RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/\r
\r
RtAudio: realtime audio i/o C++ classes\r
- Copyright (c) 2001-2011 Gary P. Scavone\r
+ Copyright (c) 2001-2012 Gary P. Scavone\r
\r
Permission is hereby granted, free of charge, to any person\r
obtaining a copy of this software and associated documentation files\r
*/\r
/************************************************************************/\r
\r
-// RtAudio: Version 4.0.9\r
+// RtAudio: Version 4.0.11\r
\r
#include "RtAudio.h"\r
#include <iostream>\r
#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
+ delete rtapi_;\r
+ rtapi_ = 0;\r
+\r
#if defined(__UNIX_JACK__)\r
if ( api == UNIX_JACK )\r
rtapi_ = new RtApiJack();\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
:deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }\r
};\r
\r
+ThreadHandle threadId;\r
+\r
RtApiCore:: RtApiCore()\r
{\r
#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )\r
\r
// We'll try higher bit rates first and then work our way down.\r
std::vector< std::pair<UInt32, UInt32> > physicalFormats;\r
- formatFlags = description.mFormatFlags | kLinearPCMFormatFlagIsFloat & ~kLinearPCMFormatFlagIsSignedInteger;\r
+ formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;\r
physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );\r
formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;\r
physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );\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
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- return;\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
stopStream();\r
}\r
\r
+// This function will be called by a spawned thread when the user\r
+// callback function signals that the stream should be stopped or\r
+// aborted. It is better to handle it this way because the\r
+// callbackEvent() function probably should return before the AudioDeviceStop()\r
+// function is called.\r
+extern "C" void *coreStopStream( void *ptr )\r
+{\r
+ CallbackInfo *info = (CallbackInfo *) ptr;\r
+ RtApiCore *object = (RtApiCore *) info->object;\r
+\r
+ object->stopStream();\r
+ pthread_exit( NULL );\r
+}\r
+\r
bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,\r
const AudioBufferList *inBufferList,\r
const AudioBufferList *outBufferList )\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
- stopStream();\r
+ pthread_create( &threadId, NULL, coreStopStream, info );\r
else // external call to stopStream()\r
pthread_cond_signal( &handle->condition );\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
AudioDeviceID outputDevice = handle->id[0];\r
\r
// Invoke user callback to get fresh output data UNLESS we are\r
handle->xrun[1] = false;\r
}\r
\r
- handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
- stream_.bufferSize, streamTime, status, info->userData );\r
- if ( handle->drainCounter == 2 ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
+ stream_.bufferSize, streamTime, status, info->userData );\r
+ if ( cbReturnValue == 2 ) {\r
+ stream_.state = STREAM_STOPPING;\r
+ handle->drainCounter = 2;\r
abortStream();\r
return SUCCESS;\r
}\r
- else if ( handle->drainCounter == 1 )\r
+ else if ( cbReturnValue == 1 ) {\r
+ handle->drainCounter = 1;\r
handle->internalDrain = true;\r
+ }\r
}\r
\r
if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {\r
}\r
\r
unlock:\r
- MUTEX_UNLOCK( &stream_.mutex );\r
+ //MUTEX_UNLOCK( &stream_.mutex );\r
\r
RtApi::tickStreamTime();\r
return SUCCESS;\r
}\r
\r
if ( device >= nDevices ) {\r
+ jack_client_close( client );\r
errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";\r
error( RtError::INVALID_USE );\r
}\r
\r
// Get the latency of the JACK port.\r
ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );\r
- if ( ports[ firstChannel ] )\r
- stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );\r
+ if ( ports[ firstChannel ] ) {\r
+ // Added by Ge Wang\r
+ jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);\r
+ // the range (usually the min and max are equal)\r
+ jack_latency_range_t latrange; latrange.min = latrange.max = 0;\r
+ // get the latency range\r
+ jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );\r
+ // be optimistic, use the min!\r
+ stream_.latency[mode] = latrange.min;\r
+ //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );\r
+ }\r
free( ports );\r
\r
