { - errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - error( RtError::WARNING ); - goto tryOutput; - } - // Do byte swapping if necessary. - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( buffer, stream_.bufferSize * channels, format ); + // Do byte swapping if necessary. + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( buffer, stream_.bufferSize * channels, format ); - // Do buffer conversion if necessary. - if ( stream_.doConvertBuffer[1] ) - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + // Do buffer conversion if necessary. + if ( stream_.doConvertBuffer[1] ) + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - // Check stream latency - result = snd_pcm_delay( handle[1], &frames ); - if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; - } + // Check stream latency + result = snd_pcm_delay( handle[1], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; + } - tryOutput: + tryOutput: - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - channels = stream_.nDeviceChannels[0]; - format = stream_.deviceFormat[0]; - } - else { - buffer = stream_.userBuffer[0]; - channels = stream_.nUserChannels[0]; - format = stream_.userFormat; - } + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + channels = stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + channels = stream_.nUserChannels[0]; + format = stream_.userFormat; + } - // Do byte swapping if necessary. - if ( stream_.doByteSwap[0] ) - byteSwapBuffer(buffer, stream_.bufferSize * channels, format); + // Do byte swapping if necessary. + if ( stream_.doByteSwap[0] ) + byteSwapBuffer(buffer, stream_.bufferSize * channels, format); - // Write samples to device in interleaved/non-interleaved format. - if ( stream_.deviceInterleaved[0] ) - result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize ); - else { - void *bufs[channels]; - size_t offset = stream_.bufferSize * formatBytes( format ); - for ( int i=0; ixrun[0] = true; - result = snd_pcm_prepare( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << "."; + // Write samples to device in interleaved/non-interleaved format. + if ( stream_.deviceInterleaved[0] ) + result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize ); + else { + void *bufs[channels]; + size_t offset = stream_.bufferSize * formatBytes( format ); + for ( int i=0; ixrun[0] = true; + result = snd_pcm_prepare( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + } + else { + errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); } } else { - errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; + errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); } + error( RtError::WARNING ); + goto unlock; } - else { - errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - error( RtError::WARNING ); - goto unlock; + + // Check stream latency + result = snd_pcm_delay( handle[0], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[0] = frames; } - // Check stream latency - result = snd_pcm_delay( handle[0], &frames ); - if ( result == 0 && frames > 0 ) stream_.latency[0] = frames; - } + unlock: + MUTEX_UNLOCK( &stream_.mutex ); - unlock: - MUTEX_UNLOCK( &stream_.mutex ); + RtApi::tickStreamTime(); + if ( doStopStream == 1 ) this->stopStream(); + } - RtApi::tickStreamTime(); - if ( doStopStream == 1 ) this->stopStream(); -} + extern "C" void *alsaCallbackHandler( void *ptr ) + { + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiAlsa *object = (RtApiAlsa *) info->object; + bool *isRunning = &info->isRunning; -extern "C" void *alsaCallbackHandler( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiAlsa *object = (RtApiAlsa *) info->object; - bool *isRunning = &info->isRunning; + while ( *isRunning == true ) { + pthread_testcancel(); + object->callbackEvent(); + } - while ( *isRunning == true ) { - pthread_testcancel(); - object->callbackEvent(); + pthread_exit( NULL ); } - pthread_exit( NULL ); -} - -//******************** End of __LINUX_ALSA__ *********************// + //******************** End of __LINUX_ALSA__ *********************// #endif @@ -6099,1594 +6200,1594 @@ extern "C" void *alsaCallbackHandler( void *ptr ) #include #include -extern "C" void *ossCallbackHandler(void * ptr); - -// A structure to hold various information related to the OSS API -// implementation. -struct OssHandle { - int id[2]; // device ids - bool xrun[2]; - bool triggered; - pthread_cond_t runnable; - - OssHandle() - :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } -}; + extern "C" void *ossCallbackHandler(void * ptr); -RtApiOss :: RtApiOss() -{ - // Nothing to do here. -} + // A structure to hold various information related to the OSS API + // implementation. + struct OssHandle { + int id[2]; // device ids + bool xrun[2]; + bool triggered; + pthread_cond_t runnable; -RtApiOss :: ~RtApiOss() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} + OssHandle() + :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } + }; -unsigned int RtApiOss :: getDeviceCount( void ) -{ - int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); - if ( mixerfd == -1 ) { - errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'."; - error( RtError::WARNING ); - return 0; + RtApiOss :: RtApiOss() + { + // Nothing to do here. } - oss_sysinfo sysinfo; - if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required."; - error( RtError::WARNING ); - return 0; + RtApiOss :: ~RtApiOss() + { + if ( stream_.state != STREAM_CLOSED ) closeStream(); } - close( mixerfd ); - return sysinfo.numaudios; -} - -RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; + unsigned int RtApiOss :: getDeviceCount( void ) + { + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'."; + error( RtError::WARNING ); + return 0; + } - int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); - if ( mixerfd == -1 ) { - errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'."; - error( RtError::WARNING ); - return info; - } + oss_sysinfo sysinfo; + if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required."; + error( RtError::WARNING ); + return 0; + } - oss_sysinfo sysinfo; - int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); - if ( result == -1 ) { close( mixerfd ); - errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required."; - error( RtError::WARNING ); - return info; + return sysinfo.numaudios; } - unsigned nDevices = sysinfo.numaudios; - if ( nDevices == 0 ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceInfo: no devices found!"; - error( RtError::INVALID_USE ); - } + RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) + { + RtAudio::DeviceInfo info; + info.probed = false; - if ( device >= nDevices ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - } + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'."; + error( RtError::WARNING ); + return info; + } - oss_audioinfo ainfo; - ainfo.dev = device; - result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); - close( mixerfd ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } + oss_sysinfo sysinfo; + int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); + if ( result == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required."; + error( RtError::WARNING ); + return info; + } - // Probe channels - if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels; - if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels; - if ( ainfo.caps & PCM_CAP_DUPLEX ) { - if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - } + unsigned nDevices = sysinfo.numaudios; + if ( nDevices == 0 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: no devices found!"; + error( RtError::INVALID_USE ); + } - // Probe data formats ... do for input - unsigned long mask = ainfo.iformats; - if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE ) - info.nativeFormats |= RTAUDIO_SINT16; - if ( mask & AFMT_S8 ) - info.nativeFormats |= RTAUDIO_SINT8; - if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE ) - info.nativeFormats |= RTAUDIO_SINT32; - if ( mask & AFMT_FLOAT ) - info.nativeFormats |= RTAUDIO_FLOAT32; - if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE ) - info.nativeFormats |= RTAUDIO_SINT24; - - // Check that we have at least one supported format - if ( info.nativeFormats == 0 ) { - errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } + if ( device >= nDevices ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } - // Probe the supported sample rates. - info.sampleRates.clear(); - if ( ainfo.nrates ) { - for ( unsigned int i=0; i 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + } + + // Probe data formats ... do for input + unsigned long mask = ainfo.iformats; + if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE ) + info.nativeFormats |= RTAUDIO_SINT16; + if ( mask & AFMT_S8 ) + info.nativeFormats |= RTAUDIO_SINT8; + if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE ) + info.nativeFormats |= RTAUDIO_SINT32; + if ( mask & AFMT_FLOAT ) + info.nativeFormats |= RTAUDIO_FLOAT32; + if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE ) + info.nativeFormats |= RTAUDIO_SINT24; + + // Check that we have at least one supported format + if ( info.nativeFormats == 0 ) { + errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // Probe the supported sample rates. + info.sampleRates.clear(); + if ( ainfo.nrates ) { + for ( unsigned int i=0; i= (int) SAMPLE_RATES[k] ) info.sampleRates.push_back( SAMPLE_RATES[k] ); - break; - } } } - } - else { - // Check min and max rate values; - for ( unsigned int k=0; k= (int) SAMPLE_RATES[k] ) - info.sampleRates.push_back( SAMPLE_RATES[k] ); + + if ( info.sampleRates.size() == 0 ) { + errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + } + else { + info.probed = true; + info.name = ainfo.name; } - } - if ( info.sampleRates.size() == 0 ) { - errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } - else { - info.probed = true; - info.name = ainfo.name; + return info; } - return info; -} + bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) + { + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'."; + return FAILURE; + } -bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); - if ( mixerfd == -1 ) { - errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'."; - return FAILURE; - } + oss_sysinfo sysinfo; + int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); + if ( result == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required."; + return FAILURE; + } - oss_sysinfo sysinfo; - int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); - if ( result == -1 ) { - close( mixerfd ); - errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required."; - return FAILURE; - } + unsigned nDevices = sysinfo.numaudios; + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: no devices found!"; + return FAILURE; + } - unsigned nDevices = sysinfo.numaudios; - if ( nDevices == 0 ) { - // This should not happen because a check is made before this function is called. - close( mixerfd ); - errorText_ = "RtApiOss::probeDeviceOpen: no devices found!"; - return FAILURE; - } + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } - if ( device >= nDevices ) { - // This should not happen because a check is made before this function is called. + oss_audioinfo ainfo; + ainfo.dev = device; + result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); close( mixerfd ); - errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } + if ( result == -1 ) { + errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; + errorText_ = errorStream_.str(); + return FAILURE; + } - oss_audioinfo ainfo; - ainfo.dev = device; - result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); - close( mixerfd ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Check if device supports input or output + if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) || + ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) { + if ( mode == OUTPUT ) + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output."; + else + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Check if device supports input or output - if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) || - ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) { + int flags = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; if ( mode == OUTPUT ) - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output."; - else - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - int flags = 0; - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( mode == OUTPUT ) - flags |= O_WRONLY; - else { // mode == INPUT - if (stream_.mode == OUTPUT && stream_.device[0] == device) { - // We just set the same device for playback ... close and reopen for duplex (OSS only). - close( handle->id[0] ); - handle->id[0] = 0; - if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) { - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode."; - errorText_ = errorStream_.str(); - return FAILURE; - } - // Check that the number previously set channels is the same. - if ( stream_.nUserChannels[0] != channels ) { - errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; + flags |= O_WRONLY; + else { // mode == INPUT + if (stream_.mode == OUTPUT && stream_.device[0] == device) { + // We just set the same device for playback ... close and reopen for duplex (OSS only). + close( handle->id[0] ); + handle->id[0] = 0; + if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) { + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode."; + errorText_ = errorStream_.str(); + return FAILURE; + } + // Check that the number previously set channels is the same. + if ( stream_.nUserChannels[0] != channels ) { + errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + flags |= O_RDWR; } - flags |= O_RDWR; + else + flags |= O_RDONLY; } - else - flags |= O_RDONLY; - } - // Set exclusive access if specified. - if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL; - - // Try to open the device. - int fd; - fd = open( ainfo.devnode, flags, 0 ); - if ( fd == -1 ) { - if ( errno == EBUSY ) - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy."; - else - errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Set exclusive access if specified. + if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL; - // For duplex operation, specifically set this mode (this doesn't seem to work). - /* - if ( flags | O_RDWR ) { - result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL ); - if ( result == -1) { - errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ")."; + // Try to open the device. + int fd; + fd = open( ainfo.devnode, flags, 0 ); + if ( fd == -1 ) { + if ( errno == EBUSY ) + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy."; + else + errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ")."; errorText_ = errorStream_.str(); return FAILURE; } - } - */ - - // Check the device channel support. - stream_.nUserChannels[mode] = channels; - if ( ainfo.max_channels < (int)(channels + firstChannel) ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the number of channels. - int deviceChannels = channels + firstChannel; - result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels ); - if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.nDeviceChannels[mode] = deviceChannels; - // Get the data format mask - int mask; - result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask ); - if ( result == -1 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // For duplex operation, specifically set this mode (this doesn't seem to work). + /* + if ( flags | O_RDWR ) { + result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL ); + if ( result == -1) { + errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + */ - // Determine how to set the device format. - stream_.userFormat = format; - int deviceFormat = -1; - stream_.doByteSwap[mode] = false; - if ( format == RTAUDIO_SINT8 ) { - if ( mask & AFMT_S8 ) { - deviceFormat = AFMT_S8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - } - else if ( format == RTAUDIO_SINT16 ) { - if ( mask & AFMT_S16_NE ) { - deviceFormat = AFMT_S16_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - else if ( mask & AFMT_S16_OE ) { - deviceFormat = AFMT_S16_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - stream_.doByteSwap[mode] = true; - } - } - else if ( format == RTAUDIO_SINT24 ) { - if ( mask & AFMT_S24_NE ) { - deviceFormat = AFMT_S24_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - } - else if ( mask & AFMT_S24_OE ) { - deviceFormat = AFMT_S24_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - stream_.doByteSwap[mode] = true; - } - } - else if ( format == RTAUDIO_SINT32 ) { - if ( mask & AFMT_S32_NE ) { - deviceFormat = AFMT_S32_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - } - else if ( mask & AFMT_S32_OE ) { - deviceFormat = AFMT_S32_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - stream_.doByteSwap[mode] = true; + // Check the device channel support. + stream_.nUserChannels[mode] = channels; + if ( ainfo.max_channels < (int)(channels + firstChannel) ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters."; + errorText_ = errorStream_.str(); + return FAILURE; } - } - if ( deviceFormat == -1 ) { - // The user requested format is not natively supported by the device. - if ( mask & AFMT_S16_NE ) { - deviceFormat = AFMT_S16_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; + // Set the number of channels. + int deviceChannels = channels + firstChannel; + result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels ); + if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; } - else if ( mask & AFMT_S32_NE ) { - deviceFormat = AFMT_S32_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; + stream_.nDeviceChannels[mode] = deviceChannels; + + // Get the data format mask + int mask; + result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats."; + errorText_ = errorStream_.str(); + return FAILURE; } - else if ( mask & AFMT_S24_NE ) { - deviceFormat = AFMT_S24_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; + + // Determine how to set the device format. + stream_.userFormat = format; + int deviceFormat = -1; + stream_.doByteSwap[mode] = false; + if ( format == RTAUDIO_SINT8 ) { + if ( mask & AFMT_S8 ) { + deviceFormat = AFMT_S8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } } - else if ( mask & AFMT_S16_OE ) { - deviceFormat = AFMT_S16_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - stream_.doByteSwap[mode] = true; + else if ( format == RTAUDIO_SINT16 ) { + if ( mask & AFMT_S16_NE ) { + deviceFormat = AFMT_S16_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + else if ( mask & AFMT_S16_OE ) { + deviceFormat = AFMT_S16_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + stream_.doByteSwap[mode] = true; + } } - else if ( mask & AFMT_S32_OE ) { - deviceFormat = AFMT_S32_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - stream_.doByteSwap[mode] = true; + else if ( format == RTAUDIO_SINT24 ) { + if ( mask & AFMT_S24_NE ) { + deviceFormat = AFMT_S24_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + } + else if ( mask & AFMT_S24_OE ) { + deviceFormat = AFMT_S24_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + stream_.doByteSwap[mode] = true; + } } - else if ( mask & AFMT_S24_OE ) { - deviceFormat = AFMT_S24_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - stream_.doByteSwap[mode] = true; + else if ( format == RTAUDIO_SINT32 ) { + if ( mask & AFMT_S32_NE ) { + deviceFormat = AFMT_S32_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + } + else if ( mask & AFMT_S32_OE ) { + deviceFormat = AFMT_S32_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + stream_.doByteSwap[mode] = true; + } } - else if ( mask & AFMT_S8) { - deviceFormat = AFMT_S8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; + + if ( deviceFormat == -1 ) { + // The user requested format is not natively supported by the device. + if ( mask & AFMT_S16_NE ) { + deviceFormat = AFMT_S16_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + else if ( mask & AFMT_S32_NE ) { + deviceFormat = AFMT_S32_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + } + else if ( mask & AFMT_S24_NE ) { + deviceFormat = AFMT_S24_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + } + else if ( mask & AFMT_S16_OE ) { + deviceFormat = AFMT_S16_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S32_OE ) { + deviceFormat = AFMT_S32_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S24_OE ) { + deviceFormat = AFMT_S24_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S8) { + deviceFormat = AFMT_S8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + } } - } - if ( stream_.deviceFormat[mode] == 0 ) { - // This really shouldn't happen ... - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - return FAILURE; - } + if ( stream_.deviceFormat[mode] == 0 ) { + // This really shouldn't happen ... + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Set the data format. - int temp = deviceFormat; - result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat ); - if ( result == -1 || deviceFormat != temp ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Set the data format. + int temp = deviceFormat; + result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat ); + if ( result == -1 || deviceFormat != temp ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Attempt to set the buffer size. According to OSS, the minimum - // number of buffers is two. The supposed minimum buffer size is 16 - // bytes, so that will be our lower bound. The argument to this - // call is in the form 0xMMMMSSSS (hex), where the buffer size (in - // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM. - // We'll check the actual value used near the end of the setup - // procedure. - int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels; - if ( ossBufferBytes < 16 ) ossBufferBytes = 16; - int buffers = 0; - if ( options ) buffers = options->numberOfBuffers; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2; - if ( buffers < 2 ) buffers = 3; - temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) ); - result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp ); - if ( result == -1 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.nBuffers = buffers; + // Attempt to set the buffer size. According to OSS, the minimum + // number of buffers is two. The supposed minimum buffer size is 16 + // bytes, so that will be our lower bound. The argument to this + // call is in the form 0xMMMMSSSS (hex), where the buffer size (in + // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM. + // We'll check the actual value used near the end of the setup + // procedure. + int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels; + if ( ossBufferBytes < 16 ) ossBufferBytes = 16; + int buffers = 0; + if ( options ) buffers = options->numberOfBuffers; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2; + if ( buffers < 2 ) buffers = 3; + temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) ); + result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.nBuffers = buffers; - // Save buffer size (in sample frames). - *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels ); - stream_.bufferSize = *bufferSize; + // Save buffer size (in sample frames). + *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels ); + stream_.bufferSize = *bufferSize; - // Set the sample rate. - int srate = sampleRate; - result = ioctl( fd, SNDCTL_DSP_SPEED, &srate ); - if ( result == -1 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Set the sample rate. + int srate = sampleRate; + result = ioctl( fd, SNDCTL_DSP_SPEED, &srate ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Verify the sample rate setup worked. - if ( abs( srate - sampleRate ) > 100 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.sampleRate = sampleRate; + // Verify the sample rate setup worked. + if ( abs( srate - sampleRate ) > 100 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.sampleRate = sampleRate; - if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) { - // We're doing duplex setup here. - stream_.deviceFormat[0] = stream_.deviceFormat[1]; - stream_.nDeviceChannels[0] = deviceChannels; - } + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) { + // We're doing duplex setup here. + stream_.deviceFormat[0] = stream_.deviceFormat[1]; + stream_.nDeviceChannels[0] = deviceChannels; + } - // Set interleaving parameters. - stream_.userInterleaved = true; - stream_.deviceInterleaved[mode] = true; - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) - stream_.userInterleaved = false; + // Set interleaving parameters. + stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) + stream_.userInterleaved = false; + + // Set flags for buffer conversion + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; + + // Allocate the stream handles if necessary and then save. + if ( stream_.apiHandle == 0 ) { + try { + handle = new OssHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory."; + goto error; + } - // Set flags for buffer conversion - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; + if ( pthread_cond_init( &handle->runnable, NULL ) ) { + errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable."; + goto error; + } - // Allocate the stream handles if necessary and then save. - if ( stream_.apiHandle == 0 ) { - try { - handle = new OssHandle; + stream_.apiHandle = (void *) handle; } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory."; - goto error; + else { + handle = (OssHandle *) stream_.apiHandle; } + handle->id[mode] = fd; - if ( pthread_cond_init( &handle->runnable, NULL ) ) { - errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable."; + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory."; goto error; } - stream_.apiHandle = (void *) handle; - } - else { - handle = (OssHandle *) stream_.apiHandle; - } - handle->id[mode] = fd; - - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - if ( stream_.doConvertBuffer[mode] ) { + if ( stream_.doConvertBuffer[mode] ) { - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; + } } - } - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory."; - goto error; + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory."; + goto error; + } } } - } - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - // Setup thread if necessary. - if ( stream_.mode == OUTPUT && mode == INPUT ) { - // We had already set up an output stream. - stream_.mode = DUPLEX; - if ( stream_.device[0] == device ) handle->id[0] = fd; - } - else { - stream_.mode = mode; + // Setup thread if necessary. + if ( stream_.mode == OUTPUT && mode == INPUT ) { + // We had already set up an output stream. + stream_.mode = DUPLEX; + if ( stream_.device[0] == device ) handle->id[0] = fd; + } + else { + stream_.mode = mode; - // Setup callback thread. - stream_.callbackInfo.object = (void *) this; + // Setup callback thread. + stream_.callbackInfo.object = (void *) this; - // Set the thread attributes for joinable and realtime scheduling - // priority. The higher priority will only take affect if the - // program is run as root or suid. - pthread_attr_t attr; - pthread_attr_init( &attr ); - pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); + // Set the thread attributes for joinable and realtime scheduling + // priority. The higher priority will only take affect if the + // program is run as root or suid. + pthread_attr_t attr; + pthread_attr_init( &attr ); + pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) - if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { - struct sched_param param; - int priority = options->priority; - int min = sched_get_priority_min( SCHED_RR ); - int max = sched_get_priority_max( SCHED_RR ); - if ( priority < min ) priority = min; - else if ( priority > max ) priority = max; - param.sched_priority = priority; - pthread_attr_setschedparam( &attr, ¶m ); - pthread_attr_setschedpolicy( &attr, SCHED_RR ); - } - else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { + struct sched_param param; + int priority = options->priority; + int min = sched_get_priority_min( SCHED_RR ); + int max = sched_get_priority_max( SCHED_RR ); + if ( priority < min ) priority = min; + else if ( priority > max ) priority = max; + param.sched_priority = priority; + pthread_attr_setschedparam( &attr, ¶m ); + pthread_attr_setschedpolicy( &attr, SCHED_RR ); + } + else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); #else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); #endif - stream_.callbackInfo.isRunning = true; - result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo ); - pthread_attr_destroy( &attr ); - if ( result ) { - stream_.callbackInfo.isRunning = false; - errorText_ = "RtApiOss::error creating callback thread!"; - goto error; + stream_.callbackInfo.isRunning = true; + result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo ); + pthread_attr_destroy( &attr ); + if ( result ) { + stream_.callbackInfo.isRunning = false; + errorText_ = "RtApiOss::error creating callback thread!"; + goto error; + } } - } - - return SUCCESS; - error: - if ( handle ) { - pthread_cond_destroy( &handle->runnable ); - if ( handle->id[0] ) close( handle->id[0] ); - if ( handle->id[1] ) close( handle->id[1] ); - delete handle; - stream_.apiHandle = 0; - } + return SUCCESS; - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; + error: + if ( handle ) { + pthread_cond_destroy( &handle->runnable ); + if ( handle->id[0] ) close( handle->id[0] ); + if ( handle->id[1] ) close( handle->id[1] ); + delete handle; + stream_.apiHandle = 0; } - } - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } - return FAILURE; -} + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } -void RtApiOss :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiOss::closeStream(): no open stream to close!"; - error( RtError::WARNING ); - return; + return FAILURE; } - OssHandle *handle = (OssHandle *) stream_.apiHandle; - stream_.callbackInfo.isRunning = false; - MUTEX_LOCK( &stream_.mutex ); - if ( stream_.state == STREAM_STOPPED ) - pthread_cond_signal( &handle->runnable ); - MUTEX_UNLOCK( &stream_.mutex ); - pthread_join( stream_.callbackInfo.thread, NULL ); + void RtApiOss :: closeStream() + { + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiOss::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } - if ( stream_.state == STREAM_RUNNING ) { - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) - ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); - else - ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); - stream_.state = STREAM_STOPPED; - } + OssHandle *handle = (OssHandle *) stream_.apiHandle; + stream_.callbackInfo.isRunning = false; + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) + pthread_cond_signal( &handle->runnable ); + MUTEX_UNLOCK( &stream_.mutex ); + pthread_join( stream_.callbackInfo.thread, NULL ); - if ( handle ) { - pthread_cond_destroy( &handle->runnable ); - if ( handle->id[0] ) close( handle->id[0] ); - if ( handle->id[1] ) close( handle->id[1] ); - delete handle; - stream_.apiHandle = 0; - } + if ( stream_.state == STREAM_RUNNING ) { + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) + ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); + else + ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + stream_.state = STREAM_STOPPED; + } - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; + if ( handle ) { + pthread_cond_destroy( &handle->runnable ); + if ( handle->id[0] ) close( handle->id[0] ); + if ( handle->id[1] ) close( handle->id[1] ); + delete handle; + stream_.apiHandle = 0; } - } - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; + } + } - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } -void RtApiOss :: startStream() -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiOss::startStream(): the stream is already running!"; - error( RtError::WARNING ); - return; + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; } - MUTEX_LOCK( &stream_.mutex ); + void RtApiOss :: startStream() + { + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiOss::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } - stream_.state = STREAM_RUNNING; + MUTEX_LOCK( &stream_.mutex ); - // No need to do anything else here ... OSS automatically starts - // when fed samples. + stream_.state = STREAM_RUNNING; - MUTEX_UNLOCK( &stream_.mutex ); + // No need to do anything else here ... OSS automatically starts + // when fed samples. - OssHandle *handle = (OssHandle *) stream_.apiHandle; - pthread_cond_signal( &handle->runnable ); -} + MUTEX_UNLOCK( &stream_.mutex ); -void RtApiOss :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiOss::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + pthread_cond_signal( &handle->runnable ); } - // Change the state before the lock to improve shutdown response - // when using a callback. - stream_.state = STREAM_STOPPED; - MUTEX_LOCK( &stream_.mutex ); + void RtApiOss :: stopStream() + { + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiOss::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } - int result = 0; - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + // Change the state before the lock to improve shutdown response + // when using a callback. + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); - // Flush the output with zeros a few times. - char *buffer; - int samples; - RtAudioFormat format; + int result = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - samples = stream_.bufferSize * stream_.nDeviceChannels[0]; - format = stream_.deviceFormat[0]; - } - else { - buffer = stream_.userBuffer[0]; - samples = stream_.bufferSize * stream_.nUserChannels[0]; - format = stream_.userFormat; - } + // Flush the output with zeros a few times. + char *buffer; + int samples; + RtAudioFormat format; - memset( buffer, 0, samples * formatBytes(format) ); - for ( unsigned int i=0; iid[0], buffer, samples * formatBytes(format) ); - if ( result == -1 ) { - errorText_ = "RtApiOss::stopStream: audio write error."; - error( RtError::WARNING ); + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + samples = stream_.bufferSize * stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + samples = stream_.bufferSize * stream_.nUserChannels[0]; + format = stream_.userFormat; } - } - result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - handle->triggered = false; - } + memset( buffer, 0, samples * formatBytes(format) ); + for ( unsigned int i=0; iid[0], buffer, samples * formatBytes(format) ); + if ( result == -1 ) { + errorText_ = "RtApiOss::stopStream: audio write error."; + error( RtError::WARNING ); + } + } - if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { - result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; + result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + handle->triggered = false; } - } - unlock: - MUTEX_UNLOCK( &stream_.mutex ); + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { + result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + } - stream_.state = STREAM_STOPPED; - if ( result != -1 ) return; - error( RtError::SYSTEM_ERROR ); -} + unlock: + MUTEX_UNLOCK( &stream_.mutex ); -void RtApiOss :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiOss::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; + stream_.state = STREAM_STOPPED; + if ( result != -1 ) return; + error( RtError::SYSTEM_ERROR ); } - // Change the state before the lock to improve shutdown response - // when using a callback. - stream_.state = STREAM_STOPPED; - MUTEX_LOCK( &stream_.mutex ); + void RtApiOss :: abortStream() + { + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiOss::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } - int result = 0; - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; + // Change the state before the lock to improve shutdown response + // when using a callback. + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + int result = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + handle->triggered = false; } - handle->triggered = false; - } - if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { - result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { + result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } } - } - unlock: - MUTEX_UNLOCK( &stream_.mutex ); + unlock: + MUTEX_UNLOCK( &stream_.mutex ); - stream_.state = STREAM_STOPPED; - if ( result != -1 ) return; - error( RtError::SYSTEM_ERROR ); -} + stream_.state = STREAM_STOPPED; + if ( result != -1 ) return; + error( RtError::SYSTEM_ERROR ); + } -void RtApiOss :: callbackEvent() -{ - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_LOCK( &stream_.mutex ); - pthread_cond_wait( &handle->runnable, &stream_.mutex ); - if ( stream_.state != STREAM_RUNNING ) { + void RtApiOss :: callbackEvent() + { + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + pthread_cond_wait( &handle->runnable, &stream_.mutex ); + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } MUTEX_UNLOCK( &stream_.mutex ); - return; } - MUTEX_UNLOCK( &stream_.mutex ); - } - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); - return; - } + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return; + } - // Invoke user callback to get fresh output data. - int doStopStream = 0; - RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; - } - doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); - if ( doStopStream == 2 ) { - this->abortStream(); - return; - } + // Invoke user callback to get fresh output data. + int doStopStream = 0; + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); + if ( doStopStream == 2 ) { + this->abortStream(); + return; + } - MUTEX_LOCK( &stream_.mutex ); + MUTEX_LOCK( &stream_.mutex ); - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) goto unlock; + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; - int result; - char *buffer; - int samples; - RtAudioFormat format; + int result; + char *buffer; + int samples; + RtAudioFormat format; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - samples = stream_.bufferSize * stream_.nDeviceChannels[0]; - format = stream_.deviceFormat[0]; - } - else { - buffer = stream_.userBuffer[0]; - samples = stream_.bufferSize * stream_.nUserChannels[0]; - format = stream_.userFormat; - } + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + samples = stream_.bufferSize * stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + samples = stream_.bufferSize * stream_.nUserChannels[0]; + format = stream_.userFormat; + } - // Do byte swapping if necessary. - if ( stream_.doByteSwap[0] ) - byteSwapBuffer( buffer, samples, format ); + // Do byte swapping if necessary. + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( buffer, samples, format ); + + if ( stream_.mode == DUPLEX && handle->triggered == false ) { + int trig = 0; + ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); + result = write( handle->id[0], buffer, samples * formatBytes(format) ); + trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT; + ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); + handle->triggered = true; + } + else + // Write samples to device. + result = write( handle->id[0], buffer, samples * formatBytes(format) ); - if ( stream_.mode == DUPLEX && handle->triggered == false ) { - int trig = 0; - ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); - result = write( handle->id[0], buffer, samples * formatBytes(format) ); - trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT; - ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); - handle->triggered = true; + if ( result == -1 ) { + // We'll assume this is an underrun, though there isn't a + // specific means for determining that. + handle->xrun[0] = true; + errorText_ = "RtApiOss::callbackEvent: audio write error."; + error( RtError::WARNING ); + // Continue on to input section. + } } - else - // Write samples to device. - result = write( handle->id[0], buffer, samples * formatBytes(format) ); - if ( result == -1 ) { - // We'll assume this is an underrun, though there isn't a - // specific means for determining that. - handle->xrun[0] = true; - errorText_ = "RtApiOss::callbackEvent: audio write error."; - error( RtError::WARNING ); - // Continue on to input section. - } - } + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + samples = stream_.bufferSize * stream_.nDeviceChannels[1]; + format = stream_.deviceFormat[1]; + } + else { + buffer = stream_.userBuffer[1]; + samples = stream_.bufferSize * stream_.nUserChannels[1]; + format = stream_.userFormat; + } - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + // Read samples from device. + result = read( handle->id[1], buffer, samples * formatBytes(format) ); - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - samples = stream_.bufferSize * stream_.nDeviceChannels[1]; - format = stream_.deviceFormat[1]; - } - else { - buffer = stream_.userBuffer[1]; - samples = stream_.bufferSize * stream_.nUserChannels[1]; - format = stream_.userFormat; - } + if ( result == -1 ) { + // We'll assume this is an overrun, though there isn't a + // specific means for determining that. + handle->xrun[1] = true; + errorText_ = "RtApiOss::callbackEvent: audio read error."; + error( RtError::WARNING ); + goto unlock; + } - // Read samples from device. - result = read( handle->id[1], buffer, samples * formatBytes(format) ); + // Do byte swapping if necessary. + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( buffer, samples, format ); - if ( result == -1 ) { - // We'll assume this is an overrun, though there isn't a - // specific means for determining that. - handle->xrun[1] = true; - errorText_ = "RtApiOss::callbackEvent: audio read error."; - error( RtError::WARNING ); - goto unlock; + // Do buffer conversion if necessary. + if ( stream_.doConvertBuffer[1] ) + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); } - // Do byte swapping if necessary. - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( buffer, samples, format ); + unlock: + MUTEX_UNLOCK( &stream_.mutex ); - // Do buffer conversion if necessary. - if ( stream_.doConvertBuffer[1] ) - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + RtApi::tickStreamTime(); + if ( doStopStream == 1 ) this->stopStream(); } - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - - RtApi::tickStreamTime(); - if ( doStopStream == 1 ) this->stopStream(); -} + extern "C" void *ossCallbackHandler( void *ptr ) + { + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiOss *object = (RtApiOss *) info->object; + bool *isRunning = &info->isRunning; -extern "C" void *ossCallbackHandler( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiOss *object = (RtApiOss *) info->object; - bool *isRunning = &info->isRunning; + while ( *isRunning == true ) { + pthread_testcancel(); + object->callbackEvent(); + } - while ( *isRunning == true ) { - pthread_testcancel(); - object->callbackEvent(); + pthread_exit( NULL ); } - pthread_exit( NULL ); -} - -//******************** End of __LINUX_OSS__ *********************// + //******************** End of __LINUX_OSS__ *********************// #endif -// *************************************************** // -// -// Protected common (OS-independent) RtAudio methods. -// -// *************************************************** // - -// This method can be modified to control the behavior of error -// message printing. -void RtApi :: error( RtError::Type type ) -{ - errorStream_.str(""); // clear the ostringstream - if ( type == RtError::WARNING && showWarnings_ == true ) - std::cerr << '\n' << errorText_ << "\n\n"; - else - throw( RtError( errorText_, type ) ); -} + // *************************************************** // + // + // Protected common (OS-independent) RtAudio methods. + // + // *************************************************** // -void RtApi :: verifyStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApi:: a stream is not open!"; - error( RtError::INVALID_USE ); + // This method can be modified to control the behavior of error + // message printing. + void RtApi :: error( RtError::Type type ) + { + errorStream_.str(""); // clear the ostringstream + if ( type == RtError::WARNING && showWarnings_ == true ) + std::cerr << '\n' << errorText_ << "\n\n"; + else + throw( RtError( errorText_, type ) ); } -} -void RtApi :: clearStreamInfo() -{ - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; - stream_.sampleRate = 0; - stream_.bufferSize = 0; - stream_.nBuffers = 0; - stream_.userFormat = 0; - stream_.userInterleaved = true; - stream_.streamTime = 0.0; - stream_.apiHandle = 0; - stream_.deviceBuffer = 0; - stream_.callbackInfo.callback = 0; - stream_.callbackInfo.userData = 0; - stream_.callbackInfo.isRunning = false; - for ( int i=0; i<2; i++ ) { - stream_.device[i] = 11111; - stream_.doConvertBuffer[i] = false; - stream_.deviceInterleaved[i] = true; - stream_.doByteSwap[i] = false; - stream_.nUserChannels[i] = 0; - stream_.nDeviceChannels[i] = 0; - stream_.channelOffset[i] = 0; - stream_.deviceFormat[i] = 0; - stream_.latency[i] = 0; - stream_.userBuffer[i] = 0; - stream_.convertInfo[i].channels = 0; - stream_.convertInfo[i].inJump = 0; - stream_.convertInfo[i].outJump = 0; - stream_.convertInfo[i].inFormat = 0; - stream_.convertInfo[i].outFormat = 0; - stream_.convertInfo[i].inOffset.clear(); - stream_.convertInfo[i].outOffset.clear(); + void RtApi :: verifyStream() + { + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApi:: a stream is not open!"; + error( RtError::INVALID_USE ); + } } -} -unsigned int RtApi :: formatBytes( RtAudioFormat format ) -{ - if ( format == RTAUDIO_SINT16 ) - return 2; - else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 || - format == RTAUDIO_FLOAT32 ) - return 4; - else if ( format == RTAUDIO_FLOAT64 ) - return 8; - else if ( format == RTAUDIO_SINT8 ) - return 1; - - errorText_ = "RtApi::formatBytes: undefined format."; - error( RtError::WARNING ); + void RtApi :: clearStreamInfo() + { + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; + stream_.sampleRate = 0; + stream_.bufferSize = 0; + stream_.nBuffers = 0; + stream_.userFormat = 0; + stream_.userInterleaved = true; + stream_.streamTime = 0.0; + stream_.apiHandle = 0; + stream_.deviceBuffer = 0; + stream_.callbackInfo.callback = 0; + stream_.callbackInfo.userData = 0; + stream_.callbackInfo.isRunning = false; + for ( int i=0; i<2; i++ ) { + stream_.device[i] = 11111; + stream_.doConvertBuffer[i] = false; + stream_.deviceInterleaved[i] = true; + stream_.doByteSwap[i] = false; + stream_.nUserChannels[i] = 0; + stream_.nDeviceChannels[i] = 0; + stream_.channelOffset[i] = 0; + stream_.deviceFormat[i] = 0; + stream_.latency[i] = 0; + stream_.userBuffer[i] = 0; + stream_.convertInfo[i].channels = 0; + stream_.convertInfo[i].inJump = 0; + stream_.convertInfo[i].outJump = 0; + stream_.convertInfo[i].inFormat = 0; + stream_.convertInfo[i].outFormat = 0; + stream_.convertInfo[i].inOffset.clear(); + stream_.convertInfo[i].outOffset.clear(); + } + } - return 0; -} + unsigned int RtApi :: formatBytes( RtAudioFormat format ) + { + if ( format == RTAUDIO_SINT16 ) + return 2; + else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 || + format == RTAUDIO_FLOAT32 ) + return 4; + else if ( format == RTAUDIO_FLOAT64 ) + return 8; + else if ( format == RTAUDIO_SINT8 ) + return 1; + + errorText_ = "RtApi::formatBytes: undefined format."; + error( RtError::WARNING ); -void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel ) -{ - if ( mode == INPUT ) { // convert device to user buffer - stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1]; - stream_.convertInfo[mode].outJump = stream_.nUserChannels[1]; - stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1]; - stream_.convertInfo[mode].outFormat = stream_.userFormat; - } - else { // convert user to device buffer - stream_.convertInfo[mode].inJump = stream_.nUserChannels[0]; - stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0]; - stream_.convertInfo[mode].inFormat = stream_.userFormat; - stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0]; + return 0; } - if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump ) - stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump; - else - stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump; - - // Set up the interleave/deinterleave offsets. - if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) { - if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) || - ( mode == INPUT && stream_.userInterleaved ) ) { - for ( int k=0; k 0 ) { - if ( stream_.deviceInterleaved[mode] ) { - if ( mode == OUTPUT ) { - for ( int k=0; k 0 ) { + if ( stream_.deviceInterleaved[mode] ) { + if ( mode == OUTPUT ) { + for ( int k=0; k>= 8; + else if (info.inFormat == RTAUDIO_SINT32) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i>= 8; + } + in += info.inJump; + out += info.outJump; } - in += info.inJump; - out += info.outJump; } - } - else if (info.inFormat == RTAUDIO_FLOAT32) { - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i> 8) & 0x0000ffff); + else if (info.inFormat == RTAUDIO_SINT24) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 8) & 0x0000ffff); + } + in += info.inJump; + out += info.outJump; } - in += info.inJump; - out += info.outJump; } - } - else if (info.inFormat == RTAUDIO_SINT32) { - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i> 16) & 0x0000ffff); + else if (info.inFormat == RTAUDIO_SINT32) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 16) & 0x0000ffff); + } + in += info.inJump; + out += info.outJump; } - in += info.inJump; - out += info.outJump; } - } - else if (info.inFormat == RTAUDIO_FLOAT32) { - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i> 8) & 0x00ff); + if (info.inFormat == RTAUDIO_SINT16) { + Int16 *in = (Int16 *)inBuffer; + for (unsigned int i=0; i> 8) & 0x00ff); + } + in += info.inJump; + out += info.outJump; } - in += info.inJump; - out += info.outJump; } - } - else if (info.inFormat == RTAUDIO_SINT24) { - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i> 16) & 0x000000ff); + else if (info.inFormat == RTAUDIO_SINT24) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 16) & 0x000000ff); + } + in += info.inJump; + out += info.outJump; } - in += info.inJump; - out += info.outJump; } - } - else if (info.inFormat == RTAUDIO_SINT32) { - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i> 24) & 0x000000ff); + else if (info.inFormat == RTAUDIO_SINT32) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 24) & 0x000000ff); + } + in += info.inJump; + out += info.outJump; } - in += info.inJump; - out += info.outJump; } - } - else if (info.inFormat == RTAUDIO_FLOAT32) { - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i>8) | (x<<8); } -//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); } -//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); } + //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); } + //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); } + //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); } -void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ) -{ - register char val; - register char *ptr; - - ptr = buffer; - if ( format == RTAUDIO_SINT16 ) { - for ( unsigned int i=0; i to quit (buffer size = " << bufferFrames << ").\n"; std::cin.get( input ); try { // Stop the stream - std::cout << "Stream latency = " << dac.getStreamLatency() << "\n" << std::endl; dac.stopStream(); } catch ( RtError& e ) { diff --git a/tests/record.cpp b/tests/record.cpp index 7fb1317..22bcaac 100644 --- a/tests/record.cpp +++ b/tests/record.cpp @@ -15,21 +15,21 @@ /* typedef char MY_TYPE; #define FORMAT RTAUDIO_SINT8 +*/ typedef signed short MY_TYPE; #define FORMAT RTAUDIO_SINT16 +/* typedef signed long MY_TYPE; #define FORMAT RTAUDIO_SINT24 typedef signed long MY_TYPE; #define FORMAT RTAUDIO_SINT32 -*/ typedef float MY_TYPE; #define FORMAT RTAUDIO_FLOAT32 -/* typedef double MY_TYPE; #define FORMAT RTAUDIO_FLOAT64 */ -- cgit v1.2.3 From d035dfe8fe72475651f0f132016726fb1de65529 Mon Sep 17 00:00:00 2001 From: Gary Scavone Date: Fri, 2 Jan 2009 20:49:04 +0000 Subject: Bug fix for mono mode in OS-X support (GS). --- RtAudio.cpp | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/RtAudio.cpp b/RtAudio.cpp index 9b186cc..b16535b 100644 --- a/RtAudio.cpp +++ b/RtAudio.cpp @@ -1098,6 +1098,7 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne stream_.doConvertBuffer[mode] = true; std::cout << "doConvert = " << stream_.doConvertBuffer[mode] << ", userInterleaved = " << stream_.userInterleaved << ", deviceInterleaved = " << stream_.deviceInterleaved[mode] << std::endl; + std::cout << "nUserChannels = " << stream_.nUserChannels[mode] << ", nDeviceChannels = " << stream_.nDeviceChannels[mode] << std::endl; // Allocate our CoreHandle structure for the stream. CoreHandle *handle = 0; @@ -1467,7 +1468,7 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize; for ( unsigned int i=0; imBuffers[handle->iStream[0]+i].mData, - &inBuffer[i*bufferBytes], bufferBytes ); + (void *)&inBuffer[i*stream_.bufferSize], bufferBytes ); } } else { // fill multiple multi-channel streams with interleaved data @@ -1477,7 +1478,7 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, bool inInterleaved = ( stream_.userInterleaved ) ? true : false; UInt32 inChannels = stream_.nUserChannels[0]; if ( stream_.doConvertBuffer[0] ) { - inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 + inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode inChannels = stream_.nDeviceChannels[0]; } @@ -1555,7 +1556,7 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, if ( stream_.deviceInterleaved[1] == false ) { // mono mode UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize; for ( unsigned int i=0; imBuffers[handle->iStream[1]+i].mData, bufferBytes ); } } @@ -1566,7 +1567,7 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, bool outInterleaved = ( stream_.userInterleaved ) ? true : false; UInt32 outChannels = stream_.nUserChannels[1]; if ( stream_.doConvertBuffer[1] ) { - outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 + outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode outChannels = stream_.nDeviceChannels[1]; } -- cgit v1.2.3 From d80e83b7145a6c2ecaa6d7fc6ade83ef124515a1 Mon Sep 17 00:00:00 2001 From: Gary Scavone Date: Thu, 15 Jan 2009 20:26:42 +0000 Subject: Various changes in preparation for new 4.0.5 release (GS). --- RtAudio.cpp | 10121 +++++++++++++++++++++++---------------------- RtAudio.h | 4 +- doc/doxygen/Doxyfile | 942 +---- doc/doxygen/footer.html | 2 +- doc/doxygen/license.txt | 2 +- doc/doxygen/tutorial.txt | 2 +- doc/release.txt | 16 +- install | 2 +- readme | 4 +- 9 files changed, 5168 insertions(+), 5927 deletions(-) diff --git a/RtAudio.cpp b/RtAudio.cpp index b16535b..f86ea12 100644 --- a/RtAudio.cpp +++ b/RtAudio.cpp @@ -10,7 +10,7 @@ RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/ RtAudio: realtime audio i/o C++ classes - Copyright (c) 2001-2008 Gary P. Scavone + Copyright (c) 2001-2009 Gary P. Scavone Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files @@ -38,7 +38,7 @@ */ /************************************************************************/ -// RtAudio: Version 4.0.4 +// RtAudio: Version 4.0.5 #include "RtAudio.h" #include @@ -880,8 +880,6 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne free( bufferList ); - std::cout << "deviceStreams = " << nStreams << ", firstStream = " << firstStream << ", streamCount = " << streamCount << ", channelOffset = " << channelOffset << std::endl; - // Determine the buffer size. AudioValueRange bufferRange; dataSize = sizeof( AudioValueRange ); @@ -1097,9 +1095,6 @@ bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne else if ( monoMode && stream_.userInterleaved ) stream_.doConvertBuffer[mode] = true; - std::cout << "doConvert = " << stream_.doConvertBuffer[mode] << ", userInterleaved = " << stream_.userInterleaved << ", deviceInterleaved = " << stream_.deviceInterleaved[mode] << std::endl; - std::cout << "nUserChannels = " << stream_.nUserChannels[mode] << ", nDeviceChannels = " << stream_.nDeviceChannels[mode] << std::endl; - // Allocate our CoreHandle structure for the stream. CoreHandle *handle = 0; if ( stream_.apiHandle == 0 ) { @@ -1326,6 +1321,11 @@ void RtApiCore :: stopStream( void ) MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + OSStatus result = noErr; CoreHandle *handle = (CoreHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { @@ -1353,10 +1353,11 @@ void RtApiCore :: stopStream( void ) } } + stream_.state = STREAM_STOPPED; + unlock: MUTEX_UNLOCK( &stream_.mutex ); - stream_.state = STREAM_STOPPED; if ( result == noErr ) return; error( RtError::SYSTEM_ERROR ); } @@ -1401,6 +1402,12 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, MUTEX_LOCK( &stream_.mutex ); + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return SUCCESS; + } + AudioDeviceID outputDevice = handle->id[0]; // Invoke user callback to get fresh output data UNLESS we are @@ -1630,799 +1637,810 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, return SUCCESS; } - const char* RtApiCore :: getErrorCode( OSStatus code ) - { - switch( code ) { +const char* RtApiCore :: getErrorCode( OSStatus code ) +{ + switch( code ) { - case kAudioHardwareNotRunningError: - return "kAudioHardwareNotRunningError"; + case kAudioHardwareNotRunningError: + return "kAudioHardwareNotRunningError"; - case kAudioHardwareUnspecifiedError: - return "kAudioHardwareUnspecifiedError"; + case kAudioHardwareUnspecifiedError: + return "kAudioHardwareUnspecifiedError"; - case kAudioHardwareUnknownPropertyError: - return "kAudioHardwareUnknownPropertyError"; + case kAudioHardwareUnknownPropertyError: + return "kAudioHardwareUnknownPropertyError"; - case kAudioHardwareBadPropertySizeError: - return "kAudioHardwareBadPropertySizeError"; + case kAudioHardwareBadPropertySizeError: + return "kAudioHardwareBadPropertySizeError"; - case kAudioHardwareIllegalOperationError: - return "kAudioHardwareIllegalOperationError"; + case kAudioHardwareIllegalOperationError: + return "kAudioHardwareIllegalOperationError"; - case kAudioHardwareBadObjectError: - return "kAudioHardwareBadObjectError"; + case kAudioHardwareBadObjectError: + return "kAudioHardwareBadObjectError"; - case kAudioHardwareBadDeviceError: - return "kAudioHardwareBadDeviceError"; + case kAudioHardwareBadDeviceError: + return "kAudioHardwareBadDeviceError"; - case kAudioHardwareBadStreamError: - return "kAudioHardwareBadStreamError"; + case kAudioHardwareBadStreamError: + return "kAudioHardwareBadStreamError"; - case kAudioHardwareUnsupportedOperationError: - return "kAudioHardwareUnsupportedOperationError"; + case kAudioHardwareUnsupportedOperationError: + return "kAudioHardwareUnsupportedOperationError"; - case kAudioDeviceUnsupportedFormatError: - return "kAudioDeviceUnsupportedFormatError"; + case kAudioDeviceUnsupportedFormatError: + return "kAudioDeviceUnsupportedFormatError"; - case kAudioDevicePermissionsError: - return "kAudioDevicePermissionsError"; + case kAudioDevicePermissionsError: + return "kAudioDevicePermissionsError"; - default: - return "CoreAudio unknown error"; - } + default: + return "CoreAudio unknown error"; } +} //******************** End of __MACOSX_CORE__ *********************// #endif #if defined(__UNIX_JACK__) - // JACK is a low-latency audio server, originally written for the - // GNU/Linux operating system and now also ported to OS-X. It can - // connect a number of different applications to an audio device, as - // well as allowing them to share audio between themselves. - // - // When using JACK with RtAudio, "devices" refer to JACK clients that - // have ports connected to the server. The JACK server is typically - // started in a terminal as follows: - // - // .jackd -d alsa -d hw:0 - // - // or through an interface program such as qjackctl. Many of the - // parameters normally set for a stream are fixed by the JACK server - // and can be specified when the JACK server is started. In - // particular, - // - // .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4 - // - // specifies a sample rate of 44100 Hz, a buffer size of 512 sample - // frames, and number of buffers = 4. Once the server is running, it - // is not possible to override these values. If the values are not - // specified in the command-line, the JACK server uses default values. - // - // The JACK server does not have to be running when an instance of - // RtApiJack is created, though the function getDeviceCount() will - // report 0 devices found until JACK has been started. When no - // devices are available (i.e., the JACK server is not running), a - // stream cannot be opened. +// JACK is a low-latency audio server, originally written for the +// GNU/Linux operating system and now also ported to OS-X. It can +// connect a number of different applications to an audio device, as +// well as allowing them to share audio between themselves. +// +// When using JACK with RtAudio, "devices" refer to JACK clients that +// have ports connected to the server. The JACK server is typically +// started in a terminal as follows: +// +// .jackd -d alsa -d hw:0 +// +// or through an interface program such as qjackctl. Many of the +// parameters normally set for a stream are fixed by the JACK server +// and can be specified when the JACK server is started. In +// particular, +// +// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4 +// +// specifies a sample rate of 44100 Hz, a buffer size of 512 sample +// frames, and number of buffers = 4. Once the server is running, it +// is not possible to override these values. If the values are not +// specified in the command-line, the JACK server uses default values. +// +// The JACK server does not have to be running when an instance of +// RtApiJack is created, though the function getDeviceCount() will +// report 0 devices found until JACK has been started. When no +// devices are available (i.e., the JACK server is not running), a +// stream cannot be opened. #include #include - // A structure to hold various information related to the Jack API - // implementation. - struct JackHandle { - jack_client_t *client; - jack_port_t **ports[2]; - std::string deviceName[2]; - bool xrun[2]; - pthread_cond_t condition; - int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - - JackHandle() - :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; } - }; - - RtApiJack :: RtApiJack() - { - // Nothing to do here. - } - - RtApiJack :: ~RtApiJack() - { - if ( stream_.state != STREAM_CLOSED ) closeStream(); - } - - unsigned int RtApiJack :: getDeviceCount( void ) - { - // See if we can become a jack client. - jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption; - jack_status_t *status = NULL; - jack_client_t *client = jack_client_open( "RtApiJackCount", options, status ); - if ( client == 0 ) return 0; - - const char **ports; - std::string port, previousPort; - unsigned int nChannels = 0, nDevices = 0; - ports = jack_get_ports( client, NULL, NULL, 0 ); - if ( ports ) { - // Parse the port names up to the first colon (:). - size_t iColon = 0; - do { - port = (char *) ports[ nChannels ]; - iColon = port.find(":"); - if ( iColon != std::string::npos ) { - port = port.substr( 0, iColon + 1 ); - if ( port != previousPort ) { - nDevices++; - previousPort = port; - } - } - } while ( ports[++nChannels] ); - free( ports ); - } +// A structure to hold various information related to the Jack API +// implementation. +struct JackHandle { + jack_client_t *client; + jack_port_t **ports[2]; + std::string deviceName[2]; + bool xrun[2]; + pthread_cond_t condition; + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. - jack_client_close( client ); - return nDevices; - } + JackHandle() + :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; } +}; - RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) - { - RtAudio::DeviceInfo info; - info.probed = false; +RtApiJack :: RtApiJack() +{ + // Nothing to do here. +} - jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption - jack_status_t *status = NULL; - jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status ); - if ( client == 0 ) { - errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!"; - error( RtError::WARNING ); - return info; - } +RtApiJack :: ~RtApiJack() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} - const char **ports; - std::string port, previousPort; - unsigned int nPorts = 0, nDevices = 0; - ports = jack_get_ports( client, NULL, NULL, 0 ); - if ( ports ) { - // Parse the port names up to the first colon (:). - size_t iColon = 0; - do { - port = (char *) ports[ nPorts ]; - iColon = port.find(":"); - if ( iColon != std::string::npos ) { - port = port.substr( 0, iColon ); - if ( port != previousPort ) { - if ( nDevices == device ) info.name = port; - nDevices++; - previousPort = port; - } +unsigned int RtApiJack :: getDeviceCount( void ) +{ + // See if we can become a jack client. + jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption; + jack_status_t *status = NULL; + jack_client_t *client = jack_client_open( "RtApiJackCount", options, status ); + if ( client == 0 ) return 0; + + const char **ports; + std::string port, previousPort; + unsigned int nChannels = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, NULL, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nChannels ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon + 1 ); + if ( port != previousPort ) { + nDevices++; + previousPort = port; } - } while ( ports[++nPorts] ); - free( ports ); - } - - if ( device >= nDevices ) { - errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - } + } + } while ( ports[++nChannels] ); + free( ports ); + } - // Get the current jack server sample rate. - info.sampleRates.clear(); - info.sampleRates.push_back( jack_get_sample_rate( client ) ); + jack_client_close( client ); + return nDevices; +} - // Count the available ports containing the client name as device - // channels. Jack "input ports" equal RtAudio output channels. - unsigned int nChannels = 0; - ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput ); - if ( ports ) { - while ( ports[ nChannels ] ) nChannels++; - free( ports ); - info.outputChannels = nChannels; - } +RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; - // Jack "output ports" equal RtAudio input channels. - nChannels = 0; - ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput ); - if ( ports ) { - while ( ports[ nChannels ] ) nChannels++; - free( ports ); - info.inputChannels = nChannels; - } + jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption + jack_status_t *status = NULL; + jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status ); + if ( client == 0 ) { + errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!"; + error( RtError::WARNING ); + return info; + } - if ( info.outputChannels == 0 && info.inputChannels == 0 ) { - jack_client_close(client); - errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!"; - error( RtError::WARNING ); - return info; - } + const char **ports; + std::string port, previousPort; + unsigned int nPorts = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, NULL, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nPorts ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon ); + if ( port != previousPort ) { + if ( nDevices == device ) info.name = port; + nDevices++; + previousPort = port; + } + } + } while ( ports[++nPorts] ); + free( ports ); + } - // If device opens for both playback and capture, we determine the channels. - if ( info.outputChannels > 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + if ( device >= nDevices ) { + errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } - // Jack always uses 32-bit floats. - info.nativeFormats = RTAUDIO_FLOAT32; + // Get the current jack server sample rate. + info.sampleRates.clear(); + info.sampleRates.push_back( jack_get_sample_rate( client ) ); + + // Count the available ports containing the client name as device + // channels. Jack "input ports" equal RtAudio output channels. + unsigned int nChannels = 0; + ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + info.outputChannels = nChannels; + } - // Jack doesn't provide default devices so we'll use the first available one. - if ( device == 0 && info.outputChannels > 0 ) - info.isDefaultOutput = true; - if ( device == 0 && info.inputChannels > 0 ) - info.isDefaultInput = true; + // Jack "output ports" equal RtAudio input channels. + nChannels = 0; + ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + info.inputChannels = nChannels; + } + if ( info.outputChannels == 0 && info.inputChannels == 0 ) { jack_client_close(client); - info.probed = true; + errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!"; + error( RtError::WARNING ); return info; } - int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer ) - { - CallbackInfo *info = (CallbackInfo *) infoPointer; + // If device opens for both playback and capture, we determine the channels. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - RtApiJack *object = (RtApiJack *) info->object; - if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1; + // Jack always uses 32-bit floats. + info.nativeFormats = RTAUDIO_FLOAT32; - return 0; - } + // Jack doesn't provide default devices so we'll use the first available one. + if ( device == 0 && info.outputChannels > 0 ) + info.isDefaultOutput = true; + if ( device == 0 && info.inputChannels > 0 ) + info.isDefaultInput = true; - void jackShutdown( void *infoPointer ) - { - CallbackInfo *info = (CallbackInfo *) infoPointer; - RtApiJack *object = (RtApiJack *) info->object; + jack_client_close(client); + info.probed = true; + return info; +} - // Check current stream state. If stopped, then we'll assume this - // was called as a result of a call to RtApiJack::stopStream (the - // deactivation of a client handle causes this function to be called). - // If not, we'll assume the Jack server is shutting down or some - // other problem occurred and we should close the stream. - if ( object->isStreamRunning() == false ) return; +int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; - object->closeStream(); - std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl; - } + RtApiJack *object = (RtApiJack *) info->object; + if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1; - int jackXrun( void *infoPointer ) - { - JackHandle *handle = (JackHandle *) infoPointer; + return 0; +} - if ( handle->ports[0] ) handle->xrun[0] = true; - if ( handle->ports[1] ) handle->xrun[1] = true; +void jackShutdown( void *infoPointer ) +{ + CallbackInfo *info = (CallbackInfo *) infoPointer; + RtApiJack *object = (RtApiJack *) info->object; - return 0; - } + // Check current stream state. If stopped, then we'll assume this + // was called as a result of a call to RtApiJack::stopStream (the + // deactivation of a client handle causes this function to be called). + // If not, we'll assume the Jack server is shutting down or some + // other problem occurred and we should close the stream. + if ( object->isStreamRunning() == false ) return; - bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) - { - JackHandle *handle = (JackHandle *) stream_.apiHandle; - - // Look for jack server and try to become a client (only do once per stream). - jack_client_t *client = 0; - if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) { - jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption; - jack_status_t *status = NULL; - if ( options && !options->streamName.empty() ) - client = jack_client_open( options->streamName.c_str(), jackoptions, status ); - else - client = jack_client_open( "RtApiJack", jackoptions, status ); - if ( client == 0 ) { - errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!"; - error( RtError::WARNING ); - return FAILURE; - } - } - else { - // The handle must have been created on an earlier pass. - client = handle->client; - } - - const char **ports; - std::string port, previousPort, deviceName; - unsigned int nPorts = 0, nDevices = 0; - ports = jack_get_ports( client, NULL, NULL, 0 ); - if ( ports ) { - // Parse the port names up to the first colon (:). - size_t iColon = 0; - do { - port = (char *) ports[ nPorts ]; - iColon = port.find(":"); - if ( iColon != std::string::npos ) { - port = port.substr( 0, iColon ); - if ( port != previousPort ) { - if ( nDevices == device ) deviceName = port; - nDevices++; - previousPort = port; - } - } - } while ( ports[++nPorts] ); - free( ports ); - } + object->closeStream(); + std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl; +} - if ( device >= nDevices ) { - errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } +int jackXrun( void *infoPointer ) +{ + JackHandle *handle = (JackHandle *) infoPointer; - // Count the available ports containing the client name as device - // channels. Jack "input ports" equal RtAudio output channels. - unsigned int nChannels = 0; - unsigned long flag = JackPortIsInput; - if ( mode == INPUT ) flag = JackPortIsOutput; - ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); - if ( ports ) { - while ( ports[ nChannels ] ) nChannels++; - free( ports ); - } + if ( handle->ports[0] ) handle->xrun[0] = true; + if ( handle->ports[1] ) handle->xrun[1] = true; - // Compare the jack ports for specified client to the requested number of channels. - if ( nChannels < (channels + firstChannel) ) { - errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } + return 0; +} - // Check the jack server sample rate. - unsigned int jackRate = jack_get_sample_rate( client ); - if ( sampleRate != jackRate ) { - jack_client_close( client ); - errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ")."; - errorText_ = errorStream_.str(); +bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + JackHandle *handle = (JackHandle *) stream_.apiHandle; + + // Look for jack server and try to become a client (only do once per stream). + jack_client_t *client = 0; + if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) { + jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption; + jack_status_t *status = NULL; + if ( options && !options->streamName.empty() ) + client = jack_client_open( options->streamName.c_str(), jackoptions, status ); + else + client = jack_client_open( "RtApiJack", jackoptions, status ); + if ( client == 0 ) { + errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!"; + error( RtError::WARNING ); return FAILURE; } - stream_.sampleRate = jackRate; - - // Get the latency of the JACK port. - ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); - if ( ports[ firstChannel ] ) - stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) ); + } + else { + // The handle must have been created on an earlier pass. + client = handle->client; + } + + const char **ports; + std::string port, previousPort, deviceName; + unsigned int nPorts = 0, nDevices = 0; + ports = jack_get_ports( client, NULL, NULL, 0 ); + if ( ports ) { + // Parse the port names up to the first colon (:). + size_t iColon = 0; + do { + port = (char *) ports[ nPorts ]; + iColon = port.find(":"); + if ( iColon != std::string::npos ) { + port = port.substr( 0, iColon ); + if ( port != previousPort ) { + if ( nDevices == device ) deviceName = port; + nDevices++; + previousPort = port; + } + } + } while ( ports[++nPorts] ); free( ports ); + } - // The jack server always uses 32-bit floating-point data. - stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - stream_.userFormat = format; + if ( device >= nDevices ) { + errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; + // Count the available ports containing the client name as device + // channels. Jack "input ports" equal RtAudio output channels. + unsigned int nChannels = 0; + unsigned long flag = JackPortIsInput; + if ( mode == INPUT ) flag = JackPortIsOutput; + ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); + if ( ports ) { + while ( ports[ nChannels ] ) nChannels++; + free( ports ); + } - // Jack always uses non-interleaved buffers. - stream_.deviceInterleaved[mode] = false; + // Compare the jack ports for specified client to the requested number of channels. + if ( nChannels < (channels + firstChannel) ) { + errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Jack always provides host byte-ordered data. - stream_.doByteSwap[mode] = false; + // Check the jack server sample rate. + unsigned int jackRate = jack_get_sample_rate( client ); + if ( sampleRate != jackRate ) { + jack_client_close( client ); + errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.sampleRate = jackRate; - // Get the buffer size. The buffer size and number of buffers - // (periods) is set when the jack server is started. - stream_.bufferSize = (int) jack_get_buffer_size( client ); - *bufferSize = stream_.bufferSize; + // Get the latency of the JACK port. + ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); + if ( ports[ firstChannel ] ) + stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) ); + free( ports ); - stream_.nDeviceChannels[mode] = channels; - stream_.nUserChannels[mode] = channels; + // The jack server always uses 32-bit floating-point data. + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + stream_.userFormat = format; - // Set flags for buffer conversion. - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; - // Allocate our JackHandle structure for the stream. - if ( handle == 0 ) { - try { - handle = new JackHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory."; - goto error; - } + // Jack always uses non-interleaved buffers. + stream_.deviceInterleaved[mode] = false; - if ( pthread_cond_init(&handle->condition, NULL) ) { - errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable."; - goto error; - } - stream_.apiHandle = (void *) handle; - handle->client = client; - } - handle->deviceName[mode] = deviceName; + // Jack always provides host byte-ordered data. + stream_.doByteSwap[mode] = false; - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } + // Get the buffer size. The buffer size and number of buffers + // (periods) is set when the jack server is started. + stream_.bufferSize = (int) jack_get_buffer_size( client ); + *bufferSize = stream_.bufferSize; - if ( stream_.doConvertBuffer[mode] ) { + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; - bool makeBuffer = true; - if ( mode == OUTPUT ) - bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - else { // mode == INPUT - bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] ); - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]); - if ( bufferBytes < bytesOut ) makeBuffer = false; - } - } + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } + // Allocate our JackHandle structure for the stream. + if ( handle == 0 ) { + try { + handle = new JackHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory."; + goto error; } - // Allocate memory for the Jack ports (channels) identifiers. - handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels ); - if ( handle->ports[mode] == NULL ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory."; + if ( pthread_cond_init(&handle->condition, NULL) ) { + errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable."; goto error; } + stream_.apiHandle = (void *) handle; + handle->client = client; + } + handle->deviceName[mode] = deviceName; - stream_.device[mode] = device; - stream_.channelOffset[mode] = firstChannel; - stream_.state = STREAM_STOPPED; - stream_.callbackInfo.object = (void *) this; + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } - if ( stream_.mode == OUTPUT && mode == INPUT ) - // We had already set up the stream for output. - stream_.mode = DUPLEX; - else { - stream_.mode = mode; - jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo ); - jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle ); - jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo ); - } + if ( stream_.doConvertBuffer[mode] ) { - // Register our ports. - char label[64]; - if ( mode == OUTPUT ) { - for ( unsigned int i=0; iports[0][i] = jack_port_register( handle->client, (const char *)label, - JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 ); + bool makeBuffer = true; + if ( mode == OUTPUT ) + bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + else { // mode == INPUT + bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] ); + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]); + if ( bufferBytes < bytesOut ) makeBuffer = false; } } - else { - for ( unsigned int i=0; iports[1][i] = jack_port_register( handle->client, (const char *)label, - JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 ); + + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory."; + goto error; } } + } - // Setup the buffer conversion information structure. We don't use - // buffers to do channel offsets, so we override that parameter - // here. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); - - return SUCCESS; + // Allocate memory for the Jack ports (channels) identifiers. + handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels ); + if ( handle->ports[mode] == NULL ) { + errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory."; + goto error; + } - error: - if ( handle ) { - pthread_cond_destroy( &handle->condition ); - jack_client_close( handle->client ); + stream_.device[mode] = device; + stream_.channelOffset[mode] = firstChannel; + stream_.state = STREAM_STOPPED; + stream_.callbackInfo.object = (void *) this; - if ( handle->ports[0] ) free( handle->ports[0] ); - if ( handle->ports[1] ) free( handle->ports[1] ); + if ( stream_.mode == OUTPUT && mode == INPUT ) + // We had already set up the stream for output. + stream_.mode = DUPLEX; + else { + stream_.mode = mode; + jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo ); + jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle ); + jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo ); + } - delete handle; - stream_.apiHandle = 0; + // Register our ports. + char label[64]; + if ( mode == OUTPUT ) { + for ( unsigned int i=0; iports[0][i] = jack_port_register( handle->client, (const char *)label, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 ); } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } + } + else { + for ( unsigned int i=0; iports[1][i] = jack_port_register( handle->client, (const char *)label, + JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 ); } + } - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } + // Setup the buffer conversion information structure. We don't use + // buffers to do channel offsets, so we override that parameter + // here. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); - return FAILURE; - } + return SUCCESS; - void RtApiJack :: closeStream( void ) - { - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiJack::closeStream(): no open stream to close!"; - error( RtError::WARNING ); - return; - } + error: + if ( handle ) { + pthread_cond_destroy( &handle->condition ); + jack_client_close( handle->client ); - JackHandle *handle = (JackHandle *) stream_.apiHandle; - if ( handle ) { + if ( handle->ports[0] ) free( handle->ports[0] ); + if ( handle->ports[1] ) free( handle->ports[1] ); - if ( stream_.state == STREAM_RUNNING ) - jack_deactivate( handle->client ); + delete handle; + stream_.apiHandle = 0; + } - jack_client_close( handle->client ); + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; } + } - if ( handle ) { - if ( handle->ports[0] ) free( handle->ports[0] ); - if ( handle->ports[1] ) free( handle->ports[1] ); - pthread_cond_destroy( &handle->condition ); - delete handle; - stream_.apiHandle = 0; - } + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } + return FAILURE; +} - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } +void RtApiJack :: closeStream( void ) +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiJack::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + if ( handle ) { + + if ( stream_.state == STREAM_RUNNING ) + jack_deactivate( handle->client ); - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; + jack_client_close( handle->client ); } - void RtApiJack :: startStream( void ) - { - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiJack::startStream(): the stream is already running!"; - error( RtError::WARNING ); - return; + if ( handle ) { + if ( handle->ports[0] ) free( handle->ports[0] ); + if ( handle->ports[1] ) free( handle->ports[1] ); + pthread_cond_destroy( &handle->condition ); + delete handle; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; } + } + + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } - MUTEX_LOCK(&stream_.mutex); + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} - JackHandle *handle = (JackHandle *) stream_.apiHandle; - int result = jack_activate( handle->client ); - if ( result ) { - errorText_ = "RtApiJack::startStream(): unable to activate JACK client!"; +void RtApiJack :: startStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiJack::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK(&stream_.mutex); + + JackHandle *handle = (JackHandle *) stream_.apiHandle; + int result = jack_activate( handle->client ); + if ( result ) { + errorText_ = "RtApiJack::startStream(): unable to activate JACK client!"; + goto unlock; + } + + const char **ports; + + // Get the list of available ports. + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + result = 1; + ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput); + if ( ports == NULL) { + errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!"; goto unlock; } - const char **ports; - - // Get the list of available ports. - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + // Now make the port connections. Since RtAudio wasn't designed to + // allow the user to select particular channels of a device, we'll + // just open the first "nChannels" ports with offset. + for ( unsigned int i=0; iclient, handle->deviceName[0].c_str(), NULL, JackPortIsInput); - if ( ports == NULL) { - errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!"; + if ( ports[ stream_.channelOffset[0] + i ] ) + result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] ); + if ( result ) { + free( ports ); + errorText_ = "RtApiJack::startStream(): error connecting output ports!"; goto unlock; } + } + free(ports); + } - // Now make the port connections. Since RtAudio wasn't designed to - // allow the user to select particular channels of a device, we'll - // just open the first "nChannels" ports with offset. - for ( unsigned int i=0; iclient, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] ); - if ( result ) { - free( ports ); - errorText_ = "RtApiJack::startStream(): error connecting output ports!"; - goto unlock; - } - } - free(ports); + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + result = 1; + ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput ); + if ( ports == NULL) { + errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!"; + goto unlock; } - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + // Now make the port connections. See note above. + for ( unsigned int i=0; iclient, handle->deviceName[1].c_str(), NULL, JackPortIsOutput ); - if ( ports == NULL) { - errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!"; + if ( ports[ stream_.channelOffset[1] + i ] ) + result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) ); + if ( result ) { + free( ports ); + errorText_ = "RtApiJack::startStream(): error connecting input ports!"; goto unlock; } - - // Now make the port connections. See note above. - for ( unsigned int i=0; iclient, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) ); - if ( result ) { - free( ports ); - errorText_ = "RtApiJack::startStream(): error connecting input ports!"; - goto unlock; - } - } - free(ports); } + free(ports); + } + + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; - handle->drainCounter = 0; - handle->internalDrain = false; - stream_.state = STREAM_RUNNING; + unlock: + MUTEX_UNLOCK(&stream_.mutex); - unlock: - MUTEX_UNLOCK(&stream_.mutex); + if ( result == 0 ) return; + error( RtError::SYSTEM_ERROR ); +} - if ( result == 0 ) return; - error( RtError::SYSTEM_ERROR ); +void RtApiJack :: stopStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiJack::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; } - void RtApiJack :: stopStream( void ) - { - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiJack::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } + MUTEX_LOCK( &stream_.mutex ); - MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } - JackHandle *handle = (JackHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + JackHandle *handle = (JackHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 1; - pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled - } + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 1; + pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled } + } - jack_deactivate( handle->client ); - stream_.state = STREAM_STOPPED; + jack_deactivate( handle->client ); + stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); + MUTEX_UNLOCK( &stream_.mutex ); +} + +void RtApiJack :: abortStream( void ) +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiJack::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; } - void RtApiJack :: abortStream( void ) - { - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiJack::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } + JackHandle *handle = (JackHandle *) stream_.apiHandle; + handle->drainCounter = 1; - JackHandle *handle = (JackHandle *) stream_.apiHandle; - handle->drainCounter = 1; + stopStream(); +} - stopStream(); +bool RtApiJack :: callbackEvent( unsigned long nframes ) +{ + if ( stream_.state == STREAM_STOPPED ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return FAILURE; + } + if ( stream_.bufferSize != nframes ) { + errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!"; + error( RtError::WARNING ); + return FAILURE; } - bool RtApiJack :: callbackEvent( unsigned long nframes ) - { - if ( stream_.state == STREAM_STOPPED ) return SUCCESS; - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); - return FAILURE; - } - if ( stream_.bufferSize != nframes ) { - errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!"; - error( RtError::WARNING ); - return FAILURE; - } + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + JackHandle *handle = (JackHandle *) stream_.apiHandle; - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - JackHandle *handle = (JackHandle *) stream_.apiHandle; + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + if ( handle->internalDrain == false ) + pthread_cond_signal( &handle->condition ); + else + stopStream(); + return SUCCESS; + } - // Check if we were draining the stream and signal is finished. - if ( handle->drainCounter > 3 ) { - if ( handle->internalDrain == false ) - pthread_cond_signal( &handle->condition ); - else - stopStream(); - return SUCCESS; - } + MUTEX_LOCK( &stream_.mutex ); - MUTEX_LOCK( &stream_.mutex ); + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return SUCCESS; + } - // Invoke user callback first, to get fresh output data. - if ( handle->drainCounter == 0 ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; - } - handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( handle->drainCounter == 2 ) { - MUTEX_UNLOCK( &stream_.mutex ); - abortStream(); - return SUCCESS; - } - else if ( handle->drainCounter == 1 ) - handle->internalDrain = true; + // Invoke user callback first, to get fresh output data. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( handle->drainCounter == 2 ) { + MUTEX_UNLOCK( &stream_.mutex ); + abortStream(); + return SUCCESS; + } + else if ( handle->drainCounter == 1 ) + handle->internalDrain = true; + } - jack_default_audio_sample_t *jackbuffer; - unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t ); - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - if ( handle->drainCounter > 0 ) { // write zeros to the output stream + jack_default_audio_sample_t *jackbuffer; + unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t ); + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); - memset( jackbuffer, 0, bufferBytes ); - } + if ( handle->drainCounter > 0 ) { // write zeros to the output stream + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memset( jackbuffer, 0, bufferBytes ); } - else if ( stream_.doConvertBuffer[0] ) { - convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + } + else if ( stream_.doConvertBuffer[0] ) { - for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); - memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes ); - } + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes ); } - else { // no buffer conversion - for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); - memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes ); - } + } + else { // no buffer conversion + for ( unsigned int i=0; iports[0][i], (jack_nframes_t) nframes ); + memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes ); } + } - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; } + } - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - if ( stream_.doConvertBuffer[1] ) { - for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); - memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes ); - } - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + if ( stream_.doConvertBuffer[1] ) { + for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); + memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes ); } - else { // no buffer conversion - for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); - memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes ); - } + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + } + else { // no buffer conversion + for ( unsigned int i=0; iports[1][i], (jack_nframes_t) nframes ); + memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes ); } } + } - unlock: - MUTEX_UNLOCK(&stream_.mutex); + unlock: + MUTEX_UNLOCK(&stream_.mutex); - RtApi::tickStreamTime(); - return SUCCESS; - } + RtApi::tickStreamTime(); + return SUCCESS; +} //******************** End of __UNIX_JACK__ *********************// #endif #if defined(__WINDOWS_ASIO__) // ASIO API on Windows - // The ASIO API is designed around a callback scheme, so this - // implementation is similar to that used for OS-X CoreAudio and Linux - // Jack. The primary constraint with ASIO is that it only allows - // access to a single driver at a time. Thus, it is not possible to - // have more than one simultaneous RtAudio stream. - // - // This implementation also requires a number of external ASIO files - // and a few global variables. The ASIO callback scheme does not - // allow for the passing of user data, so we must create a global - // pointer to our callbackInfo structure. - // - // On unix systems, we make use of a pthread condition variable. - // Since there is no equivalent in Windows, I hacked something based - // on information found in - // http://www.cs.wustl.edu/~schmidt/win32-cv-1.html. +// The ASIO API is designed around a callback scheme, so this +// implementation is similar to that used for OS-X CoreAudio and Linux +// Jack. The primary constraint with ASIO is that it only allows +// access to a single driver at a time. Thus, it is not possible to +// have more than one simultaneous RtAudio stream. +// +// This implementation also requires a number of external ASIO files +// and a few global variables. The ASIO callback scheme does not +// allow for the passing of user data, so we must create a global +// pointer to our callbackInfo structure. +// +// On unix systems, we make use of a pthread condition variable. +// Since there is no equivalent in Windows, I hacked something based +// on information found in +// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html. #include "asiosys.h" #include "asio.h" @@ -2430,941 +2448,946 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, #include "asiodrivers.h" #include - AsioDrivers drivers; - ASIOCallbacks asioCallbacks; - ASIODriverInfo driverInfo; - CallbackInfo *asioCallbackInfo; - bool asioXRun; - - struct AsioHandle { - int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - ASIOBufferInfo *bufferInfos; - HANDLE condition; - - AsioHandle() - :drainCounter(0), internalDrain(false), bufferInfos(0) {} - }; +AsioDrivers drivers; +ASIOCallbacks asioCallbacks; +ASIODriverInfo driverInfo; +CallbackInfo *asioCallbackInfo; +bool asioXRun; - // Function declarations (definitions at end of section) - static const char* getAsioErrorString( ASIOError result ); - void sampleRateChanged( ASIOSampleRate sRate ); - long asioMessages( long selector, long value, void* message, double* opt ); +struct AsioHandle { + int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + ASIOBufferInfo *bufferInfos; + HANDLE condition; - RtApiAsio :: RtApiAsio() - { - // ASIO cannot run on a multi-threaded appartment. You can call - // CoInitialize beforehand, but it must be for appartment threading - // (in which case, CoInitilialize will return S_FALSE here). - coInitialized_ = false; - HRESULT hr = CoInitialize( NULL ); - if ( FAILED(hr) ) { - errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"; - error( RtError::WARNING ); - } - coInitialized_ = true; + AsioHandle() + :drainCounter(0), internalDrain(false), bufferInfos(0) {} +}; - drivers.removeCurrentDriver(); - driverInfo.asioVersion = 2; +// Function declarations (definitions at end of section) +static const char* getAsioErrorString( ASIOError result ); +void sampleRateChanged( ASIOSampleRate sRate ); +long asioMessages( long selector, long value, void* message, double* opt ); - // See note in DirectSound implementation about GetDesktopWindow(). - driverInfo.sysRef = GetForegroundWindow(); +RtApiAsio :: RtApiAsio() +{ + // ASIO cannot run on a multi-threaded appartment. You can call + // CoInitialize beforehand, but it must be for appartment threading + // (in which case, CoInitilialize will return S_FALSE here). + coInitialized_ = false; + HRESULT hr = CoInitialize( NULL ); + if ( FAILED(hr) ) { + errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"; + error( RtError::WARNING ); } + coInitialized_ = true; - RtApiAsio :: ~RtApiAsio() - { - if ( stream_.state != STREAM_CLOSED ) closeStream(); - if ( coInitialized_ ) CoUninitialize(); - } + drivers.removeCurrentDriver(); + driverInfo.asioVersion = 2; - unsigned int RtApiAsio :: getDeviceCount( void ) - { - return (unsigned int) drivers.asioGetNumDev(); - } + // See note in DirectSound implementation about GetDesktopWindow(). + driverInfo.sysRef = GetForegroundWindow(); +} - RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) - { - RtAudio::DeviceInfo info; - info.probed = false; +RtApiAsio :: ~RtApiAsio() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); + if ( coInitialized_ ) CoUninitialize(); +} - // Get device ID - unsigned int nDevices = getDeviceCount(); - if ( nDevices == 0 ) { - errorText_ = "RtApiAsio::getDeviceInfo: no devices found!"; - error( RtError::INVALID_USE ); - } +unsigned int RtApiAsio :: getDeviceCount( void ) +{ + return (unsigned int) drivers.asioGetNumDev(); +} - if ( device >= nDevices ) { - errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - } +RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; - // If a stream is already open, we cannot probe other devices. Thus, use the saved results. - if ( stream_.state != STREAM_CLOSED ) { - if ( device >= devices_.size() ) { - errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened."; - error( RtError::WARNING ); - return info; - } - return devices_[ device ]; - } + // Get device ID + unsigned int nDevices = getDeviceCount(); + if ( nDevices == 0 ) { + errorText_ = "RtApiAsio::getDeviceInfo: no devices found!"; + error( RtError::INVALID_USE ); + } - char driverName[32]; - ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - info.name = driverName; - - if ( !drivers.loadDriver( driverName ) ) { - errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } + if ( device >= nDevices ) { + errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } - result = ASIOInit( &driverInfo ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; - errorText_ = errorStream_.str(); + // If a stream is already open, we cannot probe other devices. Thus, use the saved results. + if ( stream_.state != STREAM_CLOSED ) { + if ( device >= devices_.size() ) { + errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened."; error( RtError::WARNING ); return info; } + return devices_[ device ]; + } - // Determine the device channel information. - long inputChannels, outputChannels; - result = ASIOGetChannels( &inputChannels, &outputChannels ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } + char driverName[32]; + ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } - info.outputChannels = outputChannels; - info.inputChannels = inputChannels; - if ( info.outputChannels > 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + info.name = driverName; - // Determine the supported sample rates. - info.sampleRates.clear(); - for ( unsigned int i=0; i 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - info.probed = true; + // Determine the supported sample rates. + info.sampleRates.clear(); + for ( unsigned int i=0; iobject; - object->callbackEvent( index ); - } + info.nativeFormats = 0; + if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) + info.nativeFormats |= RTAUDIO_SINT16; + else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) + info.nativeFormats |= RTAUDIO_SINT32; + else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) + info.nativeFormats |= RTAUDIO_FLOAT32; + else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) + info.nativeFormats |= RTAUDIO_FLOAT64; - void RtApiAsio :: saveDeviceInfo( void ) - { - devices_.clear(); + if ( getDefaultOutputDevice() == device ) + info.isDefaultOutput = true; + if ( getDefaultInputDevice() == device ) + info.isDefaultInput = true; - unsigned int nDevices = getDeviceCount(); - devices_.resize( nDevices ); - for ( unsigned int i=0; iobject; + object->callbackEvent( index ); +} - char driverName[32]; - ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } +void RtApiAsio :: saveDeviceInfo( void ) +{ + devices_.clear(); - // The getDeviceInfo() function will not work when a stream is open - // because ASIO does not allow multiple devices to run at the same - // time. Thus, we'll probe the system before opening a stream and - // save the results for use by getDeviceInfo(). - this->saveDeviceInfo(); + unsigned int nDevices = getDeviceCount(); + devices_.resize( nDevices ); + for ( unsigned int i=0; isaveDeviceInfo(); - if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) || - ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ")."; + // Only load the driver once for duplex stream. + if ( mode != INPUT || stream_.mode != OUTPUT ) { + if ( !drivers.loadDriver( driverName ) ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ")."; errorText_ = errorStream_.str(); return FAILURE; } - stream_.nDeviceChannels[mode] = channels; - stream_.nUserChannels[mode] = channels; - stream_.channelOffset[mode] = firstChannel; - // Verify the sample rate is supported. - result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate ); + result = ASIOInit( &driverInfo ); if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ")."; + errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; errorText_ = errorStream_.str(); return FAILURE; } + } - // Get the current sample rate - ASIOSampleRate currentRate; - result = ASIOGetSampleRate( ¤tRate ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Check the device channel count. + long inputChannels, outputChannels; + result = ASIOGetChannels( &inputChannels, &outputChannels ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Set the sample rate only if necessary - if ( currentRate != sampleRate ) { - result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } + if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) || + ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.nDeviceChannels[mode] = channels; + stream_.nUserChannels[mode] = channels; + stream_.channelOffset[mode] = firstChannel; - // Determine the driver data type. - ASIOChannelInfo channelInfo; - channelInfo.channel = 0; - if ( mode == OUTPUT ) channelInfo.isInput = false; - else channelInfo.isInput = true; - result = ASIOGetChannelInfo( &channelInfo ); + // Verify the sample rate is supported. + result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Get the current sample rate + ASIOSampleRate currentRate; + result = ASIOGetSampleRate( ¤tRate ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the sample rate only if necessary + if ( currentRate != sampleRate ) { + result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); if ( result != ASE_OK ) { drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format."; + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ")."; errorText_ = errorStream_.str(); return FAILURE; } + } - // Assuming WINDOWS host is always little-endian. - stream_.doByteSwap[mode] = false; - stream_.userFormat = format; - stream_.deviceFormat[mode] = 0; - if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true; - } - else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true; - } - else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true; - } - else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; - if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true; - } + // Determine the driver data type. + ASIOChannelInfo channelInfo; + channelInfo.channel = 0; + if ( mode == OUTPUT ) channelInfo.isInput = false; + else channelInfo.isInput = true; + result = ASIOGetChannelInfo( &channelInfo ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format."; + errorText_ = errorStream_.str(); + return FAILURE; + } - if ( stream_.deviceFormat[mode] == 0 ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - return FAILURE; + // Assuming WINDOWS host is always little-endian. + stream_.doByteSwap[mode] = false; + stream_.userFormat = format; + stream_.deviceFormat[mode] = 0; + if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true; + } + else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; + if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true; + } + + if ( stream_.deviceFormat[mode] == 0 ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // Set the buffer size. For a duplex stream, this will end up + // setting the buffer size based on the input constraints, which + // should be ok. + long minSize, maxSize, preferSize, granularity; + result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity ); + if ( result != ASE_OK ) { + drivers.removeCurrentDriver(); + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; + else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; + else if ( granularity == -1 ) { + // Make sure bufferSize is a power of two. + int log2_of_min_size = 0; + int log2_of_max_size = 0; + + for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) { + if ( minSize & ((long)1 << i) ) log2_of_min_size = i; + if ( maxSize & ((long)1 << i) ) log2_of_max_size = i; } - // Set the buffer size. For a duplex stream, this will end up - // setting the buffer size based on the input constraints, which - // should be ok. - long minSize, maxSize, preferSize, granularity; - result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size."; - errorText_ = errorStream_.str(); - return FAILURE; + long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) ); + int min_delta_num = log2_of_min_size; + + for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) { + long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) ); + if (current_delta < min_delta) { + min_delta = current_delta; + min_delta_num = i; + } } + *bufferSize = ( (unsigned int)1 << min_delta_num ); if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; - else if ( granularity == -1 ) { - // Make sure bufferSize is a power of two. - int log2_of_min_size = 0; - int log2_of_max_size = 0; - - for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) { - if ( minSize & ((long)1 << i) ) log2_of_min_size = i; - if ( maxSize & ((long)1 << i) ) log2_of_max_size = i; - } + } + else if ( granularity != 0 ) { + // Set to an even multiple of granularity, rounding up. + *bufferSize = (*bufferSize + granularity-1) / granularity * granularity; + } + + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) { + drivers.removeCurrentDriver(); + errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!"; + return FAILURE; + } - long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) ); - int min_delta_num = log2_of_min_size; + stream_.bufferSize = *bufferSize; + stream_.nBuffers = 2; - for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) { - long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) ); - if (current_delta < min_delta) { - min_delta = current_delta; - min_delta_num = i; - } - } + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; - *bufferSize = ( (unsigned int)1 << min_delta_num ); - if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; - else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; - } - else if ( granularity != 0 ) { - // Set to an even multiple of granularity, rounding up. - *bufferSize = (*bufferSize + granularity-1) / granularity * granularity; - } + // ASIO always uses non-interleaved buffers. + stream_.deviceInterleaved[mode] = false; - if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) { + // Allocate, if necessary, our AsioHandle structure for the stream. + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( handle == 0 ) { + try { + handle = new AsioHandle; + } + catch ( std::bad_alloc& ) { + //if ( handle == NULL ) { drivers.removeCurrentDriver(); - errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!"; + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory."; return FAILURE; } + handle->bufferInfos = 0; - stream_.bufferSize = *bufferSize; - stream_.nBuffers = 2; - - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; + // Create a manual-reset event. + handle->condition = CreateEvent( NULL, // no security + TRUE, // manual-reset + FALSE, // non-signaled initially + NULL ); // unnamed + stream_.apiHandle = (void *) handle; + } - // ASIO always uses non-interleaved buffers. - stream_.deviceInterleaved[mode] = false; + // Create the ASIO internal buffers. Since RtAudio sets up input + // and output separately, we'll have to dispose of previously + // created output buffers for a duplex stream. + long inputLatency, outputLatency; + if ( mode == INPUT && stream_.mode == OUTPUT ) { + ASIODisposeBuffers(); + if ( handle->bufferInfos ) free( handle->bufferInfos ); + } - // Allocate, if necessary, our AsioHandle structure for the stream. - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - if ( handle == 0 ) { - try { - handle = new AsioHandle; - } - catch ( std::bad_alloc& ) { - //if ( handle == NULL ) { - drivers.removeCurrentDriver(); - errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory."; - return FAILURE; - } - handle->bufferInfos = 0; - - // Create a manual-reset event. - handle->condition = CreateEvent( NULL, // no security - TRUE, // manual-reset - FALSE, // non-signaled initially - NULL ); // unnamed - stream_.apiHandle = (void *) handle; - } - - // Create the ASIO internal buffers. Since RtAudio sets up input - // and output separately, we'll have to dispose of previously - // created output buffers for a duplex stream. - long inputLatency, outputLatency; - if ( mode == INPUT && stream_.mode == OUTPUT ) { - ASIODisposeBuffers(); - if ( handle->bufferInfos ) free( handle->bufferInfos ); - } - - // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. - bool buffersAllocated = false; - unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; - handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) ); - if ( handle->bufferInfos == NULL ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - goto error; - } + // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. + bool buffersAllocated = false; + unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; + handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) ); + if ( handle->bufferInfos == NULL ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ")."; + errorText_ = errorStream_.str(); + goto error; + } - ASIOBufferInfo *infos; - infos = handle->bufferInfos; - for ( i=0; iisInput = ASIOFalse; - infos->channelNum = i + stream_.channelOffset[0]; - infos->buffers[0] = infos->buffers[1] = 0; - } - for ( i=0; iisInput = ASIOTrue; - infos->channelNum = i + stream_.channelOffset[1]; - infos->buffers[0] = infos->buffers[1] = 0; - } + ASIOBufferInfo *infos; + infos = handle->bufferInfos; + for ( i=0; iisInput = ASIOFalse; + infos->channelNum = i + stream_.channelOffset[0]; + infos->buffers[0] = infos->buffers[1] = 0; + } + for ( i=0; iisInput = ASIOTrue; + infos->channelNum = i + stream_.channelOffset[1]; + infos->buffers[0] = infos->buffers[1] = 0; + } - // Set up the ASIO callback structure and create the ASIO data buffers. - asioCallbacks.bufferSwitch = &bufferSwitch; - asioCallbacks.sampleRateDidChange = &sampleRateChanged; - asioCallbacks.asioMessage = &asioMessages; - asioCallbacks.bufferSwitchTimeInfo = NULL; - result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers."; - errorText_ = errorStream_.str(); - goto error; - } - buffersAllocated = true; + // Set up the ASIO callback structure and create the ASIO data buffers. + asioCallbacks.bufferSwitch = &bufferSwitch; + asioCallbacks.sampleRateDidChange = &sampleRateChanged; + asioCallbacks.asioMessage = &asioMessages; + asioCallbacks.bufferSwitchTimeInfo = NULL; + result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers."; + errorText_ = errorStream_.str(); + goto error; + } + buffersAllocated = true; - // Set flags for buffer conversion. - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; + // Set flags for buffer conversion. + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; - // Allocate necessary internal buffers - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } + // Allocate necessary internal buffers + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } - if ( stream_.doConvertBuffer[mode] ) { + if ( stream_.doConvertBuffer[mode] ) { - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; } + } - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory."; + goto error; } } + } - stream_.sampleRate = sampleRate; - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - asioCallbackInfo = &stream_.callbackInfo; - stream_.callbackInfo.object = (void *) this; - if ( stream_.mode == OUTPUT && mode == INPUT ) - // We had already set up an output stream. - stream_.mode = DUPLEX; - else - stream_.mode = mode; + stream_.sampleRate = sampleRate; + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + asioCallbackInfo = &stream_.callbackInfo; + stream_.callbackInfo.object = (void *) this; + if ( stream_.mode == OUTPUT && mode == INPUT ) + // We had already set up an output stream. + stream_.mode = DUPLEX; + else + stream_.mode = mode; - // Determine device latencies - result = ASIOGetLatencies( &inputLatency, &outputLatency ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency."; - errorText_ = errorStream_.str(); - error( RtError::WARNING); // warn but don't fail - } - else { - stream_.latency[0] = outputLatency; - stream_.latency[1] = inputLatency; - } + // Determine device latencies + result = ASIOGetLatencies( &inputLatency, &outputLatency ); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency."; + errorText_ = errorStream_.str(); + error( RtError::WARNING); // warn but don't fail + } + else { + stream_.latency[0] = outputLatency; + stream_.latency[1] = inputLatency; + } - // Setup the buffer conversion information structure. We don't use - // buffers to do channel offsets, so we override that parameter - // here. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); + // Setup the buffer conversion information structure. We don't use + // buffers to do channel offsets, so we override that parameter + // here. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); - return SUCCESS; + return SUCCESS; - error: - if ( buffersAllocated ) - ASIODisposeBuffers(); - drivers.removeCurrentDriver(); + error: + if ( buffersAllocated ) + ASIODisposeBuffers(); + drivers.removeCurrentDriver(); - if ( handle ) { - CloseHandle( handle->condition ); - if ( handle->bufferInfos ) - free( handle->bufferInfos ); - delete handle; - stream_.apiHandle = 0; - } + if ( handle ) { + CloseHandle( handle->condition ); + if ( handle->bufferInfos ) + free( handle->bufferInfos ); + delete handle; + stream_.apiHandle = 0; + } - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; } + } - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } - return FAILURE; + return FAILURE; +} + +void RtApiAsio :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAsio::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; } - void RtApiAsio :: closeStream() - { - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAsio::closeStream(): no open stream to close!"; - error( RtError::WARNING ); - return; - } + if ( stream_.state == STREAM_RUNNING ) { + stream_.state = STREAM_STOPPED; + ASIOStop(); + } + ASIODisposeBuffers(); + drivers.removeCurrentDriver(); - if ( stream_.state == STREAM_RUNNING ) { - stream_.state = STREAM_STOPPED; - ASIOStop(); - } - ASIODisposeBuffers(); - drivers.removeCurrentDriver(); + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( handle ) { + CloseHandle( handle->condition ); + if ( handle->bufferInfos ) + free( handle->bufferInfos ); + delete handle; + stream_.apiHandle = 0; + } - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - if ( handle ) { - CloseHandle( handle->condition ); - if ( handle->bufferInfos ) - free( handle->bufferInfos ); - delete handle; - stream_.apiHandle = 0; + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; } + } - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; +void RtApiAsio :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiAsio::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; } - void RtApiAsio :: startStream() - { - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiAsio::startStream(): the stream is already running!"; - error( RtError::WARNING ); - return; - } + MUTEX_LOCK( &stream_.mutex ); - MUTEX_LOCK( &stream_.mutex ); + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + ASIOError result = ASIOStart(); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device."; + errorText_ = errorStream_.str(); + goto unlock; + } - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - ASIOError result = ASIOStart(); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device."; - errorText_ = errorStream_.str(); - goto unlock; - } + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; + asioXRun = false; - handle->drainCounter = 0; - handle->internalDrain = false; - stream_.state = STREAM_RUNNING; - asioXRun = false; + unlock: + MUTEX_UNLOCK( &stream_.mutex ); - unlock: - MUTEX_UNLOCK( &stream_.mutex ); + if ( result == ASE_OK ) return; + error( RtError::SYSTEM_ERROR ); +} - if ( result == ASE_OK ) return; - error( RtError::SYSTEM_ERROR ); +void RtApiAsio :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; } - void RtApiAsio :: stopStream() - { - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } + MUTEX_LOCK( &stream_.mutex ); - MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 1; - MUTEX_UNLOCK( &stream_.mutex ); - WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled - ResetEvent( handle->condition ); - MUTEX_LOCK( &stream_.mutex ); - } + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 1; + MUTEX_UNLOCK( &stream_.mutex ); + WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled + ResetEvent( handle->condition ); + MUTEX_LOCK( &stream_.mutex ); } + } - ASIOError result = ASIOStop(); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device."; - errorText_ = errorStream_.str(); - } + ASIOError result = ASIOStop(); + if ( result != ASE_OK ) { + errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device."; + errorText_ = errorStream_.str(); + } - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result == ASE_OK ) return; + error( RtError::SYSTEM_ERROR ); +} - if ( result == ASE_OK ) return; - error( RtError::SYSTEM_ERROR ); +void RtApiAsio :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; } - void RtApiAsio :: abortStream() - { - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } + // The following lines were commented-out because some behavior was + // noted where the device buffers need to be zeroed to avoid + // continuing sound, even when the device buffers are completely + // disposed. So now, calling abort is the same as calling stop. + // AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + // handle->drainCounter = 1; + stopStream(); +} - // The following lines were commented-out because some behavior was - // noted where the device buffers need to be zeroed to avoid - // continuing sound, even when the device buffers are completed - // disposed. So now, calling abort is the same as calling stop. - //AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - //handle->drainCounter = 1; - stopStream(); +bool RtApiAsio :: callbackEvent( long bufferIndex ) +{ + if ( stream_.state == STREAM_STOPPED ) return SUCCESS; + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return FAILURE; } - bool RtApiAsio :: callbackEvent( long bufferIndex ) - { - if ( stream_.state == STREAM_STOPPED ) return SUCCESS; - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); - return FAILURE; - } + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > 3 ) { + if ( handle->internalDrain == false ) + SetEvent( handle->condition ); + else + stopStream(); + return SUCCESS; + } + + MUTEX_LOCK( &stream_.mutex ); - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; - // Check if we were draining the stream and signal is finished. - if ( handle->drainCounter > 3 ) { - if ( handle->internalDrain == false ) - SetEvent( handle->condition ); - else - stopStream(); + // Invoke user callback to get fresh output data UNLESS we are + // draining stream. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && asioXRun == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + asioXRun = false; + } + if ( stream_.mode != OUTPUT && asioXRun == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + asioXRun = false; + } + handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( handle->drainCounter == 2 ) { + MUTEX_UNLOCK( &stream_.mutex ); + abortStream(); return SUCCESS; } + else if ( handle->drainCounter == 1 ) + handle->internalDrain = true; + } - MUTEX_LOCK( &stream_.mutex ); + unsigned int nChannels, bufferBytes, i, j; + nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) goto unlock; + bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] ); - // Invoke user callback to get fresh output data UNLESS we are - // draining stream. - if ( handle->drainCounter == 0 ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && asioXRun == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - asioXRun = false; - } - if ( stream_.mode != OUTPUT && asioXRun == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - asioXRun = false; - } - handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( handle->drainCounter == 2 ) { - MUTEX_UNLOCK( &stream_.mutex ); - abortStream(); - return SUCCESS; + if ( handle->drainCounter > 1 ) { // write zeros to the output stream + + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes ); } - else if ( handle->drainCounter == 1 ) - handle->internalDrain = true; + } + else if ( stream_.doConvertBuffer[0] ) { - unsigned int nChannels, bufferBytes, i, j; - nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( stream_.deviceBuffer, + stream_.bufferSize * stream_.nDeviceChannels[0], + stream_.deviceFormat[0] ); - bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] ); + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memcpy( handle->bufferInfos[i].buffers[bufferIndex], + &stream_.deviceBuffer[j++*bufferBytes], bufferBytes ); + } - if ( handle->drainCounter > 1 ) { // write zeros to the output stream + } + else { - for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) - memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes ); - } + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( stream_.userBuffer[0], + stream_.bufferSize * stream_.nUserChannels[0], + stream_.userFormat ); + for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) + memcpy( handle->bufferInfos[i].buffers[bufferIndex], + &stream_.userBuffer[0][bufferBytes*j++], bufferBytes ); } - else if ( stream_.doConvertBuffer[0] ) { - convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - if ( stream_.doByteSwap[0] ) - byteSwapBuffer( stream_.deviceBuffer, - stream_.bufferSize * stream_.nDeviceChannels[0], - stream_.deviceFormat[0] ); - - for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) - memcpy( handle->bufferInfos[i].buffers[bufferIndex], - &stream_.deviceBuffer[j++*bufferBytes], bufferBytes ); - } + } - } - else { + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + } - if ( stream_.doByteSwap[0] ) - byteSwapBuffer( stream_.userBuffer[0], - stream_.bufferSize * stream_.nUserChannels[0], - stream_.userFormat ); + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - for ( i=0, j=0; ibufferInfos[i].isInput != ASIOTrue ) - memcpy( handle->bufferInfos[i].buffers[bufferIndex], - &stream_.userBuffer[0][bufferBytes*j++], bufferBytes ); - } + bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]); + if (stream_.doConvertBuffer[1]) { + + // Always interleave ASIO input data. + for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) + memcpy( &stream_.deviceBuffer[j++*bufferBytes], + handle->bufferInfos[i].buffers[bufferIndex], + bufferBytes ); } - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( stream_.deviceBuffer, + stream_.bufferSize * stream_.nDeviceChannels[1], + stream_.deviceFormat[1] ); + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + + } + else { + for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) { + memcpy( &stream_.userBuffer[1][bufferBytes*j++], + handle->bufferInfos[i].buffers[bufferIndex], + bufferBytes ); + } } + + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( stream_.userBuffer[1], + stream_.bufferSize * stream_.nUserChannels[1], + stream_.userFormat ); } + } - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + unlock: + // The following call was suggested by Malte Clasen. While the API + // documentation indicates it should not be required, some device + // drivers apparently do not function correctly without it. + ASIOOutputReady(); - bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]); + MUTEX_UNLOCK( &stream_.mutex ); - if (stream_.doConvertBuffer[1]) { + RtApi::tickStreamTime(); + return SUCCESS; +} - // Always interleave ASIO input data. - for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) - memcpy( &stream_.deviceBuffer[j++*bufferBytes], - handle->bufferInfos[i].buffers[bufferIndex], - bufferBytes ); - } +void sampleRateChanged( ASIOSampleRate sRate ) +{ + // The ASIO documentation says that this usually only happens during + // external sync. Audio processing is not stopped by the driver, + // actual sample rate might not have even changed, maybe only the + // sample rate status of an AES/EBU or S/PDIF digital input at the + // audio device. + + RtApi *object = (RtApi *) asioCallbackInfo->object; + try { + object->stopStream(); + } + catch ( RtError &exception ) { + std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl; + return; + } - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( stream_.deviceBuffer, - stream_.bufferSize * stream_.nDeviceChannels[1], - stream_.deviceFormat[1] ); - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl; +} - } - else { - for ( i=0, j=0; ibufferInfos[i].isInput == ASIOTrue ) { - memcpy( &stream_.userBuffer[1][bufferBytes*j++], - handle->bufferInfos[i].buffers[bufferIndex], - bufferBytes ); - } - } +long asioMessages( long selector, long value, void* message, double* opt ) +{ + long ret = 0; + + switch( selector ) { + case kAsioSelectorSupported: + if ( value == kAsioResetRequest + || value == kAsioEngineVersion + || value == kAsioResyncRequest + || value == kAsioLatenciesChanged + // The following three were added for ASIO 2.0, you don't + // necessarily have to support them. + || value == kAsioSupportsTimeInfo + || value == kAsioSupportsTimeCode + || value == kAsioSupportsInputMonitor) + ret = 1L; + break; + case kAsioResetRequest: + // Defer the task and perform the reset of the driver during the + // next "safe" situation. You cannot reset the driver right now, + // as this code is called from the driver. Reset the driver is + // done by completely destruct is. I.e. ASIOStop(), + // ASIODisposeBuffers(), Destruction Afterwards you initialize the + // driver again. + std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl; + ret = 1L; + break; + case kAsioResyncRequest: + // This informs the application that the driver encountered some + // non-fatal data loss. It is used for synchronization purposes + // of different media. Added mainly to work around the Win16Mutex + // problems in Windows 95/98 with the Windows Multimedia system, + // which could lose data because the Mutex was held too long by + // another thread. However a driver can issue it in other + // situations, too. + // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl; + asioXRun = true; + ret = 1L; + break; + case kAsioLatenciesChanged: + // This will inform the host application that the drivers were + // latencies changed. Beware, it this does not mean that the + // buffer sizes have changed! You might need to update internal + // delay data. + std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl; + ret = 1L; + break; + case kAsioEngineVersion: + // Return the supported ASIO version of the host application. If + // a host application does not implement this selector, ASIO 1.0 + // is assumed by the driver. + ret = 2L; + break; + case kAsioSupportsTimeInfo: + // Informs the driver whether the + // asioCallbacks.bufferSwitchTimeInfo() callback is supported. + // For compatibility with ASIO 1.0 drivers the host application + // should always support the "old" bufferSwitch method, too. + ret = 0; + break; + case kAsioSupportsTimeCode: + // Informs the driver whether application is interested in time + // code info. If an application does not need to know about time + // code, the driver has less work to do. + ret = 0; + break; + } + return ret; +} - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( stream_.userBuffer[1], - stream_.bufferSize * stream_.nUserChannels[1], - stream_.userFormat ); - } - } +static const char* getAsioErrorString( ASIOError result ) +{ + struct Messages + { + ASIOError value; + const char*message; + }; - unlock: - // The following call was suggested by Malte Clasen. While the API - // documentation indicates it should not be required, some device - // drivers apparently do not function correctly without it. - ASIOOutputReady(); + static Messages m[] = + { + { ASE_NotPresent, "Hardware input or output is not present or available." }, + { ASE_HWMalfunction, "Hardware is malfunctioning." }, + { ASE_InvalidParameter, "Invalid input parameter." }, + { ASE_InvalidMode, "Invalid mode." }, + { ASE_SPNotAdvancing, "Sample position not advancing." }, + { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." }, + { ASE_NoMemory, "Not enough memory to complete the request." } + }; - MUTEX_UNLOCK( &stream_.mutex ); + for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i ) + if ( m[i].value == result ) return m[i].message; - RtApi::tickStreamTime(); - return SUCCESS; - } + return "Unknown error."; +} +//******************** End of __WINDOWS_ASIO__ *********************// +#endif - void sampleRateChanged( ASIOSampleRate sRate ) - { - // The ASIO documentation says that this usually only happens during - // external sync. Audio processing is not stopped by the driver, - // actual sample rate might not have even changed, maybe only the - // sample rate status of an AES/EBU or S/PDIF digital input at the - // audio device. - RtApi *object = (RtApi *) asioCallbackInfo->object; - try { - object->stopStream(); - } - catch ( RtError &exception ) { - std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl; - return; - } +#if defined(__WINDOWS_DS__) // Windows DirectSound API - std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl; - } - - long asioMessages( long selector, long value, void* message, double* opt ) - { - long ret = 0; - - switch( selector ) { - case kAsioSelectorSupported: - if ( value == kAsioResetRequest - || value == kAsioEngineVersion - || value == kAsioResyncRequest - || value == kAsioLatenciesChanged - // The following three were added for ASIO 2.0, you don't - // necessarily have to support them. - || value == kAsioSupportsTimeInfo - || value == kAsioSupportsTimeCode - || value == kAsioSupportsInputMonitor) - ret = 1L; - break; - case kAsioResetRequest: - // Defer the task and perform the reset of the driver during the - // next "safe" situation. You cannot reset the driver right now, - // as this code is called from the driver. Reset the driver is - // done by completely destruct is. I.e. ASIOStop(), - // ASIODisposeBuffers(), Destruction Afterwards you initialize the - // driver again. - std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl; - ret = 1L; - break; - case kAsioResyncRequest: - // This informs the application that the driver encountered some - // non-fatal data loss. It is used for synchronization purposes - // of different media. Added mainly to work around the Win16Mutex - // problems in Windows 95/98 with the Windows Multimedia system, - // which could lose data because the Mutex was held too long by - // another thread. However a driver can issue it in other - // situations, too. - // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl; - asioXRun = true; - ret = 1L; - break; - case kAsioLatenciesChanged: - // This will inform the host application that the drivers were - // latencies changed. Beware, it this does not mean that the - // buffer sizes have changed! You might need to update internal - // delay data. - std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl; - ret = 1L; - break; - case kAsioEngineVersion: - // Return the supported ASIO version of the host application. If - // a host application does not implement this selector, ASIO 1.0 - // is assumed by the driver. - ret = 2L; - break; - case kAsioSupportsTimeInfo: - // Informs the driver whether the - // asioCallbacks.bufferSwitchTimeInfo() callback is supported. - // For compatibility with ASIO 1.0 drivers the host application - // should always support the "old" bufferSwitch method, too. - ret = 0; - break; - case kAsioSupportsTimeCode: - // Informs the driver whether application is interested in time - // code info. If an application does not need to know about time - // code, the driver has less work to do. - ret = 0; - break; - } - return ret; - } - - static const char* getAsioErrorString( ASIOError result ) - { - struct Messages - { - ASIOError value; - const char*message; - }; - - static Messages m[] = - { - { ASE_NotPresent, "Hardware input or output is not present or available." }, - { ASE_HWMalfunction, "Hardware is malfunctioning." }, - { ASE_InvalidParameter, "Invalid input parameter." }, - { ASE_InvalidMode, "Invalid mode." }, - { ASE_SPNotAdvancing, "Sample position not advancing." }, - { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." }, - { ASE_NoMemory, "Not enough memory to complete the request." } - }; - - for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i ) - if ( m[i].value == result ) return m[i].message; - - return "Unknown error."; - } - //******************** End of __WINDOWS_ASIO__ *********************// -#endif - - -#if defined(__WINDOWS_DS__) // Windows DirectSound API - - // Modified by Robin Davies, October 2005 - // - Improvements to DirectX pointer chasing. - // - Backdoor RtDsStatistics hook provides DirectX performance information. - // - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30. - // - Auto-call CoInitialize for DSOUND and ASIO platforms. - // Various revisions for RtAudio 4.0 by Gary Scavone, April 2007 +// Modified by Robin Davies, October 2005 +// - Improvements to DirectX pointer chasing. +// - Backdoor RtDsStatistics hook provides DirectX performance information. +// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30. +// - Auto-call CoInitialize for DSOUND and ASIO platforms. +// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007 #include #include @@ -3383,1587 +3406,1599 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, #pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually. #endif - static inline DWORD dsPointerDifference( DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) - { - if ( laterPointer > earlierPointer ) - return laterPointer - earlierPointer; - else - return laterPointer - earlierPointer + bufferSize; - } +static inline DWORD dsPointerDifference( DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) +{ + if ( laterPointer > earlierPointer ) + return laterPointer - earlierPointer; + else + return laterPointer - earlierPointer + bufferSize; +} - static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) - { - if ( pointer > bufferSize ) pointer -= bufferSize; - if ( laterPointer < earlierPointer ) laterPointer += bufferSize; - if ( pointer < earlierPointer ) pointer += bufferSize; - return pointer >= earlierPointer && pointer < laterPointer; - } - - // A structure to hold various information related to the DirectSound - // API implementation. - struct DsHandle { - unsigned int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - void *id[2]; - void *buffer[2]; - bool xrun[2]; - UINT bufferPointer[2]; - DWORD dsBufferSize[2]; - DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by. - HANDLE condition; - - DsHandle() - :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; } - }; +static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) +{ + if ( pointer > bufferSize ) pointer -= bufferSize; + if ( laterPointer < earlierPointer ) laterPointer += bufferSize; + if ( pointer < earlierPointer ) pointer += bufferSize; + return pointer >= earlierPointer && pointer < laterPointer; +} - /* - RtApiDs::RtDsStatistics RtApiDs::statistics; +// A structure to hold various information related to the DirectSound +// API implementation. +struct DsHandle { + unsigned int drainCounter; // Tracks callback counts when draining + bool internalDrain; // Indicates if stop is initiated from callback or not. + void *id[2]; + void *buffer[2]; + bool xrun[2]; + UINT bufferPointer[2]; + DWORD dsBufferSize[2]; + DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by. + HANDLE condition; - // Provides a backdoor hook to monitor for DirectSound read overruns and write underruns. - RtApiDs::RtDsStatistics RtApiDs::getDsStatistics() - { - RtDsStatistics s = statistics; + DsHandle() + :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; } +}; + +/* +RtApiDs::RtDsStatistics RtApiDs::statistics; + +// Provides a backdoor hook to monitor for DirectSound read overruns and write underruns. +RtApiDs::RtDsStatistics RtApiDs::getDsStatistics() +{ + RtDsStatistics s = statistics; - // update the calculated fields. - if ( s.inputFrameSize != 0 ) + // update the calculated fields. + if ( s.inputFrameSize != 0 ) s.latency += s.readDeviceSafeLeadBytes * 1.0 / s.inputFrameSize / s.sampleRate; - if ( s.outputFrameSize != 0 ) + if ( s.outputFrameSize != 0 ) s.latency += (s.writeDeviceSafeLeadBytes + s.writeDeviceBufferLeadBytes) * 1.0 / s.outputFrameSize / s.sampleRate; - return s; - } - */ + return s; +} +*/ - // Declarations for utility functions, callbacks, and structures - // specific to the DirectSound implementation. - static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, - LPCTSTR description, - LPCTSTR module, - LPVOID lpContext ); +// Declarations for utility functions, callbacks, and structures +// specific to the DirectSound implementation. +static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, + LPCTSTR description, + LPCTSTR module, + LPVOID lpContext ); - static char* getErrorString( int code ); +static char* getErrorString( int code ); - extern "C" unsigned __stdcall callbackHandler( void *ptr ); +extern "C" unsigned __stdcall callbackHandler( void *ptr ); - struct EnumInfo { - bool isInput; - bool getDefault; - bool findIndex; - unsigned int counter; - unsigned int index; - LPGUID id; - std::string name; +struct EnumInfo { + bool isInput; + bool getDefault; + bool findIndex; + unsigned int counter; + unsigned int index; + LPGUID id; + std::string name; - EnumInfo() - : isInput(false), getDefault(false), findIndex(false), counter(0), index(0) {} - }; + EnumInfo() + : isInput(false), getDefault(false), findIndex(false), counter(0), index(0) {} +}; - RtApiDs :: RtApiDs() - { - // Dsound will run both-threaded. If CoInitialize fails, then just - // accept whatever the mainline chose for a threading model. - coInitialized_ = false; - HRESULT hr = CoInitialize( NULL ); - if ( !FAILED( hr ) ) coInitialized_ = true; - } +RtApiDs :: RtApiDs() +{ + // Dsound will run both-threaded. If CoInitialize fails, then just + // accept whatever the mainline chose for a threading model. + coInitialized_ = false; + HRESULT hr = CoInitialize( NULL ); + if ( !FAILED( hr ) ) coInitialized_ = true; +} - RtApiDs :: ~RtApiDs() - { - if ( coInitialized_ ) CoUninitialize(); // balanced call. - if ( stream_.state != STREAM_CLOSED ) closeStream(); +RtApiDs :: ~RtApiDs() +{ + if ( coInitialized_ ) CoUninitialize(); // balanced call. + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} + +unsigned int RtApiDs :: getDefaultInputDevice( void ) +{ + // Count output devices. + EnumInfo info; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") counting output devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return 0; } - unsigned int RtApiDs :: getDefaultInputDevice( void ) - { - // Count output devices. - EnumInfo info; - HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") counting output devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return 0; - } + // Now enumerate input devices until we find the id = NULL. + info.isInput = true; + info.getDefault = true; + result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDefaultInputDevice: error (" << getErrorString( result ) << ") enumerating input devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return 0; + } - // Now enumerate input devices until we find the id = NULL. - info.isInput = true; - info.getDefault = true; - result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDefaultInputDevice: error (" << getErrorString( result ) << ") enumerating input devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return 0; - } + if ( info.counter > 0 ) return info.counter - 1; + return 0; +} - if ( info.counter > 0 ) return info.counter - 1; +unsigned int RtApiDs :: getDefaultOutputDevice( void ) +{ + // Enumerate output devices until we find the id = NULL. + EnumInfo info; + info.getDefault = true; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") enumerating output devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); return 0; } - unsigned int RtApiDs :: getDefaultOutputDevice( void ) - { - // Enumerate output devices until we find the id = NULL. - EnumInfo info; - info.getDefault = true; - HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") enumerating output devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return 0; - } + if ( info.counter > 0 ) return info.counter - 1; + return 0; +} - if ( info.counter > 0 ) return info.counter - 1; - return 0; +unsigned int RtApiDs :: getDeviceCount( void ) +{ + // Count DirectSound devices. + EnumInfo info; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); } - unsigned int RtApiDs :: getDeviceCount( void ) - { - // Count DirectSound devices. - EnumInfo info; - HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } + // Count DirectSoundCapture devices. + info.isInput = true; + result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + } - // Count DirectSoundCapture devices. - info.isInput = true; - result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } + return info.counter; +} + +RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device ) +{ + // Because DirectSound always enumerates input and output devices + // separately (and because we don't attempt to combine devices + // internally), none of our "devices" will ever be duplex. + + RtAudio::DeviceInfo info; + info.probed = false; - return info.counter; + // Enumerate through devices to find the id (if it exists). Note + // that we have to do the output enumeration first, even if this is + // an input device, in order for the device counter to be correct. + EnumInfo dsinfo; + dsinfo.findIndex = true; + dsinfo.index = device; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating output devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); } - RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device ) - { - // Because DirectSound always enumerates input and output devices - // separately (and because we don't attempt to combine devices - // internally), none of our "devices" will ever be duplex. - - RtAudio::DeviceInfo info; - info.probed = false; - - // Enumerate through devices to find the id (if it exists). Note - // that we have to do the output enumeration first, even if this is - // an input device, in order for the device counter to be correct. - EnumInfo dsinfo; - dsinfo.findIndex = true; - dsinfo.index = device; - HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating output devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } + if ( dsinfo.name.empty() ) goto probeInput; - if ( dsinfo.name.empty() ) goto probeInput; + LPDIRECTSOUND output; + DSCAPS outCaps; + result = DirectSoundCreate( dsinfo.id, &output, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } - LPDIRECTSOUND output; - DSCAPS outCaps; - result = DirectSoundCreate( dsinfo.id, &output, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } + outCaps.dwSize = sizeof( outCaps ); + result = output->GetCaps( &outCaps ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } - outCaps.dwSize = sizeof( outCaps ); - result = output->GetCaps( &outCaps ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } + // Get output channel information. + info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; - // Get output channel information. - info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; + // Get sample rate information. + info.sampleRates.clear(); + for ( unsigned int k=0; k= (unsigned int) outCaps.dwMinSecondarySampleRate && + SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) + info.sampleRates.push_back( SAMPLE_RATES[k] ); + } - // Get sample rate information. - info.sampleRates.clear(); - for ( unsigned int k=0; k= (unsigned int) outCaps.dwMinSecondarySampleRate && - SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) - info.sampleRates.push_back( SAMPLE_RATES[k] ); - } + // Get format information. + if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16; + if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8; - // Get format information. - if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16; - if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8; + output->Release(); - output->Release(); + if ( getDefaultOutputDevice() == device ) + info.isDefaultOutput = true; - if ( getDefaultOutputDevice() == device ) - info.isDefaultOutput = true; + // Copy name and return. + info.name = dsinfo.name; - // Copy name and return. - info.name = dsinfo.name; + info.probed = true; + return info; - info.probed = true; + probeInput: + + dsinfo.isInput = true; + result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating input devices!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + } + + if ( dsinfo.name.empty() ) return info; + + LPDIRECTSOUNDCAPTURE input; + result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); return info; + } + + DSCCAPS inCaps; + inCaps.dwSize = sizeof( inCaps ); + result = input->GetCaps( &inCaps ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // Get input channel information. + info.inputChannels = inCaps.dwChannels; + + // Get sample rate and format information. + if ( inCaps.dwChannels == 2 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8; + + if ( info.nativeFormats & RTAUDIO_SINT16 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.sampleRates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.sampleRates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.sampleRates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.sampleRates.push_back( 96000 ); + } + else if ( info.nativeFormats & RTAUDIO_SINT8 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.sampleRates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.sampleRates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.sampleRates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.sampleRates.push_back( 44100 ); + } + } + else if ( inCaps.dwChannels == 1 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16; + if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8; + if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8; + + if ( info.nativeFormats & RTAUDIO_SINT16 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.sampleRates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.sampleRates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.sampleRates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.sampleRates.push_back( 96000 ); + } + else if ( info.nativeFormats & RTAUDIO_SINT8 ) { + if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.sampleRates.push_back( 11025 ); + if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.sampleRates.push_back( 22050 ); + if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.sampleRates.push_back( 44100 ); + if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.sampleRates.push_back( 96000 ); + } + } + else info.inputChannels = 0; // technically, this would be an error + + input->Release(); + + if ( info.inputChannels == 0 ) return info; + + if ( getDefaultInputDevice() == device ) + info.isDefaultInput = true; + + // Copy name and return. + info.name = dsinfo.name; + info.probed = true; + return info; +} - probeInput: +bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + if ( channels + firstChannel > 2 ) { + errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device."; + return FAILURE; + } + + // Enumerate through devices to find the id (if it exists). Note + // that we have to do the output enumeration first, even if this is + // an input device, in order for the device counter to be correct. + EnumInfo dsinfo; + dsinfo.findIndex = true; + dsinfo.index = device; + HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating output devices!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + if ( mode == OUTPUT ) { + if ( dsinfo.name.empty() ) { + errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!"; + errorText_ = errorStream_.str(); + return FAILURE; + } + } + else { // mode == INPUT dsinfo.isInput = true; - result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); + HRESULT result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating input devices!"; + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating input devices!"; errorText_ = errorStream_.str(); - error( RtError::WARNING ); + return FAILURE; + } + if ( dsinfo.name.empty() ) { + errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!"; + errorText_ = errorStream_.str(); + return FAILURE; } + } - if ( dsinfo.name.empty() ) return info; + // According to a note in PortAudio, using GetDesktopWindow() + // instead of GetForegroundWindow() is supposed to avoid problems + // that occur when the application's window is not the foreground + // window. Also, if the application window closes before the + // DirectSound buffer, DirectSound can crash. However, for console + // applications, no sound was produced when using GetDesktopWindow(). + HWND hWnd = GetForegroundWindow(); + + // Check the numberOfBuffers parameter and limit the lowest value to + // two. This is a judgement call and a value of two is probably too + // low for capture, but it should work for playback. + int nBuffers = 0; + if ( options ) nBuffers = options->numberOfBuffers; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2; + if ( nBuffers < 2 ) nBuffers = 3; + + // Create the wave format structure. The data format setting will + // be determined later. + WAVEFORMATEX waveFormat; + ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) ); + waveFormat.wFormatTag = WAVE_FORMAT_PCM; + waveFormat.nChannels = channels + firstChannel; + waveFormat.nSamplesPerSec = (unsigned long) sampleRate; + + // Determine the device buffer size. By default, 32k, but we will + // grow it to make allowances for very large software buffer sizes. + DWORD dsBufferSize = 0; + DWORD dsPointerLeadTime = 0; + long bufferBytes = MINIMUM_DEVICE_BUFFER_SIZE; // sound cards will always *knock wood* support this + + void *ohandle = 0, *bhandle = 0; + if ( mode == OUTPUT ) { - LPDIRECTSOUNDCAPTURE input; - result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL ); + LPDIRECTSOUND output; + result = DirectSoundCreate( dsinfo.id, &output, NULL ); if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!"; + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!"; errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; + return FAILURE; } - DSCCAPS inCaps; - inCaps.dwSize = sizeof( inCaps ); - result = input->GetCaps( &inCaps ); + DSCAPS outCaps; + outCaps.dwSize = sizeof( outCaps ); + result = output->GetCaps( &outCaps ); if ( FAILED( result ) ) { - input->Release(); - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsinfo.name << ")!"; + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsinfo.name << ")!"; errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; + return FAILURE; } - // Get input channel information. - info.inputChannels = inCaps.dwChannels; - - // Get sample rate and format information. - if ( inCaps.dwChannels == 2 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8; - - if ( info.nativeFormats & RTAUDIO_SINT16 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.sampleRates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.sampleRates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.sampleRates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.sampleRates.push_back( 96000 ); - } - else if ( info.nativeFormats & RTAUDIO_SINT8 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.sampleRates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.sampleRates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.sampleRates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.sampleRates.push_back( 44100 ); - } + // Check channel information. + if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) { + errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsinfo.name << ") does not support stereo playback."; + errorText_ = errorStream_.str(); + return FAILURE; } - else if ( inCaps.dwChannels == 1 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8; - - if ( info.nativeFormats & RTAUDIO_SINT16 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.sampleRates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.sampleRates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.sampleRates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.sampleRates.push_back( 96000 ); - } - else if ( info.nativeFormats & RTAUDIO_SINT8 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.sampleRates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.sampleRates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.sampleRates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.sampleRates.push_back( 96000 ); - } + + // Check format information. Use 16-bit format unless not + // supported or user requests 8-bit. + if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT && + !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) { + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + else { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; } - else info.inputChannels = 0; // technically, this would be an error + stream_.userFormat = format; - input->Release(); + // Update wave format structure and buffer information. + waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; + waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; - if ( info.inputChannels == 0 ) return info; + // If the user wants an even bigger buffer, increase the device buffer size accordingly. + while ( dsPointerLeadTime * 2U > (DWORD) bufferBytes ) + bufferBytes *= 2; - if ( getDefaultInputDevice() == device ) - info.isDefaultInput = true; + // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes. + //result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE ); + // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes. + result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Copy name and return. - info.name = dsinfo.name; - info.probed = true; - return info; - } + // Even though we will write to the secondary buffer, we need to + // access the primary buffer to set the correct output format + // (since the default is 8-bit, 22 kHz!). Setup the DS primary + // buffer description. + DSBUFFERDESC bufferDescription; + ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSBUFFERDESC ); + bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER; - bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) - { - if ( channels + firstChannel > 2 ) { - errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device."; + // Obtain the primary buffer + LPDIRECTSOUNDBUFFER buffer; + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); return FAILURE; } - // Enumerate through devices to find the id (if it exists). Note - // that we have to do the output enumeration first, even if this is - // an input device, in order for the device counter to be correct. - EnumInfo dsinfo; - dsinfo.findIndex = true; - dsinfo.index = device; - HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); + // Set the primary DS buffer sound format. + result = buffer->SetFormat( &waveFormat ); if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating output devices!"; + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsinfo.name << ")!"; errorText_ = errorStream_.str(); return FAILURE; } - if ( mode == OUTPUT ) { - if ( dsinfo.name.empty() ) { - errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - else { // mode == INPUT - dsinfo.isInput = true; - HRESULT result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); + // Setup the secondary DS buffer description. + dsBufferSize = (DWORD) bufferBytes; + ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSBUFFERDESC ); + bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | + DSBCAPS_GLOBALFOCUS | + DSBCAPS_GETCURRENTPOSITION2 | + DSBCAPS_LOCHARDWARE ); // Force hardware mixing + bufferDescription.dwBufferBytes = bufferBytes; + bufferDescription.lpwfxFormat = &waveFormat; + + // Try to create the secondary DS buffer. If that doesn't work, + // try to use software mixing. Otherwise, there's a problem. + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | + DSBCAPS_GLOBALFOCUS | + DSBCAPS_GETCURRENTPOSITION2 | + DSBCAPS_LOCSOFTWARE ); // Force software mixing + result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating input devices!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - if ( dsinfo.name.empty() ) { - errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!"; + output->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsinfo.name << ")!"; errorText_ = errorStream_.str(); return FAILURE; } } - // According to a note in PortAudio, using GetDesktopWindow() - // instead of GetForegroundWindow() is supposed to avoid problems - // that occur when the application's window is not the foreground - // window. Also, if the application window closes before the - // DirectSound buffer, DirectSound can crash. However, for console - // applications, no sound was produced when using GetDesktopWindow(). - HWND hWnd = GetForegroundWindow(); - - // Check the numberOfBuffers parameter and limit the lowest value to - // two. This is a judgement call and a value of two is probably too - // low for capture, but it should work for playback. - int nBuffers = 0; - if ( options ) nBuffers = options->numberOfBuffers; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2; - if ( nBuffers < 2 ) nBuffers = 3; - - // Create the wave format structure. The data format setting will - // be determined later. - WAVEFORMATEX waveFormat; - ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) ); - waveFormat.wFormatTag = WAVE_FORMAT_PCM; - waveFormat.nChannels = channels + firstChannel; - waveFormat.nSamplesPerSec = (unsigned long) sampleRate; - - // Determine the device buffer size. By default, 32k, but we will - // grow it to make allowances for very large software buffer sizes. - DWORD dsBufferSize = 0; - DWORD dsPointerLeadTime = 0; - long bufferBytes = MINIMUM_DEVICE_BUFFER_SIZE; // sound cards will always *knock wood* support this - - void *ohandle = 0, *bhandle = 0; - if ( mode == OUTPUT ) { - - LPDIRECTSOUND output; - result = DirectSoundCreate( dsinfo.id, &output, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - DSCAPS outCaps; - outCaps.dwSize = sizeof( outCaps ); - result = output->GetCaps( &outCaps ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Check channel information. - if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsinfo.name << ") does not support stereo playback."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Check format information. Use 16-bit format unless not - // supported or user requests 8-bit. - if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT && - !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) { - waveFormat.wBitsPerSample = 16; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - else { - waveFormat.wBitsPerSample = 8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - stream_.userFormat = format; - - // Update wave format structure and buffer information. - waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; - waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; - dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; - - // If the user wants an even bigger buffer, increase the device buffer size accordingly. - while ( dsPointerLeadTime * 2U > (DWORD) bufferBytes ) - bufferBytes *= 2; - - // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes. - //result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE ); - // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes. - result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Even though we will write to the secondary buffer, we need to - // access the primary buffer to set the correct output format - // (since the default is 8-bit, 22 kHz!). Setup the DS primary - // buffer description. - DSBUFFERDESC bufferDescription; - ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); - bufferDescription.dwSize = sizeof( DSBUFFERDESC ); - bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER; - - // Obtain the primary buffer - LPDIRECTSOUNDBUFFER buffer; - result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the primary DS buffer sound format. - result = buffer->SetFormat( &waveFormat ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Setup the secondary DS buffer description. - dsBufferSize = (DWORD) bufferBytes; - ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); - bufferDescription.dwSize = sizeof( DSBUFFERDESC ); - bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | - DSBCAPS_GLOBALFOCUS | - DSBCAPS_GETCURRENTPOSITION2 | - DSBCAPS_LOCHARDWARE ); // Force hardware mixing - bufferDescription.dwBufferBytes = bufferBytes; - bufferDescription.lpwfxFormat = &waveFormat; + // Get the buffer size ... might be different from what we specified. + DSBCAPS dsbcaps; + dsbcaps.dwSize = sizeof( DSBCAPS ); + result = buffer->GetCaps( &dsbcaps ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Try to create the secondary DS buffer. If that doesn't work, - // try to use software mixing. Otherwise, there's a problem. - result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | - DSBCAPS_GLOBALFOCUS | - DSBCAPS_GETCURRENTPOSITION2 | - DSBCAPS_LOCSOFTWARE ); // Force software mixing - result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - } + bufferBytes = dsbcaps.dwBufferBytes; - // Get the buffer size ... might be different from what we specified. - DSBCAPS dsbcaps; - dsbcaps.dwSize = sizeof( DSBCAPS ); - result = buffer->GetCaps( &dsbcaps ); - if ( FAILED( result ) ) { - output->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Lock the DS buffer + LPVOID audioPtr; + DWORD dataLen; + result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - bufferBytes = dsbcaps.dwBufferBytes; + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); - // Lock the DS buffer - LPVOID audioPtr; - DWORD dataLen; - result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - output->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + output->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Zero the DS buffer - ZeroMemory( audioPtr, dataLen ); + dsBufferSize = bufferBytes; + ohandle = (void *) output; + bhandle = (void *) buffer; + } - // Unlock the DS buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - output->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + if ( mode == INPUT ) { - dsBufferSize = bufferBytes; - ohandle = (void *) output; - bhandle = (void *) buffer; + LPDIRECTSOUNDCAPTURE input; + result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; } - if ( mode == INPUT ) { - - LPDIRECTSOUNDCAPTURE input; - result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + DSCCAPS inCaps; + inCaps.dwSize = sizeof( inCaps ); + result = input->GetCaps( &inCaps ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - DSCCAPS inCaps; - inCaps.dwSize = sizeof( inCaps ); - result = input->GetCaps( &inCaps ); - if ( FAILED( result ) ) { - input->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Check channel information. + if ( inCaps.dwChannels < channels + firstChannel ) { + errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels."; + return FAILURE; + } - // Check channel information. - if ( inCaps.dwChannels < channels + firstChannel ) { - errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels."; - return FAILURE; + // Check format information. Use 16-bit format unless user + // requests 8-bit. + DWORD deviceFormats; + if ( channels + firstChannel == 2 ) { + deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08; + if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; } - - // Check format information. Use 16-bit format unless user - // requests 8-bit. - DWORD deviceFormats; - if ( channels + firstChannel == 2 ) { - deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08; - if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { - waveFormat.wBitsPerSample = 8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - else { // assume 16-bit is supported - waveFormat.wBitsPerSample = 16; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } + else { // assume 16-bit is supported + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; } - else { // channel == 1 - deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08; - if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { - waveFormat.wBitsPerSample = 8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - else { // assume 16-bit is supported - waveFormat.wBitsPerSample = 16; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } + } + else { // channel == 1 + deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08; + if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { + waveFormat.wBitsPerSample = 8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; } - stream_.userFormat = format; - - // Update wave format structure and buffer information. - waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; - waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; - - // Setup the secondary DS buffer description. - dsBufferSize = bufferBytes; - DSCBUFFERDESC bufferDescription; - ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) ); - bufferDescription.dwSize = sizeof( DSCBUFFERDESC ); - bufferDescription.dwFlags = 0; - bufferDescription.dwReserved = 0; - bufferDescription.dwBufferBytes = bufferBytes; - bufferDescription.lpwfxFormat = &waveFormat; - - // Create the capture buffer. - LPDIRECTSOUNDCAPTUREBUFFER buffer; - result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - input->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; + else { // assume 16-bit is supported + waveFormat.wBitsPerSample = 16; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; } + } + stream_.userFormat = format; - // Lock the capture buffer - LPVOID audioPtr; - DWORD dataLen; - result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - input->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Update wave format structure and buffer information. + waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; + waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; + + // Setup the secondary DS buffer description. + dsBufferSize = bufferBytes; + DSCBUFFERDESC bufferDescription; + ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) ); + bufferDescription.dwSize = sizeof( DSCBUFFERDESC ); + bufferDescription.dwFlags = 0; + bufferDescription.dwReserved = 0; + bufferDescription.dwBufferBytes = bufferBytes; + bufferDescription.lpwfxFormat = &waveFormat; + + // Create the capture buffer. + LPDIRECTSOUNDCAPTUREBUFFER buffer; + result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL ); + if ( FAILED( result ) ) { + input->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Zero the buffer - ZeroMemory( audioPtr, dataLen ); + // Lock the capture buffer + LPVOID audioPtr; + DWORD dataLen; + result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Unlock the buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - input->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Zero the buffer + ZeroMemory( audioPtr, dataLen ); - dsBufferSize = bufferBytes; - ohandle = (void *) input; - bhandle = (void *) buffer; + // Unlock the buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + input->Release(); + buffer->Release(); + errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsinfo.name << ")!"; + errorText_ = errorStream_.str(); + return FAILURE; } - // Set various stream parameters - DsHandle *handle = 0; - stream_.nDeviceChannels[mode] = channels + firstChannel; - stream_.nUserChannels[mode] = channels; - stream_.bufferSize = *bufferSize; - stream_.channelOffset[mode] = firstChannel; - stream_.deviceInterleaved[mode] = true; - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; + dsBufferSize = bufferBytes; + ohandle = (void *) input; + bhandle = (void *) buffer; + } - // Set flag for buffer conversion - stream_.doConvertBuffer[mode] = false; - if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode]) - stream_.doConvertBuffer[mode] = true; - if (stream_.userFormat != stream_.deviceFormat[mode]) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; + // Set various stream parameters + DsHandle *handle = 0; + stream_.nDeviceChannels[mode] = channels + firstChannel; + stream_.nUserChannels[mode] = channels; + stream_.bufferSize = *bufferSize; + stream_.channelOffset[mode] = firstChannel; + stream_.deviceInterleaved[mode] = true; + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; + else stream_.userInterleaved = true; - // Allocate necessary internal buffers - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } + // Set flag for buffer conversion + stream_.doConvertBuffer[mode] = false; + if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode]) + stream_.doConvertBuffer[mode] = true; + if (stream_.userFormat != stream_.deviceFormat[mode]) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; - if ( stream_.doConvertBuffer[mode] ) { + // Allocate necessary internal buffers + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= (long) bytesOut ) makeBuffer = false; - } - } + if ( stream_.doConvertBuffer[mode] ) { - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= (long) bytesOut ) makeBuffer = false; } } - // Allocate our DsHandle structures for the stream. - if ( stream_.apiHandle == 0 ) { - try { - handle = new DsHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory."; + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory."; goto error; } + } + } - // Create a manual-reset event. - handle->condition = CreateEvent( NULL, // no security - TRUE, // manual-reset - FALSE, // non-signaled initially - NULL ); // unnamed - stream_.apiHandle = (void *) handle; + // Allocate our DsHandle structures for the stream. + if ( stream_.apiHandle == 0 ) { + try { + handle = new DsHandle; + } + catch ( std::bad_alloc& ) { + errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory."; + goto error; } - else - handle = (DsHandle *) stream_.apiHandle; - handle->id[mode] = ohandle; - handle->buffer[mode] = bhandle; - handle->dsBufferSize[mode] = dsBufferSize; - handle->dsPointerLeadTime[mode] = dsPointerLeadTime; - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - if ( stream_.mode == OUTPUT && mode == INPUT ) - // We had already set up an output stream. - stream_.mode = DUPLEX; - else - stream_.mode = mode; - stream_.nBuffers = nBuffers; - stream_.sampleRate = sampleRate; + // Create a manual-reset event. + handle->condition = CreateEvent( NULL, // no security + TRUE, // manual-reset + FALSE, // non-signaled initially + NULL ); // unnamed + stream_.apiHandle = (void *) handle; + } + else + handle = (DsHandle *) stream_.apiHandle; + handle->id[mode] = ohandle; + handle->buffer[mode] = bhandle; + handle->dsBufferSize[mode] = dsBufferSize; + handle->dsPointerLeadTime[mode] = dsPointerLeadTime; + + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; + if ( stream_.mode == OUTPUT && mode == INPUT ) + // We had already set up an output stream. + stream_.mode = DUPLEX; + else + stream_.mode = mode; + stream_.nBuffers = nBuffers; + stream_.sampleRate = sampleRate; - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - // Setup the callback thread. - unsigned threadId; - stream_.callbackInfo.object = (void *) this; - stream_.callbackInfo.isRunning = true; - stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler, - &stream_.callbackInfo, 0, &threadId ); - if ( stream_.callbackInfo.thread == 0 ) { - errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!"; - goto error; - } + // Setup the callback thread. + unsigned threadId; + stream_.callbackInfo.object = (void *) this; + stream_.callbackInfo.isRunning = true; + stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler, + &stream_.callbackInfo, 0, &threadId ); + if ( stream_.callbackInfo.thread == 0 ) { + errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!"; + goto error; + } - // Boost DS thread priority - SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST ); - return SUCCESS; + // Boost DS thread priority + SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST ); + return SUCCESS; - error: - if ( handle ) { - if ( handle->buffer[0] ) { // the object pointer can be NULL and valid - LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - if ( buffer ) buffer->Release(); - object->Release(); - } - if ( handle->buffer[1] ) { - LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - if ( buffer ) buffer->Release(); - object->Release(); - } - CloseHandle( handle->condition ); - delete handle; - stream_.apiHandle = 0; + error: + if ( handle ) { + if ( handle->buffer[0] ) { // the object pointer can be NULL and valid + LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + if ( buffer ) buffer->Release(); + object->Release(); } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } + if ( handle->buffer[1] ) { + LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + if ( buffer ) buffer->Release(); + object->Release(); } + CloseHandle( handle->condition ); + delete handle; + stream_.apiHandle = 0; + } - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; } + } - return FAILURE; + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; } - void RtApiDs :: closeStream() - { - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiDs::closeStream(): no open stream to close!"; - error( RtError::WARNING ); - return; - } + return FAILURE; +} - // Stop the callback thread. - stream_.callbackInfo.isRunning = false; - WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE ); - CloseHandle( (HANDLE) stream_.callbackInfo.thread ); +void RtApiDs :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiDs::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } - DsHandle *handle = (DsHandle *) stream_.apiHandle; - if ( handle ) { - if ( handle->buffer[0] ) { // the object pointer can be NULL and valid - LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - if ( buffer ) { - buffer->Stop(); - buffer->Release(); - } - object->Release(); - } - if ( handle->buffer[1] ) { - LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - if ( buffer ) { - buffer->Stop(); - buffer->Release(); - } - object->Release(); + // Stop the callback thread. + stream_.callbackInfo.isRunning = false; + WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE ); + CloseHandle( (HANDLE) stream_.callbackInfo.thread ); + + DsHandle *handle = (DsHandle *) stream_.apiHandle; + if ( handle ) { + if ( handle->buffer[0] ) { // the object pointer can be NULL and valid + LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + if ( buffer ) { + buffer->Stop(); + buffer->Release(); } - CloseHandle( handle->condition ); - delete handle; - stream_.apiHandle = 0; + object->Release(); } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; + if ( handle->buffer[1] ) { + LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + if ( buffer ) { + buffer->Stop(); + buffer->Release(); } + object->Release(); } + CloseHandle( handle->condition ); + delete handle; + stream_.apiHandle = 0; + } - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; } + } - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; } - void RtApiDs :: startStream() - { - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiDs::startStream(): the stream is already running!"; - error( RtError::WARNING ); - return; - } + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} + +void RtApiDs :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiDs::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } - // Increase scheduler frequency on lesser windows (a side-effect of - // increasing timer accuracy). On greater windows (Win2K or later), - // this is already in effect. + // Increase scheduler frequency on lesser windows (a side-effect of + // increasing timer accuracy). On greater windows (Win2K or later), + // this is already in effect. - MUTEX_LOCK( &stream_.mutex ); + MUTEX_LOCK( &stream_.mutex ); - DsHandle *handle = (DsHandle *) stream_.apiHandle; + DsHandle *handle = (DsHandle *) stream_.apiHandle; - timeBeginPeriod( 1 ); + timeBeginPeriod( 1 ); - /* - memset( &statistics, 0, sizeof( statistics ) ); - statistics.sampleRate = stream_.sampleRate; - statistics.writeDeviceBufferLeadBytes = handle->dsPointerLeadTime[0]; - */ + /* + memset( &statistics, 0, sizeof( statistics ) ); + statistics.sampleRate = stream_.sampleRate; + statistics.writeDeviceBufferLeadBytes = handle->dsPointerLeadTime[0]; + */ - buffersRolling = false; - duplexPrerollBytes = 0; + buffersRolling = false; + duplexPrerollBytes = 0; - if ( stream_.mode == DUPLEX ) { - // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize. - duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] ); - } + if ( stream_.mode == DUPLEX ) { + // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize. + duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] ); + } - HRESULT result = 0; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - //statistics.outputFrameSize = formatBytes( stream_.deviceFormat[0] ) * stream_.nDeviceChannels[0]; + HRESULT result = 0; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + //statistics.outputFrameSize = formatBytes( stream_.deviceFormat[0] ) * stream_.nDeviceChannels[0]; - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - result = buffer->Play( 0, 0, DSBPLAY_LOOPING ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + result = buffer->Play( 0, 0, DSBPLAY_LOOPING ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; } + } - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - //statistics.inputFrameSize = formatBytes( stream_.deviceFormat[1]) * stream_.nDeviceChannels[1]; + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + //statistics.inputFrameSize = formatBytes( stream_.deviceFormat[1]) * stream_.nDeviceChannels[1]; - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - result = buffer->Start( DSCBSTART_LOOPING ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + result = buffer->Start( DSCBSTART_LOOPING ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; } + } + + handle->drainCounter = 0; + handle->internalDrain = false; + stream_.state = STREAM_RUNNING; + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); +} + +void RtApiDs :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } - handle->drainCounter = 0; - handle->internalDrain = false; - stream_.state = STREAM_RUNNING; + MUTEX_LOCK( &stream_.mutex ); - unlock: + if ( stream_.state == STREAM_STOPPED ) { MUTEX_UNLOCK( &stream_.mutex ); - - if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); + return; } - void RtApiDs :: stopStream() - { - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; + HRESULT result = 0; + LPVOID audioPtr; + DWORD dataLen; + DsHandle *handle = (DsHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( handle->drainCounter == 0 ) { + handle->drainCounter = 1; + MUTEX_UNLOCK( &stream_.mutex ); + WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled + ResetEvent( handle->condition ); + MUTEX_LOCK( &stream_.mutex ); } - MUTEX_LOCK( &stream_.mutex ); + // Stop the buffer and clear memory + LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + result = buffer->Stop(); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } - HRESULT result = 0; - LPVOID audioPtr; - DWORD dataLen; - DsHandle *handle = (DsHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 1; - MUTEX_UNLOCK( &stream_.mutex ); - WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled - ResetEvent( handle->condition ); - MUTEX_LOCK( &stream_.mutex ); - } + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } - // Stop the buffer and clear memory - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - result = buffer->Stop(); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); - // Lock the buffer and clear it so that if we start to play again, - // we won't have old data playing. - result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking output buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } - // Zero the DS buffer - ZeroMemory( audioPtr, dataLen ); + // If we start playing again, we must begin at beginning of buffer. + handle->bufferPointer[0] = 0; + } - // Unlock the DS buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + audioPtr = NULL; + dataLen = 0; - // If we start playing again, we must begin at beginning of buffer. - handle->bufferPointer[0] = 0; + result = buffer->Stop(); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; } - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - audioPtr = NULL; - dataLen = 0; + // Lock the buffer and clear it so that if we start to play again, + // we won't have old data playing. + result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } - result = buffer->Stop(); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } + // Zero the DS buffer + ZeroMemory( audioPtr, dataLen ); - // Lock the buffer and clear it so that if we start to play again, - // we won't have old data playing. - result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } + // Unlock the DS buffer + result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking input buffer!"; + errorText_ = errorStream_.str(); + goto unlock; + } - // Zero the DS buffer - ZeroMemory( audioPtr, dataLen ); + // If we start recording again, we must begin at beginning of buffer. + handle->bufferPointer[1] = 0; + } - // Unlock the DS buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } + unlock: + timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); - // If we start recording again, we must begin at beginning of buffer. - handle->bufferPointer[1] = 0; - } + if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); +} - unlock: - timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); - if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); +void RtApiDs :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiDs::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; } - void RtApiDs :: abortStream() - { - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiDs::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } + DsHandle *handle = (DsHandle *) stream_.apiHandle; + handle->drainCounter = 1; - DsHandle *handle = (DsHandle *) stream_.apiHandle; - handle->drainCounter = 1; + stopStream(); +} - stopStream(); +void RtApiDs :: callbackEvent() +{ + if ( stream_.state == STREAM_STOPPED ) { + Sleep(50); // sleep 50 milliseconds + return; } - void RtApiDs :: callbackEvent() - { - if ( stream_.state == STREAM_STOPPED ) { - Sleep(50); // sleep 50 milliseconds - return; - } + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return; + } - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); - return; - } + CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; + DsHandle *handle = (DsHandle *) stream_.apiHandle; - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - DsHandle *handle = (DsHandle *) stream_.apiHandle; + // Check if we were draining the stream and signal is finished. + if ( handle->drainCounter > stream_.nBuffers + 2 ) { + if ( handle->internalDrain == false ) + SetEvent( handle->condition ); + else + stopStream(); + return; + } - // Check if we were draining the stream and signal is finished. - if ( handle->drainCounter > stream_.nBuffers + 2 ) { - if ( handle->internalDrain == false ) - SetEvent( handle->condition ); - else - stopStream(); - return; - } + MUTEX_LOCK( &stream_.mutex ); - MUTEX_LOCK( &stream_.mutex ); + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return SUCCESS; + } - // Invoke user callback to get fresh output data UNLESS we are - // draining stream. - if ( handle->drainCounter == 0 ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; - } - handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( handle->drainCounter == 2 ) { - MUTEX_UNLOCK( &stream_.mutex ); - abortStream(); - return; - } - else if ( handle->drainCounter == 1 ) - handle->internalDrain = true; + // Invoke user callback to get fresh output data UNLESS we are + // draining stream. + if ( handle->drainCounter == 0 ) { + RtAudioCallback callback = (RtAudioCallback) info->callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, info->userData ); + if ( handle->drainCounter == 2 ) { + MUTEX_UNLOCK( &stream_.mutex ); + abortStream(); + return; } + else if ( handle->drainCounter == 1 ) + handle->internalDrain = true; + } - HRESULT result; - DWORD currentWritePos, safeWritePos; - DWORD currentReadPos, safeReadPos; - DWORD leadPos; - UINT nextWritePos; + HRESULT result; + DWORD currentWritePos, safeWritePos; + DWORD currentReadPos, safeReadPos; + DWORD leadPos; + UINT nextWritePos; #ifdef GENERATE_DEBUG_LOG - DWORD writeTime, readTime; + DWORD writeTime, readTime; #endif - LPVOID buffer1 = NULL; - LPVOID buffer2 = NULL; - DWORD bufferSize1 = 0; - DWORD bufferSize2 = 0; + LPVOID buffer1 = NULL; + LPVOID buffer2 = NULL; + DWORD bufferSize1 = 0; + DWORD bufferSize2 = 0; - char *buffer; - long bufferBytes; + char *buffer; + long bufferBytes; - if ( stream_.mode == DUPLEX && !buffersRolling ) { - assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); + if ( stream_.mode == DUPLEX && !buffersRolling ) { + assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); - // It takes a while for the devices to get rolling. As a result, - // there's no guarantee that the capture and write device pointers - // will move in lockstep. Wait here for both devices to start - // rolling, and then set our buffer pointers accordingly. - // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600 - // bytes later than the write buffer. + // It takes a while for the devices to get rolling. As a result, + // there's no guarantee that the capture and write device pointers + // will move in lockstep. Wait here for both devices to start + // rolling, and then set our buffer pointers accordingly. + // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600 + // bytes later than the write buffer. - // Stub: a serious risk of having a pre-emptive scheduling round - // take place between the two GetCurrentPosition calls... but I'm - // really not sure how to solve the problem. Temporarily boost to - // Realtime priority, maybe; but I'm not sure what priority the - // DirectSound service threads run at. We *should* be roughly - // within a ms or so of correct. + // Stub: a serious risk of having a pre-emptive scheduling round + // take place between the two GetCurrentPosition calls... but I'm + // really not sure how to solve the problem. Temporarily boost to + // Realtime priority, maybe; but I'm not sure what priority the + // DirectSound service threads run at. We *should* be roughly + // within a ms or so of correct. - LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; + LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - DWORD initialWritePos, initialSafeWritePos; - DWORD initialReadPos, initialSafeReadPos; + DWORD initialWritePos, initialSafeWritePos; + DWORD initialReadPos, initialSafeReadPos; - result = dsWriteBuffer->GetCurrentPosition( &initialWritePos, &initialSafeWritePos ); + result = dsWriteBuffer->GetCurrentPosition( &initialWritePos, &initialSafeWritePos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + result = dsCaptureBuffer->GetCurrentPosition( &initialReadPos, &initialSafeReadPos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + while ( true ) { + result = dsWriteBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; errorText_ = errorStream_.str(); error( RtError::SYSTEM_ERROR ); } - result = dsCaptureBuffer->GetCurrentPosition( &initialReadPos, &initialSafeReadPos ); + result = dsCaptureBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); if ( FAILED( result ) ) { errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; errorText_ = errorStream_.str(); error( RtError::SYSTEM_ERROR ); } - while ( true ) { - result = dsWriteBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - result = dsCaptureBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - if ( safeWritePos != initialSafeWritePos && safeReadPos != initialSafeReadPos ) break; - Sleep( 1 ); - } - - assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); - - buffersRolling = true; - handle->bufferPointer[0] = ( safeWritePos + handle->dsPointerLeadTime[0] ); - handle->bufferPointer[1] = safeReadPos; + if ( safeWritePos != initialSafeWritePos && safeReadPos != initialSafeReadPos ) break; + Sleep( 1 ); } - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - - if ( handle->drainCounter > 1 ) { // write zeros to the output stream - bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; - bufferBytes *= formatBytes( stream_.userFormat ); - memset( stream_.userBuffer[0], 0, bufferBytes ); - } + assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0]; - bufferBytes *= formatBytes( stream_.deviceFormat[0] ); - } - else { - buffer = stream_.userBuffer[0]; - bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; - bufferBytes *= formatBytes( stream_.userFormat ); - } + buffersRolling = true; + handle->bufferPointer[0] = ( safeWritePos + handle->dsPointerLeadTime[0] ); + handle->bufferPointer[1] = safeReadPos; + } - // No byte swapping necessary in DirectSound implementation. + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + + LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - // Ahhh ... windoze. 16-bit data is signed but 8-bit data is - // unsigned. So, we need to convert our signed 8-bit data here to - // unsigned. - if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 ) - for ( int i=0; idrainCounter > 1 ) { // write zeros to the output stream + bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; + bufferBytes *= formatBytes( stream_.userFormat ); + memset( stream_.userBuffer[0], 0, bufferBytes ); + } - DWORD dsBufferSize = handle->dsBufferSize[0]; - nextWritePos = handle->bufferPointer[0]; + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0]; + bufferBytes *= formatBytes( stream_.deviceFormat[0] ); + } + else { + buffer = stream_.userBuffer[0]; + bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; + bufferBytes *= formatBytes( stream_.userFormat ); + } - DWORD endWrite; - while ( true ) { - // Find out where the read and "safe write" pointers are. - result = dsBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } + // No byte swapping necessary in DirectSound implementation. - leadPos = safeWritePos + handle->dsPointerLeadTime[0]; - if ( leadPos > dsBufferSize ) leadPos -= dsBufferSize; - if ( leadPos < nextWritePos ) leadPos += dsBufferSize; // unwrap offset - endWrite = nextWritePos + bufferBytes; - - // Check whether the entire write region is behind the play pointer. - if ( leadPos >= endWrite ) break; - - // If we are here, then we must wait until the play pointer gets - // beyond the write region. The approach here is to use the - // Sleep() function to suspend operation until safePos catches - // up. Calculate number of milliseconds to wait as: - // time = distance * (milliseconds/second) * fudgefactor / - // ((bytes/sample) * (samples/second)) - // A "fudgefactor" less than 1 is used because it was found - // that sleeping too long was MUCH worse than sleeping for - // several shorter periods. - double millis = ( endWrite - leadPos ) * 900.0; - millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate); - if ( millis < 1.0 ) millis = 1.0; - if ( millis > 50.0 ) { - static int nOverruns = 0; - ++nOverruns; - } - Sleep( (DWORD) millis ); - } + // Ahhh ... windoze. 