From 5d8514d7eb3918a947ec97b45f4105630c64468d Mon Sep 17 00:00:00 2001 From: Gary Scavone Date: Wed, 13 Jun 2012 20:27:59 +0000 Subject: [PATCH] Mutex removal from several APIs, addition of PulseAudio support, documentation updates for 4.0.11 release. --- RtAudio.cpp | 597 ++++++-- RtAudio.h | 37 +- configure.ac | 6 + doc/doxygen/Doxyfile | 2 +- doc/doxygen/acknowledge.txt | 9 + doc/doxygen/apinotes.txt | 4 +- doc/doxygen/compiling.txt | 8 + doc/doxygen/footer.html | 2 +- doc/doxygen/license.txt | 2 +- doc/doxygen/tutorial.txt | 4 +- doc/release.txt | 5 +- include/soundcard.h | 2735 ++++++++++++++++------------------- install | 3 +- readme | 2 +- tests/duplex.cpp | 4 +- tests/playsaw.cpp | 2 +- tests/testall.cpp | 2 +- tests/teststops.cpp | 2 +- 18 files changed, 1808 insertions(+), 1618 deletions(-) diff --git a/RtAudio.cpp b/RtAudio.cpp index 47e5804..823faaf 100644 --- a/RtAudio.cpp +++ b/RtAudio.cpp @@ -58,7 +58,7 @@ const unsigned int RtApi::SAMPLE_RATES[] = { #define MUTEX_DESTROY(A) DeleteCriticalSection(A) #define MUTEX_LOCK(A) EnterCriticalSection(A) #define MUTEX_UNLOCK(A) LeaveCriticalSection(A) -#elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) +#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) // pthread API #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL) #define MUTEX_DESTROY(A) pthread_mutex_destroy(A) @@ -87,6 +87,9 @@ void RtAudio :: getCompiledApi( std::vector &apis ) throw() #if defined(__LINUX_ALSA__) apis.push_back( LINUX_ALSA ); #endif +#if defined(__LINUX_PULSE__) + apis.push_back( LINUX_PULSE ); +#endif #if defined(__LINUX_OSS__) apis.push_back( LINUX_OSS ); #endif @@ -106,7 +109,7 @@ void RtAudio :: getCompiledApi( std::vector &apis ) throw() void RtAudio :: openRtApi( RtAudio::Api api ) { - if (rtapi_) + if ( rtapi_ ) delete rtapi_; rtapi_ = 0; @@ -118,6 +121,10 @@ void RtAudio :: openRtApi( RtAudio::Api api ) if ( api == LINUX_ALSA ) rtapi_ = new RtApiAlsa(); #endif +#if defined(__LINUX_PULSE__) + if ( api == LINUX_PULSE ) + rtapi_ = new RtApiPulse(); +#endif #if defined(__LINUX_OSS__) if ( api == LINUX_OSS ) rtapi_ = new RtApiOss(); @@ -1357,8 +1364,6 @@ void RtApiCore :: startStream( void ) return; } - //MUTEX_LOCK( &stream_.mutex ); - OSStatus result = noErr; CoreHandle *handle = (CoreHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { @@ -1387,8 +1392,6 @@ void RtApiCore :: startStream( void ) stream_.state = STREAM_RUNNING; unlock: - //MUTEX_UNLOCK( &stream_.mutex ); - if ( result == noErr ) return; error( RtError::SYSTEM_ERROR ); } @@ -1402,15 +1405,6 @@ void RtApiCore :: stopStream( void ) return; } - /* - 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 ) { @@ -1420,9 +1414,7 @@ void RtApiCore :: stopStream( void ) pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled } - //MUTEX_UNLOCK( &stream_.mutex ); result = AudioDeviceStop( handle->id[0], callbackHandler ); - //MUTEX_LOCK( &stream_.mutex ); if ( result != noErr ) { errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ")."; errorText_ = errorStream_.str(); @@ -1432,9 +1424,7 @@ void RtApiCore :: stopStream( void ) if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { - //MUTEX_UNLOCK( &stream_.mutex ); result = AudioDeviceStop( handle->id[1], callbackHandler ); - //MUTEX_LOCK( &stream_.mutex ); if ( result != noErr ) { errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ")."; errorText_ = errorStream_.str(); @@ -1445,8 +1435,6 @@ void RtApiCore :: stopStream( void ) stream_.state = STREAM_STOPPED; unlock: - //MUTEX_UNLOCK( &stream_.mutex ); - if ( result == noErr ) return; error( RtError::SYSTEM_ERROR ); } @@ -1477,7 +1465,6 @@ extern "C" void *coreStopStream( void *ptr ) RtApiCore *object = (RtApiCore *) info->object; object->stopStream(); - pthread_exit( NULL ); } @@ -1498,26 +1485,14 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, // Check if we were draining the stream and signal is finished. if ( handle->drainCounter > 3 ) { - if ( handle->internalDrain == true ) { - stream_.state = STREAM_STOPPING; + stream_.state = STREAM_STOPPING; + if ( handle->internalDrain == true ) pthread_create( &threadId, NULL, coreStopStream, info ); - //stopStream(); - } else // external call to stopStream() pthread_cond_signal( &handle->condition ); return SUCCESS; } - /* - 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 @@ -1539,7 +1514,7 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], stream_.bufferSize, streamTime, status, info->userData ); if ( cbReturnValue == 2 ) { - //MUTEX_UNLOCK( &stream_.mutex ); + stream_.state = STREAM_STOPPING; handle->drainCounter = 2; abortStream(); return SUCCESS; @@ -2327,8 +2302,6 @@ void RtApiJack :: startStream( void ) return; } - MUTEX_LOCK(&stream_.mutex); - JackHandle *handle = (JackHandle *) stream_.apiHandle; int result = jack_activate( handle->client ); if ( result ) { @@ -2390,8 +2363,6 @@ void RtApiJack :: startStream( void ) stream_.state = STREAM_RUNNING; unlock: - MUTEX_UNLOCK(&stream_.mutex); - if ( result == 0 ) return; error( RtError::SYSTEM_ERROR ); } @@ -2405,13 +2376,6 @@ void RtApiJack :: stopStream( void ) return; } - 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 ) { @@ -2423,8 +2387,6 @@ void RtApiJack :: stopStream( void ) jack_deactivate( handle->client ); stream_.state = STREAM_STOPPED; - - MUTEX_UNLOCK( &stream_.mutex ); } void RtApiJack :: abortStream( void ) @@ -2453,13 +2415,12 @@ extern "C" void *jackStopStream( void *ptr ) RtApiJack *object = (RtApiJack *) info->object; object->stopStream(); - pthread_exit( NULL ); } bool RtApiJack :: callbackEvent( unsigned long nframes ) { - if ( stream_.state == STREAM_STOPPED ) return SUCCESS; + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; error( RtError::WARNING ); @@ -2476,6 +2437,8 @@ bool RtApiJack :: callbackEvent( unsigned long nframes ) // Check if we were draining the stream and signal is finished. if ( handle->drainCounter > 3 ) { + + stream_.state = STREAM_STOPPING; if ( handle->internalDrain == true ) pthread_create( &threadId, NULL, jackStopStream, info ); else @@ -2483,14 +2446,6 @@ bool RtApiJack :: callbackEvent( unsigned long nframes ) return SUCCESS; } - 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; @@ -2507,9 +2462,9 @@ bool RtApiJack :: callbackEvent( unsigned long nframes ) int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], stream_.bufferSize, streamTime, status, info->userData ); if ( cbReturnValue == 2 ) { - MUTEX_UNLOCK( &stream_.mutex ); - ThreadHandle id; + stream_.state = STREAM_STOPPING; handle->drainCounter = 2; + ThreadHandle id; pthread_create( &id, NULL, jackStopStream, info ); return SUCCESS; } @@ -2571,8 +2526,6 @@ bool RtApiJack :: callbackEvent( unsigned long nframes ) } unlock: - MUTEX_UNLOCK(&stream_.mutex); - RtApi::tickStreamTime(); return SUCCESS; } @@ -3189,8 +3142,6 @@ void RtApiAsio :: startStream() return; } - //MUTEX_LOCK( &stream_.mutex ); - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; ASIOError result = ASIOStart(); if ( result != ASE_OK ) { @@ -3206,8 +3157,6 @@ void RtApiAsio :: startStream() asioXRun = false; unlock: - //MUTEX_UNLOCK( &stream_.mutex ); - stopThreadCalled = false; if ( result == ASE_OK ) return; @@ -3223,23 +3172,11 @@ void RtApiAsio :: stopStream() return; } - /* - 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 = 2; - // MUTEX_UNLOCK( &stream_.mutex ); WaitForSingleObject( handle->condition, INFINITE ); // block until signaled - //ResetEvent( handle->condition ); - // MUTEX_LOCK( &stream_.mutex ); } } @@ -3251,8 +3188,6 @@ void RtApiAsio :: stopStream() errorText_ = errorStream_.str(); } - // MUTEX_UNLOCK( &stream_.mutex ); - if ( result == ASE_OK ) return; error( RtError::SYSTEM_ERROR ); } @@ -3286,15 +3221,13 @@ extern "C" unsigned __stdcall asioStopStream( void *ptr ) RtApiAsio *object = (RtApiAsio *) info->object; object->stopStream(); - _endthreadex( 0 ); return 0; } bool RtApiAsio :: callbackEvent( long bufferIndex ) { - if ( stream_.state == STREAM_STOPPED ) return SUCCESS; - if ( stopThreadCalled ) return SUCCESS; + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS; if ( stream_.state == STREAM_CLOSED ) { errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!"; error( RtError::WARNING ); @@ -3306,22 +3239,18 @@ bool RtApiAsio :: callbackEvent( long bufferIndex ) // Check if we were draining the stream and signal if finished. if ( handle->drainCounter > 3 ) { + + stream_.state = STREAM_STOPPING; if ( handle->internalDrain == false ) SetEvent( handle->condition ); else { // spawn a thread to stop the stream unsigned threadId; - stopThreadCalled = true; stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, &stream_.callbackInfo, 0, &threadId ); } return SUCCESS; } - /*MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) goto unlock; */ - // Invoke user callback to get fresh output data UNLESS we are // draining stream. if ( handle->drainCounter == 0 ) { @@ -3339,11 +3268,9 @@ bool RtApiAsio :: callbackEvent( long bufferIndex ) int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], stream_.bufferSize, streamTime, status, info->userData ); if ( cbReturnValue == 2 ) { - // MUTEX_UNLOCK( &stream_.mutex ); - // abortStream(); - unsigned threadId; - stopThreadCalled = true; + stream_.state = STREAM_STOPPING; handle->drainCounter = 2; + unsigned threadId; stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream, &stream_.callbackInfo, 0, &threadId ); return SUCCESS; @@ -3447,8 +3374,6 @@ bool RtApiAsio :: callbackEvent( long bufferIndex ) // drivers apparently do not function correctly without it. ASIOOutputReady(); - // MUTEX_UNLOCK( &stream_.mutex ); - RtApi::tickStreamTime(); return SUCCESS; } @@ -4447,8 +4372,6 @@ void RtApiDs :: startStream() return; } - //MUTEX_LOCK( &stream_.mutex ); - DsHandle *handle = (DsHandle *) stream_.apiHandle; // Increase scheduler frequency on lesser windows (a side-effect of @@ -4493,8 +4416,6 @@ void RtApiDs :: startStream() stream_.state = STREAM_RUNNING; unlock: - // MUTEX_UNLOCK( &stream_.mutex ); - if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); } @@ -4507,15 +4428,6 @@ void RtApiDs :: stopStream() return; } - /* - MUTEX_LOCK( &stream_.mutex ); - - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - */ - HRESULT result = 0; LPVOID audioPtr; DWORD dataLen; @@ -4523,10 +4435,7 @@ void RtApiDs :: stopStream() if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { if ( handle->drainCounter == 0 ) { handle->drainCounter = 2; - // MUTEX_UNLOCK( &stream_.mutex ); WaitForSingleObject( handle->condition, INFINITE ); // block until signaled - //ResetEvent( handle->condition ); - // MUTEX_LOCK( &stream_.mutex ); } stream_.state = STREAM_STOPPED; @@ -4604,8 +4513,6 @@ void RtApiDs :: stopStream() unlock: timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. - // MUTEX_UNLOCK( &stream_.mutex ); - if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); } @@ -4626,7 +4533,7 @@ void RtApiDs :: abortStream() void RtApiDs :: callbackEvent() { - if ( stream_.state == STREAM_STOPPED ) { + if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) { Sleep( 50 ); // sleep 50 milliseconds return; } @@ -4642,6 +4549,8 @@ void RtApiDs :: callbackEvent() // Check if we were draining the stream and signal is finished. if ( handle->drainCounter > stream_.nBuffers + 2 ) { + + stream_.state = STREAM_STOPPING; if ( handle->internalDrain == false ) SetEvent( handle->condition ); else @@ -4649,16 +4558,6 @@ void RtApiDs :: callbackEvent() return; } - /* - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - */ - // Invoke user callback to get fresh output data UNLESS we are // draining stream. if ( handle->drainCounter == 0 ) { @@ -4676,7 +4575,7 @@ void RtApiDs :: callbackEvent() int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1], stream_.bufferSize, streamTime, status, info->userData ); if ( cbReturnValue == 2 ) { - // MUTEX_UNLOCK( &stream_.mutex ); + stream_.state = STREAM_STOPPING; handle->drainCounter = 2; abortStream(); return; @@ -5011,8 +4910,6 @@ void RtApiDs :: callbackEvent() } unlock: - // MUTEX_UNLOCK( &stream_.mutex ); - RtApi::tickStreamTime(); } @@ -6139,11 +6036,6 @@ void RtApiAlsa :: stopStream() 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; @@ -6169,7 +6061,6 @@ void RtApiAlsa :: stopStream() } unlock: - stream_.state = STREAM_STOPPED; MUTEX_UNLOCK( &stream_.mutex ); if ( result >= 0 ) return; @@ -6188,11 +6079,6 @@ void RtApiAlsa :: abortStream() 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; @@ -6215,7 +6101,6 @@ void RtApiAlsa :: abortStream() } unlock: - stream_.state = STREAM_STOPPED; MUTEX_UNLOCK( &stream_.mutex ); if ( result >= 0 ) return; @@ -6425,6 +6310,436 @@ extern "C" void *alsaCallbackHandler( void *ptr ) //******************** End of __LINUX_ALSA__ *********************// #endif +#if defined(__LINUX_PULSE__) + +// Code written by Peter Meerwald, pmeerw@pmeerw.net +// and Tristan Matthews. + +#include +#include +#include + +namespace { +const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000, + 44100, 48000, 96000, 0}; } + +struct rtaudio_pa_format_mapping_t { + RtAudioFormat rtaudio_format; + pa_sample_format_t pa_format; +}; + +static const rtaudio_pa_format_mapping_t supported_sampleformats[] = { + {RTAUDIO_SINT16, PA_SAMPLE_S16LE}, + {RTAUDIO_SINT32, PA_SAMPLE_S32LE}, + {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE}, + {0, PA_SAMPLE_INVALID}}; + +struct PulseAudioHandle { + pa_simple *s_play; + pa_simple *s_rec; + pthread_t thread; + pthread_cond_t runnable_cv; + bool runnable; + PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { } +}; + +RtApiPulse::~RtApiPulse() +{ + if ( stream_.state != STREAM_CLOSED ) + closeStream(); +} + +unsigned int RtApiPulse::getDeviceCount( void ) +{ + return 1; +} + +RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int device ) +{ + RtAudio::DeviceInfo info; + info.probed = true; + info.name = "PulseAudio"; + info.outputChannels = 2; + info.inputChannels = 2; + info.duplexChannels = 2; + info.isDefaultOutput = true; + info.isDefaultInput = true; + + for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) + info.sampleRates.push_back( *sr ); + + info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32; + + return info; +} + +extern "C" void *pulseaudio_callback( void * user ) +{ + CallbackInfo *cbi = static_cast( user ); + RtApiPulse *context = static_cast( cbi->object ); + volatile bool *isRunning = &cbi->isRunning; + + while ( *isRunning ) { + pthread_testcancel(); + context->callbackEvent(); + } + + pthread_exit( NULL ); +} + +void RtApiPulse::closeStream( void ) +{ + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + + stream_.callbackInfo.isRunning = false; + if ( pah ) { + MUTEX_LOCK( &stream_.mutex ); + if ( stream_.state == STREAM_STOPPED ) { + pah->runnable = true; + pthread_cond_signal( &pah->runnable_cv ); + } + MUTEX_UNLOCK( &stream_.mutex ); + + pthread_join( pah->thread, 0 ); + if ( pah->s_play ) { + pa_simple_flush( pah->s_play, NULL ); + pa_simple_free( pah->s_play ); + } + if ( pah->s_rec ) + pa_simple_free( pah->s_rec ); + + pthread_cond_destroy( &pah->runnable_cv ); + delete pah; + stream_.apiHandle = 0; + } + + if ( stream_.userBuffer[0] ) { + free( stream_.userBuffer[0] ); + stream_.userBuffer[0] = 0; + } + if ( stream_.userBuffer[1] ) { + free( stream_.userBuffer[1] ); + stream_.userBuffer[1] = 0; + } + + stream_.state = STREAM_CLOSED; + stream_.mode = UNINITIALIZED; +} + +void RtApiPulse::callbackEvent( void ) +{ + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + + if ( stream_.state == STREAM_STOPPED ) { + MUTEX_LOCK( &stream_.mutex ); + while ( !pah->runnable ) + pthread_cond_wait( &pah->runnable_cv, &stream_.mutex ); + + if ( stream_.state != STREAM_RUNNING ) { + MUTEX_UNLOCK( &stream_.mutex ); + return; + } + MUTEX_UNLOCK( &stream_.mutex ); + } + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... " + "this shouldn't happen!"; + error( RtError::WARNING ); + return; + } + + RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; + double streamTime = getStreamTime(); + RtAudioStreamStatus status = 0; + int doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], + stream_.bufferSize, streamTime, status, + stream_.callbackInfo.userData ); + + if ( doStopStream == 2 ) { + abortStream(); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + if ( stream_.state != STREAM_RUNNING ) + goto unlock; + + int pa_error; + size_t bytes; + switch ( stream_.mode ) { + case INPUT: + bytes = stream_.nUserChannels[1] * stream_.bufferSize * formatBytes( stream_.userFormat ); + if ( pa_simple_read( pah->s_rec, stream_.userBuffer[1], bytes, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::callbackEvent: audio read error, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + } + break; + case OUTPUT: + bytes = stream_.nUserChannels[0] * stream_.bufferSize * formatBytes( stream_.userFormat ); + if ( pa_simple_write( pah->s_play, stream_.userBuffer[0], bytes, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::callbackEvent: audio write error, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + } + break; + case DUPLEX: + bytes = stream_.nUserChannels[1] * stream_.bufferSize * formatBytes( stream_.userFormat ); + if ( pa_simple_read( pah->s_rec, stream_.userBuffer[1], bytes, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::callbackEvent: audio read error, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + } + bytes = stream_.nUserChannels[0] * stream_.bufferSize * formatBytes( stream_.userFormat ); + if ( pa_simple_write( pah->s_play, stream_.userBuffer[0], bytes, &pa_error ) < 0) { + errorStream_ << "RtApiPulse::callbackEvent: audio write error, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); + error( RtError::WARNING ); + } + break; + default: + // ERROR + break; + } + + unlock: + MUTEX_UNLOCK( &stream_.mutex ); + RtApi::tickStreamTime(); + + if ( doStopStream == 1 ) + stopStream(); +} + +void RtApiPulse::startStream( void ) +{ + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::startStream(): the stream is not open!"; + error( RtError::INVALID_USE ); + return; + } + if ( stream_.state == STREAM_RUNNING ) { + errorText_ = "RtApiPulse::startStream(): the stream is already running!"; + error( RtError::WARNING ); + return; + } + + MUTEX_LOCK( &stream_.mutex ); + + stream_.state = STREAM_RUNNING; + + pah->runnable = true; + pthread_cond_signal( &pah->runnable_cv ); + MUTEX_UNLOCK( &stream_.mutex ); +} + +void RtApiPulse::stopStream( void ) +{ + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::stopStream(): the stream is not open!"; + error( RtError::INVALID_USE ); + return; + } + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + if ( pah && pah->s_play ) { + int pa_error; + if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::stopStream: error draining output device, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtError::SYSTEM_ERROR ); + } + } + + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); +} + +void RtApiPulse::abortStream( void ) +{ + PulseAudioHandle *pah = static_cast( stream_.apiHandle ); + + if ( stream_.state == STREAM_CLOSED ) { + errorText_ = "RtApiPulse::abortStream(): the stream is not open!"; + error( RtError::INVALID_USE ); + return; + } + if ( stream_.state == STREAM_STOPPED ) { + errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!"; + error( RtError::WARNING ); + return; + } + + stream_.state = STREAM_STOPPED; + MUTEX_LOCK( &stream_.mutex ); + + if ( pah && pah->s_play ) { + int pa_error; + if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) { + errorStream_ << "RtApiPulse::abortStream: error flushing output device, " << + pa_strerror( pa_error ) << "."; + errorText_ = errorStream_.str(); + MUTEX_UNLOCK( &stream_.mutex ); + error( RtError::SYSTEM_ERROR ); + } + } + + stream_.state = STREAM_STOPPED; + MUTEX_UNLOCK( &stream_.mutex ); +} + +bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, + unsigned int channels, unsigned int firstChannel, + unsigned int sampleRate, RtAudioFormat format, + unsigned int *bufferSize, RtAudio::StreamOptions *options ) +{ + PulseAudioHandle *pah = 0; + unsigned long bufferBytes = 0; + pa_sample_spec ss; + + 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; + + bool sr_found = false; + for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) { + if ( sampleRate == *sr ) { + sr_found = true; + stream_.sampleRate = sampleRate; + ss.rate = sampleRate; + break; + } + } + if ( !sr_found ) { + errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate."; + return false; + } + + bool sf_found = 0; + for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats; + sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) { + if ( format == sf->rtaudio_format ) { + sf_found = true; + stream_.userFormat = sf->rtaudio_format; + ss.format = sf->pa_format; + break; + } + } + if ( !sf_found ) { + errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample format."; + return false; + } + + if ( options && ( options->flags & RTAUDIO_NONINTERLEAVED ) ) { + errorText_ = "RtApiPulse::probeDeviceOpen: only interleaved audio data supported."; + return false; + } + + stream_.userInterleaved = true; + stream_.nBuffers = 1; + + stream_.deviceInterleaved[mode] = true; + stream_.doByteSwap[mode] = false; + stream_.doConvertBuffer[mode] = false; + stream_.deviceFormat[mode] = stream_.userFormat; + stream_.nUserChannels[mode] = channels; + stream_.nDeviceChannels[mode] = channels; + stream_.channelOffset[mode] = 0; + + // 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_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory."; + goto error; + } + stream_.bufferSize = *bufferSize; + + if ( !stream_.apiHandle ) { + PulseAudioHandle *pah = new PulseAudioHandle; + if ( !pah ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle."; + goto error; + } + + stream_.apiHandle = pah; + if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable."; + goto error; + } + } + pah = static_cast( stream_.apiHandle ); + + int error; + switch ( mode ) { + case INPUT: + pah->s_rec = pa_simple_new( NULL, "RtAudio", PA_STREAM_RECORD, NULL, "Record", &ss, NULL, NULL, &error ); + if ( !pah->s_rec ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server."; + goto error; + } + break; + case OUTPUT: + pah->s_play = pa_simple_new( NULL, "RtAudio", PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error ); + if ( !pah->s_play ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server."; + goto error; + } + break; + default: + goto error; + } + + if ( stream_.mode == UNINITIALIZED ) + stream_.mode = mode; + else if ( stream_.mode == mode ) + goto error; + else + stream_.mode = DUPLEX; + + stream_.state = STREAM_STOPPED; + + if ( !stream_.callbackInfo.isRunning ) { + stream_.callbackInfo.object = this; + stream_.callbackInfo.isRunning = true; + if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) { + errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread."; + goto error; + } + } + return true; + + error: + closeStream(); + return false; +} + +//******************** End of __LINUX_PULSE__ *********************// +#endif #if defined(__LINUX_OSS__) diff --git a/RtAudio.h b/RtAudio.h index d59e6bc..31f1fa5 100644 --- a/RtAudio.h +++ b/RtAudio.h @@ -210,6 +210,7 @@ class RtAudio enum Api { UNSPECIFIED, /*!< Search for a working compiled API. */ LINUX_ALSA, /*!< The Advanced Linux Sound Architecture API. */ + LINUX_PULSE, /*!< The Linux PulseAudio API. */ LINUX_OSS, /*!< The Linux Open Sound System API. */ UNIX_JACK, /*!< The Jack Low-Latency Audio Server API. */ MACOSX_CORE, /*!< Macintosh OS-X Core Audio API. */ @@ -511,7 +512,7 @@ class RtAudio typedef unsigned long ThreadHandle; typedef CRITICAL_SECTION StreamMutex; -#elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) +#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) // Using pthread library for various flavors of unix. #include @@ -552,7 +553,7 @@ struct CallbackInfo { // Note that RtApi is an abstract base class and cannot be // explicitly instantiated. The class RtAudio will create an // instance of an RtApi subclass (RtApiOss, RtApiAlsa, -// RtApiJack, RtApiCore, RtApiAl, RtApiDs, or RtApiAsio). +// RtApiJack, RtApiCore, RtApiDs, or RtApiAsio). // // **************************************************************** // @@ -912,6 +913,38 @@ public: #endif +#if defined(__LINUX_PULSE__) + +class RtApiPulse: public RtApi +{ +public: + ~RtApiPulse(); + RtAudio::Api getCurrentApi() { return RtAudio::LINUX_PULSE; }; + unsigned int getDeviceCount( void ); + RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); + void closeStream( void ); + void startStream( void ); + void stopStream( void ); + void abortStream( void ); + + // This function is intended for internal use only. It must be + // public because it is called by the internal callback handler, + // which is not a member of RtAudio. External use of this function + // will most likely produce highly undesireable results! + void callbackEvent( void ); + + private: + + std::vector devices_; + void saveDeviceInfo( void ); + bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, + unsigned int firstChannel, unsigned int sampleRate, + RtAudioFormat format, unsigned int *bufferSize, + RtAudio::StreamOptions *options ); +}; + +#endif + #if defined(__LINUX_OSS__) class RtApiOss: public RtApi diff --git a/configure.ac b/configure.ac index 13132aa..69250e8 100644 --- a/configure.ac +++ b/configure.ac @@ -80,6 +80,12 @@ case $host in AC_MSG_RESULT(using ALSA) AC_CHECK_LIB(asound, snd_pcm_open, , AC_MSG_ERROR(ALSA support requires the asound library!))], ) + # Look for PULSE flag + AC_ARG_WITH(pulse, [ --with-pulse = choose PulseAudio API support (linux only)], [ + api="$api -D__LINUX_PULSE__" + AC_MSG_RESULT(using PulseAudio) + AC_CHECK_LIB(pulse-simple, pa_simple_new, , AC_MSG_ERROR(PulseAudio support requires the pulse-simple library!))], ) + # Look for OSS flag AC_ARG_WITH(oss, [ --with-oss = choose OSS API support (linux only)], [ api="$api -D__LINUX_OSS__" diff --git a/doc/doxygen/Doxyfile b/doc/doxygen/Doxyfile index f3c509e..2c13b09 100644 --- a/doc/doxygen/Doxyfile +++ b/doc/doxygen/Doxyfile @@ -31,7 +31,7 @@ PROJECT_NAME = RtAudio # This could be handy for archiving the generated documentation or # if some version control system is used. -PROJECT_NUMBER = 4.0.10 +PROJECT_NUMBER = 4.0.11 # The OUTPUT_DIRECTORY tag is used to specify the (relative or absolute) # base path where the generated documentation will be put. diff --git a/doc/doxygen/acknowledge.txt b/doc/doxygen/acknowledge.txt index 81b17a8..8ea92e9 100644 --- a/doc/doxygen/acknowledge.txt +++ b/doc/doxygen/acknowledge.txt @@ -2,11 +2,20 @@ Many thanks to the following people for providing bug fixes and improvements:
    +
  • Stefan Arisona
    • +
  • Vincent Bénony
    • +
  • Rasmus Ekman
  • Anders Ervik
  • Robin Davies (Windows DS and ASIO)
    • +
  • Martin Koegler
    • +
  • Dmitry Kostjuchenko
    • +
  • Oliver Larkin
  • Antoine Lefebvre
    • +
  • Carlos Luna
  • Dominic Mazzoni
  • Tristan Matthews
    • +
  • Peter Meerwald (PulseAudio)
    • +
  • Benjamin Schroeder
  • Ryan Williams (Windows non-MS compiler ASIO support)
  • Ed Wildgoose (Linux ALSA and Jack)
    • diff --git a/doc/doxygen/apinotes.txt b/doc/doxygen/apinotes.txt index 1d369b6..fc80714 100644 --- a/doc/doxygen/apinotes.txt +++ b/doc/doxygen/apinotes.txt @@ -4,14 +4,12 @@ RtAudio is designed to provide a common API across the various supported operati \section linux Linux: -RtAudio for Linux was developed under Redhat distributions 7.0 - Fedora. Three different audio APIs are supported on Linux platforms: OSS (versions >= 4.0), ALSA, and Jack. Note that RtAudio now only supports the newer version 4.0 OSS API. The ALSA API is now part of the Linux kernel and offers significantly better functionality than the OSS API. RtAudio provides support for the 1.0 and higher versions of ALSA. Jack is a low-latency audio server written primarily for the GNU/Linux operating system. It can connect a number of different applications to an audio device, as well as allow them to share audio between themselves. Input/output latency on the order of 15 milliseconds can typically be achieved using any of the Linux APIs by fine-tuning the RtAudio buffer parameters (without kernel modifications). Latencies on the order of 5 milliseconds or less can be achieved using a low-latency kernel patch and increasing FIFO scheduling priority. The pthread library, which is used for callback functionality, is a standard component of all Linux distributions. +RtAudio for Linux was developed under Redhat distributions 7.0 - Fedora. Four different audio APIs are supported on Linux platforms: OSS (versions >= 4.0), ALSA, Jack, and PulseAudio. Note that RtAudio now only supports the newer version 4.0 OSS API. The ALSA API is now part of the Linux kernel and offers significantly better functionality than the OSS API. RtAudio provides support for the 1.0 and higher versions of ALSA. Jack is a low-latency audio server written primarily for the GNU/Linux operating system. It can connect a number of different applications to an audio device, as well as allow them to share audio between themselves. Input/output latency on the order of 15 milliseconds can typically be achieved using any of the Linux APIs by fine-tuning the RtAudio buffer parameters (without kernel modifications). Latencies on the order of 5 milliseconds or less can be achieved using a low-latency kernel patch and increasing FIFO scheduling priority. The pthread library, which is used for callback functionality, is a standard component of all Linux distributions. The ALSA library includes OSS emulation support. That means that you can run programs compiled for the OSS API even when using the ALSA drivers and library. It should be noted however that OSS emulation under ALSA is not perfect. Specifically, channel number queries seem to consistently produce invalid results. While OSS emulation is successful for the majority of RtAudio tests, it is recommended that the native ALSA implementation of RtAudio be used on systems which have ALSA drivers installed. The ALSA implementation of RtAudio makes no use of the ALSA "plug" interface. All necessary data format conversions, channel compensation, de-interleaving, and byte-swapping is handled by internal RtAudio routines. -At the moment, only one RtAudio instance can be connected to the Jack server. - \section macosx Macintosh OS-X (CoreAudio and Jack): The Apple CoreAudio API is designed to use a separate callback procedure for each of its audio devices. A single RtAudio duplex stream using two different devices is supported, though it cannot be guaranteed to always behave correctly because we cannot synchronize these two callbacks. The numberOfBuffers parameter to the RtAudio::openStream() function has no affect in this implementation. diff --git a/doc/doxygen/compiling.txt b/doc/doxygen/compiling.txt index 79cb314..d9884d0 100644 --- a/doc/doxygen/compiling.txt +++ b/doc/doxygen/compiling.txt @@ -26,6 +26,14 @@ In order to compile RtAudio for a specific OS and audio API, it is necessary to asound, pthread g++ -Wall -D__LINUX_ALSA__ -o audioprobe audioprobe.cpp RtAudio.cpp -lasound -lpthread + + Linux + PulseAudio + RtApiPulse + __LINUX_PULSE__ + pthread + g++ -Wall -D__LINUX_PULSE__ -o audioprobe audioprobe.cpp RtAudio.cpp -lpthread + Linux OSS diff --git a/doc/doxygen/footer.html b/doc/doxygen/footer.html index 114cdb0..18df8f2 100644 --- a/doc/doxygen/footer.html +++ b/doc/doxygen/footer.html @@ -1,7 +1,7 @@
    - +
    ©2001-2010 Gary P. Scavone, McGill University. All Rights Reserved.
    Maintained by Gary P. Scavone.
    ©2001-2012 Gary P. Scavone, McGill University. All Rights Reserved.
    Maintained by Gary P. Scavone.
    diff --git a/doc/doxygen/license.txt b/doc/doxygen/license.txt index eda2be5..c595b4f 100644 --- a/doc/doxygen/license.txt +++ b/doc/doxygen/license.txt @@ -1,7 +1,7 @@ /*! \page license License RtAudio: a set of realtime audio i/o C++ classes
    - Copyright (c) 2001-2011 Gary P. Scavone + Copyright (c) 2001-2012 Gary P. Scavone Permission is hereby granted, free of charge, to any person obtaining a copy of this software and associated documentation files diff --git a/doc/doxygen/tutorial.txt b/doc/doxygen/tutorial.txt index e33ceec..19ed170 100644 --- a/doc/doxygen/tutorial.txt +++ b/doc/doxygen/tutorial.txt @@ -1,6 +1,6 @@ /*! \mainpage The RtAudio Home Page -RtAudio is a set of C++ classes that provide a common API (Application Programming Interface) for realtime audio input/output across Linux, Macintosh OS-X 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 provide a common API (Application Programming Interface) for realtime audio input/output across Linux, Macintosh OS-X and Windows operating systems. RtAudio significantly simplifies the process of interacting with computer audio hardware. It was designed with the following objectives:
    • object-oriented C++ design
      • @@ -32,7 +32,7 @@ Devices are now re-enumerated every time the RtAudio::getDeviceCount(), RtAudio: \section download Download -Latest Release (?? June 2012): Version 4.0.11 +Latest Release (14 June 2012): Version 4.0.11 \section documentation Documentation Links diff --git a/doc/release.txt b/doc/release.txt index e36c3e7..a29a36c 100644 --- a/doc/release.txt +++ b/doc/release.txt @@ -1,9 +1,10 @@ -RtAudio - a set of C++ classes that provide a common API for realtime audio input/output across Linux (native ALSA, JACK, and OSS), Macintosh OS X (CoreAudio and JACK), and Windows (DirectSound and ASIO) operating systems. +RtAudio - a set of C++ classes that provide a common API for realtime audio input/output across Linux (native ALSA, JACK, PulseAudio, and OSS), Macintosh OS X (CoreAudio and JACK), and Windows (DirectSound and ASIO) operating systems. By Gary P. Scavone, 2001-2012. -v4.0.11: (?? June 2012) +v4.0.11: (14 June 2012) - fixes for memory leaks in ALSA (thanks to Martin Koegler) +- PulseAudio API support added (thanks to Peter Meerwald and Tristan Matthews) - bitwise format flag fixes in OS-X (Benjamin Schroeder and Stefan Arisona) - changes to stopStream / drain flag to avoid hung state in ASIO, DS, OS-X, and Jack APIs (Rasmus Ekman and Carlos Luna) diff --git a/include/soundcard.h b/include/soundcard.h index e8fc9f6..2cf3a2c 100644 --- a/include/soundcard.h +++ b/include/soundcard.h @@ -1,213 +1,361 @@ -#ifndef SOUNDCARD_H -#define SOUNDCARD_H /* - **************************************************************************** - * Copyright by 4Front Technologies 1993-2006 + * soundcard.h + */ + +/*- + * Copyright by Hannu Savolainen 1993 / 4Front Technologies 1993-2006 + * Modified for the new FreeBSD sound driver by Luigi Rizzo, 1997 * - ****************************************************************************** - * Modifications to this file are NOT allowed. This header file controls the - * OSS API. For compatibility reasons only 4Front Technologies can make changes - * to the definitions. If you have any questions, please contact - * hannu@opensound.com. - ****************************************************************************** + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions + * are met: + * 1. Redistributions of source code must retain the above copyright + * notice, this list of conditions and the following disclaimer. + * 2. Redistributions in binary form must reproduce the above + * copyright notice, this list of conditions and the following + * disclaimer in the documentation and/or other materials provided + * with the distribution. * - * Redistribution and use in source and binary forms, without - * modification, are permitted provided that the following conditions are - * met: 1. Redistributions of source code must retain the above copyright - * notice, this list of conditions and the following disclaimer. 2. - * Redistributions in binary form must reproduce the above copyright notice, - * this list of conditions and the following disclaimer in the documentation - * and/or other materials provided with the distribution. + * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' + * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED + * TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A + * PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR + * OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, + * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT + * LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF + * USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED + * AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT + * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN + * ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE + * POSSIBILITY OF SUCH DAMAGE. * - * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND ANY - * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED - * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE - * DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR - * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL - * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR - * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER - * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT - * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY - * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF - * SUCH DAMAGE. - **************************************************************************** + * $FreeBSD: src/sys/sys/soundcard.h,v 1.48 2006/11/26 11:55:48 netchild Exp $ + */ + +/* + * Unless coordinating changes with 4Front Technologies, do NOT make any + * modifications to ioctl commands, types, etc. that would break + * compatibility with the OSS API. + */ + +#ifndef _SYS_SOUNDCARD_H_ +#define _SYS_SOUNDCARD_H_ + /* + * If you make modifications to this file, please contact me before + * distributing the modified version. There is already enough + * diversity in the world. + * + * Regards, + * Hannu Savolainen + * hannu@voxware.pp.fi + * + ********************************************************************** + * PS. The Hacker's Guide to VoxWare available from + * nic.funet.fi:pub/Linux/ALPHA/sound. The file is + * snd-sdk-doc-0.1.ps.gz (gzipped postscript). It contains + * some useful information about programming with VoxWare. + * (NOTE! The pub/Linux/ALPHA/ directories are hidden. You have + * to cd inside them before the files are accessible.) + ********************************************************************** + */ + +/* + * SOUND_VERSION is only used by the voxware driver. Hopefully apps + * should not depend on it, but rather look at the capabilities + * of the driver in the kernel! + */ +#define SOUND_VERSION 301 +#define VOXWARE /* does this have any use ? */ + +/* + * Supported card ID numbers (Should be somewhere else? We keep + * them here just for compativility with the old driver, but these + * constants are of little or no use). + */ + +#define SNDCARD_ADLIB 1 +#define SNDCARD_SB 2 +#define SNDCARD_PAS 3 +#define SNDCARD_GUS 4 +#define SNDCARD_MPU401 5 +#define SNDCARD_SB16 6 +#define SNDCARD_SB16MIDI 7 +#define SNDCARD_UART6850 8 +#define SNDCARD_GUS16 9 +#define SNDCARD_MSS 10 +#define SNDCARD_PSS 11 +#define SNDCARD_SSCAPE 12 +#define SNDCARD_PSS_MPU 13 +#define SNDCARD_PSS_MSS 14 +#define SNDCARD_SSCAPE_MSS 15 +#define SNDCARD_TRXPRO 16 +#define SNDCARD_TRXPRO_SB 17 +#define SNDCARD_TRXPRO_MPU 18 +#define SNDCARD_MAD16 19 +#define SNDCARD_MAD16_MPU 20 +#define SNDCARD_CS4232 21 +#define SNDCARD_CS4232_MPU 22 +#define SNDCARD_MAUI 23 +#define SNDCARD_PSEUDO_MSS 24 +#define SNDCARD_AWE32 25 +#define SNDCARD_NSS 26 +#define SNDCARD_UART16550 27 +#define SNDCARD_OPL 28 + +#include +#include +#ifndef _IOWR +#include +#endif /* !