#define MUTEX_DESTROY(A) DeleteCriticalSection(A)\r
#define MUTEX_LOCK(A) EnterCriticalSection(A)\r
#define MUTEX_UNLOCK(A) LeaveCriticalSection(A)\r
-#elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)\r
+#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)\r
// pthread API\r
#define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)\r
#define MUTEX_DESTROY(A) pthread_mutex_destroy(A)\r
#if defined(__LINUX_ALSA__)\r
apis.push_back( LINUX_ALSA );\r
#endif\r
+#if defined(__LINUX_PULSE__)\r
+ apis.push_back( LINUX_PULSE );\r
+#endif\r
#if defined(__LINUX_OSS__)\r
apis.push_back( LINUX_OSS );\r
#endif\r
\r
void RtAudio :: openRtApi( RtAudio::Api api )\r
{\r
- if (rtapi_)\r
+ if ( rtapi_ )\r
delete rtapi_;\r
rtapi_ = 0;\r
\r
if ( api == LINUX_ALSA )\r
rtapi_ = new RtApiAlsa();\r
#endif\r
+#if defined(__LINUX_PULSE__)\r
+ if ( api == LINUX_PULSE )\r
+ rtapi_ = new RtApiPulse();\r
+#endif\r
#if defined(__LINUX_OSS__)\r
if ( api == LINUX_OSS )\r
rtapi_ = new RtApiOss();\r
return;\r
}\r
\r
- //MUTEX_LOCK( &stream_.mutex );\r
-\r
OSStatus result = noErr;\r
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;\r
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
stream_.state = STREAM_RUNNING;\r
\r
unlock:\r
- //MUTEX_UNLOCK( &stream_.mutex );\r
-\r
if ( result == noErr ) return;\r
error( RtError::SYSTEM_ERROR );\r
}\r
return;\r
}\r
\r
- /*\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- return;\r
- }\r
- */\r
-\r
OSStatus result = noErr;\r
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;\r
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled\r
}\r
\r
- //MUTEX_UNLOCK( &stream_.mutex );\r
result = AudioDeviceStop( handle->id[0], callbackHandler );\r
- //MUTEX_LOCK( &stream_.mutex );\r
if ( result != noErr ) {\r
errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";\r
errorText_ = errorStream_.str();\r
\r
if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {\r
\r
- //MUTEX_UNLOCK( &stream_.mutex );\r
result = AudioDeviceStop( handle->id[1], callbackHandler );\r
- //MUTEX_LOCK( &stream_.mutex );\r
if ( result != noErr ) {\r
errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";\r
errorText_ = errorStream_.str();\r
stream_.state = STREAM_STOPPED;\r
\r
unlock:\r
- //MUTEX_UNLOCK( &stream_.mutex );\r
-\r
if ( result == noErr ) return;\r
error( RtError::SYSTEM_ERROR );\r
}\r
RtApiCore *object = (RtApiCore *) info->object;\r
\r
object->stopStream();\r
-\r
pthread_exit( NULL );\r
}\r
\r
// Check if we were draining the stream and signal is finished.\r
if ( handle->drainCounter > 3 ) {\r
\r
- if ( handle->internalDrain == true ) {\r
- stream_.state = STREAM_STOPPING;\r
+ stream_.state = STREAM_STOPPING;\r
+ if ( handle->internalDrain == true )\r
pthread_create( &threadId, NULL, coreStopStream, info );\r
- //stopStream();\r
- }\r
else // external call to stopStream()\r
pthread_cond_signal( &handle->condition );\r
return SUCCESS;\r
}\r
\r
- /*\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- // The state might change while waiting on a mutex.\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- return SUCCESS;\r
- }\r
- */\r
-\r
AudioDeviceID outputDevice = handle->id[0];\r
\r
// Invoke user callback to get fresh output data UNLESS we are\r
int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
stream_.bufferSize, streamTime, status, info->userData );\r
if ( cbReturnValue == 2 ) {\r
- //MUTEX_UNLOCK( &stream_.mutex );\r