// The jack server always uses 32-bit floating-point data.\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
status |= RTAUDIO_INPUT_OVERFLOW;\r
handle->xrun[1] = false;\r
}\r
- handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
- stream_.bufferSize, streamTime, status, info->userData );\r
- if ( handle->drainCounter == 2 ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
+ stream_.bufferSize, streamTime, status, info->userData );\r
+ if ( cbReturnValue == 2 ) {\r
+ stream_.state = STREAM_STOPPING;\r
+ handle->drainCounter = 2;\r
ThreadHandle id;\r
pthread_create( &id, NULL, jackStopStream, info );\r
return SUCCESS;\r
}\r
- else if ( handle->drainCounter == 1 )\r
+ else if ( cbReturnValue == 1 ) {\r
+ handle->drainCounter = 1;\r
handle->internalDrain = true;\r
+ }\r
}\r
\r
jack_default_audio_sample_t *jackbuffer;\r
}\r
\r
unlock:\r
- MUTEX_UNLOCK(&stream_.mutex);\r
-\r
RtApi::tickStreamTime();\r
return SUCCESS;\r
}\r
info.nativeFormats |= RTAUDIO_FLOAT32;\r
else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )\r
info.nativeFormats |= RTAUDIO_FLOAT64;\r
+ else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )\r
+ info.nativeFormats |= RTAUDIO_SINT24;\r
\r
if ( info.outputChannels > 0 )\r
if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;\r
stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;\r
if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;\r
}\r
+ else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {\r
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;\r
+ if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;\r
+ }\r
\r
if ( stream_.deviceFormat[mode] == 0 ) {\r
drivers.removeCurrentDriver();\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
status |= RTAUDIO_INPUT_OVERFLOW;\r
asioXRun = false;\r
}\r
- handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
stream_.bufferSize, streamTime, status, info->userData );\r
- if ( handle->drainCounter == 2 ) {\r
- // MUTEX_UNLOCK( &stream_.mutex );\r
- // abortStream();\r
+ if ( cbReturnValue == 2 ) {\r
+ stream_.state = STREAM_STOPPING;\r
+ handle->drainCounter = 2;\r
unsigned threadId;\r
- stopThreadCalled = true;\r
stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,\r
&stream_.callbackInfo, 0, &threadId );\r
return SUCCESS;\r
}\r
- else if ( handle->drainCounter == 1 )\r
+ else if ( cbReturnValue == 1 ) {\r
+ handle->drainCounter = 1;\r
handle->internalDrain = true;\r
+ }\r
}\r
\r
unsigned int nChannels, bufferBytes, i, j;\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
std::vector< int > indices;\r
for ( unsigned int i=0; i<dsDevices.size(); i++ )\r
if ( dsDevices[i].found == false ) indices.push_back( i );\r
- for ( unsigned int nErased=0, unsigned int i=0; i<indices.size(); i++, nErased++ ) {\r
- dsDevices.erase( dsDevices.begin()-nErased );\r
- }\r
+ unsigned int nErased = 0;\r
+ for ( unsigned int i=0; i<indices.size(); i++ )\r
+ dsDevices.erase( dsDevices.begin()-nErased++ );\r
\r
return dsDevices.size();\r
}\r
// Determine the device buffer size. By default, we'll use the value\r
// defined above (32K), but we will grow it to make allowances for\r
// very large software buffer sizes.\r
- DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;;\r
+ DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;\r
DWORD dsPointerLeadTime = 0;\r
\r
void *ohandle = 0, *bhandle = 0;\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
status |= RTAUDIO_INPUT_OVERFLOW;\r
handle->xrun[1] = false;\r
}\r
- handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
- stream_.bufferSize, streamTime, status, info->userData );\r
- if ( handle->drainCounter == 2 ) {\r
- // MUTEX_UNLOCK( &stream_.mutex );\r
+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
+ stream_.bufferSize, streamTime, status, info->userData );\r
+ if ( cbReturnValue == 2 ) {\r
+ stream_.state = STREAM_STOPPING;\r
+ handle->drainCounter = 2;\r
abortStream();\r
return;\r
}\r
- else if ( handle->drainCounter == 1 )\r
+ else if ( cbReturnValue == 1 ) {\r
+ handle->drainCounter = 1;\r
handle->internalDrain = true;\r
+ }\r
}\r
\r
HRESULT result;\r
}\r
}\r
else { // mode == INPUT\r
- while ( safeReadPointer < endRead ) {\r
+ while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {\r
// See comments for playback.\r
double millis = (endRead - safeReadPointer) * 1000.0;\r
millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);\r
}\r
\r
unlock:\r
