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) 2001-2013 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 4.0.12\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
//\r
// *************************************************** //\r
\r
+std::string RtAudio :: getVersion( void ) throw()\r
+{\r
+ return std::string( RTAUDIO_VERSION );\r
+}\r
+\r
void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()\r
{\r
apis.clear();\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
// definition __RTAUDIO_DUMMY__ is automatically defined if no\r
// API-specific definitions are passed to the compiler. But just in\r
// case something weird happens, we'll print out an error message.\r
- std::cerr << "\nRtAudio: no compiled API support found ... critical error!!\n\n";\r
+ std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";\r
+ throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );\r
}\r
\r
RtAudio :: ~RtAudio() throw()\r
RtAudioFormat format, unsigned int sampleRate,\r
unsigned int *bufferFrames,\r
RtAudioCallback callback, void *userData,\r
- RtAudio::StreamOptions *options )\r
+ RtAudio::StreamOptions *options,\r
+ RtAudioErrorCallback errorCallback )\r
{\r
return rtapi_->openStream( outputParameters, inputParameters, format,\r
sampleRate, bufferFrames, callback,\r
- userData, options );\r
+ userData, options, errorCallback );\r
}\r
\r
// *************************************************** //\r
stream_.userBuffer[1] = 0;\r
MUTEX_INITIALIZE( &stream_.mutex );\r
showWarnings_ = true;\r
+ firstErrorOccurred_ = false;\r
}\r
\r
RtApi :: ~RtApi()\r
RtAudioFormat format, unsigned int sampleRate,\r
unsigned int *bufferFrames,\r
RtAudioCallback callback, void *userData,\r
- RtAudio::StreamOptions *options )\r
+ RtAudio::StreamOptions *options,\r
+ RtAudioErrorCallback errorCallback )\r
{\r
if ( stream_.state != STREAM_CLOSED ) {\r
errorText_ = "RtApi::openStream: a stream is already open!";\r
- error( RtError::INVALID_USE );\r
+ error( RtAudioError::INVALID_USE );\r
+ return;\r
}\r
\r
if ( oParams && oParams->nChannels < 1 ) {\r
errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";\r
- error( RtError::INVALID_USE );\r
+ error( RtAudioError::INVALID_USE );\r
+ return;\r
}\r
\r
if ( iParams && iParams->nChannels < 1 ) {\r
errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";\r
- error( RtError::INVALID_USE );\r
+ error( RtAudioError::INVALID_USE );\r
+ return;\r
}\r
\r
if ( oParams == NULL && iParams == NULL ) {\r
errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";\r
- error( RtError::INVALID_USE );\r
+ error( RtAudioError::INVALID_USE );\r
+ return;\r
}\r
\r
if ( formatBytes(format) == 0 ) {\r
errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";\r
- error( RtError::INVALID_USE );\r
+ error( RtAudioError::INVALID_USE );\r
+ return;\r
}\r
\r
unsigned int nDevices = getDeviceCount();\r
oChannels = oParams->nChannels;\r
if ( oParams->deviceId >= nDevices ) {\r
errorText_ = "RtApi::openStream: output device parameter value is invalid.";\r
- error( RtError::INVALID_USE );\r
+ error( RtAudioError::INVALID_USE );\r
+ return;\r
}\r
}\r
\r
iChannels = iParams->nChannels;\r
if ( iParams->deviceId >= nDevices ) {\r
errorText_ = "RtApi::openStream: input device parameter value is invalid.";\r
- error( RtError::INVALID_USE );\r
+ error( RtAudioError::INVALID_USE );\r
+ return;\r
}\r
}\r
\r
\r
result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,\r
sampleRate, format, bufferFrames, options );\r
- if ( result == false ) error( RtError::SYSTEM_ERROR );\r
+ if ( result == false ) {\r
+ error( RtAudioError::SYSTEM_ERROR );\r
+ return;\r
+ }\r
}\r
\r
if ( iChannels > 0 ) {\r
sampleRate, format, bufferFrames, options );\r
if ( result == false ) {\r
if ( oChannels > 0 ) closeStream();\r
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
+ return;\r
}\r
}\r
\r
stream_.callbackInfo.callback = (void *) callback;\r
stream_.callbackInfo.userData = userData;\r
+ stream_.callbackInfo.errorCallback = (void *) errorCallback;\r
\r
if ( options ) options->numberOfBuffers = stream_.nBuffers;\r
stream_.state = STREAM_STOPPED;\r
return;\r
}\r
\r
-bool RtApi :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,\r
- unsigned int firstChannel, unsigned int sampleRate,\r
- RtAudioFormat format, unsigned int *bufferSize,\r
- RtAudio::StreamOptions *options )\r
+bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,\r
+ unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,\r
+ RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,\r
+ RtAudio::StreamOptions * /*options*/ )\r
{\r
// MUST be implemented in subclasses!\r
return FAILURE;\r
OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);\r
if ( result != noErr ) {\r
errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
}\r
#endif\r
}\r
OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );\r
if ( result != noErr ) {\r
errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return 0;\r
}\r
\r
OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );\r
if ( result != noErr ) {\r
errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return 0;\r
}\r
\r
result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );\r
if ( result != noErr ) {\r
errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return 0;\r
}\r
\r
if ( id == deviceList[i] ) return i;\r
\r
errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return 0;\r
}\r
\r
OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );\r
if ( result != noErr ) {\r
errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return 0;\r
}\r
\r
result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );\r
if ( result != noErr ) {\r
errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return 0;\r
}\r
\r
if ( id == deviceList[i] ) return i;\r
\r
errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return 0;\r