+ stream_.state = STREAM_STOPPING;\r
handle->drainCounter = 2;\r
abortStream();\r
return SUCCESS;\r
return;\r
}\r
\r
- MUTEX_LOCK(&stream_.mutex);\r
-\r
JackHandle *handle = (JackHandle *) stream_.apiHandle;\r
int result = jack_activate( handle->client );\r
if ( result ) {\r
stream_.state = STREAM_RUNNING;\r
\r
unlock:\r
- MUTEX_UNLOCK(&stream_.mutex);\r
-\r
if ( result == 0 ) return;\r
error( RtError::SYSTEM_ERROR );\r
}\r
return;\r
}\r
\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- return;\r
- }\r
-\r
JackHandle *handle = (JackHandle *) stream_.apiHandle;\r
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
\r
\r
jack_deactivate( handle->client );\r
stream_.state = STREAM_STOPPED;\r
-\r
- MUTEX_UNLOCK( &stream_.mutex );\r
}\r
\r
void RtApiJack :: abortStream( void )\r
RtApiJack *object = (RtApiJack *) info->object;\r
\r
object->stopStream();\r
-\r
pthread_exit( NULL );\r
}\r
\r
bool RtApiJack :: callbackEvent( unsigned long nframes )\r
{\r
- if ( stream_.state == STREAM_STOPPED ) return SUCCESS;\r
+ if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;\r
if ( stream_.state == STREAM_CLOSED ) {\r
errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";\r
error( RtError::WARNING );\r
\r
// Check if we were draining the stream and signal is finished.\r
if ( handle->drainCounter > 3 ) {\r
+\r
+ stream_.state = STREAM_STOPPING;\r
if ( handle->internalDrain == true )\r
pthread_create( &threadId, NULL, jackStopStream, info );\r
else\r
return SUCCESS;\r
}\r
\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- // The state might change while waiting on a mutex.\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- return SUCCESS;\r
- }\r
-\r
// Invoke user callback first, to get fresh output data.\r
if ( handle->drainCounter == 0 ) {\r
RtAudioCallback callback = (RtAudioCallback) info->callback;\r
int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
stream_.bufferSize, streamTime, status, info->userData );\r
if ( cbReturnValue == 2 ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- ThreadHandle id;\r
+ stream_.state = STREAM_STOPPING;\r
handle->drainCounter = 2;\r
+ ThreadHandle id;\r
pthread_create( &id, NULL, jackStopStream, info );\r
return SUCCESS;\r
}\r
}\r
\r
unlock:\r
- MUTEX_UNLOCK(&stream_.mutex);\r
-\r
RtApi::tickStreamTime();\r
return SUCCESS;\r
}\r
return;\r
}\r
\r
- //MUTEX_LOCK( &stream_.mutex );\r
-\r
AsioHandle *handle = (AsioHandle *) stream_.apiHandle;\r
ASIOError result = ASIOStart();\r
if ( result != ASE_OK ) {\r
asioXRun = false;\r
\r
unlock:\r
- //MUTEX_UNLOCK( &stream_.mutex );\r
-\r
stopThreadCalled = false;\r
\r
if ( result == ASE_OK ) return;\r
return;\r
}\r
\r
- /*\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- return;\r
- }\r
- */\r
-\r
AsioHandle *handle = (AsioHandle *) stream_.apiHandle;\r
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
if ( handle->drainCounter == 0 ) {\r
handle->drainCounter = 2;\r
- // MUTEX_UNLOCK( &stream_.mutex );\r
WaitForSingleObject( handle->condition, INFINITE ); // block until signaled\r
- //ResetEvent( handle->condition );\r
- // MUTEX_LOCK( &stream_.mutex );\r
}\r
}\r
\r
errorText_ = errorStream_.str();\r
}\r
\r
- // MUTEX_UNLOCK( &stream_.mutex );\r
-\r
if ( result == ASE_OK ) return;\r
error( RtError::SYSTEM_ERROR );\r
}\r
RtApiAsio *object = (RtApiAsio *) info->object;\r
\r
object->stopStream();\r
-\r
_endthreadex( 0 );\r
return 0;\r
}\r
\r
bool RtApiAsio :: callbackEvent( long bufferIndex )\r