_IOWR */ + +/* + * The first part of this file contains the new FreeBSD sound ioctl + * interface. Tries to minimize the number of different ioctls, and + * to be reasonably general. + * + * 970821: some of the new calls have not been implemented yet. */ + /* - * Purpose: The C/C++ header file that defines the OSS API. - * Description: - * This header file contains all the declarations required to compile OSS - * programs. The latest version is always installed together with OSS - * use of the latest version is strongly recommended. + * the following three calls extend the generic file descriptor + * interface. AIONWRITE is the dual of FIONREAD, i.e. returns the max + * number of bytes for a write operation to be non-blocking. * - * {!notice This header file contains many obsolete definitions - * (for compatibility with older applications that still ned them). - * Do not use this file as a reference manual of OSS. - * Please check the OSS Programmer's guide for descriptions - * of the supported API details (http://www.opensound.com/pguide).} - */ + * AIOGSIZE/AIOSSIZE are used to change the behaviour of the device, + * from a character device (default) to a block device. In block mode, + * (not to be confused with blocking mode) the main difference for the + * application is that select() will return only when a complete + * block can be read/written to the device, whereas in character mode + * select will return true when one byte can be exchanged. For audio + * devices, character mode makes select almost useless since one byte + * will always be ready by the next sample time (which is often only a + * handful of microseconds away). + * Use a size of 0 or 1 to return to character mode. + */ +#define AIONWRITE _IOR('A', 10, int) /* get # bytes to write */ +struct snd_size { + int play_size; + int rec_size; +}; +#define AIOGSIZE _IOR('A', 11, struct snd_size)/* read current blocksize */ +#define AIOSSIZE _IOWR('A', 11, struct snd_size) /* sets blocksize */ -#if defined(__cplusplus) -#define EXTERNC extern "C" +/* + * The following constants define supported audio formats. The + * encoding follows voxware conventions, i.e. 1 bit for each supported + * format. We extend it by using bit 31 (RO) to indicate full-duplex + * capability, and bit 29 (RO) to indicate that the card supports/ + * needs different formats on capture & playback channels. + * Bit 29 (RW) is used to indicate/ask stereo. + * + * The number of bits required to store the sample is: + * o 4 bits for the IDA ADPCM format, + * o 8 bits for 8-bit formats, mu-law and A-law, + * o 16 bits for the 16-bit formats, and + * o 32 bits for the 24/32-bit formats. + * o undefined for the MPEG audio format. + */ + +#define AFMT_QUERY 0x00000000 /* Return current format */ +#define AFMT_MU_LAW 0x00000001 /* Logarithmic mu-law */ +#define AFMT_A_LAW 0x00000002 /* Logarithmic A-law */ +#define AFMT_IMA_ADPCM 0x00000004 /* A 4:1 compressed format where 16-bit + * squence represented using the + * the average 4 bits per sample */ +#define AFMT_U8 0x00000008 /* Unsigned 8-bit */ +#define AFMT_S16_LE 0x00000010 /* Little endian signed 16-bit */ +#define AFMT_S16_BE 0x00000020 /* Big endian signed 16-bit */ +#define AFMT_S8 0x00000040 /* Signed 8-bit */ +#define AFMT_U16_LE 0x00000080 /* Little endian unsigned 16-bit */ +#define AFMT_U16_BE 0x00000100 /* Big endian unsigned 16-bit */ +#define AFMT_MPEG 0x00000200 /* MPEG MP2/MP3 audio */ +#define AFMT_AC3 0x00000400 /* Dolby Digital AC3 */ + +#if _BYTE_ORDER == _LITTLE_ENDIAN +#define AFMT_S16_NE AFMT_S16_LE /* native endian signed 16 */ #else -#define EXTERNC extern -#endif /* EXTERN_C_WRAPPERS */ +#define AFMT_S16_NE AFMT_S16_BE +#endif -#define OSS_VERSION 0x040002 +/* + * 32-bit formats below used for 24-bit audio data where the data is stored + * in the 24 most significant bits and the least significant bits are not used + * (should be set to 0). + */ +#define AFMT_S32_LE 0x00001000 /* Little endian signed 32-bit */ +#define AFMT_S32_BE 0x00002000 /* Big endian signed 32-bit */ +#define AFMT_U32_LE 0x00004000 /* Little endian unsigned 32-bit */ +#define AFMT_U32_BE 0x00008000 /* Big endian unsigned 32-bit */ +#define AFMT_S24_LE 0x00010000 /* Little endian signed 24-bit */ +#define AFMT_S24_BE 0x00020000 /* Big endian signed 24-bit */ +#define AFMT_U24_LE 0x00040000 /* Little endian unsigned 24-bit */ +#define AFMT_U24_BE 0x00080000 /* Big endian unsigned 24-bit */ -#define SOUND_VERSION OSS_VERSION -#define OPEN_SOUND_SYSTEM +#define AFMT_STEREO 0x10000000 /* can do/want stereo */ -#if defined(__hpux) && !defined(_HPUX_SOURCE) -# error "-D_HPUX_SOURCE must be used when compiling OSS applications" -#endif +/* + * the following are really capabilities + */ +#define AFMT_WEIRD 0x20000000 /* weird hardware... */ + /* + * AFMT_WEIRD reports that the hardware might need to operate + * with different formats in the playback and capture + * channels when operating in full duplex. + * As an example, SoundBlaster16 cards only support U8 in one + * direction and S16 in the other one, and applications should + * be aware of this limitation. + */ +#define AFMT_FULLDUPLEX 0x80000000 /* can do full duplex */ -#ifdef __hpux -#include -#endif +/* + * The following structure is used to get/set format and sampling rate. + * While it would be better to have things such as stereo, bits per + * sample, endiannes, etc split in different variables, it turns out + * that formats are not that many, and not all combinations are possible. + * So we followed the Voxware approach of associating one bit to each + * format. + */ -#ifdef linux -/* In Linux we need to be prepared for cross compiling */ -#include -#else -# ifdef __FreeBSD__ -# include -# else -# include -# endif -#endif +typedef struct _snd_chan_param { + u_long play_rate; /* sampling rate */ + u_long rec_rate; /* sampling rate */ + u_long play_format; /* everything describing the format */ + u_long rec_format; /* everything describing the format */ +} snd_chan_param; +#define AIOGFMT _IOR('f', 12, snd_chan_param) /* get format */ +#define AIOSFMT _IOWR('f', 12, snd_chan_param) /* sets format */ -#ifndef __SIOWR -#if defined(__hpux) || (defined(_IOWR) && (defined(_AIX) || (!defined(sun) && !defined(sparc) && !defined(__INCioctlh) && !defined(__Lynx__)))) - -/* - * Make sure the ioctl macros are compatible with the ones already used - * by this operating system. - */ -#define SIOCPARM_MASK IOCPARM_MASK -#define SIOC_VOID IOC_VOID -#define SIOC_OUT IOC_OUT -#define SIOC_IN IOC_IN -#define SIOC_INOUT IOC_INOUT -#define __SIOC_SIZE _IOC_SIZE -#define __SIOC_DIR _IOC_DIR -#define __SIOC_NONE _IOC_NONE -#define __SIOC_READ _IOC_READ -#define __SIOC_WRITE _IOC_WRITE -#define __SIO _IO -#define __SIOR _IOR -#define __SIOW _IOW -#define __SIOWR _IOWR -#else +/* + * The following structure is used to get/set the mixer setting. + * Up to 32 mixers are supported, each one with up to 32 channels. + */ +typedef struct _snd_mix_param { + u_char subdev; /* which output */ + u_char line; /* which input */ + u_char left,right; /* volumes, 0..255, 0 = mute */ +} snd_mix_param ; -/* #define SIOCTYPE (0xff<<8) */ -#define SIOCPARM_MASK 0x1fff /* parameters must be < 8192 bytes */ -#define SIOC_VOID 0x00000000 /* no parameters */ -#define SIOC_OUT 0x20000000 /* copy out parameters */ -#define SIOC_IN 0x40000000 /* copy in parameters */ -#define SIOC_INOUT (SIOC_IN|SIOC_OUT) - -#define __SIO(x,y) ((int)(SIOC_VOID|(x<<8)|y)) -#define __SIOR(x,y,t) ((int)(SIOC_OUT|((sizeof(t)&SIOCPARM_MASK)<<16)|(x<<8)|y)) -#define __SIOW(x,y,t) ((int)(SIOC_IN|((sizeof(t)&SIOCPARM_MASK)<<16)|(x<<8)|y)) -#define __SIOWR(x,y,t) ((int)(SIOC_INOUT|((sizeof(t)&SIOCPARM_MASK)<<16)|(x<<8)|y)) -#define __SIOC_SIZE(x) ((x>>16)&SIOCPARM_MASK) -#define __SIOC_DIR(x) (x & 0xf0000000) -#define __SIOC_NONE SIOC_VOID -#define __SIOC_READ SIOC_OUT -#define __SIOC_WRITE SIOC_IN -# endif /* _IOWR */ -#endif /* !__SIOWR */ +/* XXX AIOGMIX, AIOSMIX not implemented yet */ +#define AIOGMIX _IOWR('A', 13, snd_mix_param) /* return mixer status */ +#define AIOSMIX _IOWR('A', 14, snd_mix_param) /* sets mixer status */ -#define OSS_LONGNAME_SIZE 64 -#define OSS_LABEL_SIZE 16 -#define OSS_DEVNODE_SIZE 32 -typedef char oss_longname_t[OSS_LONGNAME_SIZE]; -typedef char oss_label_t[OSS_LABEL_SIZE]; -typedef char oss_devnode_t[OSS_DEVNODE_SIZE]; +/* + * channel specifiers used in AIOSTOP and AIOSYNC + */ +#define AIOSYNC_PLAY 0x1 /* play chan */ +#define AIOSYNC_CAPTURE 0x2 /* capture chan */ +/* AIOSTOP stop & flush a channel, returns the residual count */ +#define AIOSTOP _IOWR ('A', 15, int) + +/* alternate method used to notify the sync condition */ +#define AIOSYNC_SIGNAL 0x100 +#define AIOSYNC_SELECT 0x200 + +/* what the 'pos' field refers to */ +#define AIOSYNC_READY 0x400 +#define AIOSYNC_FREE 0x800 + +typedef struct _snd_sync_parm { + long chan ; /* play or capture channel, plus modifier */ + long pos; +} snd_sync_parm; +#define AIOSYNC _IOWR ('A', 15, snd_sync_parm) /* misc. synchronization */ -#ifndef DISABLE_SEQUENCER /* - **************************************************************************** - * IOCTL Commands for /dev/sequencer and /dev/music (AKA /dev/sequencer2) + * The following is used to return device capabilities. If the structure + * passed to the ioctl is zeroed, default values are returned for rate + * and formats, a bitmap of available mixers is returned, and values + * (inputs, different levels) for the first one are returned. * - * Note that this interface is obsolete and no longer developed. New - * applications should use /dev/midi instead. - ****************************************************************************/ -#define SNDCTL_SEQ_RESET __SIO ('Q', 0) -#define SNDCTL_SEQ_SYNC __SIO ('Q', 1) -#define SNDCTL_SYNTH_INFO __SIOWR('Q', 2, struct synth_info) -#define SNDCTL_SEQ_CTRLRATE __SIOWR('Q', 3, int) /* Set/get timer resolution (HZ) */ -#define SNDCTL_SEQ_GETOUTCOUNT __SIOR ('Q', 4, int) -#define SNDCTL_SEQ_GETINCOUNT __SIOR ('Q', 5, int) -#define SNDCTL_SEQ_PERCMODE __SIOW ('Q', 6, int) -#define SNDCTL_FM_LOAD_INSTR __SIOW ('Q', 7, struct sbi_instrument) /* Obsolete. Don't use!!!!!! */ -#define SNDCTL_SEQ_TESTMIDI __SIOW ('Q', 8, int) -#define SNDCTL_SEQ_RESETSAMPLES __SIOW ('Q', 9, int) -#define SNDCTL_SEQ_NRSYNTHS __SIOR ('Q',10, int) -#define SNDCTL_SEQ_NRMIDIS __SIOR ('Q',11, int) -#define SNDCTL_MIDI_INFO __SIOWR('Q',12, struct midi_info) /* OBSOLETE - use SNDCTL_MIDIINFO instead */ -#define SNDCTL_SEQ_THRESHOLD __SIOW ('Q',13, int) -#define SNDCTL_SYNTH_MEMAVL __SIOWR('Q',14, int) /* in=dev#, out=memsize */ -#define SNDCTL_FM_4OP_ENABLE __SIOW ('Q',15, int) /* in=dev# */ -#define SNDCTL_SEQ_PANIC __SIO ('Q',17) -#define SNDCTL_SEQ_OUTOFBAND __SIOW ('Q',18, struct seq_event_rec) -#define SNDCTL_SEQ_GETTIME __SIOR ('Q',19, int) -#define SNDCTL_SYNTH_ID __SIOWR('Q',20, struct synth_info) -#define SNDCTL_SYNTH_CONTROL __SIOWR('Q',21, struct synth_control) -#define SNDCTL_SYNTH_REMOVESAMPLE __SIOWR('Q',22, struct remove_sample) /* Reserved for future use */ -#define SNDCTL_SEQ_TIMING_ENABLE __SIO ('Q', 23) /* Enable incoming MIDI timing messages */ -#define SNDCTL_SEQ_ACTSENSE_ENABLE __SIO ('Q', 24) /* Enable incoming active sensing messages */ -#define SNDCTL_SEQ_RT_ENABLE __SIO ('Q', 25) /* Enable other incoming realtime messages */ - -typedef struct synth_control -{ - int devno; /* Synthesizer # */ - char data[4000]; /* Device spesific command/data record */ -} synth_control; + * If formats, mixers, inputs are instantiated, then detailed info + * are returned depending on the call. + */ +typedef struct _snd_capabilities { + u_long rate_min, rate_max; /* min-max sampling rate */ + u_long formats; + u_long bufsize; /* DMA buffer size */ + u_long mixers; /* bitmap of available mixers */ + u_long inputs; /* bitmap of available inputs (per mixer) */ + u_short left, right; /* how many levels are supported */ +} snd_capabilities; +#define AIOGCAP _IOWR('A', 15, snd_capabilities) /* get capabilities */ + +/* + * here is the old (Voxware) ioctl interface + */ + +/* + * IOCTL Commands for /dev/sequencer + */ + +#define SNDCTL_SEQ_RESET _IO ('Q', 0) +#define SNDCTL_SEQ_SYNC _IO ('Q', 1) +#define SNDCTL_SYNTH_INFO _IOWR('Q', 2, struct synth_info) +#define SNDCTL_SEQ_CTRLRATE _IOWR('Q', 3, int) /* Set/get timer res.(hz) */ +#define SNDCTL_SEQ_GETOUTCOUNT _IOR ('Q', 4, int) +#define SNDCTL_SEQ_GETINCOUNT _IOR ('Q', 5, int) +#define SNDCTL_SEQ_PERCMODE _IOW ('Q', 6, int) +#define SNDCTL_FM_LOAD_INSTR _IOW ('Q', 7, struct sbi_instrument) /* Valid for FM only */ +#define SNDCTL_SEQ_TESTMIDI _IOW ('Q', 8, int) +#define SNDCTL_SEQ_RESETSAMPLES _IOW ('Q', 9, int) +#define SNDCTL_SEQ_NRSYNTHS _IOR ('Q',10, int) +#define SNDCTL_SEQ_NRMIDIS _IOR ('Q',11, int) +#define SNDCTL_MIDI_INFO _IOWR('Q',12, struct midi_info) +#define SNDCTL_SEQ_THRESHOLD _IOW ('Q',13, int) +#define SNDCTL_SEQ_TRESHOLD SNDCTL_SEQ_THRESHOLD /* there was once a typo */ +#define SNDCTL_SYNTH_MEMAVL _IOWR('Q',14, int) /* in=dev#, out=memsize */ +#define SNDCTL_FM_4OP_ENABLE _IOW ('Q',15, int) /* in=dev# */ +#define SNDCTL_PMGR_ACCESS _IOWR('Q',16, struct patmgr_info) +#define SNDCTL_SEQ_PANIC _IO ('Q',17) +#define SNDCTL_SEQ_OUTOFBAND _IOW ('Q',18, struct seq_event_rec) +#define SNDCTL_SEQ_GETTIME _IOR ('Q',19, int) + +struct seq_event_rec { + u_char arr[8]; +}; -typedef struct remove_sample -{ - int devno; /* Synthesizer # */ - int bankno; /* MIDI bank # (0=General MIDI) */ - int instrno; /* MIDI instrument number */ -} remove_sample; +#define SNDCTL_TMR_TIMEBASE _IOWR('T', 1, int) +#define SNDCTL_TMR_START _IO ('T', 2) +#define SNDCTL_TMR_STOP _IO ('T', 3) +#define SNDCTL_TMR_CONTINUE _IO ('T', 4) +#define SNDCTL_TMR_TEMPO _IOWR('T', 5, int) +#define SNDCTL_TMR_SOURCE _IOWR('T', 6, int) +# define TMR_INTERNAL 0x00000001 +# define TMR_EXTERNAL 0x00000002 +# define TMR_MODE_MIDI 0x00000010 +# define TMR_MODE_FSK 0x00000020 +# define TMR_MODE_CLS 0x00000040 +# define TMR_MODE_SMPTE 0x00000080 +#define SNDCTL_TMR_METRONOME _IOW ('T', 7, int) +#define SNDCTL_TMR_SELECT _IOW ('T', 8, int) -typedef struct seq_event_rec -{ - unsigned char arr[8]; -} seq_event_rec; - -#define SNDCTL_TMR_TIMEBASE __SIOWR('T', 1, int) -#define SNDCTL_TMR_START __SIO ('T', 2) -#define SNDCTL_TMR_STOP __SIO ('T', 3) -#define SNDCTL_TMR_CONTINUE __SIO ('T', 4) -#define SNDCTL_TMR_TEMPO __SIOWR('T', 5, int) -#define SNDCTL_TMR_SOURCE __SIOWR('T', 6, int) -# define TMR_INTERNAL 0x00000001 -# define TMR_EXTERNAL 0x00000002 -# define TMR_MODE_MIDI 0x00000010 -# define TMR_MODE_FSK 0x00000020 -# define TMR_MODE_CLS 0x00000040 -# define TMR_MODE_SMPTE 0x00000080 -#define SNDCTL_TMR_METRONOME __SIOW ('T', 7, int) -#define SNDCTL_TMR_SELECT __SIOW ('T', 8, int) - -/* - * Sample loading mechanism for internal synthesizers (/dev/sequencer) - * (for the .PAT format). - */ - -struct patch_info -{ - unsigned short key; /* Use WAVE_PATCH here */ -#define WAVE_PATCH _PATCHKEY(0x04) -#define GUS_PATCH WAVE_PATCH -#define WAVEFRONT_PATCH _PATCHKEY(0x06) +/* + * Endian aware patch key generation algorithm. + */ + +#if defined(_AIX) || defined(AIX) +# define _PATCHKEY(id) (0xfd00|id) +#else +# define _PATCHKEY(id) ((id<<8)|0xfd) +#endif + +/* + * Sample loading mechanism for internal synthesizers (/dev/sequencer) + * The following patch_info structure has been designed to support + * Gravis UltraSound. It tries to be universal format for uploading + * sample based patches but is probably too limited. + */ + +struct patch_info { +/* u_short key; Use GUS_PATCH here */ + short key; /* Use GUS_PATCH here */ +#define GUS_PATCH _PATCHKEY(0x04) +#define OBSOLETE_GUS_PATCH _PATCHKEY(0x02) - short device_no; /* Synthesizer number */ - short instr_no; /* Midi pgm# */ + short device_no; /* Synthesizer number */ + short instr_no; /* Midi pgm# */ - unsigned int mode; + u_long mode; /* * The least significant byte has the same format than the GUS .PAT * files @@ -217,24 +365,19 @@ struct patch_info #define WAVE_LOOPING 0x04 /* bit 2 = looping enabled-1. */ #define WAVE_BIDIR_LOOP 0x08 /* bit 3 = Set is bidirectional looping. */ #define WAVE_LOOP_BACK 0x10 /* bit 4 = Set is looping backward. */ -#define WAVE_SUSTAIN_ON 0x20 /* bit 5 = Turn sustaining on. (Env. pts. 3) */ +#define WAVE_SUSTAIN_ON 0x20 /* bit 5 = Turn sustaining on. (Env. pts. 3)*/ #define WAVE_ENVELOPES 0x40 /* bit 6 = Enable envelopes - 1 */ -#define WAVE_FAST_RELEASE 0x80 /* bit 7 = Shut off immediately after note off */ - /* (use the env_rate/env_offs fields). */ + /* (use the env_rate/env_offs fields). */ /* Linux specific bits */ #define WAVE_VIBRATO 0x00010000 /* The vibrato info is valid */ #define WAVE_TREMOLO 0x00020000 /* The tremolo info is valid */ #define WAVE_SCALE 0x00040000 /* The scaling info is valid */ -#define WAVE_FRACTIONS 0x00080000 /* Fraction information is valid */ -/* Reserved bits */ -#define WAVE_ROM 0x40000000 /* For future use */ -#define WAVE_MULAW 0x20000000 /* For future use */ /* Other bits must be zeroed */ - int len; /* Size of the wave data in bytes */ - int loop_start, loop_end; /* Byte offsets from the beginning */ + long len; /* Size of the wave data in bytes */ + long loop_start, loop_end; /* Byte offsets from the beginning */ -/* +/* * The base_freq and base_note fields are used when computing the * playback speed for a note. The base_note defines the tone frequency * which is heard if the sample is played using the base_freq as the @@ -250,56 +393,139 @@ struct patch_info * middle A is 440*1000. */ - unsigned int base_freq; - unsigned int base_note; - unsigned int high_note; - unsigned int low_note; - int panning; /* -128=left, 127=right */ - int detuning; + u_int base_freq; + u_long base_note; + u_long high_note; + u_long low_note; + int panning; /* -128=left, 127=right */ + int detuning; - /* Envelope. Enabled by mode bit