}\r
\r
unsigned int nDevices = getDeviceCount();\r
if ( nDevices == 0 ) {\r
errorText_ = "RtApiCore::getDeviceInfo: no devices found!";\r
- error( RtError::INVALID_USE );\r
+ error( RtAudioError::INVALID_USE );\r
+ return info;\r
}\r
\r
if ( device >= nDevices ) {\r
errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";\r
- error( RtError::INVALID_USE );\r
+ error( RtAudioError::INVALID_USE );\r
+ return info;\r
}\r
\r
AudioDeviceID deviceList[ nDevices ];\r
0, NULL, &dataSize, (void *) &deviceList );\r
if ( result != noErr ) {\r
errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
if ( result != noErr ) {\r
errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
if ( result != noErr ) {\r
errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
if ( result != noErr || dataSize == 0 ) {\r
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
bufferList = (AudioBufferList *) malloc( dataSize );\r
if ( bufferList == NULL ) {\r
errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
free( bufferList );\r
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
if ( result != noErr || dataSize == 0 ) {\r
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
bufferList = (AudioBufferList *) malloc( dataSize );\r
if ( bufferList == NULL ) {\r
errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
free( bufferList );\r
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
if ( result != kAudioHardwareNoError || dataSize == 0 ) {\r
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
if ( result != kAudioHardwareNoError ) {\r
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
- Float64 minimumRate = 100000000.0, maximumRate = 0.0;\r
+ // The sample rate reporting mechanism is a bit of a mystery. It\r
+ // seems that it can either return individual rates or a range of\r
+ // rates. I assume that if the min / max range values are the same,\r
+ // then that represents a single supported rate and if the min / max\r
+ // range values are different, the device supports an arbitrary\r
+ // range of values (though there might be multiple ranges, so we'll\r
+ // use the most conservative range).\r
+ Float64 minimumRate = 1.0, maximumRate = 10000000000.0;\r
+ bool haveValueRange = false;\r
+ info.sampleRates.clear();\r
for ( UInt32 i=0; i<nRanges; i++ ) {\r
- if ( rangeList[i].mMinimum < minimumRate ) minimumRate = rangeList[i].mMinimum;\r
- if ( rangeList[i].mMaximum > maximumRate ) maximumRate = rangeList[i].mMaximum;\r
+ if ( rangeList[i].mMinimum == rangeList[i].mMaximum )\r
+ info.sampleRates.push_back( (unsigned int) rangeList[i].mMinimum );\r
+ else {\r
+ haveValueRange = true;\r
+ if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;\r
+ if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;\r
+ }\r
}\r
\r
- info.sampleRates.clear();\r
- for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {\r
- if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate )\r
- info.sampleRates.push_back( SAMPLE_RATES[k] );\r
+ if ( haveValueRange ) {\r
+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {\r
+ if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate )\r
+ info.sampleRates.push_back( SAMPLE_RATES[k] );\r
+ }\r
}\r
\r
+ // Sort and remove any redundant values\r
+ std::sort( info.sampleRates.begin(), info.sampleRates.end() );\r
+ info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );\r
+\r
if ( info.sampleRates.size() == 0 ) {\r
errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
return info;\r
}\r
\r
-OSStatus callbackHandler( AudioDeviceID inDevice,\r
- const AudioTimeStamp* inNow,\r
- const AudioBufferList* inInputData,\r
- const AudioTimeStamp* inInputTime,\r
- AudioBufferList* outOutputData,\r
- const AudioTimeStamp* inOutputTime, \r
- void* infoPointer )\r
+static OSStatus callbackHandler( AudioDeviceID inDevice,\r
+ const AudioTimeStamp* /*inNow*/,\r
+ const AudioBufferList* inInputData,\r
+ const AudioTimeStamp* /*inInputTime*/,\r
+ AudioBufferList* outOutputData,\r
+ const AudioTimeStamp* /*inOutputTime*/,\r
+ void* infoPointer )\r
{\r
CallbackInfo *info = (CallbackInfo *) infoPointer;\r
\r
return kAudioHardwareNoError;\r
}\r
\r
-OSStatus xrunListener( AudioObjectID inDevice,\r
- UInt32 nAddresses,\r
- const AudioObjectPropertyAddress properties[],\r
- void* handlePointer )\r
+static OSStatus xrunListener( AudioObjectID /*inDevice*/,\r
+ UInt32 nAddresses,\r
+ const AudioObjectPropertyAddress properties[],\r
+ void* handlePointer )\r
{\r
CoreHandle *handle = (CoreHandle *) handlePointer;\r
for ( UInt32 i=0; i<nAddresses; i++ ) {\r
return kAudioHardwareNoError;\r
}\r
\r
-OSStatus rateListener( AudioObjectID inDevice,\r
- UInt32 nAddresses,\r
- const AudioObjectPropertyAddress properties[],\r
- void* ratePointer )\r
+static OSStatus rateListener( AudioObjectID inDevice,\r
+ UInt32 /*nAddresses*/,\r
+ const AudioObjectPropertyAddress /*properties*/[],\r
+ void* ratePointer )\r
{\r
-\r
Float64 *rate = (Float64 *) ratePointer;\r
UInt32 dataSize = sizeof( Float64 );\r
AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,\r
\r
result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );\r
if (result != noErr || dataSize == 0) {\r
+ free( bufferList );\r
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";\r
errorText_ = errorStream_.str();\r
return FAILURE;\r
dataSize = sizeof( Float64 );\r
property.mSelector = kAudioDevicePropertyNominalSampleRate;\r
result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );\r
-\r
if ( result != noErr ) {\r
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";\r
errorText_ = errorStream_.str();\r
\r
nominalRate = (Float64) sampleRate;\r
result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );\r
-\r
if ( result != noErr ) {\r
+ AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );\r
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";\r