{\r
- if ( stream_.state == STREAM_STOPPED ) return SUCCESS;\r
- if ( stopThreadCalled ) return SUCCESS;\r
+ if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;\r
if ( stream_.state == STREAM_CLOSED ) {\r
errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";\r
error( RtError::WARNING );\r
\r
// Check if we were draining the stream and signal if finished.\r
if ( handle->drainCounter > 3 ) {\r
+\r
+ stream_.state = STREAM_STOPPING;\r
if ( handle->internalDrain == false )\r
SetEvent( handle->condition );\r
else { // spawn a thread to stop the stream\r
unsigned threadId;\r
- stopThreadCalled = true;\r
stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,\r
&stream_.callbackInfo, 0, &threadId );\r
}\r
return SUCCESS;\r
}\r
\r
- /*MUTEX_LOCK( &stream_.mutex );\r
-\r
- // The state might change while waiting on a mutex.\r
- if ( stream_.state == STREAM_STOPPED ) goto unlock; */\r
-\r
// Invoke user callback to get fresh output data UNLESS we are\r
// draining stream.\r
if ( handle->drainCounter == 0 ) {\r
int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
stream_.bufferSize, streamTime, status, info->userData );\r
if ( cbReturnValue == 2 ) {\r
- // MUTEX_UNLOCK( &stream_.mutex );\r
- // abortStream();\r
- unsigned threadId;\r
- stopThreadCalled = true;\r
+ stream_.state = STREAM_STOPPING;\r
handle->drainCounter = 2;\r
+ unsigned threadId;\r
stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,\r
&stream_.callbackInfo, 0, &threadId );\r
return SUCCESS;\r
// drivers apparently do not function correctly without it.\r
ASIOOutputReady();\r
\r
- // MUTEX_UNLOCK( &stream_.mutex );\r
-\r
RtApi::tickStreamTime();\r
return SUCCESS;\r
}\r
return;\r
}\r
\r
- //MUTEX_LOCK( &stream_.mutex );\r
- \r
DsHandle *handle = (DsHandle *) stream_.apiHandle;\r
\r
// Increase scheduler frequency on lesser windows (a side-effect of\r
stream_.state = STREAM_RUNNING;\r
\r
unlock:\r
- // MUTEX_UNLOCK( &stream_.mutex );\r
-\r
if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );\r
}\r
\r
return;\r
}\r
\r
- /*\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- return;\r
- }\r
- */\r
-\r
HRESULT result = 0;\r
LPVOID audioPtr;\r
DWORD dataLen;\r
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
if ( handle->drainCounter == 0 ) {\r
handle->drainCounter = 2;\r
- // MUTEX_UNLOCK( &stream_.mutex );\r
WaitForSingleObject( handle->condition, INFINITE ); // block until signaled\r
- //ResetEvent( handle->condition );\r
- // MUTEX_LOCK( &stream_.mutex );\r
}\r
\r
stream_.state = STREAM_STOPPED;\r
\r
unlock:\r
timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.\r
- // MUTEX_UNLOCK( &stream_.mutex );\r
-\r
if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );\r
}\r
\r
\r
void RtApiDs :: callbackEvent()\r
{\r
- if ( stream_.state == STREAM_STOPPED ) {\r
+ if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {\r
Sleep( 50 ); // sleep 50 milliseconds\r
return;\r
}\r
\r
// Check if we were draining the stream and signal is finished.\r
if ( handle->drainCounter > stream_.nBuffers + 2 ) {\r
+\r
+ stream_.state = STREAM_STOPPING;\r
if ( handle->internalDrain == false )\r
SetEvent( handle->condition );\r
else\r
return;\r
}\r
\r
- /*\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- // The state might change while waiting on a mutex.\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- return;\r
- }\r
- */\r
-\r