WAVE_ENVELOPES */ - unsigned char env_rate[6]; /* GUS HW ramping rate */ - unsigned char env_offset[6]; /* 255 == 100% */ +/* New fields introduced in version 1.99.5 */ - /* - * The tremolo, vibrato and scale info are not supported yet. - * Enable by setting the mode bits WAVE_TREMOLO, WAVE_VIBRATO or - * WAVE_SCALE - */ + /* Envelope. Enabled by mode bit WAVE_ENVELOPES */ + u_char env_rate[ 6 ]; /* GUS HW ramping rate */ + u_char env_offset[ 6 ]; /* 255 == 100% */ + + /* + * The tremolo, vibrato and scale info are not supported yet. + * Enable by setting the mode bits WAVE_TREMOLO, WAVE_VIBRATO or + * WAVE_SCALE + */ - unsigned char tremolo_sweep; - unsigned char tremolo_rate; - unsigned char tremolo_depth; + u_char tremolo_sweep; + u_char tremolo_rate; + u_char tremolo_depth; - unsigned char vibrato_sweep; - unsigned char vibrato_rate; - unsigned char vibrato_depth; + u_char vibrato_sweep; + u_char vibrato_rate; + u_char vibrato_depth; - int scale_frequency; - unsigned int scale_factor; /* from 0 to 2048 or 0 to 2 */ + int scale_frequency; + u_int scale_factor; /* from 0 to 2048 or 0 to 2 */ - int volume; - int fractions; - int reserved1; - int spare[2]; - char data[1]; /* The waveform data starts here */ + int volume; + int spare[4]; + char data[1]; /* The waveform data starts here */ }; -struct sysex_info -{ - short key; /* Use SYSEX_PATCH or MAUI_PATCH here */ +struct sysex_info { + short key; /* Use GUS_PATCH here */ #define SYSEX_PATCH _PATCHKEY(0x05) #define MAUI_PATCH _PATCHKEY(0x06) - short device_no; /* Synthesizer number */ - int len; /* Size of the sysex data in bytes */ - unsigned char data[1]; /* Sysex data starts here */ + short device_no; /* Synthesizer number */ + long len; /* Size of the sysex data in bytes */ + u_char data[1]; /* Sysex data starts here */ +}; + +/* + * Patch management interface (/dev/sequencer, /dev/patmgr#) + * Don't use these calls if you want to maintain compatibility with + * the future versions of the driver. + */ + +#define PS_NO_PATCHES 0 /* No patch support on device */ +#define PS_MGR_NOT_OK 1 /* Plain patch support (no mgr) */ +#define PS_MGR_OK 2 /* Patch manager supported */ +#define PS_MANAGED 3 /* Patch manager running */ + +#define SNDCTL_PMGR_IFACE _IOWR('P', 1, struct patmgr_info) + +/* + * The patmgr_info is a fixed size structure which is used for two + * different purposes. The intended use is for communication between + * the application using /dev/sequencer and the patch manager daemon + * associated with a synthesizer device (ioctl(SNDCTL_PMGR_ACCESS)). + * + * This structure is also used with ioctl(SNDCTL_PGMR_IFACE) which allows + * a patch manager daemon to read and write device parameters. This + * ioctl available through /dev/sequencer also. Avoid using it since it's + * extremely hardware dependent. In addition access trough /dev/sequencer + * may confuse the patch manager daemon. + */ + +struct patmgr_info { /* Note! size must be < 4k since kmalloc() is used */ + u_long key; /* Don't worry. Reserved for communication + between the patch manager and the driver. */ +#define PM_K_EVENT 1 /* Event from the /dev/sequencer driver */ +#define PM_K_COMMAND 2 /* Request from an application */ +#define PM_K_RESPONSE 3 /* From patmgr to application */ +#define PM_ERROR 4 /* Error returned by the patmgr */ + int device; + int command; + +/* + * Commands 0x000 to 0xfff reserved for patch manager programs + */ +#define PM_GET_DEVTYPE 1 /* Returns type of the patch mgr interface of dev */ +#define PMTYPE_FM2 1 /* 2 OP fm */ +#define PMTYPE_FM4 2 /* Mixed 4 or 2 op FM (OPL-3) */ +#define PMTYPE_WAVE 3 /* Wave table synthesizer (GUS) */ +#define PM_GET_NRPGM 2 /* Returns max # of midi programs in parm1 */ +#define PM_GET_PGMMAP 3 /* Returns map of loaded midi programs in data8 */ +#define PM_GET_PGM_PATCHES 4 /* Return list of patches of a program (parm1) */ +#define PM_GET_PATCH 5 /* Return patch header of patch parm1 */ +#define PM_SET_PATCH 6 /* Set patch header of patch parm1 */ +#define PM_READ_PATCH 7 /* Read patch (wave) data */ +#define PM_WRITE_PATCH 8 /* Write patch (wave) data */ + +/* + * Commands 0x1000 to 0xffff are for communication between the patch manager + * and the client + */ +#define _PM_LOAD_PATCH 0x100 + +/* + * Commands above 0xffff reserved for device specific use + */ + + long parm1; + long parm2; + long parm3; + + union { + u_char data8[4000]; + u_short data16[2000]; + u_long data32[1000]; + struct patch_info patch; + } data; }; +/* + * When a patch manager daemon is present, it will be informed by the + * driver when something important happens. For example when the + * /dev/sequencer is opened or closed. A record with key == PM_K_EVENT is + * returned. The command field contains the event type: + */ +#define PM_E_OPENED 1 /* /dev/sequencer opened */ +#define PM_E_CLOSED 2 /* /dev/sequencer closed */ +#define PM_E_PATCH_RESET 3 /* SNDCTL_RESETSAMPLES called */ +#define PM_E_PATCH_LOADED 4 /* A patch has been loaded by appl */ + /* * /dev/sequencer input events. * * The data written to the /dev/sequencer is a stream of events. Events - * are records of 4 or 8 bytes. The first byte defines the size. + * are records of 4 or 8 bytes. The first byte defines the size. * Any number of events can be written with a write call. There * is a set of macros for sending these events. Use these macros if you * want to maximize portability of your program. @@ -314,10 +540,6 @@ struct sysex_info * Normal events (4 bytes) * There is also a 8 byte version of most of the 4 byte events. The * 8 byte one is recommended. - * - * NOTE! All 4 byte events are now obsolete. Applications should not write - * them. However 4 byte events are still used as inputs from - * /dev/sequencer (/dev/music uses only 8 byte ones). */ #define SEQ_NOTEOFF 0 #define SEQ_FMNOTEOFF SEQ_NOTEOFF /* Just old name */ @@ -328,94 +550,93 @@ struct sysex_info #define SEQ_FMPGMCHANGE SEQ_PGMCHANGE #define SEQ_SYNCTIMER TMR_START #define SEQ_MIDIPUTC 5 -#define SEQ_DRUMON 6 /*** OBSOLETE ***/ -#define SEQ_DRUMOFF 7 /*** OBSOLETE ***/ +#define SEQ_DRUMON 6 /*** OBSOLETE ***/ +#define SEQ_DRUMOFF 7 /*** OBSOLETE ***/ #define SEQ_ECHO TMR_ECHO /* For synching programs with output */ #define SEQ_AFTERTOUCH 9 #define SEQ_CONTROLLER 10 -#define SEQ_BALANCE 11 -#define SEQ_VOLMODE 12 -/************************************ - * Midi controller numbers * - ************************************/ -/* - * Controllers 0 to 31 (0x00 to 0x1f) and - * 32 to 63 (0x20 to 0x3f) are continuous - * controllers. - * In the MIDI 1.0 these controllers are sent using - * two messages. Controller numbers 0 to 31 are used - * to send the MSB and the controller numbers 32 to 63 - * are for the LSB. Note that just 7 bits are used in MIDI bytes. - */ - -#define CTL_BANK_SELECT 0x00 -#define CTL_MODWHEEL 0x01 -#define CTL_BREATH 0x02 -/* undefined 0x03 */ -#define CTL_FOOT 0x04 -#define CTL_PORTAMENTO_TIME 0x05 -#define CTL_DATA_ENTRY 0x06 -#define CTL_MAIN_VOLUME 0x07 -#define CTL_BALANCE 0x08 -/* undefined 0x09 */ -#define CTL_PAN 0x0a -#define CTL_EXPRESSION 0x0b -/* undefined 0x0c */ -/* undefined 0x0d */ -/* undefined 0x0e */ -/* undefined 0x0f */ -#define CTL_GENERAL_PURPOSE1 0x10 -#define CTL_GENERAL_PURPOSE2 0x11 -#define CTL_GENERAL_PURPOSE3 0x12 -#define CTL_GENERAL_PURPOSE4 0x13 -/* undefined 0x14 - 0x1f */ - -/* undefined 0x20 */ -/* The controller numbers 0x21 to 0x3f are reserved for the */ -/* least significant bytes of the controllers 0x00 to 0x1f. */ -/* These controllers are not recognised by the driver. */ - -/* Controllers 64 to 69 (0x40 to 0x45) are on/off switches. */ -/* 0=OFF and 127=ON (intermediate values are possible) */ -#define CTL_DAMPER_PEDAL 0x40 -#define CTL_SUSTAIN 0x40 /* Alias */ -#define CTL_HOLD 0x40 /* Alias */ -#define CTL_PORTAMENTO 0x41 -#define CTL_SOSTENUTO 0x42 -#define CTL_SOFT_PEDAL 0x43 -/* undefined 0x44 */ -#define CTL_HOLD2 0x45 -/* undefined 0x46 - 0x4f */ - -#define CTL_GENERAL_PURPOSE5 0x50 -#define CTL_GENERAL_PURPOSE6 0x51 -#define CTL_GENERAL_PURPOSE7 0x52 -#define CTL_GENERAL_PURPOSE8 0x53 -/* undefined 0x54 - 0x5a */ -#define CTL_EXT_EFF_DEPTH 0x5b -#define CTL_TREMOLO_DEPTH 0x5c -#define CTL_CHORUS_DEPTH 0x5d -#define CTL_DETUNE_DEPTH 0x5e -#define CTL_CELESTE_DEPTH 0x5e /* Alias for the above one */ -#define CTL_PHASER_DEPTH 0x5f -#define CTL_DATA_INCREMENT 0x60 -#define CTL_DATA_DECREMENT 0x61 -#define CTL_NONREG_PARM_NUM_LSB 0x62 -#define CTL_NONREG_PARM_NUM_MSB 0x63 -#define CTL_REGIST_PARM_NUM_LSB 0x64 -#define CTL_REGIST_PARM_NUM_MSB 0x65 -/* undefined 0x66 - 0x78 */ -/* reserved 0x79 - 0x7f */ +/* + * Midi controller numbers + * + * Controllers 0 to 31 (0x00 to 0x1f) and 32 to 63 (0x20 to 0x3f) + * are continuous controllers. + * In the MIDI 1.0 these controllers are sent using two messages. + * Controller numbers 0 to 31 are used to send the MSB and the + * controller numbers 32 to 63 are for the LSB. Note that just 7 bits + * are used in MIDI bytes. + */ + +#define CTL_BANK_SELECT 0x00 +#define CTL_MODWHEEL 0x01 +#define CTL_BREATH 0x02 +/* undefined 0x03 */ +#define CTL_FOOT 0x04 +#define CTL_PORTAMENTO_TIME 0x05 +#define CTL_DATA_ENTRY 0x06 +#define CTL_MAIN_VOLUME 0x07 +#define CTL_BALANCE 0x08 +/* undefined 0x09 */ +#define CTL_PAN 0x0a +#define CTL_EXPRESSION 0x0b +/* undefined 0x0c - 0x0f */ +#define CTL_GENERAL_PURPOSE1 0x10 +#define CTL_GENERAL_PURPOSE2 0x11 +#define CTL_GENERAL_PURPOSE3 0x12 +#define CTL_GENERAL_PURPOSE4 0x13 +/* undefined 0x14 - 0x1f */ + +/* undefined 0x20 */ + +/* + * The controller numbers 0x21 to 0x3f are reserved for the + * least significant bytes of the controllers 0x00 to 0x1f. + * These controllers are not recognised by the driver. + * + * Controllers 64 to 69 (0x40 to 0x45) are on/off switches. + * 0=OFF and 127=ON (intermediate values are possible) + */ +#define CTL_DAMPER_PEDAL 0x40 +#define CTL_SUSTAIN CTL_DAMPER_PEDAL /* Alias */ +#define CTL_HOLD CTL_DAMPER_PEDAL /* Alias */ +#define CTL_PORTAMENTO 0x41 +#define CTL_SOSTENUTO 0x42 +#define CTL_SOFT_PEDAL 0x43 +/* undefined 0x44 */ +#define CTL_HOLD2 0x45 +/* undefined 0x46 - 0x4f */ + +#define CTL_GENERAL_PURPOSE5 0x50 +#define CTL_GENERAL_PURPOSE6 0x51 +#define CTL_GENERAL_PURPOSE7 0x52 +#define CTL_GENERAL_PURPOSE8 0x53 +/* undefined 0x54 - 0x5a */ +#define CTL_EXT_EFF_DEPTH 0x5b +#define CTL_TREMOLO_DEPTH 0x5c +#define CTL_CHORUS_DEPTH 0x5d +#define CTL_DETUNE_DEPTH 0x5e +#define CTL_CELESTE_DEPTH CTL_DETUNE_DEPTH /* Alias for the above one */ +#define CTL_PHASER_DEPTH 0x5f +#define CTL_DATA_INCREMENT 0x60 +#define CTL_DATA_DECREMENT 0x61 +#define CTL_NONREG_PARM_NUM_LSB 0x62 +#define CTL_NONREG_PARM_NUM_MSB 0x63 +#define CTL_REGIST_PARM_NUM_LSB 0x64 +#define CTL_REGIST_PARM_NUM_MSB 0x65 +/* undefined 0x66 - 0x78 */ +/* reserved 0x79 - 0x7f */ /* Pseudo controllers (not midi compatible) */ -#define CTRL_PITCH_BENDER 255 -#define CTRL_PITCH_BENDER_RANGE 254 -#define CTRL_EXPRESSION 253 /* Obsolete */ -#define CTRL_MAIN_VOLUME 252 /* Obsolete */ +#define CTRL_PITCH_BENDER 255 +#define CTRL_PITCH_BENDER_RANGE 254 +#define CTRL_EXPRESSION 253 /* Obsolete */ +#define CTRL_MAIN_VOLUME 252 /* Obsolete */ + +#define SEQ_BALANCE 11 +#define SEQ_VOLMODE 12 /* - * Volume mode defines how volumes are used + * Volume mode decides how volumes are used */ #define VOL_METHOD_ADAGIO 1 @@ -437,90 +658,453 @@ struct sysex_info * of the associated synthesizer device. There is no limit to the size * of the extended events. These events are not queued but executed * immediately when the write() is called (execution can take several - * seconds of time). + * seconds of time). * * When a SEQ_FULLSIZE message is written to the device, it must * be written using exactly one write() call. Other events cannot * be mixed to the same write. - * - * For FM synths (YM3812/OPL3) use struct sbi_instrument and write it to the - * /dev/sequencer. Don't write other data together with the instrument structure - * Set the key field of the structure to FM_PATCH. The device field is used to - * route the patch to the corresponding device. * - * For wave table use struct patch_info. Initialize the key field - * to WAVE_PATCH. + * For FM synths (YM3812/OPL3) use struct sbi_instrument and write + * it to the /dev/sequencer. Don't write other data together with + * the instrument structure Set the key field of the structure to + * FM_PATCH. The device field is used to route the patch to the + * corresponding device. + * + * For Gravis UltraSound use struct patch_info. Initialize the key field + * to GUS_PATCH. */ -#define SEQ_PRIVATE 0xfe /* Low level HW dependent events (8 bytes) */ -#define SEQ_EXTENDED 0xff /* Extended events (8 bytes) OBSOLETE */ +#define SEQ_PRIVATE 0xfe /* Low level HW dependent events (8 bytes) */ +#define SEQ_EXTENDED 0xff /* Extended events (8 bytes) OBSOLETE */ /* * Record for FM patches */ -typedef unsigned char sbi_instr_data[32]; +typedef u_char sbi_instr_data[32]; -struct sbi_instrument -{ - unsigned short key; /* FM_PATCH or OPL3_PATCH */ +struct sbi_instrument { + u_short key; /* FM_PATCH or OPL3_PATCH */ #define FM_PATCH _PATCHKEY(0x01) #define OPL3_PATCH _PATCHKEY(0x03) - short device; /* Synth# (0-4) */ - int channel; /* Program# to be initialized */ - sbi_instr_data operators; /* Register settings for operator cells (.SBI format) */ + short device; /* Synth# (0-4) */ + int channel; /* Program# to be initialized */ + sbi_instr_data operators; /* Reg. settings for operator cells + * (.SBI format) */ }; -struct synth_info -{ /* Read only */ - char name[30]; - int device; /* 0-N. INITIALIZE BEFORE CALLING */ - int synth_type; +struct synth_info { /* Read only */ + char name[30]; + int device; /* 0-N. INITIALIZE BEFORE CALLING */ + int synth_type; #define SYNTH_TYPE_FM 0 #define SYNTH_TYPE_SAMPLE 1 #define SYNTH_TYPE_MIDI 2 /* Midi interface */ - int synth_subtype; + int synth_subtype; #define FM_TYPE_ADLIB 0x00 #define FM_TYPE_OPL3 0x01 #define MIDI_TYPE_MPU401 0x401 #define SAMPLE_TYPE_BASIC 0x10 #define SAMPLE_TYPE_GUS SAMPLE_TYPE_BASIC -#define SAMPLE_TYPE_WAVEFRONT 0x11 - - int perc_mode; /* No longer supported */ - int nr_voices; - int nr_drums; /* Obsolete field */ - int instr_bank_size; - unsigned int capabilities; -#define SYNTH_CAP_PERCMODE 0x00000001 /* No longer used */ -#define SYNTH_CAP_OPL3 0x00000002 /* Set if OPL3 supported */ -#define SYNTH_CAP_INPUT 0x00000004 /* Input (MIDI) device */ - int dummies[19]; /* Reserve space */ +#define SAMPLE_TYPE_AWE32 0x20 + + int perc_mode; /* No longer supported */ + int nr_voices; + int nr_drums; /* Obsolete field */ + int instr_bank_size; + u_long capabilities; +#define SYNTH_CAP_PERCMODE 0x00000001 /* No longer used */ +#define SYNTH_CAP_OPL3 0x00000002 /* Set if OPL3 supported */ +#define SYNTH_CAP_INPUT 0x00000004 /* Input (MIDI) device */ + int dummies[19]; /* Reserve space */ }; -struct sound_timer_info -{ - char name[32]; - int caps; +struct sound_timer_info { + char name[32]; + int caps; }; -struct midi_info /* OBSOLETE */ -{ - char name[30]; - int device; /* 0-N. INITIALIZE BEFORE CALLING */ - unsigned int capabilities; /* To be defined later */ - int dev_type; - int dummies[18]; /* Reserve space */ +struct midi_info { + char name[30]; + int device; /* 0-N. INITIALIZE BEFORE CALLING */ + u_long capabilities; /* To be defined later */ + int dev_type; + int dummies[18]; /* Reserve space */ }; +/* + * ioctl commands for the /dev/midi## + */ +typedef struct { + u_char cmd; + char nr_args, nr_returns; + u_char data[30]; +} mpu_command_rec; + +#define SNDCTL_MIDI_PRETIME _IOWR('m', 0, int) +#define SNDCTL_MIDI_MPUMODE _IOWR('m', 1, int) +#define SNDCTL_MIDI_MPUCMD _IOWR('m', 2, mpu_command_rec) +#define MIOSPASSTHRU _IOWR('m', 3, int) +#define MIOGPASSTHRU _IOWR('m', 4, int) + +/* + * IOCTL commands for /dev/dsp and /dev/audio + */ + +#define SNDCTL_DSP_RESET _IO ('P', 0) +#define SNDCTL_DSP_SYNC _IO ('P', 1) +#define SNDCTL_DSP_SPEED _IOWR('P', 2, int) +#define SNDCTL_DSP_STEREO _IOWR('P', 3, int) +#define SNDCTL_DSP_GETBLKSIZE _IOR('P', 4, int) +#define SNDCTL_DSP_SETBLKSIZE _IOW('P', 4, int) +#define SNDCTL_DSP_SETFMT _IOWR('P',5, int) /* Selects ONE fmt*/ + +/* + * SOUND_PCM_WRITE_CHANNELS is not that different + * from SNDCTL_DSP_STEREO + */ +#define SOUND_PCM_WRITE_CHANNELS _IOWR('P', 6, int) +#define SNDCTL_DSP_CHANNELS SOUND_PCM_WRITE_CHANNELS +#define SOUND_PCM_WRITE_FILTER _IOWR('P', 7, int) +#define SNDCTL_DSP_POST _IO ('P', 8) + +/* + * SNDCTL_DSP_SETBLKSIZE and the following two calls mostly do + * the same thing, i.e. set the block size used in DMA transfers. + */ +#define SNDCTL_DSP_SUBDIVIDE _IOWR('P', 9, int) +#define SNDCTL_DSP_SETFRAGMENT _IOWR('P',10, int) + + +#define SNDCTL_DSP_GETFMTS _IOR ('P',11, int) /* Returns a mask */ +/* + * Buffer status queries. + */ +typedef struct audio_buf_info { + int fragments; /* # of avail. frags (partly used ones not counted) */ + int fragstotal; /* Total # of fragments allocated */ + int fragsize; /* Size of a fragment in bytes */ + + int bytes; /* Avail. space in bytes (includes partly used fragments) */ + /* Note! 