errorText_ = errorStream_.str();\r
return FAILURE;\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
else {\r
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
}\r
}\r
\r
stream_.deviceBuffer = 0;\r
}\r
\r
+ stream_.state = STREAM_CLOSED;\r
return FAILURE;\r
}\r
\r
{\r
if ( stream_.state == STREAM_CLOSED ) {\r
errorText_ = "RtApiCore::closeStream(): no open stream to close!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\r
\r
verifyStream();\r
if ( stream_.state == STREAM_RUNNING ) {\r
errorText_ = "RtApiCore::startStream(): the stream is already running!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\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
+ error( RtAudioError::SYSTEM_ERROR );\r
}\r
\r
void RtApiCore :: stopStream( void )\r
verifyStream();\r
if ( stream_.state == STREAM_STOPPED ) {\r
errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";\r
- error( RtError::WARNING );\r
- return;\r
- }\r
-\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\r
\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
+ error( RtAudioError::SYSTEM_ERROR );\r
}\r
\r
void RtApiCore :: abortStream( void )\r
verifyStream();\r
if ( stream_.state == STREAM_STOPPED ) {\r
errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\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
+static 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
+ error( RtAudioError::WARNING );\r
return FAILURE;\r
}\r
\r
\r
// Check if we were draining the stream and signal is finished.\r
if ( handle->drainCounter > 3 ) {\r
+ ThreadHandle threadId;\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
:client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }\r
};\r
\r
-ThreadHandle threadId;\r
-void jackSilentError( const char * ) {};\r
+static void jackSilentError( const char * ) {};\r
\r
RtApiJack :: RtApiJack()\r
{\r
jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );\r
if ( client == 0 ) {\r
errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\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
+ error( RtAudioError::INVALID_USE );\r
+ return info;\r
}\r
\r
// Get the current jack server sample rate.\r
if ( info.outputChannels == 0 && info.inputChannels == 0 ) {\r
jack_client_close(client);\r
errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
return info;\r
}\r
\r
-int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )\r
+static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )\r
{\r
CallbackInfo *info = (CallbackInfo *) infoPointer;\r
\r
// server signals that it is shutting down. It is necessary to handle\r
// it this way because the jackShutdown() function must return before\r
// the jack_deactivate() function (in closeStream()) will return.\r
-extern "C" void *jackCloseStream( void *ptr )\r
+static void *jackCloseStream( void *ptr )\r
{\r
CallbackInfo *info = (CallbackInfo *) ptr;\r
RtApiJack *object = (RtApiJack *) info->object;\r
\r
pthread_exit( NULL );\r
}\r
-void jackShutdown( void *infoPointer )\r
+static void jackShutdown( void *infoPointer )\r
{\r
CallbackInfo *info = (CallbackInfo *) infoPointer;\r
RtApiJack *object = (RtApiJack *) info->object;\r
// other problem occurred and we should close the stream.\r
if ( object->isStreamRunning() == false ) return;\r
\r
+ ThreadHandle threadId;\r
pthread_create( &threadId, NULL, jackCloseStream, info );\r
std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;\r
}\r
\r
-int jackXrun( void *infoPointer )\r
+static int jackXrun( void *infoPointer )\r
{\r
JackHandle *handle = (JackHandle *) infoPointer;\r
\r
client = jack_client_open( "RtApiJack", jackoptions, status );\r
if ( client == 0 ) {\r
errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return FAILURE;\r
}\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
{\r
if ( stream_.state == STREAM_CLOSED ) {\r
errorText_ = "RtApiJack::closeStream(): no open stream to close!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\r
\r
verifyStream();\r
if ( stream_.state == STREAM_RUNNING ) {\r
errorText_ = "RtApiJack::startStream(): the stream is already running!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\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
+ error( RtAudioError::SYSTEM_ERROR );\r
}\r
\r
void RtApiJack :: stopStream( void )\r
verifyStream();\r
if ( stream_.state == STREAM_STOPPED ) {\r
errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";\r
- error( RtError::WARNING );\r
- return;\r
- }\r
-\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\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
verifyStream();\r
if ( stream_.state == STREAM_STOPPED ) {\r
errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\r
\r
// aborted. It is necessary to handle it this way because the\r
// callbackEvent() function must return before the jack_deactivate()\r
// function will return.\r
-extern "C" void *jackStopStream( void *ptr )\r
+static void *jackStopStream( void *ptr )\r
{\r
CallbackInfo *info = (CallbackInfo *) ptr;\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
+ error( RtAudioError::WARNING );\r
return FAILURE;\r
}\r
if ( stream_.bufferSize != nframes ) {\r
errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return FAILURE;\r
}\r
\r
\r
// Check if we were draining the stream and signal is finished.\r
if ( handle->drainCounter > 3 ) {\r
+ ThreadHandle threadId;\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
#include "asiodrivers.h"\r
#include <cmath>\r
\r
-AsioDrivers drivers;\r
-ASIOCallbacks asioCallbacks;\r
-ASIODriverInfo driverInfo;\r
-CallbackInfo *asioCallbackInfo;\r
-bool asioXRun;\r
+static AsioDrivers drivers;\r
+static ASIOCallbacks asioCallbacks;\r
+static ASIODriverInfo driverInfo;\r
+static CallbackInfo *asioCallbackInfo;\r
+static bool asioXRun;\r
\r
struct AsioHandle {\r
int drainCounter; // Tracks callback counts when draining\r