// Invoke user callback to get fresh output data UNLESS we are\r
// draining stream.\r
if ( handle->drainCounter == 0 ) {\r
int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
stream_.bufferSize, streamTime, status, info->userData );\r
if ( cbReturnValue == 2 ) {\r
- // MUTEX_UNLOCK( &stream_.mutex );\r
+ stream_.state = STREAM_STOPPING;\r
handle->drainCounter = 2;\r
abortStream();\r
return;\r
}\r
\r
unlock:\r
- // MUTEX_UNLOCK( &stream_.mutex );\r
-\r
RtApi::tickStreamTime();\r
}\r
\r
stream_.state = STREAM_STOPPED;\r
MUTEX_LOCK( &stream_.mutex );\r
\r
- //if ( stream_.state == STREAM_STOPPED ) {\r
- // MUTEX_UNLOCK( &stream_.mutex );\r
- // return;\r
- //}\r
-\r
int result = 0;\r
AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;\r
snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;\r
}\r
\r
unlock:\r
- stream_.state = STREAM_STOPPED;\r
MUTEX_UNLOCK( &stream_.mutex );\r
\r
if ( result >= 0 ) return;\r
stream_.state = STREAM_STOPPED;\r
MUTEX_LOCK( &stream_.mutex );\r
\r
- //if ( stream_.state == STREAM_STOPPED ) {\r
- // MUTEX_UNLOCK( &stream_.mutex );\r
- // return;\r
- //}\r
-\r
int result = 0;\r
AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;\r
snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;\r
}\r
\r
unlock:\r
- stream_.state = STREAM_STOPPED;\r
MUTEX_UNLOCK( &stream_.mutex );\r
\r
if ( result >= 0 ) return;\r
//******************** End of __LINUX_ALSA__ *********************//\r
#endif\r
\r
+#if defined(__LINUX_PULSE__)\r
+\r
+// Code written by Peter Meerwald, pmeerw@pmeerw.net\r
+// and Tristan Matthews.\r
+\r
+#include <pulse/error.h>\r
+#include <pulse/simple.h>\r
+#include <cstdio>\r
+\r
+namespace {\r
+const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,\r
+ 44100, 48000, 96000, 0}; }\r
+\r
+struct rtaudio_pa_format_mapping_t {\r
+ RtAudioFormat rtaudio_format;\r
+ pa_sample_format_t pa_format;\r
+};\r
+\r
+static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {\r
+ {RTAUDIO_SINT16, PA_SAMPLE_S16LE},\r
+ {RTAUDIO_SINT32, PA_SAMPLE_S32LE},\r
+ {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},\r
+ {0, PA_SAMPLE_INVALID}};\r
+\r
+struct PulseAudioHandle {\r
+ pa_simple *s_play;\r
+ pa_simple *s_rec;\r
+ pthread_t thread;\r
+ pthread_cond_t runnable_cv;\r
+ bool runnable;\r
+ PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }\r
+};\r
+\r
+RtApiPulse::~RtApiPulse()\r
+{\r
+ if ( stream_.state != STREAM_CLOSED )\r
+ closeStream();\r
+}\r
+\r
+unsigned int RtApiPulse::getDeviceCount( void )\r
+{\r
+ return 1;\r
+}\r
+\r
+RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int device )\r
+{\r
+ RtAudio::DeviceInfo info;\r
+ info.probed = true;\r
+ info.name = "PulseAudio";\r
+ info.outputChannels = 2;\r
+ info.inputChannels = 2;\r
+ info.duplexChannels = 2;\r
+ info.isDefaultOutput = true;\r
+ info.isDefaultInput = true;\r
+\r
+ for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )\r
+ info.sampleRates.push_back( *sr );\r
+\r
+ info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;\r
+\r
+ return info;\r
+}\r
+\r
+extern "C" void *pulseaudio_callback( void * user )\r
+{\r
+ CallbackInfo *cbi = static_cast<CallbackInfo *>( user );\r
+ RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );\r
+ volatile bool *isRunning = &cbi->isRunning;\r
+\r
+ while ( *isRunning ) {\r
+ pthread_testcancel();\r
+ context->callbackEvent();\r
+ }\r
+\r
+ pthread_exit( NULL );\r
+}\r
+\r
+void RtApiPulse::closeStream( void )\r
+{\r
