diff options
Diffstat (limited to 'drivers/rtaudio/RtAudio.cpp')
| -rw-r--r-- | drivers/rtaudio/RtAudio.cpp | 18035 |
1 files changed, 10142 insertions, 7893 deletions
diff --git a/drivers/rtaudio/RtAudio.cpp b/drivers/rtaudio/RtAudio.cpp index 883c3a82d1..04e7b4422e 100644 --- a/drivers/rtaudio/RtAudio.cpp +++ b/drivers/rtaudio/RtAudio.cpp @@ -1,7893 +1,10142 @@ -#ifdef RTAUDIO_ENABLED -/************************************************************************/ -/*! \class RtAudio - \brief Realtime audio i/o C++ classes. - - RtAudio provides a common API (Application Programming Interface) - for realtime audio input/output across Linux (native ALSA, Jack, - and OSS), Macintosh OS X (CoreAudio and Jack), and Windows - (DirectSound and ASIO) operating systems. - - RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/ - - RtAudio: realtime audio i/o C++ classes - Copyright (c) 2001-2009 Gary P. Scavone - - Permission is hereby granted, free of charge, to any person - obtaining a copy of this software and associated documentation files - (the "Software"), to deal in the Software without restriction, - including without limitation the rights to use, copy, modify, merge, - publish, distribute, sublicense, and/or sell copies of the Software, - and to permit persons to whom the Software is furnished to do so, - subject to the following conditions: - - The above copyright notice and this permission notice shall be - included in all copies or substantial portions of the Software. - - Any person wishing to distribute modifications to the Software is - asked to send the modifications to the original developer so that - they can be incorporated into the canonical version. This is, - however, not a binding provision of this license. - - THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, - EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF - MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. - IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR - ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF - CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION - WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. -*/ -/************************************************************************/ - -// RtAudio: Version 4.0.6 - -#include "RtAudio.h" -#include <iostream> -#include <cstdlib> -#include <cstring> -#include <limits.h> - -// Static variable definitions. -const unsigned int RtApi::MAX_SAMPLE_RATES = 14; -const unsigned int RtApi::SAMPLE_RATES[] = { - 4000, 5512, 8000, 9600, 11025, 16000, 22050, - 32000, 44100, 48000, 88200, 96000, 176400, 192000 -}; - -#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) - #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A) - #define MUTEX_DESTROY(A) DeleteCriticalSection(A) - #define MUTEX_LOCK(A) EnterCriticalSection(A) - #define MUTEX_UNLOCK(A) LeaveCriticalSection(A) -#elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) - // pthread API - #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL) - #define MUTEX_DESTROY(A) pthread_mutex_destroy(A) - #define MUTEX_LOCK(A) pthread_mutex_lock(A) - #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A) -#else - #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions - #define MUTEX_DESTROY(A) abs(*A) // dummy definitions -#endif - -// *************************************************** // -// -// RtAudio definitions. -// -// *************************************************** // - -void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw() -{ - apis.clear(); - - // The order here will control the order of RtAudio's API search in - // the constructor. -#if defined(__UNIX_JACK__) - apis.push_back( UNIX_JACK ); -#endif -#if defined(__LINUX_ALSA__) - apis.push_back( LINUX_ALSA ); -#endif -#if defined(__LINUX_OSS__) - apis.push_back( LINUX_OSS ); -#endif -#if defined(__WINDOWS_ASIO__) - apis.push_back( WINDOWS_ASIO ); -#endif -#if defined(__WINDOWS_DS__) - apis.push_back( WINDOWS_DS ); -#endif -#if defined(__MACOSX_CORE__) - apis.push_back( MACOSX_CORE ); -#endif -#if defined(__RTAUDIO_DUMMY__) - apis.push_back( RTAUDIO_DUMMY ); -#endif -} - -void RtAudio :: openRtApi( RtAudio::Api api ) -{ -#if defined(__UNIX_JACK__) - if ( api == UNIX_JACK ) - rtapi_ = new RtApiJack(); -#endif -#if defined(__LINUX_ALSA__) - if ( api == LINUX_ALSA ) - rtapi_ = new RtApiAlsa(); -#endif -#if defined(__LINUX_OSS__) - if ( api == LINUX_OSS ) - rtapi_ = new RtApiOss(); -#endif -#if defined(__WINDOWS_ASIO__) - if ( api == WINDOWS_ASIO ) - rtapi_ = new RtApiAsio(); -#endif -#if defined(__WINDOWS_DS__) - if ( api == WINDOWS_DS ) - rtapi_ = new RtApiDs(); -#endif -#if defined(__MACOSX_CORE__) - if ( api == MACOSX_CORE ) - rtapi_ = new RtApiCore(); -#endif -#if defined(__RTAUDIO_DUMMY__) - if ( api == RTAUDIO_DUMMY ) - rtapi_ = new RtApiDummy(); -#endif -} - -RtAudio :: RtAudio( RtAudio::Api api ) throw() -{ - rtapi_ = 0; - - if ( api != UNSPECIFIED ) { - // Attempt to open the specified API. - openRtApi( api ); - if ( rtapi_ ) return; - - // No compiled support for specified API value. Issue a debug - // warning and continue as if no API was specified. - std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl; - } - - // Iterate through the compiled APIs and return as soon as we find - // one with at least one device or we reach the end of the list. - std::vector< RtAudio::Api > apis; - getCompiledApi( apis ); - for ( unsigned int i=0; i<apis.size(); i++ ) { - openRtApi( apis[i] ); - if ( rtapi_->getDeviceCount() ) break; - } - - if ( rtapi_ ) return; - - // It should not be possible to get here because the preprocessor - // definition __RTAUDIO_DUMMY__ is automatically defined if no - // API-specific definitions are passed to the compiler. But just in - // case something weird happens, we'll print out an error message. - std::cerr << "\nRtAudio: no compiled API support found ... critical error!!\n\n"; -} - -RtAudio :: ~RtAudio() throw() -{ - delete rtapi_; -} - -void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters, - RtAudio::StreamParameters *inputParameters, - RtAudioFormat format, unsigned int sampleRate, - unsigned int *bufferFrames, - RtAudioCallback callback, void *userData, - RtAudio::StreamOptions *options ) -{ - return rtapi_->openStream( outputParameters, inputParameters, format, - sampleRate, bufferFrames, callback, - userData, options ); -} - -// *************************************************** // -// -// Public RtApi definitions (see end of file for -// private or protected utility functions). -// -// *************************************************** // - -RtApi :: RtApi() -{ - stream_.state = STREAM_CLOSED; - stream_.mode = UNINITIALIZED; - stream_.apiHandle = 0; - stream_.userBuffer[0] = 0; - stream_.userBuffer[1] = 0; - MUTEX_INITIALIZE( &stream_.mutex ); - showWarnings_ = true; -} - -RtApi :: ~RtApi() -{ - MUTEX_DESTROY( &stream_.mutex ); -} - -void RtApi :: openStream( RtAudio::StreamParameters *oParams, - RtAudio::StreamParameters *iParams, - RtAudioFormat format, unsigned int sampleRate, - unsigned int *bufferFrames, - RtAudioCallback callback, void *userData, - RtAudio::StreamOptions *options ) -{ - if ( stream_.state != STREAM_CLOSED ) { - errorText_ = "RtApi::openStream: a stream is already open!"; - error( RtError::INVALID_USE ); - } - - if ( oParams && oParams->nChannels < 1 ) { - errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one."; - error( RtError::INVALID_USE ); - } - - if ( iParams && iParams->nChannels < 1 ) { - errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one."; - error( RtError::INVALID_USE ); - } - - if ( oParams == NULL && iParams == NULL ) { - errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!"; - error( RtError::INVALID_USE ); - } - - if ( formatBytes(format) == 0 ) { - errorText_ = "RtApi::openStream: 'format' parameter value is undefined."; - error( RtError::INVALID_USE ); - } - - unsigned int nDevices = getDeviceCount(); - unsigned int oChannels = 0; - if ( oParams ) { - oChannels = oParams->nChannels; - if ( oParams->deviceId >= nDevices ) { - errorText_ = "RtApi::openStream: output device parameter value is invalid."; - error( RtError::INVALID_USE ); - } - } - - unsigned int iChannels = 0; - if ( iParams ) { - iChannels = iParams->nChannels; - if ( iParams->deviceId >= nDevices ) { - errorText_ = "RtApi::openStream: input device parameter value is invalid."; - error( RtError::INVALID_USE ); - } - } - - clearStreamInfo(); - bool result; - - if ( oChannels > 0 ) { - - result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel, - sampleRate, format, bufferFrames, options ); - if ( result == false ) error( RtError::SYSTEM_ERROR ); - } - - if ( iChannels > 0 ) { - - result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel, - sampleRate, format, bufferFrames, options ); - if ( result == false ) { - if ( oChannels > 0 ) closeStream(); - error( RtError::SYSTEM_ERROR ); - } - } - - stream_.callbackInfo.callback = (void *) callback; - stream_.callbackInfo.userData = userData; - - if ( options ) options->numberOfBuffers = stream_.nBuffers; - stream_.state = STREAM_STOPPED; -} - -unsigned int RtApi :: getDefaultInputDevice( void ) -{ - // Should be implemented in subclasses if possible. - return 0; -} - -unsigned int RtApi :: getDefaultOutputDevice( void ) -{ - // Should be implemented in subclasses if possible. - return 0; -} - -void RtApi :: closeStream( void ) -{ - // MUST be implemented in subclasses! - return; -} - -bool RtApi :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - // MUST be implemented in subclasses! - return FAILURE; -} - -void RtApi :: tickStreamTime( void ) -{ - // Subclasses that do not provide their own implementation of - // getStreamTime should call this function once per buffer I/O to - // provide basic stream time support. - - stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate ); - -#if defined( HAVE_GETTIMEOFDAY ) - gettimeofday( &stream_.lastTickTimestamp, NULL ); -#endif -} - -long RtApi :: getStreamLatency( void ) -{ - verifyStream(); - - long totalLatency = 0; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) - totalLatency = stream_.latency[0]; - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) - totalLatency += stream_.latency[1]; - - return totalLatency; -} - -double RtApi :: getStreamTime( void ) -{ - verifyStream(); - -#if defined( HAVE_GETTIMEOFDAY ) - // Return a very accurate estimate of the stream time by - // adding in the elapsed time since the last tick. - struct timeval then; - struct timeval now; - - if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 ) - return stream_.streamTime; - - gettimeofday( &now, NULL ); - then = stream_.lastTickTimestamp; - return stream_.streamTime + - ((now.tv_sec + 0.000001 * now.tv_usec) - - (then.tv_sec + 0.000001 * then.tv_usec)); -#else - return stream_.streamTime; -#endif -} - -unsigned int RtApi :: getStreamSampleRate( void ) -{ - verifyStream(); - - return stream_.sampleRate; -} - - -// *************************************************** // -// -// OS/API-specific methods. -// -// *************************************************** // - -#if defined(__MACOSX_CORE__) - -// The OS X CoreAudio API is designed to use a separate callback -// procedure for each of its audio devices. A single RtAudio duplex -// stream using two different devices is supported here, though it -// cannot be guaranteed to always behave correctly because we cannot -// synchronize these two callbacks. -// -// A property listener is installed for over/underrun information. -// However, no functionality is currently provided to allow property -// listeners to trigger user handlers because it is unclear what could -// be done if a critical stream parameter (buffer size, sample rate, -// device disconnect) notification arrived. The listeners entail -// quite a bit of extra code and most likely, a user program wouldn't -// be prepared for the result anyway. However, we do provide a flag -// to the client callback function to inform of an over/underrun. -// -// The mechanism for querying and setting system parameters was -// updated (and perhaps simplified) in OS-X version 10.4. However, -// since 10.4 support is not necessarily available to all users, I've -// decided not to update the respective code at this time. Perhaps -// this will happen when Apple makes 10.4 free for everyone. :-) - -// A structure to hold various information related to the CoreAudio API -// implementation. -struct CoreHandle { - AudioDeviceID id[2]; // device ids -#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) - AudioDeviceIOProcID procId[2]; -#endif - UInt32 iStream[2]; // device stream index (or first if using multiple) - UInt32 nStreams[2]; // number of streams to use - bool xrun[2]; - char *deviceBuffer; - pthread_cond_t condition; - int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - - CoreHandle() - :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } -}; - -RtApiCore :: RtApiCore() -{ - // Nothing to do here. -} - -RtApiCore :: ~RtApiCore() -{ - // The subclass destructor gets called before the base class - // destructor, so close an existing stream before deallocating - // apiDeviceId memory. - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} - -unsigned int RtApiCore :: getDeviceCount( void ) -{ - // Find out how many audio devices there are, if any. - UInt32 dataSize; - OSStatus result = AudioHardwareGetPropertyInfo( kAudioHardwarePropertyDevices, &dataSize, NULL ); - if ( result != noErr ) { - errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!"; - error( RtError::WARNING ); - return 0; - } - - return dataSize / sizeof( AudioDeviceID ); -} - -unsigned int RtApiCore :: getDefaultInputDevice( void ) -{ - unsigned int nDevices = getDeviceCount(); - if ( nDevices <= 1 ) return 0; - - AudioDeviceID id; - UInt32 dataSize = sizeof( AudioDeviceID ); - OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultInputDevice, - &dataSize, &id ); - - if ( result != noErr ) { - errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device."; - error( RtError::WARNING ); - return 0; - } - - dataSize *= nDevices; - AudioDeviceID deviceList[ nDevices ]; - result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList ); - if ( result != noErr ) { - errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs."; - error( RtError::WARNING ); - return 0; - } - - for ( unsigned int i=0; i<nDevices; i++ ) - if ( id == deviceList[i] ) return i; - - errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!"; - error( RtError::WARNING ); - return 0; -} - -unsigned int RtApiCore :: getDefaultOutputDevice( void ) -{ - unsigned int nDevices = getDeviceCount(); - if ( nDevices <= 1 ) return 0; - - AudioDeviceID id; - UInt32 