\r
// Function declarations (definitions at end of section)\r
static const char* getAsioErrorString( ASIOError result );\r
-void sampleRateChanged( ASIOSampleRate sRate );\r
-long asioMessages( long selector, long value, void* message, double* opt );\r
+static void sampleRateChanged( ASIOSampleRate sRate );\r
+static long asioMessages( long selector, long value, void* message, double* opt );\r
\r
RtApiAsio :: RtApiAsio()\r
{\r
HRESULT hr = CoInitialize( NULL ); \r
if ( FAILED(hr) ) {\r
errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
}\r
coInitialized_ = true;\r
\r
unsigned int nDevices = getDeviceCount();\r
if ( nDevices == 0 ) {\r
errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";\r
- error( RtError::INVALID_USE );\r
+ error( RtAudioError::INVALID_USE );\r
+ return info;\r
}\r
\r
if ( device >= nDevices ) {\r
errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";\r
- error( RtError::INVALID_USE );\r
+ error( RtAudioError::INVALID_USE );\r
+ return info;\r
}\r
\r
// If a stream is already open, we cannot probe other devices. Thus, use the saved results.\r
if ( stream_.state != STREAM_CLOSED ) {\r
if ( device >= devices_.size() ) {\r
errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
return devices_[ device ];\r
if ( result != ASE_OK ) {\r
errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
if ( !drivers.loadDriver( driverName ) ) {\r
errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
if ( result != ASE_OK ) {\r
errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
drivers.removeCurrentDriver();\r
errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
drivers.removeCurrentDriver();\r
errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\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
return info;\r
}\r
\r
-void bufferSwitch( long index, ASIOBool processNow )\r
+static void bufferSwitch( long index, ASIOBool processNow )\r
{\r
RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;\r
object->callbackEvent( index );\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
if ( result != ASE_OK ) {\r
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING); // warn but don't fail\r
+ error( RtAudioError::WARNING); // warn but don't fail\r
}\r
else {\r
stream_.latency[0] = outputLatency;\r
{\r
if ( stream_.state == STREAM_CLOSED ) {\r
errorText_ = "RtApiAsio::closeStream(): no open stream to close!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\r
\r
verifyStream();\r
if ( stream_.state == STREAM_RUNNING ) {\r
errorText_ = "RtApiAsio::startStream(): the stream is already running!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\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
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
}\r
\r
void RtApiAsio :: stopStream()\r
verifyStream();\r
if ( stream_.state == STREAM_STOPPED ) {\r
errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";\r
- error( RtError::WARNING );\r
- return;\r
- }\r
-\r
- /*\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
+ error( RtAudioError::WARNING );\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
+ error( RtAudioError::SYSTEM_ERROR );\r
}\r
\r
void RtApiAsio :: abortStream()\r
verifyStream();\r
if ( stream_.state == STREAM_STOPPED ) {\r
errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\r
\r
// aborted. It is necessary to handle it this way because the\r
// callbackEvent() function must return before the ASIOStop()\r
// function will return.\r
-extern "C" unsigned __stdcall asioStopStream( void *ptr )\r
+static unsigned __stdcall asioStopStream( void *ptr )\r
{\r
CallbackInfo *info = (CallbackInfo *) ptr;\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
+ error( RtAudioError::WARNING );\r
return FAILURE;\r
}\r
\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
\r
-void sampleRateChanged( ASIOSampleRate sRate )\r
+static void sampleRateChanged( ASIOSampleRate sRate )\r
{\r
// The ASIO documentation says that this usually only happens during\r
// external sync. Audio processing is not stopped by the driver,\r
try {\r
object->stopStream();\r
}\r
- catch ( RtError &exception ) {\r
+ catch ( RtAudioError &exception ) {\r
std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;\r
return;\r
}\r
std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;\r
}\r
\r
-long asioMessages( long selector, long value, void* message, double* opt )\r
+static long asioMessages( long selector, long value, void* message, double* opt )\r
{\r
long ret = 0;\r
\r
const char*message;\r
};\r
\r
- static Messages m[] = \r
+ static const Messages m[] = \r
{\r
{ ASE_NotPresent, "Hardware input or output is not present or available." },\r
{ ASE_HWMalfunction, "Hardware is malfunctioning." },\r
\r
static const char* getErrorString( int code );\r
\r
-extern "C" unsigned __stdcall callbackHandler( void *ptr );\r
+static unsigned __stdcall callbackHandler( void *ptr );\r
\r
struct DsDevice {\r
LPGUID id[2];\r
: found(false) { validId[0] = false; validId[1] = false; }\r
};\r
\r
-std::vector< DsDevice > dsDevices;\r
+struct DsProbeData {\r
+ bool isInput;\r
+ std::vector<struct DsDevice>* dsDevices;\r
+};\r
\r
RtApiDs :: RtApiDs()\r
{\r
dsDevices[i].found = false;\r
\r
// Query DirectSound devices.\r
- bool isInput = false;\r
- HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &isInput );\r
+ struct DsProbeData probeInfo;\r
+ probeInfo.isInput = false;\r
+ probeInfo.dsDevices = &dsDevices;\r
+ HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );\r
if ( FAILED( result ) ) {\r
errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
}\r
\r
// Query DirectSoundCapture devices.\r
- isInput = true;\r
- result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &isInput );\r
+ probeInfo.isInput = true;\r
+ result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );\r