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );\r
+\r
+ stream_.callbackInfo.isRunning = false;\r
+ if ( pah ) {\r
+ MUTEX_LOCK( &stream_.mutex );\r
+ if ( stream_.state == STREAM_STOPPED ) {\r
+ pah->runnable = true;\r
+ pthread_cond_signal( &pah->runnable_cv );\r
+ }\r
+ MUTEX_UNLOCK( &stream_.mutex );\r
+\r
+ pthread_join( pah->thread, 0 );\r
+ if ( pah->s_play ) {\r
+ pa_simple_flush( pah->s_play, NULL );\r
+ pa_simple_free( pah->s_play );\r
+ }\r
+ if ( pah->s_rec )\r
+ pa_simple_free( pah->s_rec );\r
+\r
+ pthread_cond_destroy( &pah->runnable_cv );\r
+ delete pah;\r
+ stream_.apiHandle = 0;\r
+ }\r
+\r
+ if ( stream_.userBuffer[0] ) {\r
+ free( stream_.userBuffer[0] );\r
+ stream_.userBuffer[0] = 0;\r
+ }\r
+ if ( stream_.userBuffer[1] ) {\r
+ free( stream_.userBuffer[1] );\r
+ stream_.userBuffer[1] = 0;\r
+ }\r
+\r
+ stream_.state = STREAM_CLOSED;\r
+ stream_.mode = UNINITIALIZED;\r
+}\r
+\r
+void RtApiPulse::callbackEvent( void )\r
+{\r
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );\r
+\r
+ if ( stream_.state == STREAM_STOPPED ) {\r
+ MUTEX_LOCK( &stream_.mutex );\r
+ while ( !pah->runnable )\r
+ pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );\r
+\r
+ if ( stream_.state != STREAM_RUNNING ) {\r
+ MUTEX_UNLOCK( &stream_.mutex );\r
+ return;\r
+ }\r
+ MUTEX_UNLOCK( &stream_.mutex );\r
+ }\r
+\r
+ if ( stream_.state == STREAM_CLOSED ) {\r
+ errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "\r
+ "this shouldn't happen!";\r
+ error( RtError::WARNING );\r
+ return;\r
+ }\r
+\r
+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;\r
+ double streamTime = getStreamTime();\r
+ RtAudioStreamStatus status = 0;\r
+ int doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
+ stream_.bufferSize, streamTime, status,\r
+ stream_.callbackInfo.userData );\r
+\r
+ if ( doStopStream == 2 ) {\r
+ abortStream();\r
+ return;\r
+ }\r
+\r
+ MUTEX_LOCK( &stream_.mutex );\r
+\r
+ if ( stream_.state != STREAM_RUNNING )\r
+ goto unlock;\r
+\r
+ int pa_error;\r
+ size_t bytes;\r
+ switch ( stream_.mode ) {\r
+ case INPUT:\r
+ bytes = stream_.nUserChannels[1] * stream_.bufferSize * formatBytes( stream_.userFormat );\r
+ if ( pa_simple_read( pah->s_rec, stream_.userBuffer[1], bytes, &pa_error ) < 0 ) {\r
+ errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<\r
+ pa_strerror( pa_error ) << ".";\r
+ errorText_ = errorStream_.str();\r
+ error( RtError::WARNING );\r
+ }\r
+ break;\r
+ case OUTPUT:\r
+ bytes = stream_.nUserChannels[0] * stream_.bufferSize * formatBytes( stream_.userFormat );\r
+ if ( pa_simple_write( pah->s_play, stream_.userBuffer[0], bytes, &pa_error ) < 0 ) {\r
+ errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<\r
+ pa_strerror( pa_error ) << ".";\r
+ errorText_ = errorStream_.str();\r
+ error( RtError::WARNING );\r
+ }\r
+ break;\r
+ case DUPLEX:\r
+ bytes = stream_.nUserChannels[1] * stream_.bufferSize * formatBytes( stream_.userFormat );\r
+ if ( pa_simple_read( pah->s_rec, stream_.userBuffer[1], bytes, &pa_error ) < 0 ) {\r
+ errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<\r
+ pa_strerror( pa_error ) << ".";\r
+ errorText_ = errorStream_.str();\r
+ error( RtError::WARNING );\r
+ }\r
+ bytes = stream_.nUserChannels[0] * stream_.bufferSize * formatBytes( stream_.userFormat );\r
+ if ( pa_simple_write( pah->s_play, stream_.userBuffer[0], bytes, &pa_error ) < 0) {\r
+ errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<\r