dataSize = sizeof( AudioDeviceID ); - OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultOutputDevice, - &dataSize, &id ); - - if ( result != noErr ) { - errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device."; - error( RtError::WARNING ); - return 0; - } - - dataSize *= nDevices; - AudioDeviceID deviceList[ nDevices ]; - result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList ); - if ( result != noErr ) { - errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs."; - error( RtError::WARNING ); - return 0; - } - - for ( unsigned int i=0; i<nDevices; i++ ) - if ( id == deviceList[i] ) return i; - - errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!"; - error( RtError::WARNING ); - return 0; -} - -RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; - - // Get device ID - unsigned int nDevices = getDeviceCount(); - if ( nDevices == 0 ) { - errorText_ = "RtApiCore::getDeviceInfo: no devices found!"; - error( RtError::INVALID_USE ); - } - - if ( device >= nDevices ) { - errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - } - - AudioDeviceID deviceList[ nDevices ]; - UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; - OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList ); - if ( result != noErr ) { - errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs."; - error( RtError::WARNING ); - return info; - } - - AudioDeviceID id = deviceList[ device ]; - - // Get the device name. - info.name.erase(); - char name[256]; - dataSize = 256; - result = AudioDeviceGetProperty( id, 0, false, - kAudioDevicePropertyDeviceManufacturer, - &dataSize, name ); - - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - info.name.append( (const char *)name, strlen(name) ); - info.name.append( ": " ); - - dataSize = 256; - result = AudioDeviceGetProperty( id, 0, false, - kAudioDevicePropertyDeviceName, - &dataSize, name ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - info.name.append( (const char *)name, strlen(name) ); - - // Get the output stream "configuration". - AudioBufferList *bufferList = nil; - result = AudioDeviceGetPropertyInfo( id, 0, false, - kAudioDevicePropertyStreamConfiguration, - &dataSize, NULL ); - if (result != noErr || dataSize == 0) { - errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - // Allocate the AudioBufferList. - bufferList = (AudioBufferList *) malloc( dataSize ); - if ( bufferList == NULL ) { - errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList."; - error( RtError::WARNING ); - return info; - } - - result = AudioDeviceGetProperty( id, 0, false, - kAudioDevicePropertyStreamConfiguration, - &dataSize, bufferList ); - if ( result != noErr ) { - free( bufferList ); - errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - // Get output channel information. - unsigned int i, nStreams = bufferList->mNumberBuffers; - for ( i=0; i<nStreams; i++ ) - info.outputChannels += bufferList->mBuffers[i].mNumberChannels; - free( bufferList ); - - // Get the input stream "configuration". - result = AudioDeviceGetPropertyInfo( id, 0, true, - kAudioDevicePropertyStreamConfiguration, - &dataSize, NULL ); - if (result != noErr || dataSize == 0) { - errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - // Allocate the AudioBufferList. - bufferList = (AudioBufferList *) malloc( dataSize ); - if ( bufferList == NULL ) { - errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList."; - error( RtError::WARNING ); - return info; - } - - result = AudioDeviceGetProperty( id, 0, true, - kAudioDevicePropertyStreamConfiguration, - &dataSize, bufferList ); - if ( result != noErr ) { - free( bufferList ); - errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - // Get input channel information. - nStreams = bufferList->mNumberBuffers; - for ( i=0; i<nStreams; i++ ) - info.inputChannels += bufferList->mBuffers[i].mNumberChannels; - free( bufferList ); - - // If device opens for both playback and capture, we determine the channels. - if ( info.outputChannels > 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - - // Probe the device sample rates. - bool isInput = false; - if ( info.outputChannels == 0 ) isInput = true; - - // Determine the supported sample rates. - result = AudioDeviceGetPropertyInfo( id, 0, isInput, - kAudioDevicePropertyAvailableNominalSampleRates, - &dataSize, NULL ); - - if ( result != kAudioHardwareNoError || dataSize == 0 ) { - errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - UInt32 nRanges = dataSize / sizeof( AudioValueRange ); - AudioValueRange rangeList[ nRanges ]; - result = AudioDeviceGetProperty( id, 0, isInput, - kAudioDevicePropertyAvailableNominalSampleRates, - &dataSize, &rangeList ); - - if ( result != kAudioHardwareNoError ) { - errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - Float64 minimumRate = 100000000.0, maximumRate = 0.0; - for ( UInt32 i=0; i<nRanges; i++ ) { - if ( rangeList[i].mMinimum < minimumRate ) minimumRate = rangeList[i].mMinimum; - if ( rangeList[i].mMaximum > maximumRate ) maximumRate = rangeList[i].mMaximum; - } - - info.sampleRates.clear(); - for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) { - if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) - info.sampleRates.push_back( SAMPLE_RATES[k] ); - } - - if ( info.sampleRates.size() == 0 ) { - errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - // CoreAudio always uses 32-bit floating point data for PCM streams. - // Thus, any other "physical" formats supported by the device are of - // no interest to the client. - info.nativeFormats = RTAUDIO_FLOAT32; - - if ( getDefaultOutputDevice() == device ) - info.isDefaultOutput = true; - if ( getDefaultInputDevice() == device ) - info.isDefaultInput = true; - - info.probed = true; - return info; -} - -OSStatus callbackHandler( AudioDeviceID inDevice, - const AudioTimeStamp* inNow, - const AudioBufferList* inInputData, - const AudioTimeStamp* inInputTime, - AudioBufferList* outOutputData, - const AudioTimeStamp* inOutputTime, - void* infoPointer ) -{ - CallbackInfo *info = (CallbackInfo *) infoPointer; - - RtApiCore *object = (RtApiCore *) info->object; - if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false ) - return kAudioHardwareUnspecifiedError; - else - return kAudioHardwareNoError; -} - -OSStatus deviceListener( AudioDeviceID inDevice, - UInt32 channel, - Boolean isInput, - AudioDevicePropertyID propertyID, - void* handlePointer ) -{ - CoreHandle *handle = (CoreHandle *) handlePointer; - if ( propertyID == kAudioDeviceProcessorOverload ) { - if ( isInput ) - handle->xrun[1] = true; - else - handle->xrun[0] = true; - } - - return kAudioHardwareNoError; -} - -static bool hasProperty( AudioDeviceID id, UInt32 channel, bool isInput, AudioDevicePropertyID property ) -{ - OSStatus result = AudioDeviceGetPropertyInfo( id, channel, isInput, property, NULL, NULL ); - return result == 0; -} - -bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - // Get device ID - unsigned int nDevices = getDeviceCount(); - if ( nDevices == 0 ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiCore::probeDeviceOpen: no devices found!"; - return FAILURE; - } - - if ( device >= nDevices ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } - - AudioDeviceID deviceList[ nDevices ]; - UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices; - OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList ); - if ( result != noErr ) { - errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs."; - return FAILURE; - } - - AudioDeviceID id = deviceList[ device ]; - - // Setup for stream mode. - bool isInput = false; - if ( mode == INPUT ) isInput = true; - - // Set or disable "hog" mode. - dataSize = sizeof( UInt32 ); - UInt32 doHog = 0; - if ( options && options->flags & RTAUDIO_HOG_DEVICE ) doHog = 1; - result = AudioHardwareSetProperty( kAudioHardwarePropertyHogModeIsAllowed, dataSize, &doHog ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Get the stream "configuration". - AudioBufferList *bufferList; - result = AudioDeviceGetPropertyInfo( id, 0, isInput, - kAudioDevicePropertyStreamConfiguration, - &dataSize, NULL ); - if (result != noErr || dataSize == 0) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Allocate the AudioBufferList. - bufferList = (AudioBufferList *) malloc( dataSize ); - if ( bufferList == NULL ) { - errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList."; - return FAILURE; - } - - result = AudioDeviceGetProperty( id, 0, isInput, - kAudioDevicePropertyStreamConfiguration, - &dataSize, bufferList ); - if ( result != noErr ) { - free( bufferList ); - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Search for one or more streams that contain the desired number of - // channels. CoreAudio devices can have an arbitrary number of - // streams and each stream can have an arbitrary number of channels. - // For each stream, a single buffer of interleaved samples is - // provided. RtAudio prefers the use of one stream of interleaved - // data or multiple consecutive single-channel streams. However, we - // now support multiple consecutive multi-channel streams of - // interleaved data as well. - UInt32 iStream, offsetCounter = firstChannel; - UInt32 nStreams = bufferList->mNumberBuffers; - bool monoMode = false; - bool foundStream = false; - - // First check that the device supports the requested number of - // channels. - UInt32 deviceChannels = 0; - for ( iStream=0; iStream<nStreams; iStream++ ) - deviceChannels += bufferList->mBuffers[iStream].mNumberChannels; - - if ( deviceChannels < ( channels + firstChannel ) ) { - free( bufferList ); - errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Look for a single stream meeting our needs. - UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0; - for ( iStream=0; iStream<nStreams; iStream++ ) { - streamChannels = bufferList->mBuffers[iStream].mNumberChannels; - if ( streamChannels >= channels + offsetCounter ) { - firstStream = iStream; - channelOffset = offsetCounter; - foundStream = true; - break; - } - if ( streamChannels > offsetCounter ) break; - offsetCounter -= streamChannels; - } - - // If we didn't find a single stream above, then we should be able - // to meet the channel specification with multiple streams. - if ( foundStream == false ) { - monoMode = true; - offsetCounter = firstChannel; - for ( iStream=0; iStream<nStreams; iStream++ ) { - streamChannels = bufferList->mBuffers[iStream].mNumberChannels; - if ( streamChannels > offsetCounter ) break; - offsetCounter -= streamChannels; - } - - firstStream = iStream; - channelOffset = offsetCounter; - Int32 channelCounter = channels + offsetCounter - streamChannels; - - if ( streamChannels > 1 ) monoMode = false; - while ( channelCounter > 0 ) { - streamChannels = bufferList->mBuffers[++iStream].mNumberChannels; - if ( streamChannels > 1 ) monoMode = false; - channelCounter -= streamChannels; - streamCount++; - } - } - - free( bufferList ); - - // Determine the buffer size. - AudioValueRange bufferRange; - dataSize = sizeof( AudioValueRange ); - result = AudioDeviceGetProperty( id, 0, isInput, - kAudioDevicePropertyBufferFrameSizeRange, - &dataSize, &bufferRange ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum; - else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum; - - // Set the buffer size. For multiple streams, I'm assuming we only - // need to make this setting for the master channel. - UInt32 theSize = (UInt32) *bufferSize; - dataSize = sizeof( UInt32 ); - result = AudioDeviceSetProperty( id, NULL, 0, isInput, - kAudioDevicePropertyBufferFrameSize, - dataSize, &theSize ); - - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // If attempting to setup a duplex stream, the bufferSize parameter - // MUST be the same in both directions! - *bufferSize = theSize; - if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - stream_.bufferSize = *bufferSize; - stream_.nBuffers = 1; - - // Get the stream ID(s) so we can set the stream format. We'll have - // to do this for each stream. - AudioStreamID streamIDs[ nStreams ]; - dataSize = nStreams * sizeof( AudioStreamID ); - result = AudioDeviceGetProperty( id, 0, isInput, - kAudioDevicePropertyStreams, - &dataSize, &streamIDs ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream ID(s) for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Now set the stream format. Also, check the physical format of the - // device and change that if necessary. - AudioStreamBasicDescription description; - dataSize = sizeof( AudioStreamBasicDescription ); - - bool updateFormat; - for ( UInt32 i=0; i<streamCount; i++ ) { - - result = AudioStreamGetProperty( streamIDs[firstStream+i], 0, - kAudioStreamPropertyVirtualFormat, - &dataSize, &description ); - - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the sample rate and data format id. However, only make the - // change if the sample rate is not within 1.0 of the desired - // rate and the format is not linear pcm. - updateFormat = false; - if ( fabs( description.mSampleRate - (double)sampleRate ) > 1.0 ) { - description.mSampleRate = (double) sampleRate; - updateFormat = true; - } - - if ( description.mFormatID != kAudioFormatLinearPCM ) { - description.mFormatID = kAudioFormatLinearPCM; - updateFormat = true; - } - - if ( updateFormat ) { - result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, - kAudioStreamPropertyVirtualFormat, - dataSize, &description ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // Now check the physical format. - result = AudioStreamGetProperty( streamIDs[firstStream+i], 0, - kAudioStreamPropertyPhysicalFormat, - &dataSize, &description ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 24 ) { - description.mFormatID = kAudioFormatLinearPCM; - AudioStreamBasicDescription testDescription = description; - unsigned long formatFlags; - - // We'll try higher bit rates first and then work our way down. - testDescription.mBitsPerChannel = 32; - formatFlags = description.mFormatFlags | kLinearPCMFormatFlagIsFloat & ~kLinearPCMFormatFlagIsSignedInteger; - testDescription.mFormatFlags = formatFlags; - result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); - if ( result == noErr ) continue; - - testDescription = description; - testDescription.mBitsPerChannel = 32; - formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger) & ~kLinearPCMFormatFlagIsFloat; - testDescription.mFormatFlags = formatFlags; - result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); - if ( result == noErr ) continue; - - testDescription = description; - testDescription.mBitsPerChannel = 24; - testDescription.mFormatFlags = formatFlags; - result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); - if ( result == noErr ) continue; - - testDescription = description; - testDescription.mBitsPerChannel = 16; - testDescription.mFormatFlags = formatFlags; - result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); - if ( result == noErr ) continue; - - testDescription = description; - testDescription.mBitsPerChannel = 8; - testDescription.mFormatFlags = formatFlags; - result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - } - - // Get the stream latency. There can be latency in both the device - // and the stream. First, attempt to get the device latency on the - // master channel or the first open channel. Errors that might - // occur here are not deemed critical. - - // ***** CHECK THIS ***** // - UInt32 latency, channel = 0; - dataSize = sizeof( UInt32 ); - AudioDevicePropertyID property = kAudioDevicePropertyLatency; - if ( hasProperty( id, channel, isInput, property ) == true ) { - result = AudioDeviceGetProperty( id, channel, isInput, property, &dataSize, &latency ); - if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency; - else { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } - } - - // Now try to get the stream latency. For multiple streams, I assume the - // latency is equal for each. - result = AudioStreamGetProperty( streamIDs[firstStream], 0, property, &dataSize, &latency ); - if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] += latency; - else { - errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream latency for device (" << device << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } - - // Byte-swapping: According to AudioHardware.h, the stream data will - // always be presented in native-endian format, so we should never - // need to byte swap. - stream_.doByteSwap[mode] = false; - - // From the CoreAudio documentation, PCM data must be supplied as - // 32-bit floats. - stream_.userFormat = format; - stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - - if ( streamCount == 1 ) - stream_.nDeviceChannels[mode] = description.mChannelsPerFrame; - else // multiple streams - stream_.nDeviceChannels[mode] = channels; - stream_.nUserChannels[mode] = channels; - stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; - stream_.deviceInterleaved[mode] = true; - if ( monoMode == true ) stream_.deviceInterleaved[mode] = false; - - // Set flags for buffer conversion. - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( streamCount == 1 ) { - if ( stream_.nUserChannels[mode] > 1 && - stream_.userInterleaved != stream_.deviceInterleaved[mode] ) - stream_.doConvertBuffer[mode] = true; - } - else if ( monoMode && stream_.userInterleaved ) - stream_.doConvertBuffer[mode] = true; - - // Allocate our CoreHandle structure for the stream. - CoreHandle *handle = 0; - if ( stream_.apiHandle == 0 ) { - try { - handle = new CoreHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory."; - goto error; - } - - if ( pthread_cond_init( &handle->condition, NULL ) ) { - errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable."; - goto error; - } - stream_.apiHandle = (void *) handle; - } - else - handle = (CoreHandle *) stream_.apiHandle; - handle->iStream[mode] = firstStream; - handle->nStreams[mode] = streamCount; - handle->id[mode] = id; - - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - // If possible, we will make use of the CoreAudio stream buffers as - // "device buffers". However, we can't do this if using multiple - // streams. - if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) { - - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - stream_.sampleRate = sampleRate; - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - stream_.callbackInfo.object = (void *) this; - - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) { - if ( streamCount > 1 ) setConvertInfo( mode, 0 ); - else setConvertInfo( mode, channelOffset ); - } - - if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device ) - // Only one callback procedure per device. - stream_.mode = DUPLEX; - else { -#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) - result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] ); -#else - // deprecated in favor of AudioDeviceCreateIOProcID() - result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo ); -#endif - if ( result != noErr ) { - errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ")."; - errorText_ = errorStream_.str(); - goto error; - } - if ( stream_.mode == OUTPUT && mode == INPUT ) - stream_.mode = DUPLEX; - else - stream_.mode = mode; - } - - // Setup the device property listener for over/underload. - result = AudioDeviceAddPropertyListener( id, 0, isInput, - kAudioDeviceProcessorOverload, - deviceListener, (void *) handle ); - - return SUCCESS; - - error: - if ( handle ) { - pthread_cond_destroy( &handle->condition ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - return FAILURE; -} - -void RtApiCore :: closeStream( void ) -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiCore::closeStream(): no open stream to close!"; - error( RtError::WARNING ); - return; - } - - CoreHandle *handle = (CoreHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( stream_.state == STREAM_RUNNING ) - AudioDeviceStop( handle->id[0], callbackHandler ); -#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) - AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] ); -#else - // deprecated in favor of AudioDeviceDestroyIOProcID() - AudioDeviceRemoveIOProc( handle->id[0], callbackHandler ); -#endif - } - - if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { - if ( stream_.state == STREAM_RUNNING ) - AudioDeviceStop( handle->id[1], callbackHandler ); -#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 ) - AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] ); -#else - // deprecated in favor of AudioDeviceDestroyIOProcID() - AudioDeviceRemoveIOProc( handle->id[1], callbackHandler ); -#endif - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - // Destroy pthread condition variable. - pthread_cond_destroy( &handle->condition ); - delete handle; - stream_.apiHandle = 0; - - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} - -void RtApiCore :: startStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiCore::startStream(): the stream is already running!"; - error( RtError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - OSStatus result = noErr; - CoreHandle *handle = (CoreHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - result = AudioDeviceStart( handle->id[0], callbackHandler ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - if ( stream_.mode == INPUT || - ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { - - result = AudioDeviceStart( handle->id[1], callbackHandler ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - handle->drainCounter = 0; - handle->internalDrain = false; - stream_.state = STREAM_RUNNING; - - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result == noErr ) return; - error( RtError::SYSTEM_ERROR ); -} - -void RtApiCore :: stopStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiCore::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - - OSStatus result = noErr; - CoreHandle *handle = (CoreHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 1; - pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled - } - - result = AudioDeviceStop( handle->id[0], callbackHandler ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) { - - result = AudioDeviceStop( handle->id[1], callbackHandler ); - if ( result != noErr ) { - errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - stream_.state = STREAM_STOPPED; - - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result == noErr ) return; - error( RtError::SYSTEM_ERROR ); -} - -void RtApiCore :: abortStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiCore::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } - - CoreHandle *handle = (CoreHandle *) stream_.apiHandle; - handle->drainCounter = 1; - - stopStream(); -} - -bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, - const AudioBufferList *inBufferList, - const AudioBufferList *outBufferList ) -{ - if ( stream_.state == STREAM_STOPPED ) return SUCCESS; - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); - return FAILURE; - } - - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - CoreHandle *handle = (CoreHandle *) stream_.apiHandle; - - // Check if we were draining the stream and signal is finished. - if ( handle->drainCounter > 3 ) { - if ( handle->internalDrain == false ) - pthread_cond_signal( &handle->condition ); - else - stopStream(); - return SUCCESS; - } - - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return SUCCESS; - } - - AudioDeviceID outputDevice = handle->id[0]; - - // Invoke user callback to get fresh output data UNLESS we are - // draining stream or duplex mode AND the input/output devices are - // different AND this function is called for the input device. - if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; - } - handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( handle->drainCounter == 2 ) { - MUTEX_UNLOCK( &stream_.mutex ); - abortStream(); - return SUCCESS; - } - else if ( handle->drainCounter == 1 ) - handle->internalDrain = true; - } - - if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) { - - if ( handle->drainCounter > 1 ) { // write zeros to the output stream - - if ( handle->nStreams[0] == 1 ) { - memset( outBufferList->mBuffers[handle->iStream[0]].mData, - 0, - outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); - } - else { // fill multiple streams with zeros - for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) { - memset( outBufferList->mBuffers[handle->iStream[0]+i].mData, - 0, - outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize ); - } - } - } - else if ( handle->nStreams[0] == 1 ) { - if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer - convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData, - stream_.userBuffer[0], stream_.convertInfo[0] ); - } - else { // copy from user buffer - memcpy( outBufferList->mBuffers[handle->iStream[0]].mData, - stream_.userBuffer[0], - outBufferList->mBuffers[handle->iStream[0]].mDataByteSize ); - } - } - else { // fill multiple streams - Float32 *inBuffer = (Float32 *) stream_.userBuffer[0]; - if ( stream_.doConvertBuffer[0] ) { - convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - inBuffer = (Float32 *) stream_.deviceBuffer; - } - - if ( stream_.deviceInterleaved[0] == false ) { // mono mode - UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize; - for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) { - memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData, - (void *)&inBuffer[i*stream_.bufferSize], bufferBytes ); - } - } - else { // fill multiple multi-channel streams with interleaved data - UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset; - Float32 *out, *in; - - bool inInterleaved = ( stream_.userInterleaved ) ? true : false; - UInt32 inChannels = stream_.nUserChannels[0]; - if ( stream_.doConvertBuffer[0] ) { - inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode - inChannels = stream_.nDeviceChannels[0]; - } - - if ( inInterleaved ) inOffset = 1; - else inOffset = stream_.bufferSize; - - channelsLeft = inChannels; - for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) { - in = inBuffer; - out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData; - streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels; - - outJump = 0; - // Account for possible channel offset in first stream - if ( i == 0 && stream_.channelOffset[0] > 0 ) { - streamChannels -= stream_.channelOffset[0]; - outJump = stream_.channelOffset[0]; - out += outJump; - } - - // Account for possible unfilled channels at end of the last stream - if ( streamChannels > channelsLeft ) { - outJump = streamChannels - channelsLeft; - streamChannels = channelsLeft; - } - - // Determine input buffer offsets and skips - if ( inInterleaved ) { - inJump = inChannels; - in += inChannels - channelsLeft; - } - else { - inJump = 1; - in += (inChannels - channelsLeft) * inOffset; - } - - for ( unsigned int i=0; i<stream_.bufferSize; i++ ) { - for ( unsigned int j=0; j<streamChannels; j++ ) { - *out++ = in[j*inOffset]; - } - out += outJump; - in += inJump; - } - channelsLeft -= streamChannels; - } - } - } - - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } - } - - AudioDeviceID inputDevice; - inputDevice = handle->id[1]; - if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) { - - if ( handle->nStreams[1] == 1 ) { - if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer - convertBuffer( stream_.userBuffer[1], - (char *) inBufferList->mBuffers[handle->iStream[1]].mData, - stream_.convertInfo[1] ); - } - else { // copy to user buffer - memcpy( stream_.userBuffer[1], - inBufferList->mBuffers[handle->iStream[1]].mData, - inBufferList->mBuffers[handle->iStream[1]].mDataByteSize ); - } - } - else { // read from multiple streams - Float32 *outBuffer = (Float32 *) stream_.userBuffer[1]; - if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer; - - if ( stream_.deviceInterleaved[1] == false ) { // mono mode - UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize; - for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) { - memcpy( (void *)&outBuffer[i*stream_.bufferSize], - inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes ); - } - } - else { // read from multiple multi-channel streams - UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset; - Float32 *out, *in; - - bool outInterleaved = ( stream_.userInterleaved ) ? true : false; - UInt32 outChannels = stream_.nUserChannels[1]; - if ( stream_.doConvertBuffer[1] ) { - outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode - outChannels = stream_.nDeviceChannels[1]; - } - - if ( outInterleaved ) outOffset = 1; - else outOffset = stream_.bufferSize; - - channelsLeft = outChannels; - for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) { - out = outBuffer; - in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData; - streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels; - - inJump = 0; - // Account for possible channel offset in first stream - if ( i == 0 && stream_.channelOffset[1] > 0 ) { - streamChannels -= stream_.channelOffset[1]; - inJump = stream_.channelOffset[1]; - in += inJump; - } - - // Account for possible unread channels at end of the last stream - if ( streamChannels > channelsLeft ) { - inJump = streamChannels - channelsLeft; - streamChannels = channelsLeft; - } - - // Determine output buffer offsets and skips - if ( outInterleaved ) { - outJump = outChannels; - out += outChannels - channelsLeft; - } - else { - outJump = 1; - out += (outChannels - channelsLeft) * outOffset; - } - - for ( unsigned