'bytes' could be more than fragments*fragsize */ +} audio_buf_info; + +#define SNDCTL_DSP_GETOSPACE _IOR ('P',12, audio_buf_info) +#define SNDCTL_DSP_GETISPACE _IOR ('P',13, audio_buf_info) + +/* + * SNDCTL_DSP_NONBLOCK is the same (but less powerful, since the + * action cannot be undone) of FIONBIO. The same can be achieved + * by opening the device with O_NDELAY + */ +#define SNDCTL_DSP_NONBLOCK _IO ('P',14) + +#define SNDCTL_DSP_GETCAPS _IOR ('P',15, int) +#define DSP_CAP_REVISION 0x000000ff /* revision level (0 to 255) */ +#define DSP_CAP_DUPLEX 0x00000100 /* Full duplex record/playback */ +#define DSP_CAP_REALTIME 0x00000200 /* Real time capability */ +#define DSP_CAP_BATCH 0x00000400 + /* + * Device has some kind of internal buffers which may + * cause some delays and decrease precision of timing + */ +#define DSP_CAP_COPROC 0x00000800 + /* Has a coprocessor, sometimes it's a DSP but usually not */ +#define DSP_CAP_TRIGGER 0x00001000 /* Supports SETTRIGGER */ +#define DSP_CAP_MMAP 0x00002000 /* Supports mmap() */ + +/* + * What do these function do ? + */ +#define SNDCTL_DSP_GETTRIGGER _IOR ('P',16, int) +#define SNDCTL_DSP_SETTRIGGER _IOW ('P',16, int) +#define PCM_ENABLE_INPUT 0x00000001 +#define PCM_ENABLE_OUTPUT 0x00000002 + +typedef struct count_info { + int bytes; /* Total # of bytes processed */ + int blocks; /* # of fragment transitions since last time */ + int ptr; /* Current DMA pointer value */ +} count_info; + +/* + * GETIPTR and GETISPACE are not that different... same for out. + */ +#define SNDCTL_DSP_GETIPTR _IOR ('P',17, count_info) +#define SNDCTL_DSP_GETOPTR _IOR ('P',18, count_info) + +typedef struct buffmem_desc { + caddr_t buffer; + int size; +} buffmem_desc; + +#define SNDCTL_DSP_MAPINBUF _IOR ('P', 19, buffmem_desc) +#define SNDCTL_DSP_MAPOUTBUF _IOR ('P', 20, buffmem_desc) +#define SNDCTL_DSP_SETSYNCRO _IO ('P', 21) +#define SNDCTL_DSP_SETDUPLEX _IO ('P', 22) +#define SNDCTL_DSP_GETODELAY _IOR ('P', 23, int) + +/* + * I guess these are the readonly version of the same + * functions that exist above as SNDCTL_DSP_... + */ +#define SOUND_PCM_READ_RATE _IOR ('P', 2, int) +#define SOUND_PCM_READ_CHANNELS _IOR ('P', 6, int) +#define SOUND_PCM_READ_BITS _IOR ('P', 5, int) +#define SOUND_PCM_READ_FILTER _IOR ('P', 7, int) + +/* + * ioctl calls to be used in communication with coprocessors and + * DSP chips. + */ + +typedef struct copr_buffer { + int command; /* Set to 0 if not used */ + int flags; +#define CPF_NONE 0x0000 +#define CPF_FIRST 0x0001 /* First block */ +#define CPF_LAST 0x0002 /* Last block */ + int len; + int offs; /* If required by the device (0 if not used) */ + + u_char data[4000]; /* NOTE! 4000 is not 4k */ +} copr_buffer; + +typedef struct copr_debug_buf { + int command; /* Used internally. Set to 0 */ + int parm1; + int parm2; + int flags; + int len; /* Length of data in bytes */ +} copr_debug_buf; + +typedef struct copr_msg { + int len; + u_char data[4000]; +} copr_msg; + +#define SNDCTL_COPR_RESET _IO ('C', 0) +#define SNDCTL_COPR_LOAD _IOWR('C', 1, copr_buffer) +#define SNDCTL_COPR_RDATA _IOWR('C', 2, copr_debug_buf) +#define SNDCTL_COPR_RCODE _IOWR('C', 3, copr_debug_buf) +#define SNDCTL_COPR_WDATA _IOW ('C', 4, copr_debug_buf) +#define SNDCTL_COPR_WCODE _IOW ('C', 5, copr_debug_buf) +#define SNDCTL_COPR_RUN _IOWR('C', 6, copr_debug_buf) +#define SNDCTL_COPR_HALT _IOWR('C', 7, copr_debug_buf) +#define SNDCTL_COPR_SENDMSG _IOW ('C', 8, copr_msg) +#define SNDCTL_COPR_RCVMSG _IOR ('C', 9, copr_msg) + +/* + * IOCTL commands for /dev/mixer + */ + +/* + * Mixer devices + * + * There can be up to 20 different analog mixer channels. The + * SOUND_MIXER_NRDEVICES gives the currently supported maximum. + * The SOUND_MIXER_READ_DEVMASK returns a bitmask which tells + * the devices supported by the particular mixer. + */ + +#define SOUND_MIXER_NRDEVICES 25 +#define SOUND_MIXER_VOLUME 0 /* Master output level */ +#define SOUND_MIXER_BASS 1 /* Treble level of all output channels */ +#define SOUND_MIXER_TREBLE 2 /* Bass level of all output channels */ +#define SOUND_MIXER_SYNTH 3 /* Volume of synthesier input */ +#define SOUND_MIXER_PCM 4 /* Output level for the audio device */ +#define SOUND_MIXER_SPEAKER 5 /* Output level for the PC speaker + * signals */ +#define SOUND_MIXER_LINE 6 /* Volume level for the line in jack */ +#define SOUND_MIXER_MIC 7 /* Volume for the signal coming from + * the microphone jack */ +#define SOUND_MIXER_CD 8 /* Volume level for the input signal + * connected to the CD audio input */ +#define SOUND_MIXER_IMIX 9 /* Recording monitor. It controls the + * output volume of the selected + * recording sources while recording */ +#define SOUND_MIXER_ALTPCM 10 /* Volume of the alternative codec + * device */ +#define SOUND_MIXER_RECLEV 11 /* Global recording level */ +#define SOUND_MIXER_IGAIN 12 /* Input gain */ +#define SOUND_MIXER_OGAIN 13 /* Output gain */ +/* + * The AD1848 codec and compatibles have three line level inputs + * (line, aux1 and aux2). Since each card manufacturer have assigned + * different meanings to these inputs, it's inpractical to assign + * specific meanings (line, cd, synth etc.) to them. + */ +#define SOUND_MIXER_LINE1 14 /* Input source 1 (aux1) */ +#define SOUND_MIXER_LINE2 15 /* Input source 2 (aux2) */ +#define SOUND_MIXER_LINE3 16 /* Input source 3 (line) */ +#define SOUND_MIXER_DIGITAL1 17 /* Digital (input) 1 */ +#define SOUND_MIXER_DIGITAL2 18 /* Digital (input) 2 */ +#define SOUND_MIXER_DIGITAL3 19 /* Digital (input) 3 */ +#define SOUND_MIXER_PHONEIN 20 /* Phone input */ +#define SOUND_MIXER_PHONEOUT 21 /* Phone output */ +#define SOUND_MIXER_VIDEO 22 /* Video/TV (audio) in */ +#define SOUND_MIXER_RADIO 23 /* Radio in */ +#define SOUND_MIXER_MONITOR 24 /* Monitor (usually mic) volume */ + + +/* + * Some on/off settings (SOUND_SPECIAL_MIN - SOUND_SPECIAL_MAX) + * Not counted to SOUND_MIXER_NRDEVICES, but use the same number space + */ +#define SOUND_ONOFF_MIN 28 +#define SOUND_ONOFF_MAX 30 +#define SOUND_MIXER_MUTE 28 /* 0 or 1 */ +#define SOUND_MIXER_ENHANCE 29 /* Enhanced stereo (0, 40, 60 or 80) */ +#define SOUND_MIXER_LOUD 30 /* 0 or 1 */ + +/* Note! Number 31 cannot be used since the sign bit is reserved */ +#define SOUND_MIXER_NONE 31 + +#define SOUND_DEVICE_LABELS { \ + "Vol ", "Bass ", "Trebl", "Synth", "Pcm ", "Spkr ", "Line ", \ + "Mic ", "CD ", "Mix ", "Pcm2 ", "Rec ", "IGain", "OGain", \ + "Line1", "Line2", "Line3", "Digital1", "Digital2", "Digital3", \ + "PhoneIn", "PhoneOut", "Video", "Radio", "Monitor"} + +#define SOUND_DEVICE_NAMES { \ + "vol", "bass", "treble", "synth", "pcm", "speaker", "line", \ + "mic", "cd", "mix", "pcm2", "rec", "igain", "ogain", \ + "line1", "line2", "line3", "dig1", "dig2", "dig3", \ + "phin", "phout", "video", "radio", "monitor"} + +/* Device bitmask identifiers */ + +#define SOUND_MIXER_RECSRC 0xff /* 1 bit per recording source */ +#define SOUND_MIXER_DEVMASK 0xfe /* 1 bit per supported device */ +#define SOUND_MIXER_RECMASK 0xfd /* 1 bit per supp. recording source */ +#define SOUND_MIXER_CAPS 0xfc +#define SOUND_CAP_EXCL_INPUT 0x00000001 /* Only 1 rec. src at a time */ +#define SOUND_MIXER_STEREODEVS 0xfb /* Mixer channels supporting stereo */ + +/* Device mask bits */ + +#define SOUND_MASK_VOLUME (1 << SOUND_MIXER_VOLUME) +#define SOUND_MASK_BASS (1 << SOUND_MIXER_BASS) +#define SOUND_MASK_TREBLE (1 << SOUND_MIXER_TREBLE) +#define SOUND_MASK_SYNTH (1 << SOUND_MIXER_SYNTH) +#define SOUND_MASK_PCM (1 << SOUND_MIXER_PCM) +#define SOUND_MASK_SPEAKER (1 << SOUND_MIXER_SPEAKER) +#define SOUND_MASK_LINE (1 << SOUND_MIXER_LINE) +#define SOUND_MASK_MIC (1 << SOUND_MIXER_MIC) +#define SOUND_MASK_CD (1 << SOUND_MIXER_CD) +#define SOUND_MASK_IMIX (1 << SOUND_MIXER_IMIX) +#define SOUND_MASK_ALTPCM (1 << SOUND_MIXER_ALTPCM) +#define SOUND_MASK_RECLEV (1 << SOUND_MIXER_RECLEV) +#define SOUND_MASK_IGAIN (1 << SOUND_MIXER_IGAIN) +#define SOUND_MASK_OGAIN (1 << SOUND_MIXER_OGAIN) +#define SOUND_MASK_LINE1 (1 << SOUND_MIXER_LINE1) +#define SOUND_MASK_LINE2 (1 << SOUND_MIXER_LINE2) +#define SOUND_MASK_LINE3 (1 << SOUND_MIXER_LINE3) +#define SOUND_MASK_DIGITAL1 (1 << SOUND_MIXER_DIGITAL1) +#define SOUND_MASK_DIGITAL2 (1 << SOUND_MIXER_DIGITAL2) +#define SOUND_MASK_DIGITAL3 (1 << SOUND_MIXER_DIGITAL3) +#define SOUND_MASK_PHONEIN (1 << SOUND_MIXER_PHONEIN) +#define SOUND_MASK_PHONEOUT (1 << SOUND_MIXER_PHONEOUT) +#define SOUND_MASK_RADIO (1 << SOUND_MIXER_RADIO) +#define SOUND_MASK_VIDEO (1 << SOUND_MIXER_VIDEO) +#define SOUND_MASK_MONITOR (1 << SOUND_MIXER_MONITOR) + +/* Obsolete macros */ +#define SOUND_MASK_MUTE (1 << SOUND_MIXER_MUTE) +#define SOUND_MASK_ENHANCE (1 << SOUND_MIXER_ENHANCE) +#define SOUND_MASK_LOUD (1 << SOUND_MIXER_LOUD) + +#define MIXER_READ(dev) _IOR('M', dev, int) +#define SOUND_MIXER_READ_VOLUME MIXER_READ(SOUND_MIXER_VOLUME) +#define SOUND_MIXER_READ_BASS MIXER_READ(SOUND_MIXER_BASS) +#define SOUND_MIXER_READ_TREBLE MIXER_READ(SOUND_MIXER_TREBLE) +#define SOUND_MIXER_READ_SYNTH MIXER_READ(SOUND_MIXER_SYNTH) +#define SOUND_MIXER_READ_PCM MIXER_READ(SOUND_MIXER_PCM) +#define SOUND_MIXER_READ_SPEAKER MIXER_READ(SOUND_MIXER_SPEAKER) +#define SOUND_MIXER_READ_LINE MIXER_READ(SOUND_MIXER_LINE) +#define SOUND_MIXER_READ_MIC MIXER_READ(SOUND_MIXER_MIC) +#define SOUND_MIXER_READ_CD MIXER_READ(SOUND_MIXER_CD) +#define SOUND_MIXER_READ_IMIX MIXER_READ(SOUND_MIXER_IMIX) +#define SOUND_MIXER_READ_ALTPCM MIXER_READ(SOUND_MIXER_ALTPCM) +#define SOUND_MIXER_READ_RECLEV MIXER_READ(SOUND_MIXER_RECLEV) +#define SOUND_MIXER_READ_IGAIN MIXER_READ(SOUND_MIXER_IGAIN) +#define SOUND_MIXER_READ_OGAIN MIXER_READ(SOUND_MIXER_OGAIN) +#define SOUND_MIXER_READ_LINE1 MIXER_READ(SOUND_MIXER_LINE1) +#define SOUND_MIXER_READ_LINE2 MIXER_READ(SOUND_MIXER_LINE2) +#define SOUND_MIXER_READ_LINE3 MIXER_READ(SOUND_MIXER_LINE3) +#define SOUND_MIXER_READ_DIGITAL1 MIXER_READ(SOUND_MIXER_DIGITAL1) +#define SOUND_MIXER_READ_DIGITAL2 MIXER_READ(SOUND_MIXER_DIGITAL2) +#define SOUND_MIXER_READ_DIGITAL3 MIXER_READ(SOUND_MIXER_DIGITAL3) +#define SOUND_MIXER_READ_PHONEIN MIXER_READ(SOUND_MIXER_PHONEIN) +#define SOUND_MIXER_READ_PHONEOUT MIXER_READ(SOUND_MIXER_PHONEOUT) +#define SOUND_MIXER_READ_RADIO MIXER_READ(SOUND_MIXER_RADIO) +#define SOUND_MIXER_READ_VIDEO MIXER_READ(SOUND_MIXER_VIDEO) +#define SOUND_MIXER_READ_MONITOR MIXER_READ(SOUND_MIXER_MONITOR) + +/* Obsolete macros */ +#define SOUND_MIXER_READ_MUTE MIXER_READ(SOUND_MIXER_MUTE) +#define SOUND_MIXER_READ_ENHANCE MIXER_READ(SOUND_MIXER_ENHANCE) +#define SOUND_MIXER_READ_LOUD MIXER_READ(SOUND_MIXER_LOUD) + +#define SOUND_MIXER_READ_RECSRC MIXER_READ(SOUND_MIXER_RECSRC) +#define SOUND_MIXER_READ_DEVMASK MIXER_READ(SOUND_MIXER_DEVMASK) +#define SOUND_MIXER_READ_RECMASK MIXER_READ(SOUND_MIXER_RECMASK) +#define SOUND_MIXER_READ_STEREODEVS MIXER_READ(SOUND_MIXER_STEREODEVS) +#define SOUND_MIXER_READ_CAPS MIXER_READ(SOUND_MIXER_CAPS) + +#define MIXER_WRITE(dev) _IOWR('M', dev, int) +#define SOUND_MIXER_WRITE_VOLUME MIXER_WRITE(SOUND_MIXER_VOLUME) +#define SOUND_MIXER_WRITE_BASS MIXER_WRITE(SOUND_MIXER_BASS) +#define SOUND_MIXER_WRITE_TREBLE MIXER_WRITE(SOUND_MIXER_TREBLE) +#define SOUND_MIXER_WRITE_SYNTH MIXER_WRITE(SOUND_MIXER_SYNTH) +#define SOUND_MIXER_WRITE_PCM MIXER_WRITE(SOUND_MIXER_PCM) +#define SOUND_MIXER_WRITE_SPEAKER MIXER_WRITE(SOUND_MIXER_SPEAKER) +#define SOUND_MIXER_WRITE_LINE MIXER_WRITE(SOUND_MIXER_LINE) +#define SOUND_MIXER_WRITE_MIC MIXER_WRITE(SOUND_MIXER_MIC) +#define SOUND_MIXER_WRITE_CD MIXER_WRITE(SOUND_MIXER_CD) +#define SOUND_MIXER_WRITE_IMIX MIXER_WRITE(SOUND_MIXER_IMIX) +#define SOUND_MIXER_WRITE_ALTPCM MIXER_WRITE(SOUND_MIXER_ALTPCM) +#define SOUND_MIXER_WRITE_RECLEV MIXER_WRITE(SOUND_MIXER_RECLEV) +#define SOUND_MIXER_WRITE_IGAIN MIXER_WRITE(SOUND_MIXER_IGAIN) +#define SOUND_MIXER_WRITE_OGAIN MIXER_WRITE(SOUND_MIXER_OGAIN) +#define SOUND_MIXER_WRITE_LINE1 MIXER_WRITE(SOUND_MIXER_LINE1) +#define SOUND_MIXER_WRITE_LINE2 MIXER_WRITE(SOUND_MIXER_LINE2) +#define SOUND_MIXER_WRITE_LINE3 MIXER_WRITE(SOUND_MIXER_LINE3) +#define SOUND_MIXER_WRITE_DIGITAL1 MIXER_WRITE(SOUND_MIXER_DIGITAL1) +#define SOUND_MIXER_WRITE_DIGITAL2 MIXER_WRITE(SOUND_MIXER_DIGITAL2) +#define SOUND_MIXER_WRITE_DIGITAL3 MIXER_WRITE(SOUND_MIXER_DIGITAL3) +#define SOUND_MIXER_WRITE_PHONEIN MIXER_WRITE(SOUND_MIXER_PHONEIN) +#define SOUND_MIXER_WRITE_PHONEOUT MIXER_WRITE(SOUND_MIXER_PHONEOUT) +#define SOUND_MIXER_WRITE_RADIO MIXER_WRITE(SOUND_MIXER_RADIO) +#define SOUND_MIXER_WRITE_VIDEO MIXER_WRITE(SOUND_MIXER_VIDEO) +#define SOUND_MIXER_WRITE_MONITOR MIXER_WRITE(SOUND_MIXER_MONITOR) + +#define SOUND_MIXER_WRITE_MUTE MIXER_WRITE(SOUND_MIXER_MUTE) +#define SOUND_MIXER_WRITE_ENHANCE MIXER_WRITE(SOUND_MIXER_ENHANCE) +#define SOUND_MIXER_WRITE_LOUD MIXER_WRITE(SOUND_MIXER_LOUD) + +#define SOUND_MIXER_WRITE_RECSRC MIXER_WRITE(SOUND_MIXER_RECSRC) + +typedef struct mixer_info { + char id[16]; + char name[32]; + int modify_counter; + int fillers[10]; +} mixer_info; + +#define SOUND_MIXER_INFO _IOR('M', 101, mixer_info) + +#define LEFT_CHN 0 +#define RIGHT_CHN 1 + /* * Level 2 event types for /dev/sequencer */ /* * The 4 most significant bits of byte 0 specify the class of - * the event: + * the event: * * 0x8X = system level events, * 0x9X = device/port specific events, event[1] = device/port, @@ -537,7 +1121,6 @@ struct midi_info /* OBSOLETE */ #define EV_CHN_COMMON 0x92 #define EV_CHN_VOICE 0x93 #define EV_SYSEX 0x94 -#define EV_SYSTEM 0x95 /* MIDI system and real time messages (input only) */ /* * Event types 200 to 220 are reserved for application use. * These numbers will not be used by the driver. @@ -580,57 +1163,18 @@ struct midi_info /* OBSOLETE */ * Local event types */ #define LOCL_STARTAUDIO 1 -#define LOCL_STARTAUDIO2 2 -#define LOCL_STARTAUDIO3 3 -#define LOCL_STARTAUDIO4 4 -#if (!defined(__KERNEL__) && !defined(KERNEL) && !defined(INKERNEL) && !defined(_KERNEL)) || defined(USE_SEQ_MACROS) +#if (!defined(_KERNEL) && !defined(INKERNEL)) || defined(USE_SEQ_MACROS) /* - * Some convenience macros to simplify programming of the - * /dev/sequencer interface + * Some convenience macros to simplify programming of the + * /dev/sequencer interface * - * These macros define the API which should be used when possible. + * These macros define the API which should be used when possible. */ -#define SEQ_DECLAREBUF() SEQ_USE_EXTBUF() - -void seqbuf_dump (void); /* This function must be provided by programs */ - -EXTERNC int OSS_init (int seqfd, int buflen); -EXTERNC void OSS_seqbuf_dump (int fd, unsigned char *buf, int buflen); -EXTERNC void OSS_seq_advbuf (int len, int fd, unsigned char *buf, int buflen); -EXTERNC void OSS_seq_needbuf (int len, int fd, unsigned char *buf, - int buflen); -EXTERNC void OSS_patch_caching (int dev, int chn, int patch, int fd, - unsigned char *buf, int buflen); -EXTERNC void OSS_drum_caching (int dev, int chn, int patch, int fd, - unsigned char *buf, int buflen); -EXTERNC void OSS_write_patch (int fd, unsigned char *buf, int len); -EXTERNC int OSS_write_patch2 (int fd, unsigned char *buf, int len); - -#define SEQ_PM_DEFINES int __foo_bar___ -#ifdef OSSLIB -# define SEQ_USE_EXTBUF() \ - EXTERNC unsigned char *_seqbuf; \ - EXTERNC int _seqbuflen;EXTERNC int _seqbufptr -# define SEQ_DEFINEBUF(len) SEQ_USE_EXTBUF();static int _requested_seqbuflen=len -# define _SEQ_ADVBUF(len) OSS_seq_advbuf(len, seqfd, _seqbuf, _seqbuflen) -# define _SEQ_NEEDBUF(len) OSS_seq_needbuf(len, seqfd, _seqbuf, _seqbuflen) -# define SEQ_DUMPBUF() OSS_seqbuf_dump(seqfd, _seqbuf, _seqbuflen) - -# define SEQ_LOAD_GMINSTR(dev, instr) \ - OSS_patch_caching(dev, -1, instr, seqfd, _seqbuf, _seqbuflen) -# define SEQ_LOAD_GMDRUM(dev, drum) \ - OSS_drum_caching(dev, -1, drum, seqfd, _seqbuf, _seqbuflen) -#else /* !OSSLIB */ - -# define SEQ_LOAD_GMINSTR(dev, instr) -# define SEQ_LOAD_GMDRUM(dev, drum) - -# define SEQ_USE_EXTBUF() \ - EXTERNC unsigned char _seqbuf[]; \ - EXTERNC int _seqbuflen;EXTERNC int _seqbufptr #ifndef USE_SIMPLE_MACROS +void seqbuf_dump(void); /* This function must be provided by programs */ + /* Sample seqbuf_dump() implementation: * * SEQ_DEFINEBUF (2048); -- Defines a buffer for 2048 bytes @@ -650,88 +1194,104 @@ EXTERNC int OSS_write_patch2 (int fd, unsigned char *buf, int len); * } */ -#define SEQ_DEFINEBUF(len) \ - unsigned char _seqbuf[len]; int _seqbuflen = len;int _seqbufptr = 0 -#define _SEQ_NEEDBUF(len) \ - if ((_seqbufptr+(len)) > _seqbuflen) seqbuf_dump() -#define _SEQ_ADVBUF(len) _seqbufptr += len -#define SEQ_DUMPBUF seqbuf_dump +#define SEQ_DEFINEBUF(len) \ + u_char _seqbuf[len]; int _seqbuflen = len;int _seqbufptr = 0 +#define SEQ_USE_EXTBUF() \ + extern u_char _seqbuf[]; \ + extern int _seqbuflen;extern int _seqbufptr +#define SEQ_DECLAREBUF() SEQ_USE_EXTBUF() +#define SEQ_PM_DEFINES struct patmgr_info _pm_info +#define _SEQ_NEEDBUF(len) \ + if ((_seqbufptr+(len)) > _seqbuflen) \ + seqbuf_dump() +#define _SEQ_ADVBUF(len) _seqbufptr += len +#define SEQ_DUMPBUF seqbuf_dump #else /* * This variation of the sequencer macros is used just to format one event * using fixed buffer. - * + * * The program using the macro library must define the following macros before * using this library. * - * #define _seqbuf name of the buffer (unsigned char[]) + * #define _seqbuf name of the buffer (u_char[]) * #define _SEQ_ADVBUF(len) If the applic needs to know the exact * size of the event, this macro can be used. * Otherwise this must be defined as empty. * #define _seqbufptr Define the name of index variable or 0 if - * not required. + * not required. */ #define _SEQ_NEEDBUF(len) /* empty */ #endif -#endif /* !OSSLIB */ -#define SEQ_VOLUME_MODE(dev, mode) \ - {_SEQ_NEEDBUF(8);\ - _seqbuf[_seqbufptr] = SEQ_EXTENDED;\ - _seqbuf[_seqbufptr+1] = SEQ_VOLMODE;\ - _seqbuf[_seqbufptr+2] = (dev);\ - _seqbuf[_seqbufptr+3] = (mode);\ - _seqbuf[_seqbufptr+4] = 0;\ - _seqbuf[_seqbufptr+5] = 0;\ - _seqbuf[_seqbufptr+6] = 0;\ - _seqbuf[_seqbufptr+7] = 0;\ - _SEQ_ADVBUF(8);} +#define PM_LOAD_PATCH(dev, bank, pgm) \ + (SEQ_DUMPBUF(), _pm_info.command = _PM_LOAD_PATCH, \ + _pm_info.device=dev, _pm_info.data.data8[0]=pgm, \ + _pm_info.parm1 = bank, _pm_info.parm2 = 1, \ + ioctl(seqfd, SNDCTL_PMGR_ACCESS, &_pm_info)) +#define PM_LOAD_PATCHES(dev, bank, pgm) \ + (SEQ_DUMPBUF(), _pm_info.command = _PM_LOAD_PATCH, \ + _pm_info.device=dev, bcopy( pgm, _pm_info.data.data8, 128), \ + _pm_info.parm1 = bank, _pm_info.parm2 = 128, \ + ioctl(seqfd, SNDCTL_PMGR_ACCESS, &_pm_info)) + +#define SEQ_VOLUME_MODE(dev, mode) { \ + _SEQ_NEEDBUF(8);\ + _seqbuf[_seqbufptr] = SEQ_EXTENDED;\ + _seqbuf[_seqbufptr+1] = SEQ_VOLMODE;\ + _seqbuf[_seqbufptr+2] = (dev);\ + _seqbuf[_seqbufptr+3] = (mode);\ + _seqbuf[_seqbufptr+4] = 0;\ + _seqbuf[_seqbufptr+5] = 0;\ + _seqbuf[_seqbufptr+6] = 0;\ + _seqbuf[_seqbufptr+7] = 0;\ + _SEQ_ADVBUF(8);} /* * Midi voice messages */ -#define _CHN_VOICE(dev, event, chn, note, parm) \ - {_SEQ_NEEDBUF(8);\ - _seqbuf[_seqbufptr] = EV_CHN_VOICE;\ - _seqbuf[_seqbufptr+1] = (dev);\ - _seqbuf[_seqbufptr+2] = (event);\ - _seqbuf[_seqbufptr+3] = (chn);\ - _seqbuf[_seqbufptr+4] = (note);\ - _seqbuf[_seqbufptr+5] = (parm);\ - _seqbuf[_seqbufptr+6] = (0);\ - _seqbuf[_seqbufptr+7] = 0;\ - _SEQ_ADVBUF(8);} +#define _CHN_VOICE(dev, event, chn, note, parm) { \ + _SEQ_NEEDBUF(8);\ + _seqbuf[_seqbufptr] = EV_CHN_VOICE;\ + _seqbuf[_seqbufptr+1] = (dev);\ + _seqbuf[_seqbufptr+2] = (event);\ + _seqbuf[_seqbufptr+3] = (chn);\ + _seqbuf[_seqbufptr+4] = (note);\ + _seqbuf[_seqbufptr+5] = (parm);\ + _seqbuf[_seqbufptr+6] = (0);\ + _seqbuf[_seqbufptr+7] = 0;\ + _SEQ_ADVBUF(8);} #define SEQ_START_NOTE(dev, chn, note, vol) \ - _CHN_VOICE(dev, MIDI_NOTEON, chn, note, vol) + _CHN_VOICE(dev, MIDI_NOTEON, chn, note, vol) #define SEQ_STOP_NOTE(dev, chn, note, vol) \ - _CHN_VOICE(dev, MIDI_NOTEOFF, chn, note, vol) + _CHN_VOICE(dev, MIDI_NOTEOFF, chn, note, vol) #define SEQ_KEY_PRESSURE(dev, chn, note, pressure) \ - _CHN_VOICE(dev, MIDI_KEY_PRESSURE, chn, note, pressure) + _CHN_VOICE(dev, MIDI_KEY_PRESSURE, chn, note, pressure) /* * Midi channel messages */ -#define _CHN_COMMON(dev, event, chn, p1, p2, w14) \ - {_SEQ_NEEDBUF(8);\ - _seqbuf[_seqbufptr] = EV_CHN_COMMON;\ - _seqbuf[_seqbufptr+1] = (dev);\ - _seqbuf[_seqbufptr+2] = (event);\ - _seqbuf[_seqbufptr+3] = (chn);\ - _seqbuf[_seqbufptr+4] = (p1);\ - _seqbuf[_seqbufptr+5] = (p2);\ - *(short *)&_seqbuf[_seqbufptr+6] = (w14);\ - _SEQ_ADVBUF(8);} +#define _CHN_COMMON(dev, event, chn, p1, p2, w14) { \ + _SEQ_NEEDBUF(8);\ + _seqbuf[_seqbufptr] = EV_CHN_COMMON;\ + _seqbuf[_seqbufptr+1] = (dev);\ + _seqbuf[_seqbufptr+2] = (event);\ + _seqbuf[_seqbufptr+3] = (chn);\ + _seqbuf[_seqbufptr+4] = (p1);\ + _seqbuf[_seqbufptr+5] = (p2);\ + *(short *)&_seqbuf[_seqbufptr+6] = (w14);\ + _SEQ_ADVBUF(8);} /* * SEQ_SYSEX permits sending of sysex messages. (It may look that it permits * sending any MIDI bytes but it's absolutely not possible. Trying to do * so _will_ cause problems with MPU401 intelligent mode). * - * Sysex messages are sent in blocks of 1 to 6 bytes. Longer messages must be + * Sysex messages are sent in blocks of 1 to 6 bytes. Longer messages must be * sent by calling SEQ_SYSEX() several times (there must be no other events * between them). First sysex fragment must have 0xf0 in the first byte * and the last byte (buf[len-1] of the last fragment must be 0xf7. No byte @@ -741,53 +1301,43 @@ EXTERNC int OSS_write_patch2 (int fd, unsigned char *buf, int len); * Breaking the above rules may work with some MIDI ports but is likely to * cause fatal problems with some other devices (such as MPU401). */ -#define SEQ_SYSEX(dev, buf, len) \ - {int ii, ll=(len); \ - unsigned char *bufp=buf;\ - if (ll>6)ll=6;\ - _SEQ_NEEDBUF(8);\ - _seqbuf[_seqbufptr] = EV_SYSEX;\ - _seqbuf[_seqbufptr+1] = (dev);\ - for(ii=0;ii6)l=6;\ + _SEQ_NEEDBUF(8);\ + _seqbuf[_seqbufptr] = EV_SYSEX;\ + for(i=0;i>8)&0xff);\ - _seqbuf[_seqbufptr+7] = 0;\ - _SEQ_ADVBUF(8);} + _CHN_COMMON(dev, MIDI_PITCH_BEND, chn, 0, 0, value) + + +#define SEQ_V2_X_CONTROL(dev, voice, controller, value) { \ + _SEQ_NEEDBUF(8);\ + _seqbuf[_seqbufptr] = SEQ_EXTENDED;\ + _seqbuf[_seqbufptr+1] = SEQ_CONTROLLER;\ + _seqbuf[_seqbufptr+2] = (dev);\ + _seqbuf[_seqbufptr+3] = (voice);\ + _seqbuf[_seqbufptr+4] = (controller);\ + *(short *)&_seqbuf[_seqbufptr+5] = (value);\ + _seqbuf[_seqbufptr+7] = 0;\ + _SEQ_ADVBUF(8);} + /* * The following 5 macros are incorrectly implemented and obsolete. * Use SEQ_BENDER and SEQ_CONTROL (with proper controller) instead. */ + #define SEQ_PITCHBEND(dev, voice, value) \ SEQ_V2_X_CONTROL(dev, voice, CTRL_PITCH_BENDER, value) #define SEQ_BENDER_RANGE(dev, voice, value) \ @@ -803,13 +1353,15 @@ EXTERNC int OSS_write_patch2 (int fd, unsigned char *buf, int len); * Timing and syncronization macros */ -#define _TIMER_EVENT(ev, parm) {_SEQ_NEEDBUF(8);\ - _seqbuf[_seqbufptr+0] = EV_TIMING; \ - _seqbuf[_seqbufptr+1] = (ev); \ - _seqbuf[_seqbufptr+2] = 0;\ - _seqbuf[_seqbufptr+3] = 0;\ - *(unsigned int *)&_seqbuf[_seqbufptr+4] = (parm); \ - _SEQ_ADVBUF(8);} +#define _TIMER_EVENT(ev, parm) { \ + _SEQ_NEEDBUF(8);\ + _seqbuf[_seqbufptr+0] = EV_TIMING; \ + _seqbuf[_seqbufptr+1] = (ev); \ + _seqbuf[_seqbufptr+2] = 0;\ + _seqbuf[_seqbufptr+3] = 0;\ + *(u_int *)&_seqbuf[_seqbufptr+4] = (parm); \ + _SEQ_ADVBUF(8); \ + } #define SEQ_START_TIMER() _TIMER_EVENT(TMR_START, 0) #define SEQ_STOP_TIMER() _TIMER_EVENT(TMR_STOP, 0) @@ -825,502 +1377,126 @@ EXTERNC int OSS_write_patch2 (int fd, unsigned char *buf, int len); * Local control events */ -#define _LOCAL_EVENT(ev, parm) {_SEQ_NEEDBUF(8);\ - _seqbuf[_seqbufptr+0] = EV_SEQ_LOCAL; \ - _seqbuf[_seqbufptr+1] = (ev); \ - _seqbuf[_seqbufptr+2] = 0;\ - _seqbuf[_seqbufptr+3] = 0;\ - *(unsigned int *)&_seqbuf[_seqbufptr+4] = (parm); \ - _SEQ_ADVBUF(8);} +#define _LOCAL_EVENT(ev, parm) { \ + _SEQ_NEEDBUF(8);\ + _seqbuf[_seqbufptr+0] = EV_SEQ_LOCAL; \ + _seqbuf[_seqbufptr+1] = (ev); \ + _seqbuf[_seqbufptr+2] = 0;\ + _seqbuf[_seqbufptr+3] = 0;\ + *(u_int *)&_seqbuf[_seqbufptr+4] = (parm); \ + _SEQ_ADVBUF(8); \ + } #define SEQ_PLAYAUDIO(devmask) _LOCAL_EVENT(LOCL_STARTAUDIO, devmask) -#define SEQ_PLAYAUDIO2(devmask) _LOCAL_EVENT(LOCL_STARTAUDIO2, devmask) -#define SEQ_PLAYAUDIO3(devmask) _LOCAL_EVENT(LOCL_STARTAUDIO3, devmask) -#define SEQ_PLAYAUDIO4(devmask) _LOCAL_EVENT(LOCL_STARTAUDIO4, devmask) /* - * Events for the level 1 interface only + * Events for the level 1 interface only */ -#define SEQ_MIDIOUT(device, byte) {_SEQ_NEEDBUF(4);\ - _seqbuf[_seqbufptr] = SEQ_MIDIPUTC;\ - _seqbuf[_seqbufptr+1] = (byte);\ - _seqbuf[_seqbufptr+2] = (device);\ - _seqbuf[_seqbufptr+3] = 0;\ - _SEQ_ADVBUF(4);} +#define SEQ_MIDIOUT(device, byte) { \ + _SEQ_NEEDBUF(4);\ + _seqbuf[_seqbufptr] = SEQ_MIDIPUTC;\ + _seqbuf[_seqbufptr+1] = (byte);\ + _seqbuf[_seqbufptr+2] = (device);\ + _seqbuf[_seqbufptr+3] = 0;\ + _SEQ_ADVBUF(4);} /* * Patch loading. */ -#ifdef OSSLIB -# define SEQ_WRPATCH(patchx, len) \ - OSS_write_patch(seqfd, (char*)(patchx), len) -# define SEQ_WRPATCH2(patchx, len) \ - OSS_write_patch2(seqfd, (char*)(patchx), len) -#else -# define SEQ_WRPATCH(patchx, len) \ - {if (_seqbufptr) SEQ_DUMPBUF();\ - if (write(seqfd, (char*)(patchx), len)==-1) \ - perror("Write patch: /dev/sequencer");} -# define SEQ_WRPATCH2(patchx, len) \ - (SEQ_DUMPBUF(), write(seqfd, (char*)(patchx), len)) -#endif - -#endif -#endif /* ifndef DISABLE_SEQUENCER */ +#define SEQ_WRPATCH(patchx, len) { \ + if (_seqbufptr) seqbuf_dump(); \ + if (write(seqfd, (char*)(patchx), len)==-1) \ + perror("Write patch: /dev/sequencer"); \ + } -/* - **************************************************************************** - * ioctl commands for the /dev/midi## - ****************************************************************************/ -#define SNDCTL_MIDI_PRETIME __SIOWR('m', 0, int) - -#if 0 -/* - * The SNDCTL_MIDI_MPUMODE and SNDCTL_MIDI_MPUCMD calls - * are completely obsolete. The hardware device (MPU-401 "intelligent mode" - * and compatibles) has disappeared from the market 10 years ago so there - * is no need for this stuff. The MPU-401 "UART" mode devices don't support - * this stuff. - */ -typedef struct -{ - unsigned char cmd; - char nr_args, nr_returns; - unsigned char data[30]; -} mpu_command_rec; +#define SEQ_WRPATCH2(patchx, len) \ + ( seqbuf_dump(), write(seqfd, (char*)(patchx), len) ) -#define SNDCTL_MIDI_MPUMODE __SIOWR('m', 1, int) -#define SNDCTL_MIDI_MPUCMD __SIOWR('m', 2, mpu_command_rec) #endif /* - * SNDCTL_MIDI_MTCINPUT turns on a mode where OSS automatically inserts - * MTC quarter frame messages (F1 xx) to the input. - * The argument is the MTC mode: - * - * -1 = Turn MTC messages OFF (default) - * 24 = 24 FPS - * 25 = 25 FPS - * 29 = 30 FPS drop frame - * 30 = 30 FPS + * Here I have moved all the aliases for ioctl names. + */ + +#define SNDCTL_DSP_SAMPLESIZE SNDCTL_DSP_SETFMT +#define SOUND_PCM_WRITE_BITS SNDCTL_DSP_SETFMT +#define SOUND_PCM_SETFMT SNDCTL_DSP_SETFMT + +#define SOUND_PCM_WRITE_RATE SNDCTL_DSP_SPEED +#define SOUND_PCM_POST SNDCTL_DSP_POST +#define SOUND_PCM_RESET SNDCTL_DSP_RESET +#define SOUND_PCM_SYNC SNDCTL_DSP_SYNC +#define SOUND_PCM_SUBDIVIDE SNDCTL_DSP_SUBDIVIDE +#define SOUND_PCM_SETFRAGMENT SNDCTL_DSP_SETFRAGMENT +#define SOUND_PCM_GETFMTS SNDCTL_DSP_GETFMTS +#define SOUND_PCM_GETOSPACE SNDCTL_DSP_GETOSPACE +#define SOUND_PCM_GETISPACE SNDCTL_DSP_GETISPACE +#define SOUND_PCM_NONBLOCK SNDCTL_DSP_NONBLOCK +#define SOUND_PCM_GETCAPS SNDCTL_DSP_GETCAPS +#define SOUND_PCM_GETTRIGGER SNDCTL_DSP_GETTRIGGER +#define SOUND_PCM_SETTRIGGER SNDCTL_DSP_SETTRIGGER +#define SOUND_PCM_SETSYNCRO SNDCTL_DSP_SETSYNCRO +#define SOUND_PCM_GETIPTR SNDCTL_DSP_GETIPTR +#define SOUND_PCM_GETOPTR SNDCTL_DSP_GETOPTR +#define SOUND_PCM_MAPINBUF SNDCTL_DSP_MAPINBUF +#define SOUND_PCM_MAPOUTBUF SNDCTL_DSP_MAPOUTBUF + +/***********************************************************************/ + +/** + * XXX OSSv4 defines -- some bits taken straight out of the new + * sys/soundcard.h bundled with recent OSS releases. * - * Note that 25 FPS mode is probably the only mode that is supported. Other - * modes may be supported in the future versions of OSS, 25 FPS is handy - * because it generates 25*4=100 quarter frame messages per second which - * matches the usual 100 HZ system timer rate). - * - * The quarter frame timer will be reset to 0:00:00:00.0 at the moment this - * ioctl is made. - */ -#define SNDCTL_MIDI_MTCINPUT __SIOWR('m', 3, int) - -/* - * MTC/SMPTE time code record (for future use) - */ -typedef struct -{ - unsigned char hours, minutes, seconds, frames, qframes; - char direction; -#define MTC_DIR_STOPPED 0 -#define MTC_DIR_FORWARD 1 -#define MTC_DIR_BACKWARD -1 - unsigned char time_code_type; - unsigned int flags; -} oss_mtc_data_t; - -#define SNDCTL_MIDI_SETMODE __SIOWR('m', 6, int) -# define MIDI_MODE_TRADITIONAL 0 -# define MIDI_MODE_TIMED 1 /* Input times are in MIDI ticks */ -# define MIDI_MODE_TIMED_ABS 2 /* Input times are absolute (usecs) */ - -/* - * Packet header for MIDI_MODE_TIMED and MIDI_MODE_TIMED_ABS - */ -typedef unsigned long long oss_midi_time_t; /* Variable type for MIDI time (clock ticks) */ - -typedef struct -{ - int magic; /* Initialize to MIDI_HDR_MAGIC */ -#define MIDI_HDR_MAGIC -1 - unsigned short event_type; -#define MIDI_EV_WRITE 0 /* Write or read (with payload) */ -#define MIDI_EV_TEMPO 1 -#define MIDI_EV_ECHO 2 -#define MIDI_EV_START 3 -#define MIDI_EV_STOP 4 -#define MIDI_EV_CONTINUE 5 -#define MIDI_EV_XPRESSWRITE 6 -#define MIDI_EV_TIMEBASE 7 -#define MIDI_EV_DEVCTL 8 /* Device control read/write */ - unsigned short options; -#define MIDI_OPT_NONE 0x0000 -#define MIDI_OPT_TIMED 0x0001 -#define MIDI_OPT_CONTINUATION 0x0002 -#define MIDI_OPT_USECTIME 0x0004 /* Time is absolute (in usecs) */ -#define MIDI_OPT_BUSY 0x0008 /* Reserved for internal use */ - oss_midi_time_t time; - int parm; - int filler[3]; /* Fur future expansion - init to zeros */ -} midi_packet_header_t; -/* - * MIDI_PAYLOAD_SIZE is the maximum size of one MIDI input chunk. It must be - * less (or equal) than 1024 which is the read size recommended in the - * documentation. TODO: Explain this better. - */ -#define MIDI_PAYLOAD_SIZE 1000 - -typedef struct -{ - midi_packet_header_t hdr; - unsigned char payload[MIDI_PAYLOAD_SIZE]; -} midi_packet_t; - -#define SNDCTL_MIDI_TIMEBASE __SIOWR('m', 7, int) -#define SNDCTL_MIDI_TEMPO __SIOWR('m', 8, int) -/* - * User land MIDI servers (synths) can use SNDCTL_MIDI_SET_LATENCY - * to request MIDI events to be sent to them in advance. The parameter - * (in microseconds) tells how much before the events are submitted. + * NB: These macros and structures will be reorganized and inserted + * in appropriate places throughout this file once the code begins + * to take shape. * - * This feature is only valid for loopback devices and possibly some other - * types of virtual devices. - */ -#define SNDCTL_MIDI_SET_LATENCY __SIOW ('m', 9, int) -/* - **************************************************************************** - * IOCTL commands for /dev/dsp - ****************************************************************************/ - -#define SNDCTL_DSP_HALT __SIO ('P', 0) -#define SNDCTL_DSP_RESET SNDCTL_DSP_HALT /* Old name */ -#define SNDCTL_DSP_SYNC __SIO ('P', 1) -#define SNDCTL_DSP_SPEED __SIOWR('P', 2, int) - -/* SNDCTL_DSP_STEREO is obsolete - use SNDCTL_DSP_CHANNELS instead */ -#define SNDCTL_DSP_STEREO __SIOWR('P', 3, int) -/* SNDCTL_DSP_STEREO is obsolete - use SNDCTL_DSP_CHANNELS instead */ - -#define SNDCTL_DSP_GETBLKSIZE __SIOWR('P', 4, int) -#define SNDCTL_DSP_SAMPLESIZE SNDCTL_DSP_SETFMT -#define SNDCTL_DSP_CHANNELS __SIOWR('P', 6, int) -#define SNDCTL_DSP_POST __SIO ('P', 8) -#define SNDCTL_DSP_SUBDIVIDE __SIOWR('P', 9, int) -#define SNDCTL_DSP_SETFRAGMENT __SIOWR('P',10, int) - -/* Audio data formats (Note! U8=8 and S16_LE=16 for compatibility) */ -#define SNDCTL_DSP_GETFMTS __SIOR ('P',11, int) /* Returns a mask */ -#define SNDCTL_DSP_SETFMT __SIOWR('P',5, int) /* Selects ONE fmt */ -# define AFMT_QUERY 0x00000000 /* Return current fmt */ -# define AFMT_MU_LAW 0x00000001 -# define AFMT_A_LAW 0x00000002 -# define AFMT_IMA_ADPCM 0x00000004 -# define AFMT_U8 0x00000008 -# define AFMT_S16_LE 0x00000010 /* Little endian signed 16 */ -# define AFMT_S16_BE 0x00000020 /* Big endian signed 16 */ -# define AFMT_S8 0x00000040 -# define AFMT_U16_LE 0x00000080 /* Little endian U16 */ -# define AFMT_U16_BE 0x00000100 /* Big endian U16 */ -# define AFMT_MPEG 0x00000200 /* MPEG (2) audio */ - -/* AC3 _compressed_ bitstreams (See Programmer's Guide for details). */ -# define AFMT_AC3 0x00000400 -/* Ogg Vorbis _compressed_ bit streams */ -# define AFMT_VORBIS 0x00000800 - -/* 32 bit formats (MSB aligned) formats */ -# define AFMT_S32_LE 0x00001000 -# define AFMT_S32_BE 0x00002000 - -/* Reserved for _native_ endian double precision IEEE floating point */ -# define AFMT_FLOAT 0x00004000 - -/* 24 bit formats (LSB aligned in 32 bit word) formats */ -# define AFMT_S24_LE 0x00008000 -# define AFMT_S24_BE 0x00010000 - -/* - * S/PDIF raw format. In this format the S/PDIF frames (including all - * control and user bits) are included in the data stream. Each sample - * is stored in a 32 bit frame (see IEC-958 for more info). This format - * is supported by very few devices and it's only usable for purposes - * where full access to the control/user bits is required (real time control). + * @todo reorganize layout more like the 4Front version + * @todo ask about maintaining __SIOWR vs. _IOWR ioctl cmd defines */ -# define AFMT_SPDIF_RAW 0x00020000 - -/* 24 bit packed (3 byte) little endian format (USB compatibility) */ -# define AFMT_S24_PACKED 0x00040000 - -/* - * Some big endian/little endian handling macros (native endian and opposite - * endian formats). The usage of these macros is described in the OSS - * Programmer's Manual. +/** + * @note The @c OSSV4_EXPERIMENT macro is meant to wrap new development code + * in the sound system relevant to adopting 4Front's OSSv4 specification. + * Users should not enable this! Really! */ - -#if defined(_AIX) || defined(AIX) || defined(sparc) || defined(__hppa) || defined(PPC) || defined(__powerpc__) && !defined(i386) && !defined(__i386) && !defined(__i386__) - -/* Big endian machines */ -# define _PATCHKEY(id) (0xfd00|id) -# define AFMT_S16_NE AFMT_S16_BE -# define AFMT_U16_NE AFMT_U16_BE -# define AFMT_S32_NE AFMT_S32_BE -# define AFMT_S24_NE AFMT_S24_BE -# define AFMT_S16_OE AFMT_S16_LE -# define AFMT_S32_OE AFMT_S32_LE -# define AFMT_S24_OE AFMT_S24_LE +#if 0 +# define OSSV4_EXPERIMENT 1 #else -# define _PATCHKEY(id) ((id<<8)|0xfd) -# define AFMT_S16_NE AFMT_S16_LE -# define AFMT_U16_NE AFMT_U16_LE -# define AFMT_S32_NE AFMT_S32_LE -# define AFMT_S24_NE AFMT_S24_LE -# define AFMT_S16_OE AFMT_S16_BE -# define AFMT_S32_OE AFMT_S32_BE -# define AFMT_S24_OE AFMT_S24_BE +# undef OSSV4_EXPERIMENT #endif -/* - * Buffer status queries. - */ -typedef struct audio_buf_info -{ - int fragments; /* # of available fragments (partially usend ones not counted) */ - int fragstotal; /* Total # of fragments allocated */ - int fragsize; /* Size of a fragment in bytes */ - int bytes; /* Available space in bytes (includes partially used fragments) */ - /* Note! 'bytes' could be more than fragments*fragsize */ -} audio_buf_info; - -#define SNDCTL_DSP_GETOSPACE __SIOR ('P',12, audio_buf_info) -#define SNDCTL_DSP_GETISPACE __SIOR ('P',13, audio_buf_info) -#define SNDCTL_DSP_GETCAPS __SIOR ('P',15, int) -# define PCM_CAP_REVISION 0x000000ff /* Bits for revision level (0 to 255) */ -# define PCM_CAP_DUPLEX 0x00000100 /* Full duplex record/playback */ -# define PCM_CAP_REALTIME 0x00000200 /* Not in use */ -# define PCM_CAP_BATCH 0x00000400 /* Device has some kind of */ - /* internal buffers which may */ - /* cause some delays and */ - /* decrease precision of timing */ -# define PCM_CAP_COPROC 0x00000800 /* Has a coprocessor */ - /* Sometimes it's a DSP */ - /* but usually not */ -# define PCM_CAP_TRIGGER 0x00001000 /* Supports SETTRIGGER */ -# define PCM_CAP_MMAP 0x00002000 /* Supports mmap() */ -# define PCM_CAP_MULTI 0x00004000 /* Supports multiple open */ -# define PCM_CAP_BIND 0x00008000 /* Supports binding to front/rear/center/lfe */ -# define PCM_CAP_INPUT 0x00010000 /* Supports recording */ -# define PCM_CAP_OUTPUT 0x00020000 /* Supports playback */ -# define PCM_CAP_VIRTUAL 0x00040000 /* Virtuial device */ -/* 0x00040000 and 0x00080000 reserved for future use */ - -/* Analog/digital control capabilities */ -# define PCM_CAP_ANALOGOUT 0x00100000 -# define PCM_CAP_ANALOGIN 0x00200000 -# define PCM_CAP_DIGITALOUT 0x00400000 -# define PCM_CAP_DIGITALIN 0x00800000 -# define PCM_CAP_ADMASK 0x00f00000 -/* - * NOTE! (capabilities & PCM_CAP_ADMASK)==0 means just that the - * digital/analog interface control features are not supported by the - * device/driver. However the device still supports analog, digital or - * both inputs/outputs (depending on the device). See the OSS Programmer's - * Guide for full details. - */ -# define PCM_CAP_SHADOW 0x01000000 /* "Shadow" device */ - -/* - * Preferred channel usage. These bits can be used to - * give recommendations to the application. Used by few drivers. - * For example if ((caps & DSP_CH_MASK) == DSP_CH_MONO) means that - * the device works best in mono mode. However it doesn't necessarily mean - * that the device cannot be used in stereo. These bits should only be used - * special applications such as multi track hard disk recorders to find out - * the initial setup. However the user should be able to override this - * selection. - * - * To find out which modes are actually supported the application should - * try to select them using SNDCTL_DSP_CHANNELS. - */ -# define DSP_CH_MASK 0x06000000 /* Mask */ -# define DSP_CH_ANY 0x00000000 /* No preferred mode */ -# define DSP_CH_MONO 0x02000000 -# define DSP_CH_STEREO 0x04000000 -# define DSP_CH_MULTI 0x06000000 /* More than two channels */ - -# define PCM_CAP_HIDDEN 0x08000000 /* Hidden device */ -# define PCM_CAP_FREERATE 0x10000000 -# define PCM_CAP_MODEM 0x20000000 /* Modem device */ -# define PCM_CAP_DEFAULT 0x40000000 /* "Default" device */ - -/* - * The PCM_CAP_* capability names were known as DSP_CAP_* prior OSS 4.0 - * so it's necessary to define the older names too. - */ -#define DSP_CAP_ADMASK PCM_CAP_ADMASK -#define DSP_CAP_ANALOGIN PCM_CAP_ANALOGIN -#define DSP_CAP_ANALOGOUT PCM_CAP_ANALOGOUT -#define DSP_CAP_BATCH PCM_CAP_BATCH -#define DSP_CAP_BIND PCM_CAP_BIND -#define DSP_CAP_COPROC PCM_CAP_COPROC -#define DSP_CAP_DEFAULT PCM_CAP_DEFAULT -#define DSP_CAP_DIGITALIN PCM_CAP_DIGITALIN -#define DSP_CAP_DIGITALOUT PCM_CAP_DIGITALOUT -#define DSP_CAP_DUPLEX PCM_CAP_DUPLEX -#define DSP_CAP_FREERATE PCM_CAP_FREERATE -#define DSP_CAP_HIDDEN PCM_CAP_HIDDEN -#define DSP_CAP_INPUT PCM_CAP_INPUT -#define DSP_CAP_MMAP PCM_CAP_MMAP -#define DSP_CAP_MODEM PCM_CAP_MODEM -#define DSP_CAP_MULTI PCM_CAP_MULTI -#define DSP_CAP_OUTPUT PCM_CAP_OUTPUT -#define DSP_CAP_REALTIME PCM_CAP_REALTIME -#define DSP_CAP_REVISION PCM_CAP_REVISION -#define DSP_CAP_SHADOW PCM_CAP_SHADOW -#define DSP_CAP_TRIGGER PCM_CAP_TRIGGER -#define DSP_CAP_VIRTUAL PCM_CAP_VIRTUAL - -#define SNDCTL_DSP_GETTRIGGER __SIOR ('P',16, int) -#define SNDCTL_DSP_SETTRIGGER __SIOW ('P',16, int) -# define PCM_ENABLE_INPUT 0x00000001 -# define PCM_ENABLE_OUTPUT 0x00000002 - -typedef struct count_info -{ - unsigned int bytes; /* Total # of bytes processed */ - int blocks; /* # of fragment transitions since last time */ - int ptr; /* Current DMA pointer value */ -} count_info; - -#define SNDCTL_DSP_GETIPTR __SIOR ('P',17, count_info) -#define SNDCTL_DSP_GETOPTR __SIOR ('P',18, count_info) -typedef struct buffmem_desc -{ - unsigned *buffer; - int size; -} buffmem_desc; -#define SNDCTL_DSP_SETSYNCRO __SIO ('P', 21) -#define SNDCTL_DSP_SETDUPLEX __SIO ('P', 22) +#ifdef SOUND_VERSION +# undef SOUND_VERSION +# define SOUND_VERSION 0x040000 +#endif /* !SOUND_VERSION */ -#define SNDCTL_DSP_PROFILE __SIOW ('P', 23, int) /* OBSOLETE */ -#define APF_NORMAL 0 /* Normal applications */ -#define APF_NETWORK 1 /* Underruns probably caused by an "external" delay */ -#define APF_CPUINTENS 2 /* Underruns probably caused by "overheating" the CPU */ - -#define SNDCTL_DSP_GETODELAY __SIOR ('P', 23, int) +#define OSS_LONGNAME_SIZE 64 +#define OSS_LABEL_SIZE 16 +#define OSS_DEVNODE_SIZE 32 +typedef char oss_longname_t[OSS_LONGNAME_SIZE]; +typedef char oss_label_t[OSS_LABEL_SIZE]; +typedef char oss_devnode_t[OSS_DEVNODE_SIZE]; typedef struct audio_errinfo { - int play_underruns; - int rec_overruns; - unsigned int