+ pa_strerror( pa_error ) << ".";\r
+ errorText_ = errorStream_.str();\r
+ error( RtError::WARNING );\r
+ }\r
+ break;\r
+ default:\r
+ // ERROR\r
+ break;\r
+ }\r
+\r
+ unlock:\r
+ MUTEX_UNLOCK( &stream_.mutex );\r
+ RtApi::tickStreamTime();\r
+\r
+ if ( doStopStream == 1 )\r
+ stopStream();\r
+}\r
+\r
+void RtApiPulse::startStream( void )\r
+{\r
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );\r
+\r
+ if ( stream_.state == STREAM_CLOSED ) {\r
+ errorText_ = "RtApiPulse::startStream(): the stream is not open!";\r
+ error( RtError::INVALID_USE );\r
+ return;\r
+ }\r
+ if ( stream_.state == STREAM_RUNNING ) {\r
+ errorText_ = "RtApiPulse::startStream(): the stream is already running!";\r
+ error( RtError::WARNING );\r
+ return;\r
+ }\r
+\r
+ MUTEX_LOCK( &stream_.mutex );\r
+\r
+ stream_.state = STREAM_RUNNING;\r
+\r
+ pah->runnable = true;\r
+ pthread_cond_signal( &pah->runnable_cv );\r
+ MUTEX_UNLOCK( &stream_.mutex );\r
+}\r
+\r
+void RtApiPulse::stopStream( void )\r
+{\r
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );\r
+\r
+ if ( stream_.state == STREAM_CLOSED ) {\r
+ errorText_ = "RtApiPulse::stopStream(): the stream is not open!";\r
+ error( RtError::INVALID_USE );\r
+ return;\r
+ }\r
+ if ( stream_.state == STREAM_STOPPED ) {\r
+ errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";\r
+ error( RtError::WARNING );\r
+ return;\r
+ }\r
+\r
+ stream_.state = STREAM_STOPPED;\r
+ MUTEX_LOCK( &stream_.mutex );\r
+\r
+ if ( pah && pah->s_play ) {\r
+ int pa_error;\r
+ if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {\r
+ errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<\r
+ pa_strerror( pa_error ) << ".";\r
+ errorText_ = errorStream_.str();\r
+ MUTEX_UNLOCK( &stream_.mutex );\r
+ error( RtError::SYSTEM_ERROR );\r
+ }\r
+ }\r
+\r
+ stream_.state = STREAM_STOPPED;\r
+ MUTEX_UNLOCK( &stream_.mutex );\r
+}\r
+\r
+void RtApiPulse::abortStream( void )\r
+{\r
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );\r
+\r
+ if ( stream_.state == STREAM_CLOSED ) {\r
+ errorText_ = "RtApiPulse::abortStream(): the stream is not open!";\r
+ error( RtError::INVALID_USE );\r
+ return;\r
+ }\r
+ if ( stream_.state == STREAM_STOPPED ) {\r
+ errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";\r
+ error( RtError::WARNING );\r
+ return;\r
+ }\r
+\r
+ stream_.state = STREAM_STOPPED;\r
+ MUTEX_LOCK( &stream_.mutex );\r
+\r
+ if ( pah && pah->s_play ) {\r
+ int pa_error;\r
+ if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {\r
+ errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<\r
+ pa_strerror( pa_error ) << ".";\r
+ errorText_ = errorStream_.str();\r
+ MUTEX_UNLOCK( &stream_.mutex );\r
+ error( RtError::SYSTEM_ERROR );\r
+ }\r
+ }\r
+\r
+ stream_.state = STREAM_STOPPED;\r
+ MUTEX_UNLOCK( &stream_.mutex );\r
+}\r
+\r
+bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,\r
+ unsigned int channels, unsigned int firstChannel,\r
+ unsigned int sampleRate, RtAudioFormat format,\r
+ unsigned int *bufferSize, RtAudio::StreamOptions *options )\r
+{\r
+ PulseAudioHandle *pah = 0;\r
+ unsigned long bufferBytes = 0;\r
+ pa_sample_spec ss;\r
+\r
+ if ( device != 0 ) return false;\r
+ if ( mode != INPUT && mode != OUTPUT ) return false;\r
+ if ( channels != 1 && channels != 2 ) {\r
+ errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";\r
+ return false;\r
+ }\r
+ ss.channels = channels;\r
+\r
+ if ( firstChannel != 0 ) return false;\r