int i=0; i<stream_.bufferSize; i++ ) { - for ( unsigned int j=0; j<streamChannels; j++ ) { - out[j*outOffset] = *in++; - } - out += outJump; - in += inJump; - } - channelsLeft -= streamChannels; - } - } - - if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer - convertBuffer( stream_.userBuffer[1], - stream_.deviceBuffer, - stream_.convertInfo[1] ); - } - } - } - - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - - RtApi::tickStreamTime(); - return SUCCESS; -} - -const char* RtApiCore :: getErrorCode( OSStatus code ) -{ - switch( code ) { - - case kAudioHardwareNotRunningError: - return "kAudioHardwareNotRunningError"; - - case kAudioHardwareUnspecifiedError: - return "kAudioHardwareUnspecifiedError"; - - case kAudioHardwareUnknownPropertyError: - return "kAudioHardwareUnknownPropertyError"; - - case kAudioHardwareBadPropertySizeError: - return "kAudioHardwareBadPropertySizeError"; - - case kAudioHardwareIllegalOperationError: - return "kAudioHardwareIllegalOperationError"; - - case kAudioHardwareBadObjectError: - return "kAudioHardwareBadObjectError"; - - case kAudioHardwareBadDeviceError: - return "kAudioHardwareBadDeviceError"; - - case kAudioHardwareBadStreamError: - return "kAudioHardwareBadStreamError"; - - case kAudioHardwareUnsupportedOperationError: - return "kAudioHardwareUnsupportedOperationError"; - - case kAudioDeviceUnsupportedFormatError: - return "kAudioDeviceUnsupportedFormatError"; - - case kAudioDevicePermissionsError: - return "kAudioDevicePermissionsError"; - - default: - return "CoreAudio unknown error"; - } -} - - //******************** End of __MACOSX_CORE__ *********************// -#endif - -#if defined(__UNIX_JACK__) - -// JACK is a low-latency audio server, originally written for the -// GNU/Linux operating system and now also ported to OS-X. It can -// connect a number of different applications to an audio device, as -// well as allowing them to share audio between themselves. -// -// When using JACK with RtAudio, "devices" refer to JACK clients that -// have ports connected to the server. The JACK server is typically -// started in a terminal as follows: -// -// .jackd -d alsa -d hw:0 -// -// or through an interface program such as qjackctl. Many of the -// parameters normally set for a stream are fixed by the JACK server -// and can be specified when the JACK server is started. In -// particular, -// -// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4 -// -// specifies a sample rate of 44100 Hz, a buffer size of 512 sample -// frames, and number of buffers = 4. Once the server is running, it -// is not possible to override these values. If the values are not -// specified in the command-line, the JACK server uses default values. -// -// The JACK server does not have to be running when an instance of -// RtApiJack is created, though the function getDeviceCount() will -// report 0 devices found until JACK has been started. When no -// devices are available (i.e., the JACK server is not running), a -// stream cannot be opened. - -#include <jack/jack.h> -#include <unistd.h> - -// A structure to hold various information related to the Jack API -// implementation. -struct JackHandle { - jack_client_t *client; - jack_port_t **ports[2]; - std::string deviceName[2]; - bool xrun[2]; - pthread_cond_t condition; - int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - - JackHandle() - :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; } -}; - -void jackSilentError( const char * ) {}; - -RtApiJack :: RtApiJack() -{ - // Nothing to do here. -#if !defined(__RTAUDIO_DEBUG__) - // Turn off Jack's internal error reporting. - jack_set_error_function( &jackSilentError ); -#endif -} - -RtApiJack :: ~RtApiJack() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} - -unsigned int RtApiJack :: getDeviceCount( void ) -{ - // See if we can become a jack client. - jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption; - jack_status_t *status = NULL; - jack_client_t *client = jack_client_open( "RtApiJackCount", options, status ); - if ( client == 0 ) return 0; - - const char **ports; - std::string port, previousPort; - unsigned int nChannels = 0, nDevices = 0; - ports = jack_get_ports( client, NULL, NULL, 0 ); - if ( ports ) { - // Parse the port names up to the first colon (:). - size_t iColon = 0; - do { - port = (char *) ports[ nChannels ]; - iColon = port.find(":"); - if ( iColon != std::string::npos ) { - port = port.substr( 0, iColon + 1 ); - if ( port != previousPort ) { - nDevices++; - previousPort = port; - } - } - } while ( ports[++nChannels] ); - free( ports ); - } - - jack_client_close( client ); - return nDevices; -} - -RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; - - jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption - jack_status_t *status = NULL; - jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status ); - if ( client == 0 ) { - errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!"; - error( RtError::WARNING ); - return info; - } - - const char **ports; - std::string port, previousPort; - unsigned int nPorts = 0, nDevices = 0; - ports = jack_get_ports( client, NULL, NULL, 0 ); - if ( ports ) { - // Parse the port names up to the first colon (:). - size_t iColon = 0; - do { - port = (char *) ports[ nPorts ]; - iColon = port.find(":"); - if ( iColon != std::string::npos ) { - port = port.substr( 0, iColon ); - if ( port != previousPort ) { - if ( nDevices == device ) info.name = port; - nDevices++; - previousPort = port; - } - } - } while ( ports[++nPorts] ); - free( ports ); - } - - if ( device >= nDevices ) { - errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - } - - // Get the current jack server sample rate. - info.sampleRates.clear(); - info.sampleRates.push_back( jack_get_sample_rate( client ) ); - - // Count the available ports containing the client name as device - // channels. Jack "input ports" equal RtAudio output channels. - unsigned int nChannels = 0; - ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput ); - if ( ports ) { - while ( ports[ nChannels ] ) nChannels++; - free( ports ); - info.outputChannels = nChannels; - } - - // Jack "output ports" equal RtAudio input channels. - nChannels = 0; - ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput ); - if ( ports ) { - while ( ports[ nChannels ] ) nChannels++; - free( ports ); - info.inputChannels = nChannels; - } - - if ( info.outputChannels == 0 && info.inputChannels == 0 ) { - jack_client_close(client); - errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!"; - error( RtError::WARNING ); - return info; - } - - // If device opens for both playback and capture, we determine the channels. - if ( info.outputChannels > 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - - // Jack always uses 32-bit floats. - info.nativeFormats = RTAUDIO_FLOAT32; - - // Jack doesn't provide default devices so we'll use the first available one. - if ( device == 0 && info.outputChannels > 0 ) - info.isDefaultOutput = true; - if ( device == 0 && info.inputChannels > 0 ) - info.isDefaultInput = true; - - jack_client_close(client); - info.probed = true; - return info; -} - -int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer ) -{ - CallbackInfo *info = (CallbackInfo *) infoPointer; - - RtApiJack *object = (RtApiJack *) info->object; - if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1; - - return 