play_ptradjust; - unsigned int rec_ptradjust; - int play_errorcount; - int rec_errorcount; - int play_lasterror; - int rec_lasterror; - int play_errorparm; - int rec_errorparm; - int filler[16]; + int play_underruns; + int rec_overruns; + unsigned int play_ptradjust; + unsigned int rec_ptradjust; + int play_errorcount; + int rec_errorcount; + int play_lasterror; + int rec_lasterror; + long play_errorparm; + long rec_errorparm; + int filler[16]; } audio_errinfo; -#define SNDCTL_DSP_GETPLAYVOL __SIOR ('P', 24, int) -#define SNDCTL_DSP_SETPLAYVOL __SIOWR('P', 24, int) -#define SNDCTL_DSP_GETERROR __SIOR ('P', 25, audio_errinfo) -/* - **************************************************************************** - * Digital interface (S/PDIF) control interface - */ +#define SNDCTL_DSP_GETPLAYVOL _IOR ('P', 24, int) +#define SNDCTL_DSP_SETPLAYVOL _IOWR('P', 24, int) +#define SNDCTL_DSP_GETERROR _IOR ('P', 25, audio_errinfo) -typedef struct oss_digital_control -{ - unsigned int caps; -#define DIG_CBITIN_NONE 0x00000000 -#define DIG_CBITIN_LIMITED 0x00000001 -#define DIG_CBITIN_DATA 0x00000002 -#define DIG_CBITIN_BYTE0 0x00000004 -#define DIG_CBITIN_FULL 0x00000008 -#define DIG_CBITIN_MASK 0x0000000f -#define DIG_CBITOUT_NONE 0x00000000 -#define DIG_CBITOUT_LIMITED 0x00000010 -#define DIG_CBITOUT_BYTE0 0x00000020 -#define DIG_CBITOUT_FULL 0x00000040 -#define DIG_CBITOUT_DATA 0x00000080 -#define DIG_CBITOUT_MASK 0x000000f0 -#define DIG_UBITIN 0x00000100 -#define DIG_UBITOUT 0x00000200 -#define DIG_VBITOUT 0x00000400 -#define DIG_OUTRATE 0x00000800 -#define DIG_INRATE 0x00001000 -#define DIG_INBITS 0x00002000 -#define DIG_OUTBITS 0x00004000 -#define DIG_EXACT 0x00010000 -#define DIG_PRO 0x00020000 -#define DIG_CONSUMER 0x00040000 -#define DIG_PASSTHROUGH 0x00080000 -#define DIG_OUTSEL 0x00100000 - - unsigned int valid; -#define VAL_CBITIN 0x00000001 -#define VAL_UBITIN 0x00000002 -#define VAL_CBITOUT 0x00000004 -#define VAL_UBITOUT 0x00000008 -#define VAL_ISTATUS 0x00000010 -#define VAL_IRATE 0x00000020 -#define VAL_ORATE 0x00000040 -#define VAL_INBITS 0x00000080 -#define VAL_OUTBITS 0x00000100 -#define VAL_REQUEST 0x00000200 -#define VAL_OUTSEL 0x00000400 - -#define VAL_OUTMASK (VAL_CBITOUT|VAL_UBITOUT|VAL_ORATE|VAL_OUTBITS|VAL_OUTSEL) - - unsigned int request, param; -#define SPD_RQ_PASSTHROUGH 1 - - unsigned char cbitin[24]; - unsigned char ubitin[24]; - unsigned char cbitout[24]; - unsigned char ubitout[24]; - - unsigned int outsel; -#define OUTSEL_DIGITAL 1 -#define OUTSEL_ANALOG 2 -#define OUTSEL_BOTH (OUTSEL_DIGITAL|OUTSEL_ANALOG) - - int in_data; /* Audio/data if autodetectable by the receiver */ -#define IND_UNKNOWN 0 -#define IND_AUDIO 1 -#define IND_DATA 2 - - int in_locked; /* Receiver locked */ -#define LOCK_NOT_INDICATED 0 -#define LOCK_UNLOCKED 1 -#define LOCK_LOCKED 2 - - int in_quality; /* Input signal quality */ -#define IN_QUAL_NOT_INDICATED 0 -#define IN_QUAL_POOR 1 -#define IN_QUAL_GOOD 2 - - int in_vbit, out_vbit; /* V bits */ -#define VBIT_NOT_INDICATED 0 -#define VBIT_OFF 1 -#define VBIT_ON 2 - - unsigned int in_errors; /* Various input errro conditions */ -#define INERR_CRC 0x0001 -#define INERR_QCODE_CRC 0x0002 -#define INERR_PARITY 0x0004 -#define INERR_BIPHASE 0x0008 - - int srate_in, srate_out; - int bits_in, bits_out; - - int filler[32]; -} oss_digital_control; - -#define SNDCTL_DSP_READCTL __SIOWR('P', 26, oss_digital_control) -#define SNDCTL_DSP_WRITECTL __SIOWR('P', 27, oss_digital_control) /* **************************************************************************** @@ -1333,8 +1509,8 @@ typedef struct oss_syncgroup int filler[16]; } oss_syncgroup; -#define SNDCTL_DSP_SYNCGROUP __SIOWR('P', 28, oss_syncgroup) -#define SNDCTL_DSP_SYNCSTART __SIOW ('P', 29, int) +#define SNDCTL_DSP_SYNCGROUP _IOWR('P', 28, oss_syncgroup) +#define SNDCTL_DSP_SYNCSTART _IOW ('P', 29, int) /* ************************************************************************** @@ -1350,7 +1526,7 @@ typedef struct oss_syncgroup * SNDCTL_DSP_COOKEDMODE must be called immediately after open before doing * anything else. Otherwise the call will not have any effect. */ -#define SNDCTL_DSP_COOKEDMODE __SIOW ('P', 30, int) +#define SNDCTL_DSP_COOKEDMODE _IOW ('P', 30, int) /* ************************************************************************** @@ -1358,21 +1534,25 @@ typedef struct oss_syncgroup * that can be used to implement pause/continue during playback (no effect * on recording). */ -#define SNDCTL_DSP_SILENCE __SIO ('P', 31) -#define SNDCTL_DSP_SKIP __SIO ('P', 32) +#define SNDCTL_DSP_SILENCE _IO ('P', 31) +#define SNDCTL_DSP_SKIP _IO ('P', 32) + /* **************************************************************************** * Abort transfer (reset) functions for input and output */ -#define SNDCTL_DSP_HALT_INPUT __SIO ('P', 33) +#define SNDCTL_DSP_HALT_INPUT _IO ('P', 33) #define SNDCTL_DSP_RESET_INPUT SNDCTL_DSP_HALT_INPUT /* Old name */ -#define SNDCTL_DSP_HALT_OUTPUT __SIO ('P', 34) +#define SNDCTL_DSP_HALT_OUTPUT _IO ('P', 34) #define SNDCTL_DSP_RESET_OUTPUT SNDCTL_DSP_HALT_OUTPUT /* Old name */ + /* **************************************************************************** * Low water level control */ -#define SNDCTL_DSP_LOW_WATER __SIOW ('P', 34, int) +#define SNDCTL_DSP_LOW_WATER _IOW ('P', 34, int) + +/** @todo Get rid of OSS_NO_LONG_LONG references? */ /* **************************************************************************** @@ -1387,373 +1567,79 @@ typedef struct int filler[32]; /* For future use */ } oss_count_t; -#define SNDCTL_DSP_CURRENT_IPTR __SIOR ('P', 35, oss_count_t) -#define SNDCTL_DSP_CURRENT_OPTR __SIOR ('P', 36, oss_count_t) +#define SNDCTL_DSP_CURRENT_IPTR _IOR ('P', 35, oss_count_t) +#define SNDCTL_DSP_CURRENT_OPTR _IOR ('P', 36, oss_count_t) #endif /* **************************************************************************** * Interface for selecting recording sources and playback output routings. */ -#define SNDCTL_DSP_GET_RECSRC_NAMES __SIOR ('P', 37, oss_mixer_enuminfo) -#define SNDCTL_DSP_GET_RECSRC __SIOR ('P', 38, int) -#define SNDCTL_DSP_SET_RECSRC __SIOWR('P', 38, int) +#define SNDCTL_DSP_GET_RECSRC_NAMES _IOR ('P', 37, oss_mixer_enuminfo) +#define SNDCTL_DSP_GET_RECSRC _IOR ('P', 38, int) +#define SNDCTL_DSP_SET_RECSRC _IOWR('P', 38, int) -#define SNDCTL_DSP_GET_PLAYTGT_NAMES __SIOR ('P', 39, oss_mixer_enuminfo) -#define SNDCTL_DSP_GET_PLAYTGT __SIOR ('P', 40, int) -#define SNDCTL_DSP_SET_PLAYTGT __SIOWR('P', 40, int) -#define SNDCTL_DSP_GETRECVOL __SIOR ('P', 41, int) -#define SNDCTL_DSP_SETRECVOL __SIOWR('P', 41, int) +#define SNDCTL_DSP_GET_PLAYTGT_NAMES _IOR ('P', 39, oss_mixer_enuminfo) +#define SNDCTL_DSP_GET_PLAYTGT _IOR ('P', 40, int) +#define SNDCTL_DSP_SET_PLAYTGT _IOWR('P', 40, int) +#define SNDCTL_DSP_GETRECVOL _IOR ('P', 41, int) +#define SNDCTL_DSP_SETRECVOL _IOWR('P', 41, int) /* *************************************************************************** * Some calls for setting the channel assignment with multi channel devices - * (see the manual for details). - */ -#ifndef OSS_NO_LONG_LONG -#define SNDCTL_DSP_GET_CHNORDER __SIOR ('P', 42, unsigned long long) -#define SNDCTL_DSP_SET_CHNORDER __SIOWR('P', 42, unsigned long long) -# define CHID_UNDEF 0 -# define CHID_L 1 -# define CHID_R 2 -# define CHID_C 3 -# define CHID_LFE 4 -# define CHID_LS 5 -# define CHID_RS 6 -# define CHID_LR 7 -# define CHID_RR 8 -#define CHNORDER_UNDEF 0x0000000000000000ULL -#define CHNORDER_NORMAL 0x0000000087654321ULL -#endif + * (see the manual for details). */ +#define SNDCTL_DSP_GET_CHNORDER _IOR ('P', 42, unsigned long long) +#define SNDCTL_DSP_SET_CHNORDER _IOWR('P', 42, unsigned long long) +# define CHID_UNDEF 0 +# define CHID_L 1 # define CHID_R 2 +# define CHID_C 3 +# define CHID_LFE 4 +# define CHID_LS 5 +# define CHID_RS 6 +# define CHID_LR 7 +# define CHID_RR 8 +#define CHNORDER_UNDEF 0x0000000000000000ULL +#define CHNORDER_NORMAL 0x0000000087654321ULL #define MAX_PEAK_CHANNELS 128 typedef unsigned short oss_peaks_t[MAX_PEAK_CHANNELS]; -#define SNDCTL_DSP_GETIPEAKS __SIOR('P', 43, oss_peaks_t) -#define SNDCTL_DSP_GETOPEAKS __SIOR('P', 44, oss_peaks_t) - -#define SNDCTL_DSP_POLICY __SIOW('P', 45, int) /* See the manual */ - -/* - **************************************************************************** - * Few ioctl calls that are not official parts of OSS. They have been used - * by few freeware implementations of OSS. - */ -#define SNDCTL_DSP_GETCHANNELMASK __SIOWR('P', 64, int) -#define SNDCTL_DSP_BIND_CHANNEL __SIOWR('P', 65, int) -# define DSP_BIND_QUERY 0x00000000 -# define DSP_BIND_FRONT 0x00000001 -# define DSP_BIND_SURR 0x00000002 -# define DSP_BIND_CENTER_LFE 0x00000004 -# define DSP_BIND_HANDSET 0x00000008 -# define DSP_BIND_MIC 0x00000010 -# define DSP_BIND_MODEM1 0x00000020 -# define DSP_BIND_MODEM2 0x00000040 -# define DSP_BIND_I2S 0x00000080 -# define DSP_BIND_SPDIF 0x00000100 -# define DSP_BIND_REAR 0x00000200 - -#ifndef NO_LEGACY_MIXER -/* - **************************************************************************** - * IOCTL commands for the "legacy " /dev/mixer API (obsolete) - * - * Mixer controls - * - * There can be up to 20 different analog mixer channels. The - * SOUND_MIXER_NRDEVICES gives the currently supported maximum. - * The SOUND_MIXER_READ_DEVMASK returns a bitmask which tells - * the devices supported by the particular mixer. - * - * {!notice This "legacy" mixer API is obsolete. It has been superceded - * by a new one (see below). - */ - -#define SOUND_MIXER_NRDEVICES 28 -#define SOUND_MIXER_VOLUME 0 -#define SOUND_MIXER_BASS 1 -#define SOUND_MIXER_TREBLE 2 -#define SOUND_MIXER_SYNTH 3 -#define SOUND_MIXER_PCM 4 -#define SOUND_MIXER_SPEAKER 5 -#define SOUND_MIXER_LINE 6 -#define SOUND_MIXER_MIC 7 -#define SOUND_MIXER_CD 8 -#define SOUND_MIXER_IMIX 9 /* Recording monitor */ -#define SOUND_MIXER_ALTPCM 10 -#define SOUND_MIXER_RECLEV 11 /* Recording level */ -#define SOUND_MIXER_IGAIN 12 /* Input gain */ -#define SOUND_MIXER_OGAIN 13 /* Output gain */ -/* - * Some soundcards have three line level inputs (line, aux1 and aux2). - * Since each card manufacturer has assigned different meanings to - * these inputs, it's impractical to assign specific meanings - * (eg line, cd, synth etc.) to them. - */ -#define SOUND_MIXER_LINE1 14 /* Input source 1 (aux1) */ -#define SOUND_MIXER_LINE2 15 /* Input source 2 (aux2) */ -#define SOUND_MIXER_LINE3 16 /* Input source 3 (line) */ -#define SOUND_MIXER_DIGITAL1 17 /* Digital I/O 1 */ -#define SOUND_MIXER_DIGITAL2 18 /* Digital I/O 2 */ -#define SOUND_MIXER_DIGITAL3 19 /* Digital I/O 3 */ -#define SOUND_MIXER_PHONE 20 /* Phone */ -#define SOUND_MIXER_MONO 21 /* Mono Output */ -#define SOUND_MIXER_VIDEO 22 /* Video/TV (audio) in */ -#define SOUND_MIXER_RADIO 23 /* Radio in */ -#define SOUND_MIXER_DEPTH 24 /* Surround depth */ -#define SOUND_MIXER_REARVOL 25 /* Rear/Surround speaker vol */ -#define SOUND_MIXER_CENTERVOL 26 /* Center/LFE speaker vol */ -#define SOUND_MIXER_SIDEVOL 27 /* Side-Surround (8speaker) vol */ - -/* - * Warning: SOUND_MIXER_SURRVOL is an old name of SOUND_MIXER_SIDEVOL. - * They are both assigned to the same mixer control. Don't - * use both control names in the same program/driver. - */ -#define SOUND_MIXER_SURRVOL SOUND_MIXER_SIDEVOL - -/* Some on/off settings (SOUND_SPECIAL_MIN - SOUND_SPECIAL_MAX) */ -/* Not counted to SOUND_MIXER_NRDEVICES, but use the same number space */ -#define SOUND_ONOFF_MIN 28 -#define SOUND_ONOFF_MAX 30 - -/* Note! Number 31 cannot be used since the sign bit is reserved */ -#define SOUND_MIXER_NONE 31 - -/* - * The following unsupported macros are no longer functional. - * Use SOUND_MIXER_PRIVATE# macros in future. - */ -#define SOUND_MIXER_ENHANCE SOUND_MIXER_NONE -#define SOUND_MIXER_MUTE SOUND_MIXER_NONE -#define SOUND_MIXER_LOUD SOUND_MIXER_NONE - -#define SOUND_DEVICE_LABELS \ - {"Vol ", "Bass ", "Treble", "Synth", "Pcm ", "Speaker ", "Line ", \ - "Mic ", "CD ", "Mix ", "Pcm2 ", "Rec ", "IGain", "OGain", \ - "Aux1", "Aux2", "Aux3", "Digital1", "Digital2", "Digital3", \ - "Phone", "Mono", "Video", "Radio", "Depth", \ - "Rear", "Center", "Side"} - -#define SOUND_DEVICE_NAMES \ - {"vol", "bass", "treble", "synth", "pcm", "speaker", "line", \ - "mic", "cd", "mix", "pcm2", "rec", "igain", "ogain", \ - "aux1", "aux2", "aux3", "dig1", "dig2", "dig3", \ - "phone", "mono", "video", "radio", "depth", \ - "rear", "center", "side"} - -/* Device bitmask identifiers */ - -#define SOUND_MIXER_RECSRC 0xff /* Arg contains a bit for each recording source */ -#define SOUND_MIXER_DEVMASK 0xfe /* Arg contains a bit for each supported device */ -#define SOUND_MIXER_RECMASK 0xfd /* Arg contains a bit for each supported recording source */ -#define SOUND_MIXER_CAPS 0xfc -# define SOUND_CAP_EXCL_INPUT 0x00000001 /* Only one recording source at a time */ -# define SOUND_CAP_NOLEGACY 0x00000004 /* For internal use only */ -# define SOUND_CAP_NORECSRC 0x00000008 -#define SOUND_MIXER_STEREODEVS 0xfb /* Mixer channels supporting stereo */ - -/* OSS/Free ONLY */ -#define SOUND_MIXER_OUTSRC 0xfa /* Arg contains a bit for each input source to output */ -#define SOUND_MIXER_OUTMASK 0xf9 /* Arg contains a bit for each supported input source to output */ -/* OSS/Free ONLY */ - -/* Device mask bits */ - -#define SOUND_MASK_VOLUME (1 << SOUND_MIXER_VOLUME) -#define SOUND_MASK_BASS (1 << SOUND_MIXER_BASS) -#define SOUND_MASK_TREBLE (1 << SOUND_MIXER_TREBLE) -#define SOUND_MASK_SYNTH (1 << SOUND_MIXER_SYNTH) -#define SOUND_MASK_PCM (1 << SOUND_MIXER_PCM) -#define SOUND_MASK_SPEAKER (1 << SOUND_MIXER_SPEAKER) -#define SOUND_MASK_LINE (1 << SOUND_MIXER_LINE) -#define SOUND_MASK_MIC (1 << SOUND_MIXER_MIC) -#define SOUND_MASK_CD (1 << SOUND_MIXER_CD) -#define SOUND_MASK_IMIX (1 << SOUND_MIXER_IMIX) -#define SOUND_MASK_ALTPCM (1 << SOUND_MIXER_ALTPCM) -#define SOUND_MASK_RECLEV (1 << SOUND_MIXER_RECLEV) -#define SOUND_MASK_IGAIN (1 << SOUND_MIXER_IGAIN) -#define SOUND_MASK_OGAIN (1 << SOUND_MIXER_OGAIN) -#define SOUND_MASK_LINE1 (1 << SOUND_MIXER_LINE1) -#define SOUND_MASK_LINE2 (1 << SOUND_MIXER_LINE2) -#define SOUND_MASK_LINE3 (1 << SOUND_MIXER_LINE3) -#define SOUND_MASK_DIGITAL1 (1 << SOUND_MIXER_DIGITAL1) -#define SOUND_MASK_DIGITAL2 (1 << SOUND_MIXER_DIGITAL2) -#define SOUND_MASK_DIGITAL3 (1 << SOUND_MIXER_DIGITAL3) -#define SOUND_MASK_MONO (1 << SOUND_MIXER_MONO) -#define SOUND_MASK_PHONE (1 << SOUND_MIXER_PHONE) -#define SOUND_MASK_RADIO (1 << SOUND_MIXER_RADIO) -#define SOUND_MASK_VIDEO (1 << SOUND_MIXER_VIDEO) -#define SOUND_MASK_DEPTH (1 << SOUND_MIXER_DEPTH) -#define SOUND_MASK_REARVOL (1 << SOUND_MIXER_REARVOL) -#define SOUND_MASK_CENTERVOL (1 << SOUND_MIXER_CENTERVOL) -#define SOUND_MASK_SIDEVOL (1 << SOUND_MIXER_SIDEVOL) - -/* Note! SOUND_MASK_SURRVOL is alias of SOUND_MASK_SIDEVOL */ -#define SOUND_MASK_SURRVOL (1 << SOUND_MIXER_SIDEVOL) - -/* Obsolete macros */ -#define SOUND_MASK_MUTE (1 << SOUND_MIXER_MUTE) -#define SOUND_MASK_ENHANCE (1 << SOUND_MIXER_ENHANCE) -#define SOUND_MASK_LOUD (1 << SOUND_MIXER_LOUD) - -#define MIXER_READ(dev) __SIOR('M', dev, int) -#define SOUND_MIXER_READ_VOLUME MIXER_READ(SOUND_MIXER_VOLUME) -#define SOUND_MIXER_READ_BASS MIXER_READ(SOUND_MIXER_BASS) -#define SOUND_MIXER_READ_TREBLE MIXER_READ(SOUND_MIXER_TREBLE) -#define SOUND_MIXER_READ_SYNTH MIXER_READ(SOUND_MIXER_SYNTH) -#define SOUND_MIXER_READ_PCM MIXER_READ(SOUND_MIXER_PCM) -#define SOUND_MIXER_READ_SPEAKER MIXER_READ(SOUND_MIXER_SPEAKER) -#define SOUND_MIXER_READ_LINE MIXER_READ(SOUND_MIXER_LINE) -#define SOUND_MIXER_READ_MIC MIXER_READ(SOUND_MIXER_MIC) -#define SOUND_MIXER_READ_CD MIXER_READ(SOUND_MIXER_CD) -#define SOUND_MIXER_READ_IMIX MIXER_READ(SOUND_MIXER_IMIX) -#define SOUND_MIXER_READ_ALTPCM MIXER_READ(SOUND_MIXER_ALTPCM) -#define SOUND_MIXER_READ_RECLEV MIXER_READ(SOUND_MIXER_RECLEV) -#define SOUND_MIXER_READ_IGAIN MIXER_READ(SOUND_MIXER_IGAIN) -#define SOUND_MIXER_READ_OGAIN MIXER_READ(SOUND_MIXER_OGAIN) -#define SOUND_MIXER_READ_LINE1 MIXER_READ(SOUND_MIXER_LINE1) -#define SOUND_MIXER_READ_LINE2 MIXER_READ(SOUND_MIXER_LINE2) -#define SOUND_MIXER_READ_LINE3 MIXER_READ(SOUND_MIXER_LINE3) - -/* Obsolete macros */ -#define SOUND_MIXER_READ_MUTE MIXER_READ(SOUND_MIXER_MUTE) -#define SOUND_MIXER_READ_ENHANCE MIXER_READ(SOUND_MIXER_ENHANCE) -#define SOUND_MIXER_READ_LOUD MIXER_READ(SOUND_MIXER_LOUD) - -#define SOUND_MIXER_READ_RECSRC MIXER_READ(SOUND_MIXER_RECSRC) -#define SOUND_MIXER_READ_DEVMASK MIXER_READ(SOUND_MIXER_DEVMASK) -#define SOUND_MIXER_READ_RECMASK MIXER_READ(SOUND_MIXER_RECMASK) -#define SOUND_MIXER_READ_STEREODEVS MIXER_READ(SOUND_MIXER_STEREODEVS) -#define SOUND_MIXER_READ_CAPS MIXER_READ(SOUND_MIXER_CAPS) - -#define MIXER_WRITE(dev) __SIOWR('M', dev, int) -#define SOUND_MIXER_WRITE_VOLUME MIXER_WRITE(SOUND_MIXER_VOLUME) -#define SOUND_MIXER_WRITE_BASS MIXER_WRITE(SOUND_MIXER_BASS) -#define SOUND_MIXER_WRITE_TREBLE MIXER_WRITE(SOUND_MIXER_TREBLE) -#define SOUND_MIXER_WRITE_SYNTH MIXER_WRITE(SOUND_MIXER_SYNTH) -#define SOUND_MIXER_WRITE_PCM MIXER_WRITE(SOUND_MIXER_PCM) -#define SOUND_MIXER_WRITE_SPEAKER MIXER_WRITE(SOUND_MIXER_SPEAKER) -#define SOUND_MIXER_WRITE_LINE MIXER_WRITE(SOUND_MIXER_LINE) -#define SOUND_MIXER_WRITE_MIC MIXER_WRITE(SOUND_MIXER_MIC) -#define SOUND_MIXER_WRITE_CD MIXER_WRITE(SOUND_MIXER_CD) -#define SOUND_MIXER_WRITE_IMIX MIXER_WRITE(SOUND_MIXER_IMIX) -#define SOUND_MIXER_WRITE_ALTPCM MIXER_WRITE(SOUND_MIXER_ALTPCM) -#define SOUND_MIXER_WRITE_RECLEV MIXER_WRITE(SOUND_MIXER_RECLEV) -#define SOUND_MIXER_WRITE_IGAIN MIXER_WRITE(SOUND_MIXER_IGAIN) -#define SOUND_MIXER_WRITE_OGAIN MIXER_WRITE(SOUND_MIXER_OGAIN) -#define SOUND_MIXER_WRITE_LINE1 MIXER_WRITE(SOUND_MIXER_LINE1) -#define SOUND_MIXER_WRITE_LINE2 MIXER_WRITE(SOUND_MIXER_LINE2) -#define SOUND_MIXER_WRITE_LINE3 MIXER_WRITE(SOUND_MIXER_LINE3) - -/* Obsolete macros */ -#define SOUND_MIXER_WRITE_MUTE MIXER_WRITE(SOUND_MIXER_MUTE) -#define SOUND_MIXER_WRITE_ENHANCE MIXER_WRITE(SOUND_MIXER_ENHANCE) -#define SOUND_MIXER_WRITE_LOUD MIXER_WRITE(SOUND_MIXER_LOUD) - -#define SOUND_MIXER_WRITE_RECSRC MIXER_WRITE(SOUND_MIXER_RECSRC) - -typedef struct mixer_info /* OBSOLETE */ -{ - char id[16]; - char name[32]; - int modify_counter; - int card_number; - int port_number; - char handle[32]; -} mixer_info; - -/* SOUND_MIXER_INFO is obsolete - use SNDCTL_MIXERINFO instead */ -#define SOUND_MIXER_INFO __SIOR ('M', 101, mixer_info) - -/* - * Two ioctls for special souncard function (OSS/Free only) - */ -#define SOUND_MIXER_AGC _SIOWR('M', 103, int) -#define SOUND_MIXER_3DSE _SIOWR('M', 104, int) -/* - * The SOUND_MIXER_PRIVATE# commands can be redefined by low level drivers. - * These features can be used when accessing device specific features. - */ -#define SOUND_MIXER_PRIVATE1 __SIOWR('M', 111, int) -#define SOUND_MIXER_PRIVATE2 __SIOWR('M', 112, int) -#define SOUND_MIXER_PRIVATE3 __SIOWR('M', 113, int) -#define SOUND_MIXER_PRIVATE4 __SIOWR('M', 114, int) -#define SOUND_MIXER_PRIVATE5 __SIOWR('M', 115, int) - -/* The following two controls were never implemented and they should not be used. */ -#define SOUND_MIXER_READ_MAINVOL __SIOR ('M', 116, int) -#define SOUND_MIXER_WRITE_MAINVOL __SIOWR('M', 116, int) - -/* - * SOUND_MIXER_GETLEVELS