+\r
+ bool sr_found = false;\r
+ for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {\r
+ if ( sampleRate == *sr ) {\r
+ sr_found = true;\r
+ stream_.sampleRate = sampleRate;\r
+ ss.rate = sampleRate;\r
+ break;\r
+ }\r
+ }\r
+ if ( !sr_found ) {\r
+ errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";\r
+ return false;\r
+ }\r
+\r
+ bool sf_found = 0;\r
+ for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;\r
+ sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {\r
+ if ( format == sf->rtaudio_format ) {\r
+ sf_found = true;\r
+ stream_.userFormat = sf->rtaudio_format;\r
+ ss.format = sf->pa_format;\r
+ break;\r
+ }\r
+ }\r
+ if ( !sf_found ) {\r
+ errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample format.";\r
+ return false;\r
+ }\r
+\r
+ if ( options && ( options->flags & RTAUDIO_NONINTERLEAVED ) ) {\r
+ errorText_ = "RtApiPulse::probeDeviceOpen: only interleaved audio data supported.";\r
+ return false;\r
+ }\r
+\r
+ stream_.userInterleaved = true;\r
+ stream_.nBuffers = 1;\r
+\r
+ stream_.deviceInterleaved[mode] = true;\r
+ stream_.doByteSwap[mode] = false;\r
+ stream_.doConvertBuffer[mode] = false;\r
+ stream_.deviceFormat[mode] = stream_.userFormat;\r
+ stream_.nUserChannels[mode] = channels;\r
+ stream_.nDeviceChannels[mode] = channels;\r
+ stream_.channelOffset[mode] = 0;\r
+\r
+ // Allocate necessary internal buffers.\r
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );\r
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );\r
+ if ( stream_.userBuffer[mode] == NULL ) {\r
+ errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";\r
+ goto error;\r
+ }\r
+ stream_.bufferSize = *bufferSize;\r
+\r
+ if ( !stream_.apiHandle ) {\r
+ PulseAudioHandle *pah = new PulseAudioHandle;\r
+ if ( !pah ) {\r
+ errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";\r
+ goto error;\r
+ }\r
+\r
+ stream_.apiHandle = pah;\r
+ if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {\r
+ errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";\r
+ goto error;\r
+ }\r
+ }\r
+ pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );\r
+\r
+ int error;\r
+ switch ( mode ) {\r
+ case INPUT:\r
+ pah->s_rec = pa_simple_new( NULL, "RtAudio", PA_STREAM_RECORD, NULL, "Record", &ss, NULL, NULL, &error );\r
+ if ( !pah->s_rec ) {\r
+ errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";\r
+ goto error;\r
+ }\r
+ break;\r
+ case OUTPUT:\r
+ pah->s_play = pa_simple_new( NULL, "RtAudio", PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );\r
+ if ( !pah->s_play ) {\r
+ errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";\r
+ goto error;\r
+ }\r
+ break;\r
+ default:\r
+ goto error;\r
+ }\r
+\r
+ if ( stream_.mode == UNINITIALIZED )\r
+ stream_.mode = mode;\r
+ else if ( stream_.mode == mode )\r
+ goto error;\r
+ else\r
+ stream_.mode = DUPLEX;\r
+\r
+ stream_.state = STREAM_STOPPED;\r
+\r
+ if ( !stream_.callbackInfo.isRunning ) {\r
+ stream_.callbackInfo.object = this;\r
+ stream_.callbackInfo.isRunning = true;\r
+ if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) {\r
+ errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";\r
+ goto error;\r
+ }\r
+ }\r
+ return true;\r
+ \r
+ error:\r
+ closeStream();\r
+ return false;\r
+}\r
+\r
+//******************** End of __LINUX_PULSE__ *********************//\r
+#endif\r
\r
#if defined(__LINUX_OSS__)\r
\r
-#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 <sys/types.h>