16-bit data is signed but 8-bit data is + // unsigned. So, we need to convert our signed 8-bit data here to + // unsigned. + if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 ) + for ( int i=0; idsBufferSize[0] ) ) { - // statistics.writeDeviceSafeLeadBytes = dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ); - //} - - if ( dsPointerBetween( nextWritePos, safeWritePos, currentWritePos, dsBufferSize ) - || dsPointerBetween( endWrite, safeWritePos, currentWritePos, dsBufferSize ) ) { - // We've strayed into the forbidden zone ... resync the read pointer. - //++statistics.numberOfWriteUnderruns; - handle->xrun[0] = true; - nextWritePos = safeWritePos + handle->dsPointerLeadTime[0] - bufferBytes + dsBufferSize; - while ( nextWritePos >= dsBufferSize ) nextWritePos -= dsBufferSize; - handle->bufferPointer[0] = nextWritePos; - endWrite = nextWritePos + bufferBytes; - } + DWORD dsBufferSize = handle->dsBufferSize[0]; + nextWritePos = handle->bufferPointer[0]; - // Lock free space in the buffer - result = dsBuffer->Lock( nextWritePos, bufferBytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0 ); + DWORD endWrite; + while ( true ) { + // Find out where the read and "safe write" pointers are. + result = dsBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos ); if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!"; + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; errorText_ = errorStream_.str(); error( RtError::SYSTEM_ERROR ); } - // Copy our buffer into the DS buffer - CopyMemory( buffer1, buffer, bufferSize1 ); - if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 ); - - // Update our buffer offset and unlock sound buffer - dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - nextWritePos = ( nextWritePos + bufferSize1 + bufferSize2 ) % dsBufferSize; + leadPos = safeWritePos + handle->dsPointerLeadTime[0]; + if ( leadPos > dsBufferSize ) leadPos -= dsBufferSize; + if ( leadPos < nextWritePos ) leadPos += dsBufferSize; // unwrap offset + endWrite = nextWritePos + bufferBytes; + + // Check whether the entire write region is behind the play pointer. + if ( leadPos >= endWrite ) break; + + // If we are here, then we must wait until the play pointer gets + // beyond the write region. The approach here is to use the + // Sleep() function to suspend operation until safePos catches + // up. Calculate number of milliseconds to wait as: + // time = distance * (milliseconds/second) * fudgefactor / + // ((bytes/sample) * (samples/second)) + // A "fudgefactor" less than 1 is used because it was found + // that sleeping too long was MUCH worse than sleeping for + // several shorter periods. + double millis = ( endWrite - leadPos ) * 900.0; + millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate); + if ( millis < 1.0 ) millis = 1.0; + if ( millis > 50.0 ) { + static int nOverruns = 0; + ++nOverruns; + } + Sleep( (DWORD) millis ); + } + + //if ( statistics.writeDeviceSafeLeadBytes < dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ) ) { + // statistics.writeDeviceSafeLeadBytes = dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ); + //} + + if ( dsPointerBetween( nextWritePos, safeWritePos, currentWritePos, dsBufferSize ) + || dsPointerBetween( endWrite, safeWritePos, currentWritePos, dsBufferSize ) ) { + // We've strayed into the forbidden zone ... resync the read pointer. + //++statistics.numberOfWriteUnderruns; + handle->xrun[0] = true; + nextWritePos = safeWritePos + handle->dsPointerLeadTime[0] - bufferBytes + dsBufferSize; + while ( nextWritePos >= dsBufferSize ) nextWritePos -= dsBufferSize; handle->bufferPointer[0] = nextWritePos; + endWrite = nextWritePos + bufferBytes; + } - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } + // Lock free space in the buffer + result = dsBuffer->Lock( nextWritePos, bufferBytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); } - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + // Copy our buffer into the DS buffer + CopyMemory( buffer1, buffer, bufferSize1 ); + if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 ); - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1]; - bufferBytes *= formatBytes( stream_.deviceFormat[1] ); - } - else { - buffer = stream_.userBuffer[1]; - bufferBytes = stream_.bufferSize * stream_.nUserChannels[1]; - bufferBytes *= formatBytes( stream_.userFormat ); - } + // Update our buffer offset and unlock sound buffer + dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + nextWritePos = ( nextWritePos + bufferSize1 + bufferSize2 ) % dsBufferSize; + handle->bufferPointer[0] = nextWritePos; - LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - long nextReadPos = handle->bufferPointer[1]; - DWORD dsBufferSize = handle->dsBufferSize[1]; + if ( handle->drainCounter ) { + handle->drainCounter++; + goto unlock; + } + } - // Find out where the write and "safe read" pointers are. - result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset - DWORD endRead = nextReadPos + bufferBytes; - - // Handling depends on whether we are INPUT or DUPLEX. - // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode, - // then a wait here will drag the write pointers into the forbidden zone. - // - // In DUPLEX mode, rather than wait, we will back off the read pointer until - // it's in a safe position. This causes dropouts, but it seems to be the only - // practical way to sync up the read and write pointers reliably, given the - // the very complex relationship between phase and increment of the read and write - // pointers. - // - // In order to minimize audible dropouts in DUPLEX mode, we will - // provide a pre-roll period of 0.5 seconds in which we return - // zeros from the read buffer while the pointers sync up. - - if ( stream_.mode == DUPLEX ) { - if ( safeReadPos < endRead ) { - if ( duplexPrerollBytes <= 0 ) { - // Pre-roll time over. Be more agressive. - int adjustment = endRead-safeReadPos; - - handle->xrun[1] = true; - //++statistics.numberOfReadOverruns; - // Two cases: - // - large adjustments: we've probably run out of CPU cycles, so just resync exactly, - // and perform fine adjustments later. - // - small adjustments: back off by twice as much. - if ( adjustment >= 2*bufferBytes ) - nextReadPos = safeReadPos-2*bufferBytes; - else - nextReadPos = safeReadPos-bufferBytes-adjustment; - - //statistics.readDeviceSafeLeadBytes = currentReadPos-nextReadPos; - //if ( statistics.readDeviceSafeLeadBytes < 0) statistics.readDeviceSafeLeadBytes += dsBufferSize; - if ( nextReadPos < 0 ) nextReadPos += dsBufferSize; + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1]; + bufferBytes *= formatBytes( stream_.deviceFormat[1] ); + } + else { + buffer = stream_.userBuffer[1]; + bufferBytes = stream_.bufferSize * stream_.nUserChannels[1]; + bufferBytes *= formatBytes( stream_.userFormat ); + } - } - else { - // In pre=roll time. Just do it. - nextReadPos = safeReadPos-bufferBytes; - while ( nextReadPos < 0 ) nextReadPos += dsBufferSize; - } - endRead = nextReadPos + bufferBytes; - } - } - else { // mode == INPUT - while ( safeReadPos < endRead ) { - // See comments for playback. - double millis = (endRead - safeReadPos) * 900.0; - millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate); - if ( millis < 1.0 ) millis = 1.0; - Sleep( (DWORD) millis ); - - // Wake up, find out where we are now - result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - - if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset - } - } + LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; + long nextReadPos = handle->bufferPointer[1]; + DWORD dsBufferSize = handle->dsBufferSize[1]; + + // Find out where the write and "safe read" pointers are. + result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } - //if (statistics.readDeviceSafeLeadBytes < dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ) ) - // statistics.readDeviceSafeLeadBytes = dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ); + if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset + DWORD endRead = nextReadPos + bufferBytes; - // Lock free space in the buffer - result = dsBuffer->Lock( nextReadPos, bufferBytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } + // Handling depends on whether we are INPUT or DUPLEX. + // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode, + // then a wait here will drag the write pointers into the forbidden zone. + // + // In DUPLEX mode, rather than wait, we will back off the read pointer until + // it's in a safe position. This causes dropouts, but it seems to be the only + // practical way to sync up the read and write pointers reliably, given the + // the very complex relationship between phase and increment of the read and write + // pointers. + // + // In order to minimize audible dropouts in DUPLEX mode, we will + // provide a pre-roll period of 0.5 seconds in which we return + // zeros from the read buffer while the pointers sync up. - if ( duplexPrerollBytes <= 0 ) { - // Copy our buffer into the DS buffer - CopyMemory( buffer, buffer1, bufferSize1 ); - if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 ); - } - else { - memset( buffer, 0, bufferSize1 ); - if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 ); - duplexPrerollBytes -= bufferSize1 + bufferSize2; + if ( stream_.mode == DUPLEX ) { + if ( safeReadPos < endRead ) { + if ( duplexPrerollBytes <= 0 ) { + // Pre-roll time over. Be more agressive. + int adjustment = endRead-safeReadPos; + + handle->xrun[1] = true; + //++statistics.numberOfReadOverruns; + // Two cases: + // - large adjustments: we've probably run out of CPU cycles, so just resync exactly, + // and perform fine adjustments later. + // - small adjustments: back off by twice as much. + if ( adjustment >= 2*bufferBytes ) + nextReadPos = safeReadPos-2*bufferBytes; + else + nextReadPos = safeReadPos-bufferBytes-adjustment; + + //statistics.readDeviceSafeLeadBytes = currentReadPos-nextReadPos; + //if ( statistics.readDeviceSafeLeadBytes < 0) statistics.readDeviceSafeLeadBytes += dsBufferSize; + if ( nextReadPos < 0 ) nextReadPos += dsBufferSize; + + } + else { + // In pre=roll time. Just do it. + nextReadPos = safeReadPos-bufferBytes; + while ( nextReadPos < 0 ) nextReadPos += dsBufferSize; + } + endRead = nextReadPos + bufferBytes; } + } + else { // mode == INPUT + while ( safeReadPos < endRead ) { + // See comments for playback. + double millis = (endRead - safeReadPos) * 900.0; + millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate); + if ( millis < 1.0 ) millis = 1.0; + Sleep( (DWORD) millis ); - // Update our buffer offset and unlock sound buffer - nextReadPos = ( nextReadPos + bufferSize1 + bufferSize2 ) % dsBufferSize; - dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); + // Wake up, find out where we are now + result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + + if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset } - handle->bufferPointer[1] = nextReadPos; + } - // No byte swapping necessary in DirectSound implementation. + //if (statistics.readDeviceSafeLeadBytes < dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ) ) + // statistics.readDeviceSafeLeadBytes = dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ); - // If necessary, convert 8-bit data from unsigned to signed. - if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 ) - for ( int j=0; jLock( nextReadPos, bufferBytes, &buffer1, + &bufferSize1, &buffer2, &bufferSize2, 0 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } - // Do buffer conversion if necessary. - if ( stream_.doConvertBuffer[1] ) - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + if ( duplexPrerollBytes <= 0 ) { + // Copy our buffer into the DS buffer + CopyMemory( buffer, buffer1, bufferSize1 ); + if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 ); + } + else { + memset( buffer, 0, bufferSize1 ); + if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 ); + duplexPrerollBytes -= bufferSize1 + bufferSize2; } -#ifdef GENERATE_DEBUG_LOG - if ( currentDebugLogEntry < debugLog.size() ) - { - TTickRecord &r = debugLog[currentDebugLogEntry++]; - r.currentReadPointer = currentReadPos; - r.safeReadPointer = safeReadPos; - r.currentWritePointer = currentWritePos; - r.safeWritePointer = safeWritePos; - r.readTime = readTime; - r.writeTime = writeTime; - r.nextReadPointer = handles[1].bufferPointer; - r.nextWritePointer = handles[0].bufferPointer; - } -#endif - unlock: - MUTEX_UNLOCK( &stream_.mutex ); + // Update our buffer offset and unlock sound buffer + nextReadPos = ( nextReadPos + bufferSize1 + bufferSize2 ) % dsBufferSize; + dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); + if ( FAILED( result ) ) { + errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!"; + errorText_ = errorStream_.str(); + error( RtError::SYSTEM_ERROR ); + } + handle->bufferPointer[1] = nextReadPos; + + // No byte swapping necessary in DirectSound implementation. + + // If necessary, convert 8-bit data from unsigned to signed. + if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 ) + for ( int j=0; jobject; - bool* isRunning = &info->isRunning; + RtApi::tickStreamTime(); +} - while ( *isRunning == true ) { - object->callbackEvent(); - } +// Definitions for utility functions and callbacks +// specific to the DirectSound implementation. - _endthreadex( 0 ); - return 0; +extern "C" unsigned __stdcall callbackHandler( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiDs *object = (RtApiDs *) info->object; + bool* isRunning = &info->isRunning; + + while ( *isRunning == true ) { + object->callbackEvent(); } + _endthreadex( 0 ); + return 0; +} + #include "tchar.h" - std::string convertTChar( LPCTSTR name ) - { - std::string s; +std::string convertTChar( LPCTSTR name ) +{ + std::string s; #if defined( UNICODE ) || defined( _UNICODE ) - // Yes, this conversion doesn't make sense for two-byte characters - // but RtAudio is currently written to return an std::string of - // one-byte chars for the device name. - for ( unsigned int i=0; iisInput == true ) { - DSCCAPS caps; - LPDIRECTSOUNDCAPTURE object; + HRESULT hr; + if ( info->isInput == true ) { + DSCCAPS caps; + LPDIRECTSOUNDCAPTURE object; - hr = DirectSoundCaptureCreate( lpguid, &object, NULL ); - if ( hr != DS_OK ) return TRUE; + hr = DirectSoundCaptureCreate( lpguid, &object, NULL ); + if ( hr != DS_OK ) return TRUE; - caps.dwSize = sizeof(caps); - hr = object->GetCaps( &caps ); - if ( hr == DS_OK ) { - if ( caps.dwChannels > 0 && caps.dwFormats > 0 ) - info->counter++; - } - object->Release(); - } - else { - DSCAPS caps; - LPDIRECTSOUND object; - hr = DirectSoundCreate( lpguid, &object, NULL ); - if ( hr != DS_OK ) return TRUE; - - caps.dwSize = sizeof(caps); - hr = object->GetCaps( &caps ); - if ( hr == DS_OK ) { - if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO ) - info->counter++; - } - object->Release(); + caps.dwSize = sizeof(caps); + hr = object->GetCaps( &caps ); + if ( hr == DS_OK ) { + if ( caps.dwChannels > 0 && caps.dwFormats > 0 ) + info->counter++; } + object->Release(); + } + else { + DSCAPS caps; + LPDIRECTSOUND object; + hr = DirectSoundCreate( lpguid, &object, NULL ); + if ( hr != DS_OK ) return TRUE; - if ( info->getDefault && lpguid == NULL ) return FALSE; - - if ( info->findIndex && info->counter > info->index ) { - info->id = lpguid; - info->name = convertTChar( description ); - return FALSE; + caps.dwSize = sizeof(caps); + hr = object->GetCaps( &caps ); + if ( hr == DS_OK ) { + if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO ) + info->counter++; } + object->Release(); + } - return TRUE; + if ( info->getDefault && lpguid == NULL ) return FALSE; + + if ( info->findIndex && info->counter > info->index ) { + info->id = lpguid; + info->name = convertTChar( description ); + return FALSE; } - static char* getErrorString( int code ) - { - switch ( code ) { + return TRUE; +} + +static char* getErrorString( int code ) +{ + switch ( code ) { - case DSERR_ALLOCATED: - return "Already allocated"; + case DSERR_ALLOCATED: + return "Already allocated"; - case DSERR_CONTROLUNAVAIL: - return "Control unavailable"; + case DSERR_CONTROLUNAVAIL: + return "Control unavailable"; - case DSERR_INVALIDPARAM: - return "Invalid parameter"; + case DSERR_INVALIDPARAM: + return "Invalid parameter"; - case DSERR_INVALIDCALL: - return "Invalid call"; + case DSERR_INVALIDCALL: + return "Invalid call"; - case DSERR_GENERIC: - return "Generic error"; + case DSERR_GENERIC: + return "Generic error"; - case DSERR_PRIOLEVELNEEDED: - return "Priority level needed"; + case DSERR_PRIOLEVELNEEDED: + return "Priority level needed"; - case DSERR_OUTOFMEMORY: - return "Out of memory"; + case DSERR_OUTOFMEMORY: + return "Out of memory"; - case DSERR_BADFORMAT: - return "The sample rate or the channel format is not supported"; + case DSERR_BADFORMAT: + return "The sample rate or the channel format is not supported"; - case DSERR_UNSUPPORTED: - return "Not supported"; + case DSERR_UNSUPPORTED: + return "Not supported"; - case DSERR_NODRIVER: - return "No driver"; + case DSERR_NODRIVER: + return "No driver"; - case DSERR_ALREADYINITIALIZED: - return "Already initialized"; + case DSERR_ALREADYINITIALIZED: + return "Already initialized"; - case DSERR_NOAGGREGATION: - return "No aggregation"; + case DSERR_NOAGGREGATION: + return "No aggregation"; - case DSERR_BUFFERLOST: - return "Buffer lost"; + case DSERR_BUFFERLOST: + return "Buffer lost"; - case DSERR_OTHERAPPHASPRIO: - return "Another application already has priority"; + case DSERR_OTHERAPPHASPRIO: + return "Another application already has priority"; - case DSERR_UNINITIALIZED: - return "Uninitialized"; + case DSERR_UNINITIALIZED: + return "Uninitialized"; - default: - return "DirectSound unknown error"; - } + default: + return "DirectSound unknown error"; } - //******************** End of __WINDOWS_DS__ *********************// +} +//******************** End of __WINDOWS_DS__ *********************// #endif @@ -4974,1220 +5009,1228 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, // A structure to hold various information related to the ALSA API // implementation. - struct AlsaHandle { - snd_pcm_t *handles[2]; - bool synchronized; - bool xrun[2]; - pthread_cond_t runnable; - - AlsaHandle() - :synchronized(false) { xrun[0] = false; xrun[1] = false; } - }; +struct AlsaHandle { + snd_pcm_t *handles[2]; + bool synchronized; + bool xrun[2]; + pthread_cond_t runnable; - extern "C" void *alsaCallbackHandler( void * ptr ); + AlsaHandle() + :synchronized(false) { xrun[0] = false; xrun[1] = false; } +}; - RtApiAlsa :: RtApiAlsa() - { - // Nothing to do here. - } +extern "C" void *alsaCallbackHandler( void * ptr ); - RtApiAlsa :: ~RtApiAlsa() - { - if ( stream_.state != STREAM_CLOSED ) closeStream(); - } +RtApiAlsa :: RtApiAlsa() +{ + // Nothing to do here. +} - unsigned int RtApiAlsa :: getDeviceCount( void ) - { - unsigned nDevices = 0; - int result, subdevice, card; - char name[64]; - snd_ctl_t *handle; +RtApiAlsa :: ~RtApiAlsa() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} - // Count cards and devices - card = -1; - snd_card_next( &card ); - while ( card >= 0 ) { - sprintf( name, "hw:%d", card ); - result = snd_ctl_open( &handle, name, 0 ); +unsigned int RtApiAlsa :: getDeviceCount( void ) +{ + unsigned nDevices = 0; + int result, subdevice, card; + char name[64]; + snd_ctl_t *handle; + + // Count cards and devices + card = -1; + snd_card_next( &card ); + while ( card >= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &handle, name, 0 ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + goto nextcard; + } + subdevice = -1; + while( 1 ) { + result = snd_ctl_pcm_next_device( handle, &subdevice ); if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << "."; + errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtError::WARNING ); - goto nextcard; - } - subdevice = -1; - while( 1 ) { - result = snd_ctl_pcm_next_device( handle, &subdevice ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - break; - } - if ( subdevice < 0 ) - break; - nDevices++; + break; } - nextcard: - snd_ctl_close( handle ); - snd_card_next( &card ); + if ( subdevice < 0 ) + break; + nDevices++; } - - return nDevices; + nextcard: + snd_ctl_close( handle ); + snd_card_next( &card ); } - RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) - { - RtAudio::DeviceInfo info; - info.probed = false; + return nDevices; +} - unsigned nDevices = 0; - int result, subdevice, card; - char name[64]; - snd_ctl_t *chandle; +RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; - // Count cards and devices - card = -1; - snd_card_next( &card ); - while ( card >= 0 ) { - sprintf( name, "hw:%d", card ); - result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); + unsigned nDevices = 0; + int result, subdevice, card; + char name[64]; + snd_ctl_t *chandle; + + // Count cards and devices + card = -1; + snd_card_next( &card ); + while ( card >= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + goto nextcard; + } + subdevice = -1; + while( 1 ) { + result = snd_ctl_pcm_next_device( chandle, &subdevice ); if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << "."; + errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtError::WARNING ); - goto nextcard; + break; } - subdevice = -1; - while( 1 ) { - result = snd_ctl_pcm_next_device( chandle, &subdevice ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - break; - } - if ( subdevice < 0 ) break; - if ( nDevices == device ) { - sprintf( name, "hw:%d,%d", card, subdevice ); - goto foundDevice; - } - nDevices++; + if ( subdevice < 0 ) break; + if ( nDevices == device ) { + sprintf( name, "hw:%d,%d", card, subdevice ); + goto foundDevice; } - nextcard: - snd_ctl_close( chandle ); - snd_card_next( &card ); + nDevices++; } + nextcard: + snd_ctl_close( chandle ); + snd_card_next( &card ); + } - if ( nDevices == 0 ) { - errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!"; - error( RtError::INVALID_USE ); - } + if ( nDevices == 0 ) { + errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!"; + error( RtError::INVALID_USE ); + } - if ( device >= nDevices ) { - errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - } + if ( device >= nDevices ) { + errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } - foundDevice: + foundDevice: - // If a stream is already open, we cannot probe the stream devices. - // Thus, use the saved results. - if ( stream_.state != STREAM_CLOSED && - ( stream_.device[0] == device || stream_.device[1] == device ) ) { - if ( device >= devices_.size() ) { - errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened."; - error( RtError::WARNING ); - return info; - } - return devices_[ device ]; + // If a stream is already open, we cannot probe the stream devices. + // Thus, use the saved results. + if ( stream_.state != STREAM_CLOSED && + ( stream_.device[0] == device || stream_.device[1] == device ) ) { + if ( device >= devices_.size() ) { + errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened."; + error( RtError::WARNING ); + return info; } + return devices_[ device ]; + } - int openMode = SND_PCM_ASYNC; - snd_pcm_stream_t stream; - snd_pcm_info_t *pcminfo; - snd_pcm_info_alloca( &pcminfo ); - snd_pcm_t *phandle; - snd_pcm_hw_params_t *params; - snd_pcm_hw_params_alloca( ¶ms ); - - // First try for playback - stream = SND_PCM_STREAM_PLAYBACK; - snd_pcm_info_set_device( pcminfo, subdevice ); - snd_pcm_info_set_subdevice( pcminfo, 0 ); - snd_pcm_info_set_stream( pcminfo, stream ); + int openMode = SND_PCM_ASYNC; + snd_pcm_stream_t stream; + snd_pcm_info_t *pcminfo; + snd_pcm_info_alloca( &pcminfo ); + snd_pcm_t *phandle; + snd_pcm_hw_params_t *params; + snd_pcm_hw_params_alloca( ¶ms ); - result = snd_ctl_pcm_info( chandle, pcminfo ); - if ( result < 0 ) { - // Device probably doesn't support playback. - goto captureProbe; - } + // First try for playback + stream = SND_PCM_STREAM_PLAYBACK; + snd_pcm_info_set_device( pcminfo, subdevice ); + snd_pcm_info_set_subdevice( pcminfo, 0 ); + snd_pcm_info_set_stream( pcminfo, stream ); - result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto captureProbe; - } + result = snd_ctl_pcm_info( chandle, pcminfo ); + if ( result < 0 ) { + // Device probably doesn't support playback. + goto captureProbe; + } - // The device is open ... fill the parameter structure. - result = snd_pcm_hw_params_any( phandle, params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto captureProbe; - } + result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + goto captureProbe; + } - // Get output channel information. - unsigned int value; - result = snd_pcm_hw_params_get_channels_max( params, &value ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto captureProbe; - } - info.outputChannels = value; + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + goto captureProbe; + } - captureProbe: - // Now try for capture - stream = SND_PCM_STREAM_CAPTURE; - snd_pcm_info_set_stream( pcminfo, stream ); + // Get output channel information. + unsigned int value; + result = snd_pcm_hw_params_get_channels_max( params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + goto captureProbe; + } + info.outputChannels = value; + snd_pcm_close( phandle ); - result = snd_ctl_pcm_info( chandle, pcminfo ); - snd_ctl_close( chandle ); - if ( result < 0 ) { - // Device probably doesn't support capture. - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } + captureProbe: + // Now try for capture + stream = SND_PCM_STREAM_CAPTURE; + snd_pcm_info_set_stream( pcminfo, stream ); - result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } + result = snd_ctl_pcm_info( chandle, pcminfo ); + snd_ctl_close( chandle ); + if ( result < 0 ) { + // Device probably doesn't support capture. + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } - // The device is open ... fill the parameter structure. - result = snd_pcm_hw_params_any( phandle, params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } + result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } - result = snd_pcm_hw_params_get_channels_max( params, &value ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } - info.inputChannels = value; + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } - // If device opens for both playback and capture, we determine the channels. - if ( info.outputChannels > 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - - // ALSA doesn't provide default devices so we'll use the first available one. - if ( device == 0 && info.outputChannels > 0 ) - info.isDefaultOutput = true; - if ( device == 0 && info.inputChannels > 0 ) - info.isDefaultInput = true; - - probeParameters: - // At this point, we just need to figure out the supported data - // formats and sample rates. We'll proceed by opening the device in - // the direction with the maximum number of channels, or playback if - // they are equal. This might limit our sample rate options, but so - // be it. - - if ( info.outputChannels >= info.inputChannels ) - stream = SND_PCM_STREAM_PLAYBACK; - else - stream = SND_PCM_STREAM_CAPTURE; - snd_pcm_info_set_stream( pcminfo, stream ); + result = snd_pcm_hw_params_get_channels_max( params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + if ( info.outputChannels == 0 ) return info; + goto probeParameters; + } + info.inputChannels = value; + snd_pcm_close( phandle ); - result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } + // If device opens for both playback and capture, we determine the channels. + if ( info.outputChannels > 0 && info.inputChannels > 0 ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - // The device is open ... fill the parameter structure. - result = snd_pcm_hw_params_any( phandle, params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } + // ALSA doesn't provide default devices so we'll use the first available one. + if ( device == 0 && info.outputChannels > 0 ) + info.isDefaultOutput = true; + if ( device == 0 && info.inputChannels > 0 ) + info.isDefaultInput = true; - // Test our discrete set of sample rate values. - info.sampleRates.clear(); - for ( unsigned int i=0; i= info.inputChannels ) + stream = SND_PCM_STREAM_PLAYBACK; + else + stream = SND_PCM_STREAM_CAPTURE; + snd_pcm_info_set_stream( pcminfo, stream ); - // Get the device name - char *cardname; - result = snd_card_get_name( card, &cardname ); - if ( result >= 0 ) - sprintf( name, "hw:%s,%d", cardname, subdevice ); - info.name = name; + result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } - // That's all ... close the device and return + // The device is open ... fill the parameter structure. + result = snd_pcm_hw_params_any( phandle, params ); + if ( result < 0 ) { snd_pcm_close( phandle ); - info.probed = true; + errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); return info; } - void RtApiAlsa :: saveDeviceInfo( void ) - { - devices_.clear(); + // Test our discrete set of sample rate values. + info.sampleRates.clear(); + for ( unsigned int i=0; i= 0 ) + sprintf( name, "hw:%s,%d", cardname, subdevice ); + info.name = name; - { + // That's all ... close the device and return + snd_pcm_close( phandle ); + info.probed = true; + return info; +} + +void RtApiAlsa :: saveDeviceInfo( void ) +{ + devices_.clear(); + + unsigned int nDevices = getDeviceCount(); + devices_.resize( nDevices ); + for ( unsigned int i=0; i= 0 ) { - sprintf( name, "hw:%d", card ); - result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - subdevice = -1; - while( 1 ) { - result = snd_ctl_pcm_next_device( chandle, &subdevice ); - if ( result < 0 ) break; - if ( subdevice < 0 ) break; - if ( nDevices == device ) { - sprintf( name, "hw:%d,%d", card, subdevice ); - snd_ctl_close( chandle ); - goto foundDevice; - } - nDevices++; + // Count cards and devices + card = -1; + snd_card_next( &card ); + while ( card >= 0 ) { + sprintf( name, "hw:%d", card ); + result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + subdevice = -1; + while( 1 ) { + result = snd_ctl_pcm_next_device( chandle, &subdevice ); + if ( result < 0 ) break; + if ( subdevice < 0 ) break; + if ( nDevices == device ) { + sprintf( name, "hw:%d,%d", card, subdevice ); + snd_ctl_close( chandle ); + goto foundDevice; } - snd_ctl_close( chandle ); - snd_card_next( &card ); + nDevices++; } + snd_ctl_close( chandle ); + snd_card_next( &card ); + } - if ( nDevices == 0 ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!"; - return FAILURE; - } + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!"; + return FAILURE; + } - if ( device >= nDevices ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. + errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } + + foundDevice: - foundDevice: + // The getDeviceInfo() function will not work for a device that is + // already open. Thus, we'll probe the system before opening a + // stream and save the results for use by getDeviceInfo(). + if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once + this->saveDeviceInfo(); - // The getDeviceInfo() function will not work for a device that is - // already open. Thus, we'll probe the system before opening a - // stream and save the results for use by getDeviceInfo(). - if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once - this->saveDeviceInfo(); + snd_pcm_stream_t stream; + if ( mode == OUTPUT ) + stream = SND_PCM_STREAM_PLAYBACK; + else + stream = SND_PCM_STREAM_CAPTURE; - snd_pcm_stream_t stream; + snd_pcm_t *phandle; + int openMode = SND_PCM_ASYNC; + result = snd_pcm_open( &phandle, name, stream, openMode ); + if ( result < 0 ) { if ( mode == OUTPUT ) - stream = SND_PCM_STREAM_PLAYBACK; + errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output."; else - stream = SND_PCM_STREAM_CAPTURE; - - snd_pcm_t *phandle; - int openMode = SND_PCM_ASYNC; - result = snd_pcm_open( &phandle, name, stream, openMode ); - if ( result < 0 ) { - if ( mode == OUTPUT ) - errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output."; - else - errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input."; - errorText_ = errorStream_.str(); - return FAILURE; - } + errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Fill the parameter structure. - snd_pcm_hw_params_t *hw_params; - snd_pcm_hw_params_alloca( &hw_params ); - result = snd_pcm_hw_params_any( phandle, hw_params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Fill the parameter structure. + snd_pcm_hw_params_t *hw_params; + snd_pcm_hw_params_alloca( &hw_params ); + result = snd_pcm_hw_params_any( phandle, hw_params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } #if defined(__RTAUDIO_DEBUG__) - fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" ); - snd_pcm_hw_params_dump( hw_params, out ); + fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" ); + snd_pcm_hw_params_dump( hw_params, out ); #endif - // Set access ... check user preference. - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) { - stream_.userInterleaved = false; - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); - if ( result < 0 ) { - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); - stream_.deviceInterleaved[mode] = true; - } - else - stream_.deviceInterleaved[mode] = false; - } - else { - stream_.userInterleaved = true; + // Set access ... check user preference. + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) { + stream_.userInterleaved = false; + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); + if ( result < 0 ) { result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); - if ( result < 0 ) { - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); - stream_.deviceInterleaved[mode] = false; - } - else - stream_.deviceInterleaved[mode] = true; + stream_.deviceInterleaved[mode] = true; } - + else + stream_.deviceInterleaved[mode] = false; + } + else { + stream_.userInterleaved = true; + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; + result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); + stream_.deviceInterleaved[mode] = false; } + else + stream_.deviceInterleaved[mode] = true; + } - // Determine how to set the device format. - stream_.userFormat = format; - snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN; - - if ( format == RTAUDIO_SINT8 ) - deviceFormat = SND_PCM_FORMAT_S8; - else if ( format == RTAUDIO_SINT16 ) - deviceFormat = SND_PCM_FORMAT_S16; - else if ( format == RTAUDIO_SINT24 ) - deviceFormat = SND_PCM_FORMAT_S24; - else if ( format == RTAUDIO_SINT32 ) - deviceFormat = SND_PCM_FORMAT_S32; - else if ( format == RTAUDIO_FLOAT32 ) - deviceFormat = SND_PCM_FORMAT_FLOAT; - else if ( format == RTAUDIO_FLOAT64 ) - deviceFormat = SND_PCM_FORMAT_FLOAT64; - - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) { - stream_.deviceFormat[mode] = format; - goto setFormat; - } + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // The user requested format is not natively supported by the device. - deviceFormat = SND_PCM_FORMAT_FLOAT64; - if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; - goto setFormat; - } + // Determine how to set the device format. + stream_.userFormat = format; + snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN; + if ( format == RTAUDIO_SINT8 ) + deviceFormat = SND_PCM_FORMAT_S8; + else if ( format == RTAUDIO_SINT16 ) + deviceFormat = SND_PCM_FORMAT_S16; + else if ( format == RTAUDIO_SINT24 ) + deviceFormat = SND_PCM_FORMAT_S24; + else if ( format == RTAUDIO_SINT32 ) + deviceFormat = SND_PCM_FORMAT_S32; + else if ( format == RTAUDIO_FLOAT32 ) deviceFormat = SND_PCM_FORMAT_FLOAT; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - goto setFormat; - } + else if ( format == RTAUDIO_FLOAT64 ) + deviceFormat = SND_PCM_FORMAT_FLOAT64; - deviceFormat = SND_PCM_FORMAT_S32; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - goto setFormat; - } + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) { + stream_.deviceFormat[mode] = format; + goto setFormat; + } - deviceFormat = SND_PCM_FORMAT_S24; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - goto setFormat; - } + // The user requested format is not natively supported by the device. + deviceFormat = SND_PCM_FORMAT_FLOAT64; + if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; + goto setFormat; + } - deviceFormat = SND_PCM_FORMAT_S16; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - goto setFormat; - } + deviceFormat = SND_PCM_FORMAT_FLOAT; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; + goto setFormat; + } - deviceFormat = SND_PCM_FORMAT_S8; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - goto setFormat; - } + deviceFormat = SND_PCM_FORMAT_S32; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S24; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S16; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + goto setFormat; + } + + deviceFormat = SND_PCM_FORMAT_S8; + if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { + stream_.deviceFormat[mode] = RTAUDIO_SINT8; + goto setFormat; + } + + // If we get here, no supported format was found. + errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + return FAILURE; - // If we get here, no supported format was found. - errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio."; + setFormat: + result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); return FAILURE; + } - setFormat: - result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat ); - if ( result < 0 ) { + // Determine whether byte-swaping is necessary. + stream_.doByteSwap[mode] = false; + if ( deviceFormat != SND_PCM_FORMAT_S8 ) { + result = snd_pcm_format_cpu_endian( deviceFormat ); + if ( result == 0 ) + stream_.doByteSwap[mode] = true; + else if (result < 0) { snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << "."; + errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); return FAILURE; } + } - // Determine whether byte-swaping is necessary. - stream_.doByteSwap[mode] = false; - if ( deviceFormat != SND_PCM_FORMAT_S8 ) { - result = snd_pcm_format_cpu_endian( deviceFormat ); - if ( result == 0 ) - stream_.doByteSwap[mode] = true; - else if (result < 0) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // Set