- // MUTEX_UNLOCK( &stream_.mutex );\r
-\r
RtApi::tickStreamTime();\r
}\r
\r
snd_card_next( &card );\r
}\r
\r
+ result = snd_ctl_open( &handle, "default", 0 );\r
+ if (result == 0) {\r
+ nDevices++;\r
+ snd_ctl_close( handle );\r
+ }\r
+\r
return nDevices;\r
}\r
\r
snd_card_next( &card );\r
}\r
\r
+ result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );\r
+ if ( result == 0 ) {\r
+ if ( nDevices == device ) {\r
+ strcpy( name, "default" );\r
+ goto foundDevice;\r
+ }\r
+ nDevices++;\r
+ }\r
+\r
if ( nDevices == 0 ) {\r
errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";\r
error( RtError::INVALID_USE );\r
// Thus, use the saved results.\r
if ( stream_.state != STREAM_CLOSED &&\r
( stream_.device[0] == device || stream_.device[1] == device ) ) {\r
+ snd_ctl_close( chandle );\r
if ( device >= devices_.size() ) {\r
errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";\r
error( RtError::WARNING );\r
snd_pcm_hw_params_t *params;\r
snd_pcm_hw_params_alloca( ¶ms );\r
\r
- // First try for playback\r
+ // First try for playback unless default device (which has subdev -1)\r
stream = SND_PCM_STREAM_PLAYBACK;\r
- snd_pcm_info_set_device( pcminfo, subdevice );\r
- snd_pcm_info_set_subdevice( pcminfo, 0 );\r
snd_pcm_info_set_stream( pcminfo, stream );\r
+ if ( subdevice != -1 ) {\r
+ snd_pcm_info_set_device( pcminfo, subdevice );\r
+ snd_pcm_info_set_subdevice( pcminfo, 0 );\r
\r
- result = snd_ctl_pcm_info( chandle, pcminfo );\r
- if ( result < 0 ) {\r
- // Device probably doesn't support playback.\r
- goto captureProbe;\r
+ result = snd_ctl_pcm_info( chandle, pcminfo );\r
+ if ( result < 0 ) {\r
+ // Device probably doesn't support playback.\r
+ goto captureProbe;\r
+ }\r
}\r
\r
result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );\r
snd_pcm_close( phandle );\r
\r
captureProbe:\r
- // Now try for capture\r
stream = SND_PCM_STREAM_CAPTURE;\r
snd_pcm_info_set_stream( pcminfo, stream );\r
\r
- result = snd_ctl_pcm_info( chandle, pcminfo );\r
- snd_ctl_close( chandle );\r
- if ( result < 0 ) {\r
- // Device probably doesn't support capture.\r
- if ( info.outputChannels == 0 ) return info;\r
- goto probeParameters;\r
+ // Now try for capture unless default device (with subdev = -1)\r
+ if ( subdevice != -1 ) {\r
+ result = snd_ctl_pcm_info( chandle, pcminfo );\r
+ snd_ctl_close( chandle );\r
+ if ( result < 0 ) {\r
+ // Device probably doesn't support capture.\r
+ if ( info.outputChannels == 0 ) return info;\r
+ goto probeParameters;\r
+ }\r
}\r
\r
result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);\r
snd_card_next( &card );\r
}\r
\r
+ result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );\r
+ if ( result == 0 ) {\r
+ if ( nDevices == device ) {\r
+ strcpy( name, "default" );\r
+ goto foundDevice;\r
+ }\r
+ nDevices++;\r
+ }\r
+\r
if ( nDevices == 0 ) {\r
// This should not happen because a check is made before this function is called.\r
errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";\r
}\r
\r
// If we get here, no supported format was found.\r
+ snd_pcm_close( phandle );\r
errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";\r
errorText_ = errorStream_.str();\r
return FAILURE;\r
// If attempting to setup a duplex stream, the bufferSize parameter\r
// MUST be the same in both directions!\r
if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {\r
+ snd_pcm_close( phandle );\r
errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";\r
errorText_ = errorStream_.str();\r
return FAILURE;\r
apiInfo = (AlsaHandle *) stream_.apiHandle;\r
}\r
apiInfo->handles[mode] = phandle;\r
+ phandle = 0;\r
\r
// Allocate necessary internal buffers.\r
unsigned long bufferBytes;\r
pthread_attr_t attr;\r
pthread_attr_init( &attr );\r
pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );\r
+\r
#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)\r
if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {\r
- struct sched_param param;\r
+ // We previously attempted to increase the audio callback priority\r
+ // to SCHED_RR here via the attributes. However, while no errors\r
+ // were reported in doing so, it did not work. So, now this is\r
+ // done in the alsaCallbackHandler function.\r
+ stream_.callbackInfo.doRealtime = true;\r
int priority = options->priority;\r
int min = sched_get_priority_min( SCHED_RR );\r
int max = sched_get_priority_max( SCHED_RR );\r