if ( FAILED( result ) ) {\r
errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
}\r
\r
// Clean out any devices that may have disappeared.\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
getDeviceCount();\r
if ( dsDevices.size() == 0 ) {\r
errorText_ = "RtApiDs::getDeviceInfo: no devices found!";\r
- error( RtError::INVALID_USE );\r
+ error( RtAudioError::INVALID_USE );\r
+ return info;\r
}\r
}\r
\r
if ( device >= dsDevices.size() ) {\r
errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";\r
- error( RtError::INVALID_USE );\r
+ error( RtAudioError::INVALID_USE );\r
+ return info;\r
}\r
\r
HRESULT result;\r
if ( FAILED( result ) ) {\r
errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
goto probeInput;\r
}\r
\r
output->Release();\r
errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
goto probeInput;\r
}\r
\r
if ( FAILED( result ) ) {\r
errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
input->Release();\r
errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\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
stream_.deviceBuffer = 0;\r
}\r
\r
+ stream_.state = STREAM_CLOSED;\r
return FAILURE;\r
}\r
\r
{\r
if ( stream_.state == STREAM_CLOSED ) {\r
errorText_ = "RtApiDs::closeStream(): no open stream to close!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\r
\r
verifyStream();\r
if ( stream_.state == STREAM_RUNNING ) {\r
errorText_ = "RtApiDs::startStream(): the stream is already running!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\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
+ if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );\r
}\r
\r
void RtApiDs :: stopStream()\r
verifyStream();\r
if ( stream_.state == STREAM_STOPPED ) {\r
errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";\r
- error( RtError::WARNING );\r
- return;\r
- }\r
-\r
- /*\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\r
- */\r
\r
HRESULT result = 0;\r
LPVOID audioPtr;\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
+ if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );\r
}\r
\r
void RtApiDs :: abortStream()\r
verifyStream();\r
if ( stream_.state == STREAM_STOPPED ) {\r
errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\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
if ( stream_.state == STREAM_CLOSED ) {\r
errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\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
if ( FAILED( result ) ) {\r
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";\r
errorText_ = errorStream_.str();\r
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
+ return;\r
}\r
result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );\r
if ( FAILED( result ) ) {\r
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";\r
errorText_ = errorStream_.str();\r
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
+ return;\r
}\r
while ( true ) {\r
result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );\r
if ( FAILED( result ) ) {\r
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";\r
errorText_ = errorStream_.str();\r
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
+ return;\r
}\r
result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );\r
if ( FAILED( result ) ) {\r
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";\r
errorText_ = errorStream_.str();\r
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
+ return;\r
}\r
if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;\r
Sleep( 1 );\r
if ( FAILED( result ) ) {\r
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";\r
errorText_ = errorStream_.str();\r
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
+ return;\r
}\r
handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];\r
if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];\r
if ( FAILED( result ) ) {\r
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";\r
errorText_ = errorStream_.str();\r
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
+ return;\r
}\r
\r
// We will copy our output buffer into the region between\r
if ( FAILED( result ) ) {\r
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";\r
errorText_ = errorStream_.str();\r
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
+ return;\r
}\r
\r
// Copy our buffer into the DS buffer\r
if ( FAILED( result ) ) {\r
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";\r
errorText_ = errorStream_.str();\r
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
+ return;\r
}\r
nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;\r
handle->bufferPointer[0] = nextWritePointer;\r
if ( FAILED( result ) ) {\r
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";\r
errorText_ = errorStream_.str();\r
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
+ return;\r
}\r
\r
if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset\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
if ( FAILED( result ) ) {\r
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";\r
errorText_ = errorStream_.str();\r
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
+ return;\r
}\r
\r
if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset\r
if ( FAILED( result ) ) {\r
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";\r
errorText_ = errorStream_.str();\r
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
+ return;\r
}\r
\r
if ( duplexPrerollBytes <= 0 ) {\r
if ( FAILED( result ) ) {\r
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";\r
errorText_ = errorStream_.str();\r
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
+ return;\r
}\r
handle->bufferPointer[1] = nextReadPointer;\r
\r
}\r
\r
unlock:\r
- // MUTEX_UNLOCK( &stream_.mutex );\r
-\r
RtApi::tickStreamTime();\r
}\r
\r
// Definitions for utility functions and callbacks\r
// specific to the DirectSound implementation.\r
\r
