X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=RtAudio.cpp;h=18d1fd0af29b264fe835aa78649b09e9af5324fa;hb=HEAD;hp=7c87572fd79964819720eab69ccbce4cbaf1dd45;hpb=79c306a90943f00bd29a725fc25d3ee3fb3f0d34;p=rtaudio-cdist.git diff --git a/RtAudio.cpp b/RtAudio.cpp index 7c87572..18d1fd0 100644 --- a/RtAudio.cpp +++ b/RtAudio.cpp @@ -47,6 +47,7 @@ #include #include #include +#include // Static variable definitions. const unsigned int RtApi::MAX_SAMPLE_RATES = 14; @@ -113,7 +114,7 @@ const char* rtaudio_api_names[][2] = { { "ds" , "DirectSound" }, { "dummy" , "Dummy" }, }; -const unsigned int rtaudio_num_api_names = +const unsigned int rtaudio_num_api_names = sizeof(rtaudio_api_names)/sizeof(rtaudio_api_names[0]); // The order here will control the order of RtAudio's API search in @@ -465,14 +466,14 @@ double RtApi :: getStreamTime( void ) struct timeval then; struct timeval now; - if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 ) + if ( stream_.state != STREAM_RUNNING || (stream_.lastTickTimestamp.tv_sec == 0 && stream_.lastTickTimestamp.tv_usec == 0) ) return stream_.streamTime; gettimeofday( &now, NULL ); then = stream_.lastTickTimestamp; return stream_.streamTime + ((now.tv_sec + 0.000001 * now.tv_usec) - - (then.tv_sec + 0.000001 * then.tv_usec)); + (then.tv_sec + 0.000001 * then.tv_usec)); #else return stream_.streamTime; #endif @@ -496,6 +497,14 @@ unsigned int RtApi :: getStreamSampleRate( void ) return stream_.sampleRate; } +void RtApi :: startStream( void ) +{ +#if defined( HAVE_GETTIMEOFDAY ) + stream_.lastTickTimestamp.tv_sec = 0; + stream_.lastTickTimestamp.tv_usec = 0; +#endif +} + // *************************************************** // // @@ -1535,12 +1544,17 @@ void RtApiCore :: closeStream( void ) void RtApiCore :: startStream( void ) { verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiCore::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); return; } + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + OSStatus result = noErr; CoreHandle *handle = (CoreHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { @@ -1889,7 +1903,7 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, channelsLeft -= streamChannels; } } - + if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, @@ -2493,12 +2507,17 @@ void RtApiJack :: closeStream( void ) void RtApiJack :: startStream( void ) { verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiJack::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); return; } + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + JackHandle *handle = (JackHandle *) stream_.apiHandle; int result = jack_activate( handle->client ); if ( result ) { @@ -2782,7 +2801,7 @@ RtApiAsio :: RtApiAsio() // CoInitialize beforehand, but it must be for appartment threading // (in which case, CoInitilialize will return S_FALSE here). coInitialized_ = false; - HRESULT hr = CoInitialize( NULL ); + HRESULT hr = CoInitialize( NULL ); if ( FAILED(hr) ) { errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"; error( RtAudioError::WARNING ); @@ -3233,7 +3252,7 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne errorText_ = errorStream_.str(); goto error; } - buffersAllocated = true; + buffersAllocated = true; stream_.state = STREAM_STOPPED; // Set flags for buffer conversion. @@ -3372,12 +3391,17 @@ bool stopThreadCalled = false; void RtApiAsio :: startStream() { verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiAsio::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); return; } + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; ASIOError result = ASIOStart(); if ( result != ASE_OK ) { @@ -3707,13 +3731,13 @@ static long asioMessages( long selector, long value, void* /*message*/, double* static const char* getAsioErrorString( ASIOError result ) { - struct Messages + struct Messages { ASIOError value; const char*message; }; - static const Messages m[] = + static const Messages m[] = { { ASE_NotPresent, "Hardware input or output is not present or available." }, { ASE_HWMalfunction, "Hardware is malfunctioning." }, @@ -3829,8 +3853,9 @@ public: relOutIndex += bufferSize_; } - // "in" index can end on the "out" index but cannot begin at it - if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) { + // the "IN" index CAN BEGIN at the "OUT" index + // the "IN" index CANNOT END at the "OUT" index + if ( inIndex_ < relOutIndex && inIndexEnd >= relOutIndex ) { return false; // not enough space between "in" index and "out" index } @@ -3890,8 +3915,9 @@ public: relInIndex += bufferSize_; } - // "out" index can begin at and end on the "in" index - if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) { + // the "OUT" index CANNOT BEGIN at the "IN" index + // the "OUT" index CAN END at the "IN" index + if ( outIndex_ <= relInIndex && outIndexEnd > relInIndex ) { return false; // not enough space between "out" index and "in" index } @@ -3946,7 +3972,7 @@ private: // In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate // between HW and the user. The WasapiResampler class is used to perform this conversion between -// HwIn->UserIn and UserOut->HwOut during the stream callback loop. +// HwIn->UserIn and UserO ut->HwOut during the stream callback loop. class WasapiResampler { public: @@ -4388,7 +4414,9 @@ RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device ) hr = audioClient->GetMixFormat( &deviceFormat ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format."; + char error[256]; + snprintf(error, sizeof(error), "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format (%d)", hr); + errorText_ = error; goto Exit; } @@ -4544,6 +4572,7 @@ void RtApiWasapi::closeStream( void ) void RtApiWasapi::startStream( void ) { verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiWasapi::startStream: The stream is already running."; @@ -4551,6 +4580,10 @@ void RtApiWasapi::startStream( void ) return; } + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + // update stream state stream_.state = STREAM_RUNNING; @@ -4590,26 +4623,6 @@ void RtApiWasapi::stopStream( void ) // Wait for the last buffer to play before stopping. Sleep( 1000 * stream_.bufferSize / stream_.sampleRate ); - // stop capture client if applicable - if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) { - HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop(); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream."; - error( RtAudioError::DRIVER_ERROR ); - return; - } - } - - // stop render client if applicable - if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) { - HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop(); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream."; - error( RtAudioError::DRIVER_ERROR ); - return; - } - } - // close thread handle if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) { errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread."; @@ -4640,26 +4653,6 @@ void RtApiWasapi::abortStream( void ) Sleep( 1 ); } - // stop capture client if applicable - if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) { - HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop(); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream."; - error( RtAudioError::DRIVER_ERROR ); - return; - } - } - - // stop render client if applicable - if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) { - HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop(); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream."; - error( RtAudioError::DRIVER_ERROR ); - return; - } - } - // close thread handle if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) { errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread."; @@ -4860,8 +4853,7 @@ bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigne stream_.doConvertBuffer[mode] = false; if ( stream_.userFormat != stream_.deviceFormat[mode] || stream_.nUserChannels[0] != stream_.nDeviceChannels[0] || - stream_.nUserChannels[1] != stream_.nDeviceChannels[1] || - stream_.userInterleaved ) + stream_.nUserChannels[1] != stream_.nDeviceChannels[1] ) stream_.doConvertBuffer[mode] = true; else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && stream_.nUserChannels[mode] > 1 ) @@ -4982,7 +4974,8 @@ void RtApiWasapi::wasapiThread() HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" ); if ( AvrtDll ) { DWORD taskIndex = 0; - TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" ); + TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = + ( TAvSetMmThreadCharacteristicsPtr ) (void(*)()) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" ); AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex ); FreeLibrary( AvrtDll ); } @@ -5042,6 +5035,20 @@ void RtApiWasapi::wasapiThread() } ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient; + + // reset the capture stream + hr = captureAudioClient->Reset(); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to reset capture stream."; + goto Exit; + } + + // start the capture stream + hr = captureAudioClient->Start(); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to start capture stream."; + goto Exit; + } } unsigned int inBufferSize = 0; @@ -5057,20 +5064,6 @@ void RtApiWasapi::wasapiThread() // set captureBuffer size captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) ); - - // reset the capture stream - hr = captureAudioClient->Reset(); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to reset capture stream."; - goto Exit; - } - - // start the capture stream - hr = captureAudioClient->Start(); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to start capture stream."; - goto Exit; - } } // start render stream if applicable @@ -5123,6 +5116,20 @@ void RtApiWasapi::wasapiThread() ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient; ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent; + + // reset the render stream + hr = renderAudioClient->Reset(); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to reset render stream."; + goto Exit; + } + + // start the render stream + hr = renderAudioClient->Start(); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to start render stream."; + goto Exit; + } } unsigned int outBufferSize = 0; @@ -5138,20 +5145,6 @@ void RtApiWasapi::wasapiThread() // set renderBuffer size renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) ); - - // reset the render stream - hr = renderAudioClient->Reset(); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to reset render stream."; - goto Exit; - } - - // start the render stream - hr = renderAudioClient->Start(); - if ( FAILED( hr ) ) { - errorText = "RtApiWasapi::wasapiThread: Unable to start render stream."; - goto Exit; - } } // malloc buffer memory @@ -5175,8 +5168,8 @@ void RtApiWasapi::wasapiThread() } convBuffSize *= 2; // allow overflow for *SrRatio remainders - convBuffer = ( char* ) malloc( convBuffSize ); - stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize ); + convBuffer = ( char* ) calloc( convBuffSize, 1 ); + stream_.deviceBuffer = ( char* ) calloc( deviceBuffSize, 1 ); if ( !convBuffer || !stream_.deviceBuffer ) { errorType = RtAudioError::MEMORY_ERROR; errorText = "RtApiWasapi::wasapiThread: Error allocating device buffer memory."; @@ -5263,6 +5256,9 @@ void RtApiWasapi::wasapiThread() captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0, stream_.callbackInfo.userData ); + // tick stream time + RtApi::tickStreamTime(); + // Handle return value from callback if ( callbackResult == 1 ) { // instantiate a thread to stop this thread @@ -5318,6 +5314,12 @@ void RtApiWasapi::wasapiThread() stream_.convertInfo[OUTPUT] ); } + else { + // no further conversion, simple copy userBuffer to deviceBuffer + memcpy( stream_.deviceBuffer, + stream_.userBuffer[OUTPUT], + stream_.bufferSize * stream_.nUserChannels[OUTPUT] * formatBytes( stream_.userFormat ) ); + } // Convert callback buffer to stream sample rate renderResampler->Convert( convBuffer, @@ -5465,9 +5467,6 @@ void RtApiWasapi::wasapiThread() // unsetting the callbackPulled flag lets the stream know that // the audio device is ready for another callback output buffer. callbackPulled = false; - - // tick stream time - RtApi::tickStreamTime(); } } @@ -5500,7 +5499,7 @@ Exit: #if defined(__WINDOWS_DS__) // Windows DirectSound API // Modified by Robin Davies, October 2005 -// - Improvements to DirectX pointer chasing. +// - Improvements to DirectX pointer chasing. // - 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 @@ -5544,7 +5543,7 @@ struct DsHandle { void *id[2]; void *buffer[2]; bool xrun[2]; - UINT bufferPointer[2]; + UINT bufferPointer[2]; DWORD dsBufferSize[2]; DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by. HANDLE condition; @@ -6381,18 +6380,23 @@ void RtApiDs :: closeStream() void RtApiDs :: startStream() { verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiDs::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); return; } + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + DsHandle *handle = (DsHandle *) stream_.apiHandle; // Increase scheduler frequency on lesser windows (a side-effect of // increasing timer accuracy). On greater windows (Win2K or later), // this is already in effect. - timeBeginPeriod( 1 ); + timeBeginPeriod( 1 ); buffersRolling = false; duplexPrerollBytes = 0; @@ -6437,6 +6441,7 @@ void RtApiDs :: startStream() void RtApiDs :: stopStream() { verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; error( RtAudioError::WARNING ); @@ -6713,7 +6718,7 @@ void RtApiDs :: callbackEvent() } if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - + LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; if ( handle->drainCounter > 1 ) { // write zeros to the output stream @@ -6780,7 +6785,7 @@ void RtApiDs :: callbackEvent() } if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize ) - || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) { + || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) { // We've strayed into the forbidden zone ... resync the read pointer. handle->xrun[0] = true; nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes; @@ -6854,14 +6859,14 @@ void RtApiDs :: callbackEvent() if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset DWORD endRead = nextReadPointer + bufferBytes; - // Handling depends on whether we are INPUT or DUPLEX. + // 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 + // + // 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 @@ -6912,7 +6917,7 @@ void RtApiDs :: callbackEvent() error( RtAudioError::SYSTEM_ERROR ); return; } - + if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset } } @@ -7154,7 +7159,7 @@ unsigned int RtApiAlsa :: getDeviceCount( void ) unsigned nDevices = 0; int result, subdevice, card; char name[64]; - snd_ctl_t *handle; + snd_ctl_t *handle = 0; // Count cards and devices card = -1; @@ -7163,6 +7168,7 @@ unsigned int RtApiAlsa :: getDeviceCount( void ) sprintf( name, "hw:%d", card ); result = snd_ctl_open( &handle, name, 0 ); if ( result < 0 ) { + handle = 0; errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); @@ -7182,7 +7188,8 @@ unsigned int RtApiAlsa :: getDeviceCount( void ) nDevices++; } nextcard: - snd_ctl_close( handle ); + if ( handle ) + snd_ctl_close( handle ); snd_card_next( &card ); } @@ -7203,7 +7210,7 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) unsigned nDevices = 0; int result, subdevice, card; char name[64]; - snd_ctl_t *chandle; + snd_ctl_t *chandle = 0; // Count cards and devices card = -1; @@ -7213,6 +7220,7 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) sprintf( name, "hw:%d", card ); result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); if ( result < 0 ) { + chandle = 0; errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << "."; errorText_ = errorStream_.str(); error( RtAudioError::WARNING ); @@ -7235,7 +7243,8 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) nDevices++; } nextcard: - snd_ctl_close( chandle ); + if ( chandle ) + snd_ctl_close( chandle ); snd_card_next( &card ); } @@ -8071,6 +8080,7 @@ void RtApiAlsa :: startStream() // This method calls snd_pcm_prepare if the device isn't already in that state. verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiAlsa::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); @@ -8079,6 +8089,10 @@ void RtApiAlsa :: startStream() MUTEX_LOCK( &stream_.mutex ); + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + int result = 0; snd_pcm_state_t state; AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; @@ -8135,7 +8149,7 @@ void RtApiAlsa :: stopStream() AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( apiInfo->synchronized ) + if ( apiInfo->synchronized ) result = snd_pcm_drop( handle[0] ); else result = snd_pcm_drain( handle[0] ); @@ -8400,8 +8414,8 @@ static void *alsaCallbackHandler( void *ptr ) #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) if ( info->doRealtime ) { - std::cerr << "RtAudio alsa: " << - (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") << + std::cerr << "RtAudio alsa: " << + (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") << "running realtime scheduling" << std::endl; } #endif @@ -8424,6 +8438,7 @@ static void *alsaCallbackHandler( void *ptr ) #include #include +#include #include static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000, @@ -8460,8 +8475,33 @@ unsigned int RtApiPulse::getDeviceCount( void ) return 1; } +void RtApiPulse::sinkInfoCallback(pa_context*, const pa_sink_info* info, int, void* arg) +{ + RtApiPulse* api = (RtApiPulse *) arg; + if (info) { + api->channels_ = info->sample_spec.channels; + } + pa_threaded_mainloop_signal(api->mainloop_, 0); +} + +void RtApiPulse::contextStateCallback(pa_context* c, void* arg) +{ + pa_threaded_mainloop* mainloop = (pa_threaded_mainloop*) arg; + + switch (pa_context_get_state(c)) { + case PA_CONTEXT_READY: + case PA_CONTEXT_TERMINATED: + case PA_CONTEXT_FAILED: + pa_threaded_mainloop_signal(mainloop, 0); + break; + default: + break; + } +} + RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ ) { + /* Set up some defaults in case we crash and burn */ RtAudio::DeviceInfo info; info.probed = true; info.name = "PulseAudio"; @@ -8477,6 +8517,72 @@ RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ ) info.preferredSampleRate = 48000; info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32; + /* Get the number of output channels from pulseaudio. A simple task, you say? + "What is your mainloop?" */ + mainloop_ = pa_threaded_mainloop_new(); + if (!mainloop_) { + return info; + } + + pa_threaded_mainloop_start(mainloop_); + pa_threaded_mainloop_lock(mainloop_); + + /* "And what is your context?" */ + pa_context* context = pa_context_new(pa_threaded_mainloop_get_api(mainloop_), "RtAudio"); + if (!context) { + pa_threaded_mainloop_unlock(mainloop_); + pa_threaded_mainloop_stop(mainloop_); + pa_threaded_mainloop_free(mainloop_); + mainloop_ = 0; + return info; + } + + pa_context_set_state_callback(context, contextStateCallback, mainloop_); + + pa_context_connect(context, 0, (pa_context_flags_t) 0, 0); + + /* "And what is your favourite colour?" */ + int connected = 0; + pa_context_state_t state = pa_context_get_state(context); + for (; !connected; state = pa_context_get_state(context)) { + switch (state) { + case PA_CONTEXT_READY: + connected = 1; + continue; + case PA_CONTEXT_FAILED: + case PA_CONTEXT_TERMINATED: + /* Blue! No, I mean red! */ + pa_threaded_mainloop_unlock(mainloop_); + pa_context_disconnect(context); + pa_context_unref(context); + pa_threaded_mainloop_stop(mainloop_); + pa_threaded_mainloop_free(mainloop_); + mainloop_ = 0; + return info; + default: + pa_threaded_mainloop_wait(mainloop_); + break; + } + } + + pa_operation* op = pa_context_get_sink_info_by_index(context, 0, sinkInfoCallback, this); + + if (op) { + pa_operation_unref(op); + } + + pa_threaded_mainloop_wait(mainloop_); + pa_threaded_mainloop_unlock(mainloop_); + + pa_context_disconnect(context); + pa_context_unref(context); + + pa_threaded_mainloop_stop(mainloop_); + pa_threaded_mainloop_free(mainloop_); + mainloop_ = 0; + + info.outputChannels = channels_; + return info; } @@ -8485,15 +8591,15 @@ static void *pulseaudio_callback( void * user ) CallbackInfo *cbi = static_cast( user ); RtApiPulse *context = static_cast( cbi->object ); volatile bool *isRunning = &cbi->isRunning; - + #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) if (cbi->doRealtime) { - std::cerr << "RtAudio pulse: " << - (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") << + std::cerr << "RtAudio pulse: " << + (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") << "running realtime scheduling" << std::endl; } #endif - + while ( *isRunning ) { pthread_testcancel(); context->callbackEvent(); @@ -8611,7 +8717,7 @@ void RtApiPulse::callbackEvent( void ) else bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize * formatBytes( stream_.userFormat ); - + if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) { errorStream_ << "RtApiPulse::callbackEvent: audio read error, " << pa_strerror( pa_error ) << "."; @@ -8629,12 +8735,21 @@ void RtApiPulse::callbackEvent( void ) MUTEX_UNLOCK( &stream_.mutex ); RtApi::tickStreamTime(); + if (pah->s_play) { + int e = 0; + pa_usec_t const lat = pa_simple_get_latency(pah->s_play, &e); + if (e == 0) { + stream_.latency[0] = lat * stream_.sampleRate / 1000000; + } + } + if ( doStopStream == 1 ) stopStream(); } void RtApiPulse::startStream( void ) { + RtApi::startStream(); PulseAudioHandle *pah = static_cast( stream_.apiHandle ); if ( stream_.state == STREAM_CLOSED ) { @@ -8650,6 +8765,10 @@ void RtApiPulse::startStream( void ) MUTEX_LOCK( &stream_.mutex ); + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + stream_.state = STREAM_RUNNING; pah->runnable = true; @@ -8673,6 +8792,7 @@ void RtApiPulse::stopStream( void ) } stream_.state = STREAM_STOPPED; + pah->runnable = false; MUTEX_LOCK( &stream_.mutex ); if ( pah && pah->s_play ) { @@ -8707,6 +8827,7 @@ void RtApiPulse::abortStream( void ) } stream_.state = STREAM_STOPPED; + pah->runnable = false; MUTEX_LOCK( &stream_.mutex ); if ( pah && pah->s_play ) { @@ -8736,10 +8857,6 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, if ( device != 0 ) return false; if ( mode != INPUT && mode != OUTPUT ) return false; - if ( channels != 1 && channels != 2 ) { - errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels."; - return false; - } ss.channels = channels; if ( firstChannel != 0 ) return false; @@ -8859,7 +8976,38 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, } break; case OUTPUT: - pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error ); + /* XXX: hard-coded for DCP-o-matic */ + pa_channel_map map; + pa_channel_map_init(&map); + /* XXX: need to check 7.1 */ + map.channels = channels; + + if (channels > 0) { + map.map[0] = PA_CHANNEL_POSITION_FRONT_LEFT; + } + if (channels > 1) { + map.map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT; + } + if (channels > 2) { + map.map[2] = PA_CHANNEL_POSITION_FRONT_CENTER; + } + if (channels > 3) { + map.map[3] = PA_CHANNEL_POSITION_LFE; + } + if (channels > 4) { + map.map[4] = PA_CHANNEL_POSITION_REAR_LEFT; + } + if (channels > 5) { + map.map[5] = PA_CHANNEL_POSITION_REAR_RIGHT; + } + if (channels > 6) { + map.map[6] = PA_CHANNEL_POSITION_SIDE_LEFT; + } + if (channels > 7) { + map.map[7] = PA_CHANNEL_POSITION_SIDE_RIGHT; + } + + pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, &map, NULL, &error ); if ( !pah->s_play ) { errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server."; goto error; @@ -8878,7 +9026,7 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, if ( !stream_.callbackInfo.isRunning ) { stream_.callbackInfo.object = this; - + stream_.state = STREAM_STOPPED; // Set the thread attributes for joinable and realtime scheduling // priority (optional). The higher priority will only take affect @@ -8899,7 +9047,7 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, if ( priority < min ) priority = min; else if ( priority > max ) priority = max; param.sched_priority = priority; - + // Set the policy BEFORE the priority. Otherwise it fails. pthread_attr_setschedpolicy(&attr, SCHED_RR); pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM); @@ -8928,7 +9076,7 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, } return SUCCESS; - + error: if ( pah && stream_.callbackInfo.isRunning ) { pthread_cond_destroy( &pah->runnable_cv ); @@ -9520,7 +9668,7 @@ bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned if ( priority < min ) priority = min; else if ( priority > max ) priority = max; param.sched_priority = priority; - + // Set the policy BEFORE the priority. Otherwise it fails. pthread_attr_setschedpolicy(&attr, SCHED_RR); pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM); @@ -9626,6 +9774,7 @@ void RtApiOss :: closeStream() void RtApiOss :: startStream() { verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiOss::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); @@ -9634,6 +9783,10 @@ void RtApiOss :: startStream() MUTEX_LOCK( &stream_.mutex ); + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + stream_.state = STREAM_RUNNING; // No need to do anything else here ... OSS automatically starts @@ -9903,8 +10056,8 @@ static void *ossCallbackHandler( void *ptr ) #ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) if (info->doRealtime) { - std::cerr << "RtAudio oss: " << - (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") << + std::cerr << "RtAudio oss: " << + (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") << "running realtime scheduling" << std::endl; } #endif @@ -10625,4 +10778,3 @@ void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat // End: // // vim: et sts=2 sw=2 -