the sample rate. - result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Set the sample rate. + result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Determine the number of channels for this device. We support a possible - // minimum device channel number > than the value requested by the user. - stream_.nUserChannels[mode] = channels; - unsigned int value; - result = snd_pcm_hw_params_get_channels_max( hw_params, &value ); - unsigned int deviceChannels = value; - if ( result < 0 || deviceChannels < channels + firstChannel ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Determine the number of channels for this device. We support a possible + // minimum device channel number > than the value requested by the user. + stream_.nUserChannels[mode] = channels; + unsigned int value; + result = snd_pcm_hw_params_get_channels_max( hw_params, &value ); + unsigned int deviceChannels = value; + if ( result < 0 || deviceChannels < channels + firstChannel ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } - result = snd_pcm_hw_params_get_channels_min( hw_params, &value ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - deviceChannels = value; - if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel; - stream_.nDeviceChannels[mode] = deviceChannels; + result = snd_pcm_hw_params_get_channels_min( hw_params, &value ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + deviceChannels = value; + if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel; + stream_.nDeviceChannels[mode] = deviceChannels; - // Set the device channels. - result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Set the device channels. + result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Set the buffer number, which in ALSA is referred to as the "period". - int totalSize, dir; - unsigned int periods = 0; - if ( options ) periods = options->numberOfBuffers; - totalSize = *bufferSize * periods; + // Set the buffer number, which in ALSA is referred to as the "period". + int totalSize, dir; + unsigned int periods = 0; + if ( options ) periods = options->numberOfBuffers; + totalSize = *bufferSize * periods; - // Set the buffer (or period) size. - snd_pcm_uframes_t periodSize = *bufferSize; - result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - *bufferSize = periodSize; + // Set the buffer (or period) size. + snd_pcm_uframes_t periodSize = *bufferSize; + result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } + *bufferSize = periodSize; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; - else periods = totalSize / *bufferSize; - // Even though the hardware might allow 1 buffer, it won't work reliably. - if ( periods < 2 ) periods = 2; - result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; + else periods = totalSize / *bufferSize; + // Even though the hardware might allow 1 buffer, it won't work reliably. + if ( periods < 2 ) periods = 2; + result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // If attempting to setup a duplex stream, the bufferSize parameter - // MUST be the same in both directions! - if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { - errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // If attempting to setup a duplex stream, the bufferSize parameter + // MUST be the same in both directions! + if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { + errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } - stream_.bufferSize = *bufferSize; + stream_.bufferSize = *bufferSize; - // Install the hardware configuration - result = snd_pcm_hw_params( phandle, hw_params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Install the hardware configuration + result = snd_pcm_hw_params( phandle, hw_params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } #if defined(__RTAUDIO_DEBUG__) - fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n"); - snd_pcm_hw_params_dump( hw_params, out ); + fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n"); + snd_pcm_hw_params_dump( hw_params, out ); #endif - // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns. - snd_pcm_sw_params_t *sw_params = NULL; - snd_pcm_sw_params_alloca( &sw_params ); - snd_pcm_sw_params_current( phandle, sw_params ); - snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize ); - snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX ); - snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 ); - - // The following two settings were suggested by Theo Veenker - //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize ); - //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 ); - - // here are two options for a fix - //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX ); - snd_pcm_uframes_t val; - snd_pcm_sw_params_get_boundary( sw_params, &val ); - snd_pcm_sw_params_set_silence_size( phandle, sw_params, val ); - - result = snd_pcm_sw_params( phandle, sw_params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns. + snd_pcm_sw_params_t *sw_params = NULL; + snd_pcm_sw_params_alloca( &sw_params ); + snd_pcm_sw_params_current( phandle, sw_params ); + snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize ); + snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX ); + snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 ); + + // The following two settings were suggested by Theo Veenker + //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize ); + //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 ); + + // here are two options for a fix + //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX ); + snd_pcm_uframes_t val; + snd_pcm_sw_params_get_boundary( sw_params, &val ); + snd_pcm_sw_params_set_silence_size( phandle, sw_params, val ); + + result = snd_pcm_sw_params( phandle, sw_params ); + if ( result < 0 ) { + snd_pcm_close( phandle ); + errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + return FAILURE; + } #if defined(__RTAUDIO_DEBUG__) - fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n"); - snd_pcm_sw_params_dump( sw_params, out ); + fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n"); + snd_pcm_sw_params_dump( sw_params, out ); #endif - // Set flags for buffer conversion - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - // Allocate the ApiHandle if necessary and then save. - AlsaHandle *apiInfo = 0; - if ( stream_.apiHandle == 0 ) { - try { - apiInfo = (AlsaHandle *) new AlsaHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory."; - goto error; - } - - if ( pthread_cond_init( &apiInfo->runnable, NULL ) ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable."; - goto error; - } + // Set flags for buffer conversion + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; - stream_.apiHandle = (void *) apiInfo; - apiInfo->handles[0] = 0; - apiInfo->handles[1] = 0; - } - else { - apiInfo = (AlsaHandle *) stream_.apiHandle; + // Allocate the ApiHandle if necessary and then save. + AlsaHandle *apiInfo = 0; + if ( stream_.apiHandle == 0 ) { + try { + apiInfo = (AlsaHandle *) new AlsaHandle; } - apiInfo->handles[mode] = phandle; - - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory."; + catch ( std::bad_alloc& ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory."; goto error; } - if ( stream_.doConvertBuffer[mode] ) { - - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } + if ( pthread_cond_init( &apiInfo->runnable, NULL ) ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable."; + goto error; } - stream_.sampleRate = sampleRate; - stream_.nBuffers = periods; - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + stream_.apiHandle = (void *) apiInfo; + apiInfo->handles[0] = 0; + apiInfo->handles[1] = 0; + } + else { + apiInfo = (AlsaHandle *) stream_.apiHandle; + } + apiInfo->handles[mode] = phandle; - // Setup thread if necessary. - if ( stream_.mode == OUTPUT && mode == INPUT ) { - // We had already set up an output stream. - stream_.mode = DUPLEX; - // Link the streams if possible. - apiInfo->synchronized = false; - if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 ) - apiInfo->synchronized = true; - else { - errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices."; - error( RtError::WARNING ); - } - } - else { - stream_.mode = mode; + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } - // Setup callback thread. - stream_.callbackInfo.object = (void *) this; - - // Set the thread attributes for joinable and realtime scheduling - // priority (optional). The higher priority will only take affect - // if the program is run as root or suid. Note, under Linux - // processes with CAP_SYS_NICE privilege, a user can change - // scheduling policy and priority (thus need not be root). See - // POSIX "capabilities". - pthread_attr_t attr; - pthread_attr_init( &attr ); - pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); -#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) - if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { - struct sched_param param; - int priority = options->priority; - int min = sched_get_priority_min( SCHED_RR ); - int max = sched_get_priority_max( SCHED_RR ); - if ( priority < min ) priority = min; - else if ( priority > max ) priority = max; - param.sched_priority = priority; - pthread_attr_setschedparam( &attr, ¶m ); - pthread_attr_setschedpolicy( &attr, SCHED_RR ); - } - else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#endif + if ( stream_.doConvertBuffer[mode] ) { - stream_.callbackInfo.isRunning = true; - result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo ); - pthread_attr_destroy( &attr ); - if ( result ) { - stream_.callbackInfo.isRunning = false; - errorText_ = "RtApiAlsa::error creating callback thread!"; - goto error; + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; } } - return SUCCESS; - - error: - if ( apiInfo ) { - pthread_cond_destroy( &apiInfo->runnable ); - if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); - if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); - delete apiInfo; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory."; + goto error; } } + } - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } + stream_.sampleRate = sampleRate; + stream_.nBuffers = periods; + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; - return FAILURE; - } + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - void RtApiAlsa :: closeStream() - { - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAlsa::closeStream(): no open stream to close!"; + // Setup thread if necessary. + if ( stream_.mode == OUTPUT && mode == INPUT ) { + // We had already set up an output stream. + stream_.mode = DUPLEX; + // Link the streams if possible. + apiInfo->synchronized = false; + if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 ) + apiInfo->synchronized = true; + else { + errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices."; error( RtError::WARNING ); - return; } + } + else { + stream_.mode = mode; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - stream_.callbackInfo.isRunning = false; - MUTEX_LOCK( &stream_.mutex ); - if ( stream_.state == STREAM_STOPPED ) - pthread_cond_signal( &apiInfo->runnable ); - MUTEX_UNLOCK( &stream_.mutex ); - pthread_join( stream_.callbackInfo.thread, NULL ); + // Setup callback thread. + stream_.callbackInfo.object = (void *) this; - if ( stream_.state == STREAM_RUNNING ) { - stream_.state = STREAM_STOPPED; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) - snd_pcm_drop( apiInfo->handles[0] ); - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) - snd_pcm_drop( apiInfo->handles[1] ); + // Set the thread attributes for joinable and realtime scheduling + // priority (optional). The higher priority will only take affect + // if the program is run as root or suid. Note, under Linux + // processes with CAP_SYS_NICE privilege, a user can change + // scheduling policy and priority (thus need not be root). See + // POSIX "capabilities". + pthread_attr_t attr; + pthread_attr_init( &attr ); + pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); +#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { + struct sched_param param; + int priority = options->priority; + int min = sched_get_priority_min( SCHED_RR ); + int max = sched_get_priority_max( SCHED_RR ); + if ( priority < min ) priority = min; + else if ( priority > max ) priority = max; + param.sched_priority = priority; + pthread_attr_setschedparam( &attr, ¶m ); + pthread_attr_setschedpolicy( &attr, SCHED_RR ); } + else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#endif - if ( apiInfo ) { - pthread_cond_destroy( &apiInfo->runnable ); - if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); - if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); - delete apiInfo; - stream_.apiHandle = 0; + stream_.callbackInfo.isRunning = true; + result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo ); + pthread_attr_destroy( &attr ); + if ( result ) { + stream_.callbackInfo.isRunning = false; + errorText_ = "RtApiAlsa::error creating callback thread!"; + goto error; } + } - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } + return SUCCESS; - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; + error: + if ( apiInfo ) { + pthread_cond_destroy( &apiInfo->runnable ); + if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); + if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); + delete apiInfo; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; } + } - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; } - void RtApiAlsa :: startStream() - { - // This method calls snd_pcm_prepare if the device isn't already in that state. + return FAILURE; +} - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiAlsa::startStream(): the stream is already running!"; - error( RtError::WARNING ); - return; - } +void RtApiAlsa :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAlsa::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } - MUTEX_LOCK( &stream_.mutex ); + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + stream_.callbackInfo.isRunning = false; + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) + pthread_cond_signal( &apiInfo->runnable ); + MUTEX_UNLOCK( &stream_.mutex ); + pthread_join( stream_.callbackInfo.thread, NULL ); - int result = 0; - snd_pcm_state_t state; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - state = snd_pcm_state( handle[0] ); - if ( state != SND_PCM_STATE_PREPARED ) { - result = snd_pcm_prepare( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - } + if ( stream_.state == STREAM_RUNNING ) { + stream_.state = STREAM_STOPPED; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) + snd_pcm_drop( apiInfo->handles[0] ); + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) + snd_pcm_drop( apiInfo->handles[1] ); + } - if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { - state = snd_pcm_state( handle[1] ); - if ( state != SND_PCM_STATE_PREPARED ) { - result = snd_pcm_prepare( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } + if ( apiInfo ) { + pthread_cond_destroy( &apiInfo->runnable ); + if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); + if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); + delete apiInfo; + stream_.apiHandle = 0; + } + + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; } + } - stream_.state = STREAM_RUNNING; + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; + } - unlock: - MUTEX_UNLOCK( &stream_.mutex ); + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} - pthread_cond_signal( &apiInfo->runnable ); +void RtApiAlsa :: startStream() +{ + // This method calls snd_pcm_prepare if the device isn't already in that state. - if ( result >= 0 ) return; - error( RtError::SYSTEM_ERROR ); + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiAlsa::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; } - void RtApiAlsa :: stopStream() - { - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } - - // Change the state before the lock to improve shutdown response. - stream_.state = STREAM_STOPPED; - MUTEX_LOCK( &stream_.mutex ); + MUTEX_LOCK( &stream_.mutex ); - int result = 0; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( apiInfo->synchronized ) - result = snd_pcm_drop( handle[0] ); - else - result = snd_pcm_drain( handle[0] ); + int result = 0; + snd_pcm_state_t state; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + state = snd_pcm_state( handle[0] ); + if ( state != SND_PCM_STATE_PREPARED ) { + result = snd_pcm_prepare( handle[0] ); if ( result < 0 ) { - errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << "."; + errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); goto unlock; } } + } - if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { - result = snd_pcm_drop( handle[1] ); + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + state = snd_pcm_state( handle[1] ); + if ( state != SND_PCM_STATE_PREPARED ) { + result = snd_pcm_prepare( handle[1] ); if ( result < 0 ) { - errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << "."; + errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); goto unlock; } } + } - unlock: - MUTEX_UNLOCK( &stream_.mutex ); + stream_.state = STREAM_RUNNING; - if ( result >= 0 ) return; - error( RtError::SYSTEM_ERROR ); + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + + pthread_cond_signal( &apiInfo->runnable ); + + if ( result >= 0 ) return; + error( RtError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; } - void RtApiAlsa :: abortStream() - { - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } + MUTEX_LOCK( &stream_.mutex ); - // Change the state before the lock to improve shutdown response. - stream_.state = STREAM_STOPPED; - MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } - int result = 0; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + int result = 0; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + if ( apiInfo->synchronized ) result = snd_pcm_drop( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } + else + result = snd_pcm_drain( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; } + } - if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { - result = snd_pcm_drop( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + result = snd_pcm_drop( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; } + } - unlock: - MUTEX_UNLOCK( &stream_.mutex ); + unlock: + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtError::SYSTEM_ERROR ); +} - if ( result >= 0 ) return; - error( RtError::SYSTEM_ERROR ); +void RtApiAlsa :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; } - void RtApiAlsa :: callbackEvent() - { - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_LOCK( &stream_.mutex ); - pthread_cond_wait( &apiInfo->runnable, &stream_.mutex ); - if ( stream_.state != STREAM_RUNNING ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - MUTEX_UNLOCK( &stream_.mutex ); - } + MUTEX_LOCK( &stream_.mutex ); - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); - return; - } + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } - int doStopStream = 0; - RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - apiInfo->xrun[0] = false; + int result = 0; + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + result = snd_pcm_drop( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; } - if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - apiInfo->xrun[1] = false; + } + + if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { + result = snd_pcm_drop( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + goto unlock; } - doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); + } - if ( doStopStream == 2 ) { - abortStream(); + unlock: + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result >= 0 ) return; + error( RtError::SYSTEM_ERROR ); +} + +void RtApiAlsa :: callbackEvent() +{ + AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + pthread_cond_wait( &apiInfo->runnable, &stream_.mutex ); + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); return; } + MUTEX_UNLOCK( &stream_.mutex ); + } - MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return; + } - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) goto unlock; + int doStopStream = 0; + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + apiInfo->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + apiInfo->xrun[1] = false; + } + doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); - int result; - char *buffer; - int channels; - snd_pcm_t **handle; - snd_pcm_sframes_t frames; - RtAudioFormat format; - handle = (snd_pcm_t **) apiInfo->handles; + if ( doStopStream == 2 ) { + abortStream(); + return; + } - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + MUTEX_LOCK( &stream_.mutex ); - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - channels = stream_.nDeviceChannels[1]; - format = stream_.deviceFormat[1]; - } - else { - buffer = stream_.userBuffer[1]; - channels = stream_.nUserChannels[1]; - format = stream_.userFormat; - } + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; - // Read samples from device in interleaved/non-interleaved format. - if ( stream_.deviceInterleaved[1] ) - result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize ); - else { - void *bufs[channels]; - size_t offset = stream_.bufferSize * formatBytes( format ); - for ( int i=0; ihandles; - if ( result < (int) stream_.bufferSize ) { - // Either an error or overrun occured. - if ( result == -EPIPE ) { - snd_pcm_state_t state = snd_pcm_state( handle[1] ); - if ( state == SND_PCM_STATE_XRUN ) { - apiInfo->xrun[1] = true; - result = snd_pcm_prepare( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + channels = stream_.nDeviceChannels[1]; + format = stream_.deviceFormat[1]; + } + else { + buffer = stream_.userBuffer[1]; + channels = stream_.nUserChannels[1]; + format = stream_.userFormat; + } + + // Read samples from device in interleaved/non-interleaved format. + if ( stream_.deviceInterleaved[1] ) + result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize ); + else { + void *bufs[channels]; + size_t offset = stream_.bufferSize * formatBytes( format ); + for ( int i=0; ixrun[1] = true; + result = snd_pcm_prepare( handle[1] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); } } else { - errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << "."; + errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); } - error( RtError::WARNING ); - goto tryOutput; } + else { + errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + error( RtError::WARNING ); + goto tryOutput; + } - // Do byte swapping if necessary. - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( buffer, stream_.bufferSize * channels, format ); - - // Do buffer conversion if necessary. - if ( stream_.doConvertBuffer[1] ) - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + // Do byte swapping if necessary. + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( buffer, stream_.bufferSize * channels, format ); - // Check stream latency - result = snd_pcm_delay( handle[1], &frames ); - if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; - } + // Do buffer conversion if necessary. + if ( stream_.doConvertBuffer[1] ) + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - tryOutput: + // Check stream latency + result = snd_pcm_delay( handle[1], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; + } - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + tryOutput: - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - channels = stream_.nDeviceChannels[0]; - format = stream_.deviceFormat[0]; - } - else { - buffer = stream_.userBuffer[0]; - channels = stream_.nUserChannels[0]; - format = stream_.userFormat; - } + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - // Do byte swapping if necessary. - if ( stream_.doByteSwap[0] ) - byteSwapBuffer(buffer, stream_.bufferSize * channels, format); + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + channels = stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + channels = stream_.nUserChannels[0]; + format = stream_.userFormat; + } - // Write samples to device in interleaved/non-interleaved format. - if ( stream_.deviceInterleaved[0] ) - result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize ); - else { - void *bufs[channels]; - size_t offset = stream_.bufferSize * formatBytes( format ); - for ( int i=0; ixrun[0] = true; - result = snd_pcm_prepare( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; + // Write samples to device in interleaved/non-interleaved format. + if ( stream_.deviceInterleaved[0] ) + result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize ); + else { + void *bufs[channels]; + size_t offset = stream_.bufferSize * formatBytes( format ); + for ( int i=0; ixrun[0] = true; + result = snd_pcm_prepare( handle[0] ); + if ( result < 0 ) { + errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); } } else { - errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << "."; + errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); } - error( RtError::WARNING ); - goto unlock; } - - // Check stream latency - result = snd_pcm_delay( handle[0], &frames ); - if ( result == 0 && frames > 0 ) stream_.latency[0] = frames; + else { + errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << "."; + errorText_ = errorStream_.str(); + } + error( RtError::WARNING ); + goto unlock; } - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - - RtApi::tickStreamTime(); - if ( doStopStream == 1 ) this->stopStream(); + // Check stream latency + result = snd_pcm_delay( handle[0], &frames ); + if ( result == 0 && frames > 0 ) stream_.latency[0] = frames; } - extern "C" void *alsaCallbackHandler( void *ptr ) - { - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiAlsa *object = (RtApiAlsa *) info->object; - bool *isRunning = &info->isRunning; + unlock: + MUTEX_UNLOCK( &stream_.mutex ); - while ( *isRunning == true ) { - pthread_testcancel(); - object->callbackEvent(); - } + RtApi::tickStreamTime(); + if ( doStopStream == 1 ) this->stopStream(); +} + +extern "C" void *alsaCallbackHandler( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiAlsa *object = (RtApiAlsa *) info->object; + bool *isRunning = &info->isRunning; - pthread_exit( NULL ); + while ( *isRunning == true ) { + pthread_testcancel(); + object->callbackEvent(); } - //******************** End of __LINUX_ALSA__ *********************// + pthread_exit( NULL ); +} + +//******************** End of __LINUX_ALSA__ *********************// #endif @@ -6201,1587 +6244,1593 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, #include #include - extern "C" void *ossCallbackHandler(void * ptr); +extern "C" void *ossCallbackHandler(void * ptr); - // A structure to hold various information related to the OSS API - // implementation. - struct OssHandle { - int id[2]; // device ids - bool xrun[2]; - bool triggered; - pthread_cond_t runnable; - - OssHandle() - :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } - }; +// A structure to hold various information related to the OSS API +// implementation. +struct OssHandle { + int id[2]; // device ids + bool xrun[2]; + bool triggered; + pthread_cond_t runnable; - RtApiOss :: RtApiOss() - { - // Nothing to do here. - } + OssHandle() + :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } +}; - RtApiOss :: ~RtApiOss() - { - if ( stream_.state != STREAM_CLOSED ) closeStream(); - } +RtApiOss :: RtApiOss() +{ + // Nothing to do here. +} - unsigned int RtApiOss :: getDeviceCount( void ) - { - int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); - if ( mixerfd == -1 ) { - errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'."; - error( RtError::WARNING ); - return 0; - } +RtApiOss :: ~RtApiOss() +{ + if ( stream_.state != STREAM_CLOSED ) closeStream(); +} - oss_sysinfo sysinfo; - if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required."; - error( RtError::WARNING ); - return 0; - } +unsigned int RtApiOss :: getDeviceCount( void ) +{ + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'."; + error( RtError::WARNING ); + return 0; + } + oss_sysinfo sysinfo; + if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) { close( mixerfd ); - return sysinfo.numaudios; + errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required."; + error( RtError::WARNING ); + return 0; } - RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) - { - RtAudio::DeviceInfo info; - info.probed = false; + close( mixerfd ); + return sysinfo.numaudios; +} - int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); - if ( mixerfd == -1 ) { - errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'."; - error( RtError::WARNING ); - return info; - } +RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = false; - oss_sysinfo sysinfo; - int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); - if ( result == -1 ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required."; - error( RtError::WARNING ); - return info; - } + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'."; + error( RtError::WARNING ); + return info; + } - unsigned nDevices = sysinfo.numaudios; - if ( nDevices == 0 ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceInfo: no devices found!"; - error( RtError::INVALID_USE ); - } + oss_sysinfo sysinfo; + int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); + if ( result == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required."; + error( RtError::WARNING ); + return info; + } - if ( device >= nDevices ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - } + unsigned nDevices = sysinfo.numaudios; + if ( nDevices == 0 ) { + close( mixerfd ); + errorText_ = "RtApiOss::getDeviceInfo: no devices found!"; + error( RtError::INVALID_USE ); + } - oss_audioinfo ainfo; - ainfo.dev = device; - result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); + if ( device >= nDevices ) { close( mixerfd ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } + errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!"; + error( RtError::INVALID_USE ); + } - // Probe channels - if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels; - if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels; - if ( ainfo.caps & PCM_CAP_DUPLEX ) { - if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - } - - // Probe data formats ... do for input - unsigned long mask = ainfo.iformats; - if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE ) - info.nativeFormats |= RTAUDIO_SINT16; - if ( mask & AFMT_S8 ) - info.nativeFormats |= RTAUDIO_SINT8; - if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE ) - info.nativeFormats |= RTAUDIO_SINT32; - if ( mask & AFMT_FLOAT ) - info.nativeFormats |= RTAUDIO_FLOAT32; - if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE ) - info.nativeFormats |= RTAUDIO_SINT24; - - // Check that we have at least one supported format - if ( info.nativeFormats == 0 ) { - errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } + oss_audioinfo ainfo; + ainfo.dev = device; + result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); + close( mixerfd ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } - // Probe the supported sample rates. - info.sampleRates.clear(); - if ( ainfo.nrates ) { - for ( unsigned int i=0; i 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX ) + info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; + } + + // Probe data formats ... do for input + unsigned long mask = ainfo.iformats; + if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE ) + info.nativeFormats |= RTAUDIO_SINT16; + if ( mask & AFMT_S8 ) + info.nativeFormats |= RTAUDIO_SINT8; + if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE ) + info.nativeFormats |= RTAUDIO_SINT32; + if ( mask & AFMT_FLOAT ) + info.nativeFormats |= RTAUDIO_FLOAT32; + if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE ) + info.nativeFormats |= RTAUDIO_SINT24; + + // Check that we have at least one supported format + if ( info.nativeFormats == 0 ) { + errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + return info; + } + + // Probe the supported sample rates. + info.sampleRates.clear(); + if ( ainfo.nrates ) { + for ( unsigned int i=0; i= (int) SAMPLE_RATES[k] ) + if ( ainfo.rates[i] == SAMPLE_RATES[k] ) { info.sampleRates.push_back( SAMPLE_RATES[k] ); + break; + } } } - - if ( info.sampleRates.size() == 0 ) { - errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } - else { - info.probed = true; - info.name = ainfo.name; + } + else { + // Check min and max rate values; + for ( unsigned int k=0; k= (int) SAMPLE_RATES[k] ) + info.sampleRates.push_back( SAMPLE_RATES[k] ); } + } - return info; + if ( info.sampleRates.size() == 0 ) { + errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + } + else { + info.probed = true; + info.name = ainfo.name; } + return info; +} - bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) - { - int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); - if ( mixerfd == -1 ) { - errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'."; - return FAILURE; - } - oss_sysinfo sysinfo; - int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); - if ( result == -1 ) { - close( mixerfd ); - errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required."; - return FAILURE; - } +bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ) +{ + int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); + if ( mixerfd == -1 ) { + errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'."; + return FAILURE; + } - unsigned nDevices = sysinfo.numaudios; - if ( nDevices == 0 ) { - // This should not happen because a check is made before this function is called. - close( mixerfd ); - errorText_ = "RtApiOss::probeDeviceOpen: no devices found!"; - return FAILURE; - } + oss_sysinfo sysinfo; + int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); + if ( result == -1 ) { + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required."; + return FAILURE; + } - if ( device >= nDevices ) { - // This should not happen because a check is made before this function is called. - close( mixerfd ); - errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } + unsigned nDevices = sysinfo.numaudios; + if ( nDevices == 0 ) { + // This should not happen because a check is made before this function is called. + close( mixerfd ); + errorText_ = "RtApiOss::probeDeviceOpen: no devices found!"; + return FAILURE; + } - oss_audioinfo ainfo; - ainfo.dev = device; - result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); + if ( device >= nDevices ) { + // This should not happen because a check is made before this function is called. close( mixerfd ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; - errorText_ = errorStream_.str(); - return FAILURE; - } + errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!"; + return FAILURE; + } - // Check if device supports input or output - if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) || - ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) { - if ( mode == OUTPUT ) - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output."; - else - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input."; - errorText_ = errorStream_.str(); - return FAILURE; - } + oss_audioinfo ainfo; + ainfo.dev = device; + result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); + close( mixerfd ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; + errorText_ = errorStream_.str(); + return FAILURE; + } - int flags = 0; - OssHandle *handle = (OssHandle *) stream_.apiHandle; + // Check if device supports input or output + if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) || + ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) { if ( mode == OUTPUT ) - flags |= O_WRONLY; - else { // mode == INPUT - if (stream_.mode == OUTPUT && stream_.device[0] == device) { - // We just set the same device for playback ... close and reopen for duplex (OSS only). - close( handle->id[0] ); - handle->id[0] = 0; - if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) { - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode."; - errorText_ = errorStream_.str(); - return FAILURE; - } - // Check that the number previously set channels is the same. - if ( stream_.nUserChannels[0] != channels ) { - errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - flags |= O_RDWR; + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output."; + else + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + int flags = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( mode == OUTPUT ) + flags |= O_WRONLY; + else { // mode == INPUT + if (stream_.mode == OUTPUT && stream_.device[0] == device) { + // We just set the same device for playback ... close and reopen for duplex (OSS only). + close( handle->id[0] ); + handle->id[0] = 0; + if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) { + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode."; + errorText_ = errorStream_.str(); + return FAILURE; } - else - flags |= O_RDONLY; + // Check that the number previously set channels is the same. + if ( stream_.nUserChannels[0] != channels ) { + errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + flags |= O_RDWR; } + else + flags |= O_RDONLY; + } - // Set exclusive access if specified. - if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL; + // Set exclusive access if specified. + if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL; - // Try to open the device. - int fd; - fd = open( ainfo.devnode, flags, 0 ); - if ( fd == -1 ) { - if ( errno == EBUSY ) - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy."; - else - errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; + // Try to open the device. + int fd; + fd = open( ainfo.devnode, flags, 0 ); + if ( fd == -1 ) { + if ( errno == EBUSY ) + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy."; + else + errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + + // For duplex operation, specifically set this mode (this doesn't seem to work). + /* + if ( flags | O_RDWR ) { + result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL ); + if ( result == -1) { + errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } } + */ - // For duplex operation, specifically set this mode (this doesn't seem to work). - /* - if ( flags | O_RDWR ) { - result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL ); - if ( result == -1) { - errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - */ + // Check the device channel support. + stream_.nUserChannels[mode] = channels; + if ( ainfo.max_channels < (int)(channels + firstChannel) ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Check the device channel support. - stream_.nUserChannels[mode] = channels; - if ( ainfo.max_channels < (int)(channels + firstChannel) ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Set the number of channels. + int deviceChannels = channels + firstChannel; + result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels ); + if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.nDeviceChannels[mode] = deviceChannels; - // Set the number of channels. - int deviceChannels = channels + firstChannel; - result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels ); - if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.nDeviceChannels[mode] = deviceChannels; + // Get the data format mask + int mask; + result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Get the data format mask - int mask; - result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask ); - if ( result == -1 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats."; - errorText_ = errorStream_.str(); - return FAILURE; + // Determine how to set the device format. + stream_.userFormat = format; + int deviceFormat = -1; + stream_.doByteSwap[mode] = false; + if ( format == RTAUDIO_SINT8 ) { + if ( mask & AFMT_S8 ) { + deviceFormat = AFMT_S8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; } - - // Determine how to set the device format. - stream_.userFormat = format; - int deviceFormat = -1; - stream_.doByteSwap[mode] = false; - if ( format == RTAUDIO_SINT8 ) { - if ( mask & AFMT_S8 ) { - deviceFormat = AFMT_S8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } + } + else if ( format == RTAUDIO_SINT16 ) { + if ( mask & AFMT_S16_NE ) { + deviceFormat = AFMT_S16_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; } - else if ( format == RTAUDIO_SINT16 ) { - if ( mask & AFMT_S16_NE ) { - deviceFormat = AFMT_S16_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - else if ( mask & AFMT_S16_OE ) { - deviceFormat = AFMT_S16_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - stream_.doByteSwap[mode] = true; - } + else if ( mask & AFMT_S16_OE ) { + deviceFormat = AFMT_S16_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + stream_.doByteSwap[mode] = true; } - else if ( format == RTAUDIO_SINT24 ) { - if ( mask & AFMT_S24_NE ) { - deviceFormat = AFMT_S24_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - } - else if ( mask & AFMT_S24_OE ) { - deviceFormat = AFMT_S24_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - stream_.doByteSwap[mode] = true; - } + } + else if ( format == RTAUDIO_SINT24 ) { + if ( mask & AFMT_S24_NE ) { + deviceFormat = AFMT_S24_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; } - else if ( format == RTAUDIO_SINT32 ) { - if ( mask & AFMT_S32_NE ) { - deviceFormat = AFMT_S32_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - } - else if ( mask & AFMT_S32_OE ) { - deviceFormat = AFMT_S32_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - stream_.doByteSwap[mode] = true; - } + else if ( mask & AFMT_S24_OE ) { + deviceFormat = AFMT_S24_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + stream_.doByteSwap[mode] = true; } - - if ( deviceFormat == -1 ) { - // The user requested format is not natively supported by the device. - if ( mask & AFMT_S16_NE ) { - deviceFormat = AFMT_S16_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - else if ( mask & AFMT_S32_NE ) { - deviceFormat = AFMT_S32_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - } - else if ( mask & AFMT_S24_NE ) { - deviceFormat = AFMT_S24_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - } - else if ( mask & AFMT_S16_OE ) { - deviceFormat = AFMT_S16_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - stream_.doByteSwap[mode] = true; - } - else if ( mask & AFMT_S32_OE ) { - deviceFormat = AFMT_S32_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - stream_.doByteSwap[mode] = true; - } - else if ( mask & AFMT_S24_OE ) { - deviceFormat = AFMT_S24_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - stream_.doByteSwap[mode] = true; - } - else if ( mask & AFMT_S8) { - deviceFormat = AFMT_S8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } + } + else if ( format == RTAUDIO_SINT32 ) { + if ( mask & AFMT_S32_NE ) { + deviceFormat = AFMT_S32_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; } - - if ( stream_.deviceFormat[mode] == 0 ) { - // This really shouldn't happen ... - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - return FAILURE; + else if ( mask & AFMT_S32_OE ) { + deviceFormat = AFMT_S32_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + stream_.doByteSwap[mode] = true; } + } - // Set the data format. - int temp = deviceFormat; - result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat ); - if ( result == -1 || deviceFormat != temp ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; + if ( deviceFormat == -1 ) { + // The user requested format is not natively supported by the device. + if ( mask & AFMT_S16_NE ) { + deviceFormat = AFMT_S16_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + } + else if ( mask & AFMT_S32_NE ) { + deviceFormat = AFMT_S32_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + } + else if ( mask & AFMT_S24_NE ) { + deviceFormat = AFMT_S24_NE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + } + else if ( mask & AFMT_S16_OE ) { + deviceFormat = AFMT_S16_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT16; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S32_OE ) { + deviceFormat = AFMT_S32_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT32; + stream_.doByteSwap[mode] = true; + } + else if ( mask & AFMT_S24_OE ) { + deviceFormat = AFMT_S24_OE; + stream_.deviceFormat[mode] = RTAUDIO_SINT24; + stream_.doByteSwap[mode] = true; } - - // Attempt to set the buffer size. According to OSS, the minimum - // number of buffers is two. The supposed minimum buffer size is 16 - // bytes, so that will be our lower bound. The argument to this - // call is in the form 0xMMMMSSSS (hex), where the buffer size (in - // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM. - // We'll check the actual value used near the end of the setup - // procedure. - int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels; - if ( ossBufferBytes < 16 ) ossBufferBytes = 16; - int buffers = 0; - if ( options ) buffers = options->numberOfBuffers; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2; - if ( buffers < 2 ) buffers = 3; - temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) ); - result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp ); - if ( result == -1 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; + else if ( mask & AFMT_S8) { + deviceFormat = AFMT_S8; + stream_.deviceFormat[mode] = RTAUDIO_SINT8; } - stream_.nBuffers = buffers; + } - // Save buffer size (in sample frames). - *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels ); - stream_.bufferSize = *bufferSize; + if ( stream_.deviceFormat[mode] == 0 ) { + // This really shouldn't happen ... + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Set the sample rate. - int srate = sampleRate; - result = ioctl( fd, SNDCTL_DSP_SPEED, &srate ); - if ( result == -1 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } + // Set the data format. + int temp = deviceFormat; + result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat ); + if ( result == -1 || deviceFormat != temp ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Verify the sample rate setup worked. - if ( abs( srate - sampleRate ) > 100 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.sampleRate = sampleRate; + // Attempt to set the buffer size. According to OSS, the minimum + // number of buffers is two. The supposed minimum buffer size is 16 + // bytes, so that will be our lower bound. The argument to this + // call is in the form 0xMMMMSSSS (hex), where the buffer size (in + // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM. + // We'll check the actual value used near the end of the setup + // procedure. + int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels; + if ( ossBufferBytes < 16 ) ossBufferBytes = 16; + int buffers = 0; + if ( options ) buffers = options->numberOfBuffers; + if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2; + if ( buffers < 2 ) buffers = 3; + temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) ); + result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.nBuffers = buffers; - if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) { - // We're doing duplex setup here. - stream_.deviceFormat[0] = stream_.deviceFormat[1]; - stream_.nDeviceChannels[0] = deviceChannels; - } + // Save buffer size (in sample frames). + *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels ); + stream_.bufferSize = *bufferSize; - // Set interleaving parameters. - stream_.userInterleaved = true; - stream_.deviceInterleaved[mode] = true; - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) - stream_.userInterleaved = false; + // Set the sample rate. + int srate = sampleRate; + result = ioctl( fd, SNDCTL_DSP_SPEED, &srate ); + if ( result == -1 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } - // Set flags for buffer conversion - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; + // Verify the sample rate setup worked. + if ( abs( srate - sampleRate ) > 100 ) { + close( fd ); + errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ")."; + errorText_ = errorStream_.str(); + return FAILURE; + } + stream_.sampleRate = sampleRate; - // Allocate the stream handles if necessary and then save. - if ( stream_.apiHandle == 0 ) { - try { - handle = new OssHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory."; - goto error; - } + if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) { + // We're doing duplex setup here. + stream_.deviceFormat[0] = stream_.deviceFormat[1]; + stream_.nDeviceChannels[0] = deviceChannels; + } - if ( pthread_cond_init( &handle->runnable, NULL ) ) { - errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable."; - goto error; - } + // Set interleaving parameters. + stream_.userInterleaved = true; + stream_.deviceInterleaved[mode] = true; + if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) + stream_.userInterleaved = false; + + // Set flags for buffer conversion + stream_.doConvertBuffer[mode] = false; + if ( stream_.userFormat != stream_.deviceFormat[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) + stream_.doConvertBuffer[mode] = true; + if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && + stream_.nUserChannels[mode] > 1 ) + stream_.doConvertBuffer[mode] = true; - stream_.apiHandle = (void *) handle; + // Allocate the stream handles if necessary and then save. + if ( stream_.apiHandle == 0 ) { + try { + handle = new OssHandle; } - else { - handle = (OssHandle *) stream_.apiHandle; + catch ( std::bad_alloc& ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory."; + goto error; } - handle->id[mode] = fd; - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory."; + if ( pthread_cond_init( &handle->runnable, NULL ) ) { + errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable."; goto error; } - if ( stream_.doConvertBuffer[mode] ) { + stream_.apiHandle = (void *) handle; + } + else { + handle = (OssHandle *) stream_.apiHandle; + } + handle->id[mode] = fd; - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } + // Allocate necessary internal buffers. + unsigned long bufferBytes; + bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); + stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); + if ( stream_.userBuffer[mode] == NULL ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + + if ( stream_.doConvertBuffer[mode] ) { + + bool makeBuffer = true; + bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); + if ( mode == INPUT ) { + if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { + unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); + if ( bufferBytes <= bytesOut ) makeBuffer = false; } + } - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } + if ( makeBuffer ) { + bufferBytes *= *bufferSize; + if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); + stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); + if ( stream_.deviceBuffer == NULL ) { + errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory."; + goto error; } } + } - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; + stream_.device[mode] = device; + stream_.state = STREAM_STOPPED; - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); + // Setup the buffer conversion information structure. + if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - // Setup thread if necessary. - if ( stream_.mode == OUTPUT && mode == INPUT ) { - // We had already set up an output stream. - stream_.mode = DUPLEX; - if ( stream_.device[0] == device ) handle->id[0] = fd; - } - else { - stream_.mode = mode; + // Setup thread if necessary. + if ( stream_.mode == OUTPUT && mode == INPUT ) { + // We had already set up an output stream. + stream_.mode = DUPLEX; + if ( stream_.device[0] == device ) handle->id[0] = fd; + } + else { + stream_.mode = mode; - // Setup callback thread. - stream_.callbackInfo.object = (void *) this; + // Setup callback thread. + stream_.callbackInfo.object = (void *) this; - // Set the thread attributes for joinable and realtime scheduling - // priority. The higher priority will only take affect if the - // program is run as root or suid. - pthread_attr_t attr; - pthread_attr_init( &attr ); - pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); + // Set the thread attributes for joinable and realtime scheduling + // priority. The higher priority will only take affect if the + // program is run as root or suid. + pthread_attr_t attr; + pthread_attr_init( &attr ); + pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); #ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) - if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { - struct sched_param param; - int priority = options->priority; - int min = sched_get_priority_min( SCHED_RR ); - int max = sched_get_priority_max( SCHED_RR ); - if ( priority < min ) priority = min; - else if ( priority > max ) priority = max; - param.sched_priority = priority; - pthread_attr_setschedparam( &attr, ¶m ); - pthread_attr_setschedpolicy( &attr, SCHED_RR ); - } - else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#else + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { + struct sched_param param; + int priority = options->priority; + int min = sched_get_priority_min( SCHED_RR ); + int max = sched_get_priority_max( SCHED_RR ); + if ( priority < min ) priority = min; + else if ( priority > max ) priority = max; + param.sched_priority = priority; + pthread_attr_setschedparam( &attr, ¶m ); + pthread_attr_setschedpolicy( &attr, SCHED_RR ); + } + else pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); #endif - stream_.callbackInfo.isRunning = true; - result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo ); - pthread_attr_destroy( &attr ); - if ( result ) { - stream_.callbackInfo.isRunning = false; - errorText_ = "RtApiOss::error creating callback thread!"; - goto error; - } + stream_.callbackInfo.isRunning = true; + result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo ); + pthread_attr_destroy( &attr ); + if ( result ) { + stream_.callbackInfo.isRunning = false; + errorText_ = "RtApiOss::error creating callback thread!"; + goto error; } + } - return SUCCESS; - - error: - if ( handle ) { - pthread_cond_destroy( &handle->runnable ); - if ( handle->id[0] ) close( handle->id[0] ); - if ( handle->id[1] ) close( handle->id[1] ); - delete handle; - stream_.apiHandle = 0; - } + return SUCCESS; - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } + error: + if ( handle ) { + pthread_cond_destroy( &handle->runnable ); + if ( handle->id[0] ) close( handle->id[0] ); + if ( handle->id[1] ) close( handle->id[1] ); + delete handle; + stream_.apiHandle = 0; + } - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; } + } - return FAILURE; + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; } - void RtApiOss :: closeStream() - { - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiOss::closeStream(): no open stream to close!"; - error( RtError::WARNING ); - return; - } + return FAILURE; +} - OssHandle *handle = (OssHandle *) stream_.apiHandle; - stream_.callbackInfo.isRunning = false; - MUTEX_LOCK( &stream_.mutex ); - if ( stream_.state == STREAM_STOPPED ) - pthread_cond_signal( &handle->runnable ); - MUTEX_UNLOCK( &stream_.mutex ); - pthread_join( stream_.callbackInfo.thread, NULL ); +void RtApiOss :: closeStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiOss::closeStream(): no open stream to close!"; + error( RtError::WARNING ); + return; + } - if ( stream_.state == STREAM_RUNNING ) { - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) - ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); - else - ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); - stream_.state = STREAM_STOPPED; - } + OssHandle *handle = (OssHandle *) stream_.apiHandle; + stream_.callbackInfo.isRunning = false; + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) + pthread_cond_signal( &handle->runnable ); + MUTEX_UNLOCK( &stream_.mutex ); + pthread_join( stream_.callbackInfo.thread, NULL ); - if ( handle ) { - pthread_cond_destroy( &handle->runnable ); - if ( handle->id[0] ) close( handle->id[0] ); - if ( handle->id[1] ) close( handle->id[1] ); - delete handle; - stream_.apiHandle = 0; - } + if ( stream_.state == STREAM_RUNNING ) { + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) + ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); + else + ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + stream_.state = STREAM_STOPPED; + } - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } + if ( handle ) { + pthread_cond_destroy( &handle->runnable ); + if ( handle->id[0] ) close( handle->id[0] ); + if ( handle->id[1] ) close( handle->id[1] ); + delete handle; + stream_.apiHandle = 0; + } - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; + for ( int i=0; i<2; i++ ) { + if ( stream_.userBuffer[i] ) { + free( stream_.userBuffer[i] ); + stream_.userBuffer[i] = 0; } + } - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; + if ( stream_.deviceBuffer ) { + free( stream_.deviceBuffer ); + stream_.deviceBuffer = 0; } - void RtApiOss :: startStream() - { - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiOss::startStream(): the stream is already running!"; - error( RtError::WARNING ); - return; - } + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; +} - MUTEX_LOCK( &stream_.mutex ); +void RtApiOss :: startStream() +{ + verifyStream(); + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiOss::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } - stream_.state = STREAM_RUNNING; + MUTEX_LOCK( &stream_.mutex ); - // No need to do anything else here ... OSS automatically starts - // when fed samples. + stream_.state = STREAM_RUNNING; - MUTEX_UNLOCK( &stream_.mutex ); + // No need to do anything else here ... OSS automatically starts + // when fed samples. - OssHandle *handle = (OssHandle *) stream_.apiHandle; - pthread_cond_signal( &handle->runnable ); - } + MUTEX_UNLOCK( &stream_.mutex ); - void RtApiOss :: stopStream() - { - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiOss::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } + OssHandle *handle = (OssHandle *) stream_.apiHandle; + pthread_cond_signal( &handle->runnable ); +} - // Change the state before the lock to improve shutdown response - // when using a callback. - stream_.state = STREAM_STOPPED; - MUTEX_LOCK( &stream_.mutex ); +void RtApiOss :: stopStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiOss::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } - int result = 0; - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + MUTEX_LOCK( &stream_.mutex ); - // Flush the output with zeros a few times. - char *buffer; - int samples; - RtAudioFormat format; + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - samples = stream_.bufferSize * stream_.nDeviceChannels[0]; - format = stream_.deviceFormat[0]; - } - else { - buffer = stream_.userBuffer[0]; - samples = stream_.bufferSize * stream_.nUserChannels[0]; - format = stream_.userFormat; - } + int result = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - memset( buffer, 0, samples * formatBytes(format) ); - for ( unsigned int i=0; iid[0], buffer, samples * formatBytes(format) ); - if ( result == -1 ) { - errorText_ = "RtApiOss::stopStream: audio write error."; - error( RtError::WARNING ); - } - } + // Flush the output with zeros a few times. + char *buffer; + int samples; + RtAudioFormat format; - result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - handle->triggered = false; + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + samples = stream_.bufferSize * stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + samples = stream_.bufferSize * stream_.nUserChannels[0]; + format = stream_.userFormat; } - if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { - result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + memset( buffer, 0, samples * formatBytes(format) ); + for ( unsigned int i=0; iid[0], buffer, samples * formatBytes(format) ); if ( result == -1 ) { - errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; + errorText_ = "RtApiOss::stopStream: audio write error."; + error( RtError::WARNING ); } } - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - - stream_.state = STREAM_STOPPED; - if ( result != -1 ) return; - error( RtError::SYSTEM_ERROR ); + result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; + } + handle->triggered = false; } - void RtApiOss :: abortStream() - { - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiOss::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { + result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; } + } - // Change the state before the lock to improve shutdown response - // when using a callback. - stream_.state = STREAM_STOPPED; - MUTEX_LOCK( &stream_.mutex ); + unlock: + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); - int result = 0; - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - handle->triggered = false; - } + if ( result != -1 ) return; + error( RtError::SYSTEM_ERROR ); +} - if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { - result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - } +void RtApiOss :: abortStream() +{ + verifyStream(); + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiOss::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } - unlock: - MUTEX_UNLOCK( &stream_.mutex ); + MUTEX_LOCK( &stream_.mutex ); - stream_.state = STREAM_STOPPED; - if ( result != -1 ) return; - error( RtError::SYSTEM_ERROR ); + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; } - void RtApiOss :: callbackEvent() - { - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_LOCK( &stream_.mutex ); - pthread_cond_wait( &handle->runnable, &stream_.mutex ); - if ( stream_.state != STREAM_RUNNING ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - MUTEX_UNLOCK( &stream_.mutex ); + int result = 0; + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; } + handle->triggered = false; + } - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); - return; + if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { + result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); + if ( result == -1 ) { + errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; + errorText_ = errorStream_.str(); + goto unlock; } + } - // Invoke user callback to get fresh output data. - int doStopStream = 0; - RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; - } - doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); - if ( doStopStream == 2 ) { - this->abortStream(); + unlock: + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); + + if ( result != -1 ) return; + error( RtError::SYSTEM_ERROR ); +} + +void RtApiOss :: callbackEvent() +{ + OssHandle *handle = (OssHandle *) stream_.apiHandle; + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + pthread_cond_wait( &handle->runnable, &stream_.mutex ); + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); return; } + MUTEX_UNLOCK( &stream_.mutex ); + } - MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!"; + error( RtError::WARNING ); + return; + } - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) goto unlock; + // Invoke user callback to get fresh output data. + int doStopStream = 0; + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + if ( stream_.mode != INPUT && handle->xrun[0] == true ) { + status |= RTAUDIO_OUTPUT_UNDERFLOW; + handle->xrun[0] = false; + } + if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { + status |= RTAUDIO_INPUT_OVERFLOW; + handle->xrun[1] = false; + } + doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); + if ( doStopStream == 2 ) { + this->abortStream(); + return; + } - int result; - char *buffer; - int samples; - RtAudioFormat format; + MUTEX_LOCK( &stream_.mutex ); - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { + // The state might change while waiting on a mutex. + if ( stream_.state == STREAM_STOPPED ) goto unlock; - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - samples = stream_.bufferSize * stream_.nDeviceChannels[0]; - format = stream_.deviceFormat[0]; - } - else { - buffer = stream_.userBuffer[0]; - samples = stream_.bufferSize * stream_.nUserChannels[0]; - format = stream_.userFormat; - } + int result; + char *buffer; + int samples; + RtAudioFormat format; - // Do byte swapping if necessary. - if ( stream_.doByteSwap[0] ) - byteSwapBuffer( buffer, samples, format ); - - if ( stream_.mode == DUPLEX && handle->triggered == false ) { - int trig = 0; - ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); - result = write( handle->id[0], buffer, samples * formatBytes(format) ); - trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT; - ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); - handle->triggered = true; - } - else - // Write samples to device. - result = write( handle->id[0], buffer, samples * formatBytes(format) ); + if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( result == -1 ) { - // We'll assume this is an underrun, though there isn't a - // specific means for determining that. - handle->xrun[0] = true; - errorText_ = "RtApiOss::callbackEvent: audio write error."; - error( RtError::WARNING ); - // Continue on to input section. - } + // Setup parameters and do buffer conversion if necessary. + if ( stream_.doConvertBuffer[0] ) { + buffer = stream_.deviceBuffer; + convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); + samples = stream_.bufferSize * stream_.nDeviceChannels[0]; + format = stream_.deviceFormat[0]; + } + else { + buffer = stream_.userBuffer[0]; + samples = stream_.bufferSize * stream_.nUserChannels[0]; + format = stream_.userFormat; } - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { + // Do byte swapping if necessary. + if ( stream_.doByteSwap[0] ) + byteSwapBuffer( buffer, samples, format ); - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - samples = stream_.bufferSize * stream_.nDeviceChannels[1]; - format = stream_.deviceFormat[1]; - } - else { - buffer = stream_.userBuffer[1]; - samples = stream_.bufferSize * stream_.nUserChannels[1]; - format = stream_.userFormat; - } + if ( stream_.mode == DUPLEX && handle->triggered == false ) { + int trig = 0; + ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); + result = write( handle->id[0], buffer, samples * formatBytes(format) ); + trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT; + ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); + handle->triggered = true; + } + else + // Write samples to device. + result = write( handle->id[0], buffer, samples * formatBytes(format) ); + + if ( result == -1 ) { + // We'll assume this is an underrun, though there isn't a + // specific means for determining that. + handle->xrun[0] = true; + errorText_ = "RtApiOss::callbackEvent: audio write error."; + error( RtError::WARNING ); + // Continue on to input section. + } + } - // Read samples from device. - result = read( handle->id[1], buffer, samples * formatBytes(format) ); + if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - if ( result == -1 ) { - // We'll assume this is an overrun, though there isn't a - // specific means for determining that. - handle->xrun[1] = true; - errorText_ = "RtApiOss::callbackEvent: audio read error."; - error( RtError::WARNING ); - goto unlock; - } + // Setup parameters. + if ( stream_.doConvertBuffer[1] ) { + buffer = stream_.deviceBuffer; + samples = stream_.bufferSize * stream_.nDeviceChannels[1]; + format = stream_.deviceFormat[1]; + } + else { + buffer = stream_.userBuffer[1]; + samples = stream_.bufferSize * stream_.nUserChannels[1]; + format = stream_.userFormat; + } - // Do byte swapping if necessary. - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( buffer, samples, format ); + // Read samples from device. + result = read( handle->id[1], buffer, samples * formatBytes(format) ); - // Do buffer conversion if necessary. - if ( stream_.doConvertBuffer[1] ) - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); + if ( result == -1 ) { + // We'll assume this is an overrun, though there isn't a + // specific means for determining that. + handle->xrun[1] = true; + errorText_ = "RtApiOss::callbackEvent: audio read error."; + error( RtError::WARNING ); + goto unlock; } - unlock: - MUTEX_UNLOCK( &stream_.mutex ); + // Do byte swapping if necessary. + if ( stream_.doByteSwap[1] ) + byteSwapBuffer( buffer, samples, format ); - RtApi::tickStreamTime(); - if ( doStopStream == 1 ) this->stopStream(); + // Do buffer conversion if necessary. + if ( stream_.doConvertBuffer[1] ) + convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); } - extern "C" void *ossCallbackHandler( void *ptr ) - { - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiOss *object = (RtApiOss *) info->object; - bool *isRunning = &info->isRunning; + unlock: + MUTEX_UNLOCK( &stream_.mutex ); - while ( *isRunning == true ) { - pthread_testcancel(); - object->callbackEvent(); - } + RtApi::tickStreamTime(); + if ( doStopStream == 1 ) this->stopStream(); +} + +extern "C" void *ossCallbackHandler( void *ptr ) +{ + CallbackInfo *info = (CallbackInfo *) ptr; + RtApiOss *object = (RtApiOss *) info->object; + bool *isRunning = &info->isRunning; - pthread_exit( NULL ); + while ( *isRunning == true ) { + pthread_testcancel(); + object->callbackEvent(); } - //******************** End of __LINUX_OSS__ *********************// + pthread_exit( NULL ); +} + +//******************** End of __LINUX_OSS__ *********************// #endif - // *************************************************** // - // - // Protected common (OS-independent) RtAudio methods. - // - // *************************************************** // +// *************************************************** // +// +// Protected common (OS-independent) RtAudio methods. +// +// *************************************************** // - // This method can be modified to control the behavior of error - // message printing. - void RtApi :: error( RtError::Type type ) - { - errorStream_.str(""); // clear the ostringstream - if ( type == RtError::WARNING && showWarnings_ == true ) - std::cerr << '\n' << errorText_ << "\n\n"; - else - throw( RtError( errorText_, type ) ); - } +// This method can be modified to control the behavior of error +// message printing. +void RtApi :: error( RtError::Type type ) +{ + errorStream_.str(""); // clear the ostringstream + if ( type == RtError::WARNING && showWarnings_ == true ) + std::cerr << '\n' << errorText_ << "\n\n"; + else + throw( RtError( errorText_, type ) ); +} - void RtApi :: verifyStream() - { - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApi:: a stream is not open!"; - error( RtError::INVALID_USE ); - } +void RtApi :: verifyStream() +{ + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApi:: a stream is not open!"; + error( RtError::INVALID_USE ); } +} - void RtApi :: clearStreamInfo() - { - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; - stream_.sampleRate = 0; - stream_.bufferSize = 0; - stream_.nBuffers = 0; - stream_.userFormat = 0; - stream_.userInterleaved = true; - stream_.streamTime = 0.0; - stream_.apiHandle = 0; - stream_.deviceBuffer = 0; - stream_.callbackInfo.callback = 0; - stream_.callbackInfo.userData = 0; - stream_.callbackInfo.isRunning = false; - for ( int i=0; i<2; i++ ) { - stream_.device[i] = 11111; - stream_.doConvertBuffer[i] = false; - stream_.deviceInterleaved[i] = true; - stream_.doByteSwap[i] = false; - stream_.nUserChannels[i] = 0; - stream_.nDeviceChannels[i] = 0; - stream_.channelOffset[i] = 0; - stream_.deviceFormat[i] = 0; - stream_.latency[i] = 0; - stream_.userBuffer[i] = 0; - stream_.convertInfo[i].channels = 0; - stream_.convertInfo[i].inJump = 0; - stream_.convertInfo[i].outJump = 0; - stream_.convertInfo[i].inFormat = 0; - stream_.convertInfo[i].outFormat = 0; - stream_.convertInfo[i].inOffset.clear(); - stream_.convertInfo[i].outOffset.clear(); - } +void RtApi :: clearStreamInfo() +{ + stream_.mode = UNINITIALIZED; + stream_.state = STREAM_CLOSED; + stream_.sampleRate = 0; + stream_.bufferSize = 0; + stream_.nBuffers = 0; + stream_.userFormat = 0; + stream_.userInterleaved = true; + stream_.streamTime = 0.0; + stream_.apiHandle = 0; + stream_.deviceBuffer = 0; + stream_.callbackInfo.callback = 0; + stream_.callbackInfo.userData = 0; + stream_.callbackInfo.isRunning = false; + for ( int i=0; i<2; i++ ) { + stream_.device[i] = 11111; + stream_.doConvertBuffer[i] = false; + stream_.deviceInterleaved[i] = true; + stream_.doByteSwap[i] = false; + stream_.nUserChannels[i] = 0; + stream_.nDeviceChannels[i] = 0; + stream_.channelOffset[i] = 0; + stream_.deviceFormat[i] = 0; + stream_.latency[i] = 0; + stream_.userBuffer[i] = 0; + stream_.convertInfo[i].channels = 0; + stream_.convertInfo[i].inJump = 0; + stream_.convertInfo[i].outJump = 0; + stream_.convertInfo[i].inFormat = 0; + stream_.convertInfo[i].outFormat = 0; + stream_.convertInfo[i].inOffset.clear(); + stream_.convertInfo[i].outOffset.clear(); } +} - unsigned int RtApi :: formatBytes( RtAudioFormat format ) - { - if ( format == RTAUDIO_SINT16 ) - return 2; - else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 || - format == RTAUDIO_FLOAT32 ) - return 4; - else if ( format == RTAUDIO_FLOAT64 ) - return 8; - else if ( format == RTAUDIO_SINT8 ) - return 1; - - errorText_ = "RtApi::formatBytes: undefined format."; - error( RtError::WARNING ); +unsigned int RtApi :: formatBytes( RtAudioFormat format ) +{ + if ( format == RTAUDIO_SINT16 ) + return 2; + else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 || + format == RTAUDIO_FLOAT32 ) + return 4; + else if ( format == RTAUDIO_FLOAT64 ) + return 8; + else if ( format == RTAUDIO_SINT8 ) + return 1; + + errorText_ = "RtApi::formatBytes: undefined format."; + error( RtError::WARNING ); - return 0; + return 0; +} + +void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel ) +{ + if ( mode == INPUT ) { // convert device to user buffer + stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1]; + stream_.convertInfo[mode].outJump = stream_.nUserChannels[1]; + stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1]; + stream_.convertInfo[mode].outFormat = stream_.userFormat; + } + else { // convert user to device buffer + stream_.convertInfo[mode].inJump = stream_.nUserChannels[0]; + stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0]; + stream_.convertInfo[mode].inFormat = stream_.userFormat; + stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0]; } - void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel ) - { - if ( mode == INPUT ) { // convert device to user buffer - stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1]; - stream_.convertInfo[mode].outJump = stream_.nUserChannels[1]; - stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1]; - stream_.convertInfo[mode].outFormat = stream_.userFormat; + if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump ) + stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump; + else + stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump; + + // Set up the interleave/deinterleave offsets. + if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) { + if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) || + ( mode == INPUT && stream_.userInterleaved ) ) { + for ( int k=0; k 0 ) { + if ( stream_.deviceInterleaved[mode] ) { + if ( mode == OUTPUT ) { + for ( int k=0; k 0 ) { - if ( stream_.deviceInterleaved[mode] ) { - if ( mode == OUTPUT ) { - for ( int k=0; k>= 8; - } - in += info.inJump; - out += info.outJump; + } + else if (info.inFormat == RTAUDIO_SINT32) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i>= 8; } + in += info.inJump; + out += info.outJump; } - else if (info.inFormat == RTAUDIO_FLOAT32) { - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i> 8) & 0x0000ffff); - } - in += info.inJump; - out += info.outJump; + } + else if (info.inFormat == RTAUDIO_SINT24) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 8) & 0x0000ffff); } + in += info.inJump; + out += info.outJump; } - else if (info.inFormat == RTAUDIO_SINT32) { - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i> 16) & 0x0000ffff); - } - in += info.inJump; - out += info.outJump; + } + else if (info.inFormat == RTAUDIO_SINT32) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 16) & 0x0000ffff); } + in += info.inJump; + out += info.outJump; } - else if (info.inFormat == RTAUDIO_FLOAT32) { - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i> 8) & 0x00ff); - } - in += info.inJump; - out += info.outJump; + } + if (info.inFormat == RTAUDIO_SINT16) { + Int16 *in = (Int16 *)inBuffer; + for (unsigned int i=0; i> 8) & 0x00ff); } + in += info.inJump; + out += info.outJump; } - else if (info.inFormat == RTAUDIO_SINT24) { - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i> 16) & 0x000000ff); - } - in += info.inJump; - out += info.outJump; + } + else if (info.inFormat == RTAUDIO_SINT24) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 16) & 0x000000ff); } + in += info.inJump; + out += info.outJump; } - else if (info.inFormat == RTAUDIO_SINT32) { - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i> 24) & 0x000000ff); - } - in += info.inJump; - out += info.outJump; + } + else if (info.inFormat == RTAUDIO_SINT32) { + Int32 *in = (Int32 *)inBuffer; + for (unsigned int i=0; i> 24) & 0x000000ff); } + in += info.inJump; + out += info.outJump; } - else if (info.inFormat == RTAUDIO_FLOAT32) { - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i>8) | (x<<8); } //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); } //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); } - void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ) - { - register char val; - register char *ptr; - - ptr = buffer; - if ( format == RTAUDIO_SINT16 ) { - for ( unsigned int i=0; i