-extern "C" unsigned __stdcall callbackHandler( void *ptr )\r
+static unsigned __stdcall callbackHandler( void *ptr )\r
{\r
CallbackInfo *info = (CallbackInfo *) ptr;\r
RtApiDs *object = (RtApiDs *) info->object;\r
\r
#include "tchar.h"\r
\r
-std::string convertTChar( LPCTSTR name )\r
+static std::string convertTChar( LPCTSTR name )\r
{\r
#if defined( UNICODE ) || defined( _UNICODE )\r
int length = WideCharToMultiByte(CP_UTF8, 0, name, -1, NULL, 0, NULL, NULL);\r
- std::string s( length, 0 );\r
- length = WideCharToMultiByte(CP_UTF8, 0, name, wcslen(name), &s[0], length, NULL, NULL);\r
+ std::string s( length-1, '\0' );\r
+ WideCharToMultiByte(CP_UTF8, 0, name, -1, &s[0], length, NULL, NULL);\r
#else\r
std::string s( name );\r
#endif\r
LPCTSTR module,\r
LPVOID lpContext )\r
{\r
- bool *isInput = (bool *) lpContext;\r
+ struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;\r
+ std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;\r
\r
HRESULT hr;\r
bool validDevice = false;\r
- if ( *isInput == true ) {\r
+ if ( probeInfo.isInput == true ) {\r
DSCCAPS caps;\r
LPDIRECTSOUNDCAPTURE object;\r
\r
\r
// If good device, then save its name and guid.\r
std::string name = convertTChar( description );\r
- if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )\r
+ //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )\r
+ if ( lpguid == NULL )\r
name = "Default Device";\r
if ( validDevice ) {\r
for ( unsigned int i=0; i<dsDevices.size(); i++ ) {\r
if ( dsDevices[i].name == name ) {\r
dsDevices[i].found = true;\r
- if ( *isInput ) {\r
+ if ( probeInfo.isInput ) {\r
dsDevices[i].id[1] = lpguid;\r
dsDevices[i].validId[1] = true;\r
}\r
DsDevice device;\r
device.name = name;\r
device.found = true;\r
- if ( *isInput ) {\r
+ if ( probeInfo.isInput ) {\r
device.id[1] = lpguid;\r
device.validId[1] = true;\r
}\r
:synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }\r
};\r
\r
-extern "C" void *alsaCallbackHandler( void * ptr );\r
+static void *alsaCallbackHandler( void * ptr );\r
\r
RtApiAlsa :: RtApiAlsa()\r
{\r
if ( result < 0 ) {\r
errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
goto nextcard;\r
}\r
subdevice = -1;\r
if ( result < 0 ) {\r
errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
break;\r
}\r
if ( subdevice < 0 )\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
if ( result < 0 ) {\r
errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
goto nextcard;\r
}\r
subdevice = -1;\r
if ( result < 0 ) {\r
errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
break;\r
}\r
if ( subdevice < 0 ) break;\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
+ error( RtAudioError::INVALID_USE );\r
+ return info;\r
}\r
\r
if ( device >= nDevices ) {\r
errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";\r
- error( RtError::INVALID_USE );\r
+ error( RtAudioError::INVALID_USE );\r
+ return info;\r
}\r
\r
foundDevice:\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
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
return devices_[ device ];\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
if ( result < 0 ) {\r
errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
goto captureProbe;\r
}\r
\r
snd_pcm_close( phandle );\r
errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
goto captureProbe;\r
}\r
\r
snd_pcm_close( phandle );\r
errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
goto captureProbe;\r
}\r
info.outputChannels = value;\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
+ else\r
+ snd_ctl_close( chandle );\r
\r
result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);\r
if ( result < 0 ) {\r
errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
if ( info.outputChannels == 0 ) return info;\r
goto probeParameters;\r
}\r
snd_pcm_close( phandle );\r
errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
if ( info.outputChannels == 0 ) return info;\r
goto probeParameters;\r
}\r
snd_pcm_close( phandle );\r
errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
if ( info.outputChannels == 0 ) return info;\r
goto probeParameters;\r
}\r
if ( result < 0 ) {\r
errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
snd_pcm_close( phandle );\r
errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
snd_pcm_close( phandle );\r
errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
\r
// Check that we have at least one supported format\r
if ( info.nativeFormats == 0 ) {\r
+ snd_pcm_close( phandle );\r
errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
// Get the device name\r
char *cardname;\r
result = snd_card_get_name( card, &cardname );\r
- if ( result >= 0 )\r
+ if ( result >= 0 ) {\r
sprintf( name, "hw:%s,%d", cardname, subdevice );\r
+ free( cardname );\r
+ }\r
info.name = name;\r
\r
// That's all ... close the device and return\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
apiInfo->synchronized = true;\r
else {\r
errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
}\r
}\r
else {\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_.deviceBuffer = 0;\r
}\r
\r
+ stream_.state = STREAM_CLOSED;\r
return FAILURE;\r
}\r
\r
{\r
if ( stream_.state == STREAM_CLOSED ) {\r
errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\r
\r
verifyStream();\r
if ( stream_.state == STREAM_RUNNING ) {\r
errorText_ = "RtApiAlsa::startStream(): the stream is already running!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\r
\r
}\r
\r
if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {\r
+ result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open\r
state = snd_pcm_state( handle[1] );\r
if ( state != SND_PCM_STATE_PREPARED ) {\r
result = snd_pcm_prepare( handle[1] );\r
stream_.state = STREAM_RUNNING;\r
\r
unlock:\r
- apiInfo->runnable = true;\r
+ apiInfo->runnable = false; // fixes high CPU usage when stopped\r
pthread_cond_signal( &apiInfo->runnable_cv );\r