and SOUND_MIXER_SETLEVELS calls can be used - * for querying current mixer settings from the driver and for loading - * default volume settings _prior_ activating the mixer (loading - * doesn't affect current state of the mixer hardware). These calls - * are for internal use by the driver software only. - */ - -typedef struct mixer_vol_table -{ - int num; /* Index to volume table */ - char name[32]; - int levels[32]; -} mixer_vol_table; - -#define SOUND_MIXER_GETLEVELS __SIOWR('M', 116, mixer_vol_table) -#define SOUND_MIXER_SETLEVELS __SIOWR('M', 117, mixer_vol_table) - -#define OSS_GETVERSION __SIOR ('M', 118, int) +#define SNDCTL_DSP_GETIPEAKS _IOR('P', 43, oss_peaks_t) +#define SNDCTL_DSP_GETOPEAKS _IOR('P', 44, oss_peaks_t) +#define SNDCTL_DSP_POLICY _IOW('P', 45, int) /* See the manual */ /* - * Calls to set/get the recording gain for the currently active - * recording source. These calls automatically map to the right control. - * Note that these calls are not supported by all drivers. In this case - * the call will return -1 with errno set to EINVAL - * - * The _MONGAIN work in similar way but set/get the monitoring gain for - * the currently selected recording source. + * OSS_SYSIFO is obsolete. Use SNDCTL_SYSINFO insteads. */ -#define SOUND_MIXER_READ_RECGAIN __SIOR ('M', 119, int) -#define SOUND_MIXER_WRITE_RECGAIN __SIOWR('M', 119, int) -#define SOUND_MIXER_READ_MONGAIN __SIOR ('M', 120, int) -#define SOUND_MIXER_WRITE_MONGAIN __SIOWR('M', 120, int) +#define OSS_GETVERSION _IOR ('M', 118, int) -/* The following call is for driver development time purposes. It's not - * present in any released drivers. - */ -typedef unsigned char oss_reserved_t[512]; -#define SOUND_MIXER_RESERVED __SIOWR('M', 121, oss_reserved_t) -#endif /* ifndef NO_LEGACY_MIXER */ - -/* - ************************************************************************* - * The "new" mixer API of OSS 4.0 and later. +/** + * @brief Argument for SNDCTL_SYSINFO ioctl. * - * This improved mixer API makes it possible to access every possible feature - * of every possible device. However you should read the mixer programming - * section of the OSS API Developer's Manual. There is no chance that you - * could use this interface correctly just by examining this header. + * For use w/ the SNDCTL_SYSINFO ioctl available on audio (/dev/dsp*), + * mixer, and MIDI devices. */ - typedef struct oss_sysinfo { - char product[32]; /* For example OSS/Free, OSS/Linux or OSS/Solaris */ - char version[32]; /* For example 4.0a */ - int versionnum; /* See OSS_GETVERSION */ - char options[128]; /* Reserved */ - - int numaudios; /* # of audio/dsp devices */ - int openedaudio[8]; /* Bit mask telling which audio devices are busy */ - - int numsynths; /* # of availavle synth devices */ - int nummidis; /* # of available MIDI ports */ - int numtimers; /* # of available timer devices */ - int nummixers; /* # of mixer devices */ - - int openedmidi[8]; /* Bit mask telling which midi devices are busy */ - int numcards; /* Number of sound cards in the system */ - int numaudioengines; /* Number of audio engines in the system */ - int filler[240]; /* For future expansion (set to -1) */ + char product[32]; /* For example OSS/Free, OSS/Linux or + OSS/Solaris */ + char version[32]; /* For example 4.0a */ + int versionnum; /* See OSS_GETVERSION */ + char options[128]; /* Reserved */ + + int numaudios; /* # of audio/dsp devices */ + int openedaudio[8]; /* Bit mask telling which audio devices + are busy */ + + int numsynths; /* # of availavle synth devices */ + int nummidis; /* # of available MIDI ports */ + int numtimers; /* # of available timer devices */ + int nummixers; /* # of mixer devices */ + + int openedmidi[8]; /* Bit mask telling which midi devices + are busy */ + int numcards; /* Number of sound cards in the system */ + int filler[241]; /* For future expansion (set to -1) */ } oss_sysinfo; typedef struct oss_mixext @@ -1765,8 +1651,8 @@ typedef struct oss_mixext # define MIXT_GROUP 1 /* Controller group */ # define MIXT_ONOFF 2 /* OFF (0) or ON (1) */ # define MIXT_ENUM 3 /* Enumerated (0 to maxvalue) */ -# define MIXT_MONOSLIDER 4 /* Mono slider (0 to 255) */ -# define MIXT_STEREOSLIDER 5 /* Stereo slider (dual 0 to 255) */ +# define MIXT_MONOSLIDER 4 /* Mono slider (0 to 100) */ +# define MIXT_STEREOSLIDER 5 /* Stereo slider (dual 0 to 100) */ # define MIXT_MESSAGE 6 /* (Readable) textual message */ # define MIXT_MONOVU 7 /* VU meter value (mono) */ # define MIXT_STEREOVU 8 /* VU meter value (stereo) */ @@ -1776,17 +1662,11 @@ typedef struct oss_mixext # define MIXT_MARKER 12 /* Separator between normal and extension entries */ # define MIXT_VALUE 13 /* Decimal value entry */ # define MIXT_HEXVALUE 14 /* Hexadecimal value entry */ -# define MIXT_MONODB 15 /* OBSOLETE */ -# define MIXT_STEREODB 16 /* OBSOLETE */ -# define MIXT_SLIDER 17 /* Slider (mono) with full (31 bit) postitive integer range */ +# define MIXT_MONODB 15 /* Mono atten. slider (0 to -144) */ +# define MIXT_STEREODB 16 /* Stereo atten. slider (dual 0 to -144) */ +# define MIXT_SLIDER 17 /* Slider (mono) with full integer range */ # define MIXT_3D 18 -/* - * Sliders with range expanded to 15 bits per channel (0-32767) - */ -# define MIXT_MONOSLIDER16 19 -# define MIXT_STEREOSLIDER16 20 - /* Possible value range (minvalue to maxvalue) */ /* Note that maxvalue may also be smaller than minvalue */ int maxvalue; @@ -1802,11 +1682,6 @@ typedef struct oss_mixext # define MIXF_OKFAIL 0x00000020 /* Interpret value as 1=OK, 0=FAIL */ # define MIXF_FLAT 0x00000040 /* Flat vertical space requirements */ # define MIXF_LEGACY 0x00000080 /* Legacy mixer control group */ -# define MIXF_CENTIBEL 0x00000100 /* Centibel (0.1 dB) step size */ -# define MIXF_DECIBEL 0x00000200 /* Step size of 1 dB */ -# define MIXF_MAINVOL 0x00000400 /* Main volume control */ -# define MIXF_PCMVOL 0x00000800 /* PCM output volume control */ -# define MIXF_RECVOL 0x00001000 /* PCM recording volume control */ char id[16]; /* Mnemonic ID (mainly for internal use) */ int parent; /* Entry# of parent (group) node (-1 if root) */ @@ -1852,51 +1727,62 @@ typedef struct oss_mixer_value int filler[8]; /* Reserved for future use. Initialize to 0 */ } oss_mixer_value; -#define OSS_ENUM_MAXVALUE 255 +#define OSS_ENUM_MAXVALUE 255 typedef struct oss_mixer_enuminfo { - int dev; - int ctrl; - int nvalues; - int version; /* Read the manual */ - short strindex[OSS_ENUM_MAXVALUE]; - char strings[3000]; + int dev; + int ctrl; + int nvalues; + int version; /* Read the manual */ + short strindex[OSS_ENUM_MAXVALUE]; + char strings[3000]; } oss_mixer_enuminfo; -#define OPEN_READ PCM_ENABLE_INPUT -#define OPEN_WRITE PCM_ENABLE_OUTPUT -#define OPEN_READWRITE (OPEN_READ|OPEN_WRITE) +#define OPEN_READ PCM_ENABLE_INPUT +#define OPEN_WRITE PCM_ENABLE_OUTPUT +#define OPEN_READWRITE (OPEN_READ|OPEN_WRITE) +/** + * @brief Argument for SNDCTL_AUDIOINFO ioctl. + * + * For use w/ the SNDCTL_AUDIOINFO ioctl available on audio (/dev/dsp*) + * devices. + */ typedef struct oss_audioinfo { - int dev; /* Audio device number */ - char name[64]; - int busy; /* 0, OPEN_READ, OPEN_WRITE or OPEN_READWRITE */ - int pid; - int caps; /* PCM_CAP_INPUT, PCM_CAP_OUTPUT */ - int iformats, oformats; - int magic; /* Reserved for internal use */ - char cmd[64]; /* Command using the device (if known) */ - int card_number; - int port_number; - int mixer_dev; - int legacy_device; /* Obsolete field. Replaced by devnode */ - int enabled; /* 1=enabled, 0=device not ready at this moment */ - int flags; /* For internal use only - no practical meaning */ - int min_rate, max_rate; /* Sample rate limits */ - int min_channels, max_channels; /* Number of channels supported */ - int binding; /* DSP_BIND_FRONT, etc. 0 means undefined */ - int rate_source; - char handle[32]; -#define OSS_MAX_SAMPLE_RATES 20 /* Cannot be changed */ - unsigned int nrates, rates[OSS_MAX_SAMPLE_RATES]; /* Please read the manual before using these */ - oss_longname_t song_name; /* Song name (if given) */ - oss_label_t label; /* Device label (if given) */ - int latency; /* In usecs, -1=unknown */ - oss_devnode_t devnode; /* Device special file name (absolute path) */ - int next_play_engine; /* Read the documentation for more info */ - int next_rec_engine; /* Read the documentation for more info */ - int filler[184]; + int dev; /* Audio device number */ + char name[64]; + int busy; /* 0, OPEN_READ, OPEN_WRITE or OPEN_READWRITE */ + int pid; + int caps; /* DSP_CAP_INPUT, DSP_CAP_OUTPUT */ + int iformats; + int oformats; + int magic; /* Reserved for internal use */ + char cmd[64]; /* Command using the device (if known) */ + int card_number; + int port_number; + int mixer_dev; + int real_device; /* Obsolete field. Replaced by devnode */ + int enabled; /* 1=enabled, 0=device not ready at this + moment */ + int flags; /* For internal use only - no practical + meaning */ + int min_rate; /* Sample rate limits */ + int max_rate; + int min_channels; /* Number of channels supported */ + int max_channels; + int binding; /* DSP_BIND_FRONT, etc. 0 means undefined */ + int rate_source; + char handle[32]; + #define OSS_MAX_SAMPLE_RATES 20 /* Cannot be changed */ + unsigned int nrates; + unsigned int rates[OSS_MAX_SAMPLE_RATES]; /* Please read the manual before using these */ + oss_longname_t song_name; /* Song name (if given) */ + oss_label_t label; /* Device label (if given) */ + int latency; /* In usecs, -1=unknown */ + oss_devnode_t devnode; /* Device special file name (inside + /dev) */ + int filler[186]; } oss_audioinfo; typedef struct oss_mixerinfo @@ -1911,9 +1797,7 @@ typedef struct oss_mixerinfo int magic; /* Reserved */ int enabled; /* Reserved */ int caps; -#define MIXER_CAP_VIRTUAL 0x00000001 -#define MIXER_CAP_LAYOUT_B 0x00000002 /* For internal use only */ -#define MIXER_CAP_NARROW 0x00000004 /* Conserve horiz space */ +#define MIXER_CAP_VIRTUAL 0x00000001 int flags; /* Reserved */ int nrext; /* @@ -1923,9 +1807,7 @@ typedef struct oss_mixerinfo * as the default mixer. */ int priority; - oss_devnode_t devnode; /* Device special file name (absolute path) */ - int legacy_device; - int filler[245]; /* Reserved */ + int filler[254]; /* Reserved */ } oss_mixerinfo; typedef struct oss_midi_info @@ -1957,9 +1839,7 @@ typedef struct oss_midi_info oss_longname_t song_name; /* Song name (if known) */ oss_label_t label; /* Device label (if given) */ int latency; /* In usecs, -1=unknown */ - oss_devnode_t devnode; /* Device special file name (absolute path) */ - int legacy_device; /* Legacy device mapping */ - int filler[235]; + int filler[244]; } oss_midi_info; typedef struct oss_card_info @@ -1971,91 +1851,28 @@ typedef struct oss_card_info int filler[256]; } oss_card_info; -#define SNDCTL_SYSINFO __SIOR ('X', 1, oss_sysinfo) -#define OSS_SYSINFO SNDCTL_SYSINFO /* Old name */ - -#define SNDCTL_MIX_NRMIX __SIOR ('X', 2, int) -#define SNDCTL_MIX_NREXT __SIOWR('X', 3, int) -#define SNDCTL_MIX_EXTINFO __SIOWR('X', 4, oss_mixext) -#define SNDCTL_MIX_READ __SIOWR('X', 5, oss_mixer_value) -#define SNDCTL_MIX_WRITE __SIOWR('X', 6, oss_mixer_value) +#define SNDCTL_SYSINFO _IOR ('X', 1, oss_sysinfo) +#define OSS_SYSINFO SNDCTL_SYSINFO /* Old name */ -#define SNDCTL_AUDIOINFO __SIOWR('X', 7, oss_audioinfo) -#define SNDCTL_MIX_ENUMINFO __SIOWR('X', 8, oss_mixer_enuminfo) -#define SNDCTL_MIDIINFO __SIOWR('X', 9, oss_midi_info) -#define SNDCTL_MIXERINFO __SIOWR('X',10, oss_mixerinfo) -#define SNDCTL_CARDINFO __SIOWR('X',11, oss_card_info) -#define SNDCTL_ENGINEINFO __SIOWR('X',12, oss_audioinfo) -#define SNDCTL_AUDIOINFO_EX __SIOWR('X',13, oss_audioinfo) +#define SNDCTL_MIX_NRMIX _IOR ('X', 2, int) +#define SNDCTL_MIX_NREXT _IOWR('X', 3, int) +#define SNDCTL_MIX_EXTINFO _IOWR('X', 4, oss_mixext) +#define SNDCTL_MIX_READ _IOWR('X', 5, oss_mixer_value) +#define SNDCTL_MIX_WRITE _IOWR('X', 6, oss_mixer_value) -/* ioctl codes 'X', 200-255 are reserved for internal use */ +#define SNDCTL_AUDIOINFO _IOWR('X', 7, oss_audioinfo) +#define SNDCTL_MIX_ENUMINFO _IOWR('X', 8, oss_mixer_enuminfo) +#define SNDCTL_MIDIINFO _IOWR('X', 9, oss_midi_info) +#define SNDCTL_MIXERINFO _IOWR('X',10, oss_mixerinfo) +#define SNDCTL_CARDINFO _IOWR('X',11, oss_card_info) /* * Few more "globally" available ioctl calls. */ -#define SNDCTL_SETSONG __SIOW ('Y', 2, oss_longname_t) -#define SNDCTL_GETSONG __SIOR ('Y', 2, oss_longname_t) -#define SNDCTL_SETNAME __SIOW ('Y', 3, oss_longname_t) -#define SNDCTL_SETLABEL __SIOW ('Y', 4, oss_label_t) -#define SNDCTL_GETLABEL __SIOR ('Y', 4, oss_label_t) -/* - * The "new" mixer API definitions end here. - *************************************** - */ - -/* - ********************************************************* - * Few routines that are included in -lOSSlib - * - * At this moment this interface is not used. OSSlib contains just - * stubs that call the related system calls directly. - */ -#ifdef OSSLIB -extern int osslib_open (const char *path, int flags, int dummy); -extern void osslib_close (int fd); -extern int osslib_write (int fd, const void *buf, int count); -extern int osslib_read (int fd, void *buf, int count); -extern int osslib_ioctl (int fd, unsigned int request, void *arg); -#else -# define osslib_open open -# define osslib_close close -# define osslib_write write -# define osslib_read read -# define osslib_ioctl ioctl -#endif - -#if 1 -#define SNDCTL_DSP_NONBLOCK __SIO ('P',14) /* Obsolete. Not supported any more */ -#endif - -#if 1 -/* - * Some obsolete macros that are not part of Open Sound System API. - */ -#define SOUND_PCM_READ_RATE SOUND_PCM_READ_RATE_is_obsolete -#define SOUND_PCM_READ_BITS SOUND_PCM_READ_BITS_is_obsolete -#define SOUND_PCM_READ_CHANNELS SOUND_PCM_READ_CHANNELS_is_obsolete -#define SOUND_PCM_WRITE_RATE SOUND_PCM_WRITE_RATE_is_obsolet_use_SNDCTL_DSP_SPEED_instead -#define SOUND_PCM_WRITE_CHANNELS SOUND_PCM_WRITE_CHANNELS_is_obsolete_use_SNDCTL_DSP_CHANNELS_instead -#define SOUND_PCM_WRITE_BITS SOUND_PCM_WRITE_BITS_is_obsolete_use_SNDCTL_DSP_SETFMT_instead -#define SOUND_PCM_POST SOUND_PCM_POST_is_obsolete_use_SNDCTL_DSP_POST_instead -#define SOUND_PCM_RESET SOUND_PCM_RESET_is_obsolete_use_SNDCTL_DSP_HALT_instead -#define SOUND_PCM_SYNC SOUND_PCM_SYNC_is_obsolete_use_SNDCTL_DSP_SYNC_instead -#define SOUND_PCM_SUBDIVIDE SOUND_PCM_SUBDIVIDE_is_obsolete_use_SNDCTL_DSP_SUBDIVIDE_instead -#define SOUND_PCM_SETFRAGMENT SOUND_PCM_SETFRAGMENT_is_obsolete_use_SNDCTL_DSP_SETFRAGMENT_instead -#define SOUND_PCM_GETFMTS SOUND_PCM_GETFMTS_is_obsolete_use_SNDCTL_DSP_GETFMTS_instead -#define SOUND_PCM_SETFMT SOUND_PCM_SETFMT_is_obsolete_use_SNDCTL_DSP_SETFMT_instead -#define SOUND_PCM_GETOSPACE SOUND_PCM_GETOSPACE_is_obsolete_use_SNDCTL_DSP_GETOSPACE_instead -#define SOUND_PCM_GETISPACE SOUND_PCM_GETISPACE_is_obsolete_use_SNDCTL_DSP_GETISPACE_instead -#define SOUND_PCM_NONBLOCK SOUND_PCM_NONBLOCK_is_obsolete_use_SNDCTL_DSP_NONBLOCK_instead -#define SOUND_PCM_GETCAPS SOUND_PCM_GETCAPS_is_obsolete_use_SNDCTL_DSP_GETCAPS_instead -#define SOUND_PCM_GETTRIGGER SOUND_PCM_GETTRIGGER_is_obsolete_use_SNDCTL_DSP_GETTRIGGER_instead -#define SOUND_PCM_SETTRIGGER SOUND_PCM_SETTRIGGER_is_obsolete_use_SNDCTL_DSP_SETTRIGGER_instead -#define SOUND_PCM_SETSYNCRO SOUND_PCM_SETSYNCRO_is_obsolete_use_SNDCTL_DSP_SETSYNCRO_instead -#define SOUND_PCM_GETIPTR SOUND_PCM_GETIPTR_is_obsolete_use_SNDCTL_DSP_GETIPTR_instead -#define SOUND_PCM_GETOPTR SOUND_PCM_GETOPTR_is_obsolete_use_SNDCTL_DSP_GETOPTR_instead -#define SOUND_PCM_MAPINBUF SOUND_PCM_MAPINBUF_is_obsolete_use_SNDCTL_DSP_MAPINBUF_instead -#define SOUND_PCM_MAPOUTBUF SOUND_PCM_MAPOUTBUF_is_obsolete_use_SNDCTL_DSP_MAPOUTBUF_instead -#endif +#define SNDCTL_SETSONG _IOW ('Y', 2, oss_longname_t) +#define SNDCTL_GETSONG _IOR ('Y', 2, oss_longname_t) +#define SNDCTL_SETNAME _IOW ('Y', 3, oss_longname_t) +#define SNDCTL_SETLABEL _IOW ('Y', 4, oss_label_t) +#define SNDCTL_GETLABEL _IOR ('Y', 4, oss_label_t) -#endif +#endif /* !_SYS_SOUNDCARD_H_ */ diff --git a/install b/install index 0e01d12..e590d96 100644 --- a/install +++ b/install @@ -1,4 +1,4 @@ -RtAudio - a set of C++ classes which provide a common API for realtime audio input/output across Linux (native ALSA, JACK, and OSS), Macintosh OS X (CoreAudio and JACK), and Windows (DirectSound and ASIO) operating systems. +RtAudio - a set of C++ classes which provide a common API for realtime audio input/output across Linux (native ALSA, JACK, PulseAudio, and OSS), Macintosh OS X (CoreAudio and JACK), and Windows (DirectSound and ASIO) operating systems. By Gary P. Scavone, 2001-2012. @@ -17,6 +17,7 @@ A few options can be passed to configure, including: --enable-debug = enable various debug output --with-alsa = choose native ALSA API support (linux only) + --with-pulse = choose native PulseAudio API support (linux only) --with-oss = choose OSS API support (linux only) --with-jack = choose JACK server support (linux or Macintosh OS-X) --with-core = choose CoreAudio API support (Macintosh OS-X only) diff --git a/readme b/readme index ae73eb2..2b88884 100644 --- a/readme +++ b/readme @@ -1,4 +1,4 @@ -RtAudio - a set of C++ classes that provide a common API for realtime audio input/output across Linux (native ALSA, JACK, and OSS), Macintosh OS X (CoreAudio and JACK), and Windows (DirectSound and ASIO) operating systems. +RtAudio - a set of C++ classes that provide a common API for realtime audio input/output across Linux (native ALSA, JACK, PulseAudio and OSS), Macintosh OS X (CoreAudio and JACK), and Windows (DirectSound and ASIO) operating systems. By Gary P. Scavone, 2001-2012. diff --git a/tests/duplex.cpp b/tests/duplex.cpp index 938a7f8..9991dcf 100644 --- a/tests/duplex.cpp +++ b/tests/duplex.cpp @@ -19,19 +19,21 @@ typedef signed long MY_TYPE; 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_SINT32 typedef float MY_TYPE; #define FORMAT RTAUDIO_FLOAT32 -*/ typedef double MY_TYPE; #define FORMAT RTAUDIO_FLOAT64 +*/ void usage( void ) { // Error function in case of incorrect command-line diff --git a/tests/playsaw.cpp b/tests/playsaw.cpp index 51f8ad9..2117b54 100644 --- a/tests/playsaw.cpp +++ b/tests/playsaw.cpp @@ -157,7 +157,7 @@ int main( int argc, char *argv[] ) dac.showWarnings( true ); // Set our stream parameters for output only. - bufferFrames = 256; + bufferFrames = 512; RtAudio::StreamParameters oParams; oParams.deviceId = device; oParams.nChannels = channels; diff --git a/tests/testall.cpp b/tests/testall.cpp index 695488d..f518a57 100644 --- a/tests/testall.cpp +++ b/tests/testall.cpp @@ -122,7 +122,7 @@ int main( int argc, char *argv[] ) dac.showWarnings( true ); // Set our stream parameters for output only. - bufferFrames = 256; + bufferFrames = 512; RtAudio::StreamParameters oParams, iParams; oParams.deviceId = oDevice; oParams.nChannels = channels; diff --git a/tests/teststops.cpp b/tests/teststops.cpp index 6122734..ab71dbd 100644 --- a/tests/teststops.cpp +++ b/tests/teststops.cpp @@ -111,7 +111,7 @@ int main( int argc, char *argv[] ) pausetime = PAUSETIME * 1000; // Set our stream parameters for a duplex stream. - bufferFrames = 256; + bufferFrames = 512; RtAudio::StreamParameters oParams, iParams; oParams.deviceId = oDevice; oParams.nChannels = mydata.channels; -- 2.30.2