+#include <machine/endian.h>
+#ifndef _IOWR
+#include <sys/ioccom.h>
+#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 <sys/ioctl.h>
-#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 <linux/ioctl.h>
-#else
-# ifdef __FreeBSD__
-# include <sys/ioccom.h>
-# else
-# include <sys/ioctl.h>
-# 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
#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
* 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.
* 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 */
#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
* 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,
#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.
* 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
* }
*/
-#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
* 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;ii<ll;ii++)\
- _seqbuf[_seqbufptr+ii+2] = bufp[ii];\
- for(ii=ll;ii<6;ii++)\
- _seqbuf[_seqbufptr+ii+2] = 0xff;\
- _SEQ_ADVBUF(8);}
+#define SEQ_SYSEX(dev, buf, len) { \
+ int i, l=(len); if (l>6)l=6;\
+ _SEQ_NEEDBUF(8);\
+ _seqbuf[_seqbufptr] = EV_SYSEX;\
+ for(i=0;i<l;i++)_seqbuf[_seqbufptr+i+1] = (buf)[i];\
+ for(i=l;i<6;i++)_seqbuf[_seqbufptr+i+1] = 0xff;\
+ _SEQ_ADVBUF(8);}
#define SEQ_CHN_PRESSURE(dev, chn, pressure) \
- _CHN_COMMON(dev, MIDI_CHN_PRESSURE, chn, pressure, 0, 0)
+ _CHN_COMMON(dev, MIDI_CHN_PRESSURE, chn, pressure, 0, 0)
-#define SEQ_SET_PATCH SEQ_PGM_CHANGE
-#ifdef OSSLIB
-# define SEQ_PGM_CHANGE(dev, chn, patch) \
- {OSS_patch_caching(dev, chn, patch, seqfd, _seqbuf, _seqbuflen); \
- _CHN_COMMON(dev, MIDI_PGM_CHANGE, chn, patch, 0, 0);}
-#else
-# define SEQ_PGM_CHANGE(dev, chn, patch) \
- _CHN_COMMON(dev, MIDI_PGM_CHANGE, chn, patch, 0, 0)
-#endif
+#define SEQ_SET_PATCH(dev, chn, patch) \
+ _CHN_COMMON(dev, MIDI_PGM_CHANGE, chn, patch, 0, 0)
#define SEQ_CONTROL(dev, chn, controller, value) \
- _CHN_COMMON(dev, MIDI_CTL_CHANGE, chn, controller, 0, value)
+ _CHN_COMMON(dev, MIDI_CTL_CHANGE, chn, controller, 0, value)
#define SEQ_BENDER(dev, chn, value) \
- _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);\
- _seqbuf[_seqbufptr+5] = ((value)&0xff);\
- _seqbuf[_seqbufptr+6] = ((value>>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) \
* 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)
* 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)
/*
****************************************************************************
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)
/*
**************************************************************************
* 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)
/*
**************************************************************************
* 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? */
/*
****************************************************************************
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
# 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) */
# 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;
# 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) */
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
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;
/*
* 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
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
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_ */
projects / rtaudio-cdist.git / commitdiff