MUTEX_UNLOCK( &stream_.mutex );\r
\r
if ( result >= 0 ) return;\r
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
}\r
\r
void RtApiAlsa :: stopStream()\r
verifyStream();\r
if ( stream_.state == STREAM_STOPPED ) {\r
errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\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
+ apiInfo->runnable = false; // fixes high CPU usage when stopped\r
MUTEX_UNLOCK( &stream_.mutex );\r
\r
if ( result >= 0 ) return;\r
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
}\r
\r
void RtApiAlsa :: abortStream()\r
verifyStream();\r
if ( stream_.state == STREAM_STOPPED ) {\r
errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\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
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
}\r
\r
void RtApiAlsa :: callbackEvent()\r
\r
if ( stream_.state == STREAM_CLOSED ) {\r
errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\r
\r
errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";\r
errorText_ = errorStream_.str();\r
}\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
goto tryOutput;\r
}\r
\r
errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";\r
errorText_ = errorStream_.str();\r
}\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
goto unlock;\r
}\r
\r
if ( doStopStream == 1 ) this->stopStream();\r
}\r
\r
-extern "C" void *alsaCallbackHandler( void *ptr )\r
+static void *alsaCallbackHandler( void *ptr )\r
{\r
CallbackInfo *info = (CallbackInfo *) ptr;\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
-#if defined(__LINUX_OSS__)\r
+// Code written by Peter Meerwald, pmeerw@pmeerw.net\r
+// and Tristan Matthews.\r
\r
-#include <unistd.h>\r
-#include <sys/ioctl.h>\r
-#include <unistd.h>\r
-#include <fcntl.h>\r
-#include "soundcard.h"\r
-#include <errno.h>\r
-#include <math.h>\r
+#include <pulse/error.h>\r
+#include <pulse/simple.h>\r
+#include <cstdio>\r
\r
-extern "C" void *ossCallbackHandler(void * ptr);\r
+static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,\r
+ 44100, 48000, 96000, 0};\r
\r
-// A structure to hold various information related to the OSS API\r
-// implementation.\r
-struct OssHandle {\r
- int id[2]; // device ids\r
- bool xrun[2];\r
- bool triggered;\r
- pthread_cond_t runnable;\r
+struct rtaudio_pa_format_mapping_t {\r
+ RtAudioFormat rtaudio_format;\r
+ pa_sample_format_t pa_format;\r
+};\r
\r
- OssHandle()\r
- :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }\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
-RtApiOss :: RtApiOss()\r
+RtApiPulse::~RtApiPulse()\r
{\r
- // Nothing to do here.\r
+ if ( stream_.state != STREAM_CLOSED )\r
+ closeStream();\r
}\r
\r
-RtApiOss :: ~RtApiOss()\r
+unsigned int RtApiPulse::getDeviceCount( void )\r
{\r
- if ( stream_.state != STREAM_CLOSED ) closeStream();\r
+ return 1;\r
}\r
\r
-unsigned int RtApiOss :: getDeviceCount( void )\r
+RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int device )\r
{\r
- int mixerfd = open( "/dev/mixer", O_RDWR, 0 );\r
- if ( mixerfd == -1 ) {\r
- errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";\r
- error( RtError::WARNING );\r
- return 0;\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
- oss_sysinfo sysinfo;\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
+static 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( RtAudioError::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( RtAudioError::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( RtAudioError::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( RtAudioError::INVALID_USE );\r
+ return;\r
+ }\r
+ if ( stream_.state == STREAM_RUNNING ) {\r
+ errorText_ = "RtApiPulse::startStream(): the stream is already running!";\r
+ error( RtAudioError::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( RtAudioError::INVALID_USE );\r
+ return;\r
+ }\r
+ if ( stream_.state == STREAM_STOPPED ) {\r
+ errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";\r
+ error( RtAudioError::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( RtAudioError::SYSTEM_ERROR );\r
+ return;\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( RtAudioError::INVALID_USE );\r
+ return;\r
+ }\r
+ if ( stream_.state == STREAM_STOPPED ) {\r
+ errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";\r
+ error( RtAudioError::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( RtAudioError::SYSTEM_ERROR );\r
+ return;\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
+\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
+ std::string streamName = "RtAudio";\r
+ if ( !options->streamName.empty() ) streamName = options->streamName;\r
+ switch ( mode ) {\r
+ case INPUT:\r
+ pa_buffer_attr buffer_attr;\r
+ buffer_attr.fragsize = bufferBytes;\r
+ buffer_attr.maxlength = -1;\r
+\r
+ pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &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
+ 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
+\r
+ stream_.state = STREAM_STOPPED;\r
+ return true;\r
+ \r
+ error:\r
+ if ( pah && stream_.callbackInfo.isRunning ) {\r
+ pthread_cond_destroy( &pah->runnable_cv );\r
+ delete pah;\r
+ stream_.apiHandle = 0;\r
+ }\r
+\r
+ for ( int i=0; i<2; i++ ) {\r
+ if ( stream_.userBuffer[i] ) {\r
+ free( stream_.userBuffer[i] );\r
+ stream_.userBuffer[i] = 0;\r
+ }\r
+ }\r
+\r
+ if ( stream_.deviceBuffer ) {\r
+ free( stream_.deviceBuffer );\r
+ stream_.deviceBuffer = 0;\r
+ }\r
+\r
+ return FAILURE;\r
+}\r
+\r
+//******************** End of __LINUX_PULSE__ *********************//\r
+#endif\r
+\r
+#if defined(__LINUX_OSS__)\r
+\r
+#include <unistd.h>\r
+#include <sys/ioctl.h>\r
+#include <unistd.h>\r
+#include <fcntl.h>\r
+#include <sys/soundcard.h>\r
+#include <errno.h>\r
+#include <math.h>\r
+\r
+static void *ossCallbackHandler(void * ptr);\r
+\r
+// A structure to hold various information related to the OSS API\r