if ( priority < min ) priority = min;\r
else if ( priority > max ) priority = max;\r
- param.sched_priority = priority;\r
- pthread_attr_setschedparam( &attr, ¶m );\r
- pthread_attr_setschedpolicy( &attr, SCHED_RR );\r
+ stream_.callbackInfo.priority = priority;\r
}\r
- else\r
- pthread_attr_setschedpolicy( &attr, SCHED_OTHER );\r
-#else\r
- pthread_attr_setschedpolicy( &attr, SCHED_OTHER );\r
#endif\r
\r
stream_.callbackInfo.isRunning = true;\r
stream_.apiHandle = 0;\r
}\r
\r
+ if ( phandle) snd_pcm_close( phandle );\r
+\r
for ( int i=0; i<2; i++ ) {\r
if ( stream_.userBuffer[i] ) {\r
free( stream_.userBuffer[i] );\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
RtApiAlsa *object = (RtApiAlsa *) info->object;\r
bool *isRunning = &info->isRunning;\r
\r
+#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)\r
+ if ( &info->doRealtime ) {\r
+ pthread_t tID = pthread_self(); // ID of this thread\r
+ sched_param prio = { info->priority }; // scheduling priority of thread\r
+ pthread_setschedparam( tID, SCHED_RR, &prio );\r
+ }\r
+#endif\r
+\r
while ( *isRunning == true ) {\r
pthread_testcancel();\r
object->callbackEvent();\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[OUTPUT], stream_.userBuffer[INPUT],\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
+ void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];\r
+ void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];\r
+\r
+ if ( stream_.state != STREAM_RUNNING )\r
+ goto unlock;\r
+\r
+ int pa_error;\r
+ size_t bytes;\r
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
+ if ( stream_.doConvertBuffer[OUTPUT] ) {\r
+ convertBuffer( stream_.deviceBuffer,\r
+ stream_.userBuffer[OUTPUT],\r
+ stream_.convertInfo[OUTPUT] );\r
+ bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *\r
+ formatBytes( stream_.deviceFormat[OUTPUT] );\r
+ } else\r
+ bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *\r
+ formatBytes( stream_.userFormat );\r
+\r
+ if ( pa_simple_write( pah->s_play, pulse_out, 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
+ }\r
+\r
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {\r
+ if ( stream_.doConvertBuffer[INPUT] )\r
+ bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *\r
+ formatBytes( stream_.deviceFormat[INPUT] );\r
+ else\r
+ bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *\r
+ formatBytes( stream_.userFormat );\r
+ \r
+ if ( pa_simple_read( pah->s_rec, pulse_in, 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
+ if ( stream_.doConvertBuffer[INPUT] ) {\r
+ convertBuffer( stream_.userBuffer[INPUT],\r
+ stream_.deviceBuffer,\r
+ stream_.convertInfo[INPUT] );\r
+ }\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
+ // Set interleaving parameters.\r
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;\r
+ else stream_.userInterleaved = true;\r
+ stream_.deviceInterleaved[mode] = true;\r
+ stream_.nBuffers = 1;\r
+ stream_.doByteSwap[mode] = false;\r
+ stream_.doConvertBuffer[mode] = channels > 1 && !stream_.userInterleaved;\r
+ stream_.deviceFormat[mode] = stream_.userFormat;\r
+ stream_.nUserChannels[mode] = channels;\r
+ stream_.nDeviceChannels[mode] = channels + firstChannel;\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_.doConvertBuffer[mode] ) {\r
+\r
+ bool makeBuffer = true;\r
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );\r
+ if ( mode == INPUT ) {\r
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {\r
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );\r
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;\r
+ }\r
+ }\r
+\r
+ if ( makeBuffer ) {\r
+ bufferBytes *= *bufferSize;\r
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );\r
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );\r
+ if ( stream_.deviceBuffer == NULL ) {\r
+ errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";\r
+ goto error;\r
+ }\r
+ }\r
+ }\r
+ \r
+ stream_.device[mode] = device;\r
+ stream_.state = STREAM_STOPPED;\r
+\r
+ // Setup the buffer conversion information structure.\r
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );\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
{\r
if ( format == RTAUDIO_SINT16 )\r
return 2;\r
- else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 ||\r
- format == RTAUDIO_FLOAT32 )\r
+ else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )\r
return 4;\r