+// implementation.\r
+struct OssHandle {\r
+ int id[2]; // device ids\r
+ bool xrun[2];\r
+ bool triggered;\r
+ pthread_cond_t runnable;\r
+\r
+ OssHandle()\r
+ :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }\r
+};\r
+\r
+RtApiOss :: RtApiOss()\r
+{\r
+ // Nothing to do here.\r
+}\r
+\r
+RtApiOss :: ~RtApiOss()\r
+{\r
+ if ( stream_.state != STREAM_CLOSED ) closeStream();\r
+}\r
+\r
+unsigned int RtApiOss :: getDeviceCount( void )\r
+{\r
+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );\r
+ if ( mixerfd == -1 ) {\r
+ errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";\r
+ error( RtAudioError::WARNING );\r
+ return 0;\r
+ }\r
+\r
+ oss_sysinfo sysinfo;\r
if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {\r
close( mixerfd );\r
errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return 0;\r
}\r
\r
int mixerfd = open( "/dev/mixer", O_RDWR, 0 );\r
if ( mixerfd == -1 ) {\r
errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
if ( result == -1 ) {\r
close( mixerfd );\r
errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
if ( nDevices == 0 ) {\r
close( mixerfd );\r
errorText_ = "RtApiOss::getDeviceInfo: no devices found!";\r
- error( RtError::INVALID_USE );\r
+ error( RtAudioError::INVALID_USE );\r
+ return info;\r
}\r
\r
if ( device >= nDevices ) {\r
close( mixerfd );\r
errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";\r
- error( RtError::INVALID_USE );\r
+ error( RtAudioError::INVALID_USE );\r
+ return info;\r
}\r
\r
oss_audioinfo ainfo;\r
if ( result == -1 ) {\r
errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
if ( info.nativeFormats == 0 ) {\r
errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return info;\r
}\r
\r
if ( info.sampleRates.size() == 0 ) {\r
errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";\r
errorText_ = errorStream_.str();\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
}\r
else {\r
info.probed = true;\r
{\r
if ( stream_.state == STREAM_CLOSED ) {\r
errorText_ = "RtApiOss::closeStream(): no open stream to close!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\r
\r
verifyStream();\r
if ( stream_.state == STREAM_RUNNING ) {\r
errorText_ = "RtApiOss::startStream(): the stream is already running!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\r
\r
verifyStream();\r
if ( stream_.state == STREAM_STOPPED ) {\r
errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\r
\r
result = write( handle->id[0], buffer, samples * formatBytes(format) );\r
if ( result == -1 ) {\r
errorText_ = "RtApiOss::stopStream: audio write error.";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
}\r
}\r
\r
MUTEX_UNLOCK( &stream_.mutex );\r
\r
if ( result != -1 ) return;\r
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
}\r
\r
void RtApiOss :: abortStream()\r
verifyStream();\r
if ( stream_.state == STREAM_STOPPED ) {\r
errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\r
\r
MUTEX_UNLOCK( &stream_.mutex );\r
\r
if ( result != -1 ) return;\r
- error( RtError::SYSTEM_ERROR );\r
+ error( RtAudioError::SYSTEM_ERROR );\r
}\r
\r
void RtApiOss :: callbackEvent()\r
\r
if ( stream_.state == STREAM_CLOSED ) {\r
errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
return;\r
}\r
\r
// specific means for determining that.\r
handle->xrun[0] = true;\r
errorText_ = "RtApiOss::callbackEvent: audio write error.";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
// Continue on to input section.\r
}\r
}\r
// specific means for determining that.\r
handle->xrun[1] = true;\r
errorText_ = "RtApiOss::callbackEvent: audio read error.";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
goto unlock;\r
}\r
\r
if ( doStopStream == 1 ) this->stopStream();\r
}\r
\r
-extern "C" void *ossCallbackHandler( void *ptr )\r
+static void *ossCallbackHandler( void *ptr )\r
{\r
CallbackInfo *info = (CallbackInfo *) ptr;\r
RtApiOss *object = (RtApiOss *) info->object;\r
\r
// This method can be modified to control the behavior of error\r
// message printing.\r
-void RtApi :: error( RtError::Type type )\r
+void RtApi :: error( RtAudioError::Type type )\r
{\r
errorStream_.str(""); // clear the ostringstream\r
- if ( type == RtError::WARNING && showWarnings_ == true )\r
+\r
+ RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;\r
+ if ( errorCallback ) {\r
+ // abortStream() can generate new error messages. Ignore them. Just keep original one.\r
+\r
+ if ( firstErrorOccurred_ )\r
+ return;\r
+\r
+ firstErrorOccurred_ = true;\r
+ const std::string errorMessage = errorText_;\r
+\r
+ if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {\r
+ stream_.callbackInfo.isRunning = false; // exit from the thread\r
+ abortStream();\r
+ }\r
+\r
+ errorCallback( type, errorMessage );\r
+ firstErrorOccurred_ = false;\r
+ return;\r
+ }\r
+\r
+ if ( type == RtAudioError::WARNING && showWarnings_ == true )\r
std::cerr << '\n' << errorText_ << "\n\n";\r
- else if ( type != RtError::WARNING )\r
- throw( RtError( errorText_, type ) );\r
+ else if ( type != RtAudioError::WARNING )\r
+ throw( RtAudioError( errorText_, type ) );\r
}\r
\r
void RtApi :: verifyStream()\r
{\r
if ( stream_.state == STREAM_CLOSED ) {\r
errorText_ = "RtApi:: a stream is not open!";\r
- error( RtError::INVALID_USE );\r
+ error( RtAudioError::INVALID_USE );\r
}\r
}\r
\r
stream_.callbackInfo.callback = 0;\r
stream_.callbackInfo.userData = 0;\r
stream_.callbackInfo.isRunning = false;\r
+ stream_.callbackInfo.errorCallback = 0;\r
for ( int i=0; i<2; i++ ) {\r
stream_.device[i] = 11111;\r
stream_.doConvertBuffer[i] = false;\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
errorText_ = "RtApi::formatBytes: undefined format.";\r
- error( RtError::WARNING );\r
+ error( RtAudioError::WARNING );\r
\r
return 0;\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-cdist.git / blobdiff