else if ( format == RTAUDIO_FLOAT64 )\r
return 8;\r
+ else if ( format == RTAUDIO_SINT24 )\r
+ return 3;\r
else if ( format == RTAUDIO_SINT8 )\r
return 1;\r
\r
}\r
}\r
else if (info.inFormat == RTAUDIO_SINT24) {\r
- Int32 *in = (Int32 *)inBuffer;\r
+ Int24 *in = (Int24 *)inBuffer;\r
scale = 1.0 / 8388607.5;\r
for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]] & 0x00ffffff);\r
+ out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());\r
out[info.outOffset[j]] += 0.5;\r
out[info.outOffset[j]] *= scale;\r
}\r
}\r
}\r
else if (info.inFormat == RTAUDIO_SINT24) {\r
- Int32 *in = (Int32 *)inBuffer;\r
+ Int24 *in = (Int24 *)inBuffer;\r
scale = (Float32) ( 1.0 / 8388607.5 );\r
for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]] & 0x00ffffff);\r
+ out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());\r
out[info.outOffset[j]] += 0.5;\r
out[info.outOffset[j]] *= scale;\r
}\r
out += info.outJump;\r
}\r
}\r
- else if (info.inFormat == RTAUDIO_SINT24) { // Hmmm ... we could just leave it in the lower 3 bytes\r
- Int32 *in = (Int32 *)inBuffer;\r
+ else if (info.inFormat == RTAUDIO_SINT24) {\r
+ Int24 *in = (Int24 *)inBuffer;\r
for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];\r
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();\r
out[info.outOffset[j]] <<= 8;\r
}\r
in += info.inJump;\r
}\r
}\r
else if (info.outFormat == RTAUDIO_SINT24) {\r
- Int32 *out = (Int32 *)outBuffer;\r
+ Int24 *out = (Int24 *)outBuffer;\r
if (info.inFormat == RTAUDIO_SINT8) {\r
signed char *in = (signed char *)inBuffer;\r
for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];\r
- out[info.outOffset[j]] <<= 16;\r
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);\r
+ //out[info.outOffset[j]] <<= 16;\r
}\r
in += info.inJump;\r
out += info.outJump;\r
Int16 *in = (Int16 *)inBuffer;\r
for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];\r
- out[info.outOffset[j]] <<= 8;\r
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);\r
+ //out[info.outOffset[j]] <<= 8;\r
}\r
in += info.inJump;\r
out += info.outJump;\r
}\r
else if (info.inFormat == RTAUDIO_SINT24) {\r
// Channel compensation and/or (de)interleaving only.\r
- Int32 *in = (Int32 *)inBuffer;\r
+ Int24 *in = (Int24 *)inBuffer;\r
for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
for (j=0; j<info.channels; j++) {\r
out[info.outOffset[j]] = in[info.inOffset[j]];\r
Int32 *in = (Int32 *)inBuffer;\r
for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];\r
- out[info.outOffset[j]] >>= 8;\r
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);\r
+ //out[info.outOffset[j]] >>= 8;\r
}\r
in += info.inJump;\r
out += info.outJump;\r
}\r
}\r
else if (info.inFormat == RTAUDIO_SINT24) {\r
- Int32 *in = (Int32 *)inBuffer;\r
+ Int24 *in = (Int24 *)inBuffer;\r
for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 8) & 0x0000ffff);\r
+ out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);\r
}\r
in += info.inJump;\r
out += info.outJump;\r
}\r
}\r
else if (info.inFormat == RTAUDIO_SINT24) {\r
- Int32 *in = (Int32 *)inBuffer;\r
+ Int24 *in = (Int24 *)inBuffer;\r
for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 16) & 0x000000ff);\r
+ out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);\r
}\r
in += info.inJump;\r
out += info.outJump;\r
}\r
}\r
\r
- //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }\r
- //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }\r
- //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }\r
+//static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }\r
+//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }\r
+//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }\r
\r
void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )\r
{\r
ptr += 2;\r
}\r
}\r
- else if ( format == RTAUDIO_SINT24 ||\r
- format == RTAUDIO_SINT32 ||\r
+ else if ( format == RTAUDIO_SINT32 ||\r
format == RTAUDIO_FLOAT32 ) {\r
for ( unsigned int i=0; i<samples; i++ ) {\r
// Swap 1st and 4th bytes.\r
ptr += 3;\r
}\r
}\r
+ else if ( format == RTAUDIO_SINT24 ) {\r
+ for ( unsigned int i=0; i<samples; i++ ) {\r
+ // Swap 1st and 3rd bytes.\r
+ val = *(ptr);\r
+ *(ptr) = *(ptr+2);\r
+ *(ptr+2) = val;\r
+\r
+ // Increment 2 more bytes.\r
+ ptr += 2;\r
+ }\r
+ }\r
else if ( format == RTAUDIO_FLOAT64 ) {\r
for ( unsigned int i=0; i<samples; i++ ) {\r
// Swap 1st and 8th bytes\r
projects / rtaudio.git / blobdiff