0; -} - -void jackShutdown( void *infoPointer ) -{ - CallbackInfo *info = (CallbackInfo *) infoPointer; - RtApiJack *object = (RtApiJack *) info->object; - - // Check current stream state. If stopped, then we'll assume this - // was called as a result of a call to RtApiJack::stopStream (the - // deactivation of a client handle causes this function to be called). - // If not, we'll assume the Jack server is shutting down or some - // other problem occurred and we should close the stream. - if ( object->isStreamRunning() == false ) return; - - object->closeStream(); - std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl; -} - -int jackXrun( void *infoPointer ) -{ - JackHandle *handle = (JackHandle *) infoPointer; - - if ( handle->ports[0] ) handle->xrun[0] = true; - if ( handle->ports[1] ) handle->xrun[1] = true; - - return 0; -} - -bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - JackHandle *handle = (JackHandle *) stream_.apiHandle; - - // Look for jack server and try to become a client (only do once per stream). - jack_client_t *client = 0; - if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) { - jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption; - jack_status_t *status = NULL; - if ( options && !options->streamName.empty() ) - client = jack_client_open( options->streamName.c_str(), jackoptions, status ); - else - client = jack_client_open( "RtApiJack", jackoptions, status ); - if ( client == 0 ) { - errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!"; - error( RtError::WARNING ); - return FAILURE; - } - } - else { - // The handle must have been created on an earlier pass. - client = handle->client; - } - - const char **ports; - std::string port, previousPort, deviceName; - unsigned int nPorts = 0, nDevices = 0; - ports = jack_get_ports( client, NULL, NULL, 0 ); - if ( ports ) { - // Parse the port names up to the first colon (:). - size_t iColon = 0; - do { - port = (char *) ports[ nPorts ]; - iColon = port.find(":"); - if ( iColon != std::string::npos ) { - port = port.substr( 0, iColon ); - if ( port != previousPort ) { - if ( nDevices == device ) deviceName = port; - nDevices++; - previousPort = port; - } - } - } while ( ports[++nPorts] ); - free( ports ); - } - - if ( device >= nDevices ) { - errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } - - // Count the available ports containing the client name as device - // channels. Jack "input ports" equal RtAudio output channels. - unsigned int nChannels = 0; - unsigned long flag = JackPortIsInput; - if ( mode == INPUT ) flag = JackPortIsOutput; - ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); - if ( ports ) { - while ( ports[ nChannels ] ) nChannels++; - free( ports ); - } - - // Compare the jack ports for specified client to the requested number of channels. - if ( nChannels < (channels + firstChannel) ) { - errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Check the jack server sample rate. - unsigned int jackRate = jack_get_sample_rate( client ); - if ( sampleRate != jackRate ) { - jack_client_close( client ); - errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.sampleRate = jackRate; - - // Get the latency of the JACK port. - ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); - if ( ports[ firstChannel ] ) - stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) ); - free( ports ); - - // The jack server always uses 32-bit floating-point data. - stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - stream_.userFormat = format; - - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; - - // Jack always uses non-interleaved buffers. - stream_.deviceInterleaved[mode] = false; - - // Jack always provides host byte-ordered data. - stream_.doByteSwap[mode] = false; - - // Get the buffer size. The buffer size and number of buffers - // (periods) is set when the jack server is started. - stream_.bufferSize = (int) jack_get_buffer_size( client ); - *bufferSize = stream_.bufferSize; - - stream_.nDeviceChannels[mode] = channels; - stream_.nUserChannels[mode] = channels; - - // Set flags for buffer conversion. - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - // Allocate our JackHandle structure for the stream. - if ( handle == 0 ) { - try { - handle = new JackHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory."; - goto error; - } - - if ( pthread_cond_init(&handle->condition, NULL) ) { - errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable."; - goto error; - } - stream_.apiHandle = (void *) handle; - handle->client = client; - } - handle->deviceName[mode] = deviceName; - - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - if ( stream_.doConvertBuffer[mode] ) { - - bool makeBuffer = true; - if ( mode == OUTPUT ) - bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - else { // mode == INPUT - bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] ); - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]); - if ( bufferBytes < bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - // Allocate memory for the Jack ports (channels) identifiers. - handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels ); - if ( handle->ports[mode] == NULL ) { - errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory."; - goto error; - } - - stream_.device[mode] = device; - stream_.channelOffset[mode] = firstChannel; - stream_.state = STREAM_STOPPED; - stream_.callbackInfo.object = (void *) this; - - if ( stream_.mode == OUTPUT && mode == INPUT ) - // We had already set up the stream for output. - stream_.mode = DUPLEX; - else { - stream_.mode = mode; - jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo ); - jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle ); - jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo ); - } - - // Register our ports. - char label[64]; - if ( mode == OUTPUT ) { - for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) { - snprintf( label, 64, "outport %d", i ); - handle->ports[0][i] = jack_port_register( handle->client, (const char *)label, - JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 ); - } - } - else { - for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) { - snprintf( label, 64, "inport %d", i ); - handle->ports[1][i] = jack_port_register( handle->client, (const char *)label, - JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 ); - } - } - - // Setup the buffer conversion information structure. We don't use - // buffers to do channel offsets, so we override that parameter - // here. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); - - return SUCCESS; - - error: - if ( handle ) { - pthread_cond_destroy( &handle->condition ); - jack_client_close( handle->client ); - - if ( handle->ports[0] ) free( handle->ports[0] ); - if ( handle->ports[1] ) free( handle->ports[1] ); - - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - return FAILURE; -} - -void RtApiJack :: closeStream( void ) -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiJack::closeStream(): no open stream to close!"; - error( RtError::WARNING ); - return; - } - - JackHandle *handle = (JackHandle *) stream_.apiHandle; - if ( handle ) { - - if ( stream_.state == STREAM_RUNNING ) - jack_deactivate( handle->client ); - - jack_client_close( handle->client ); - } - - if ( handle ) { - if ( handle->ports[0] ) free( handle->ports[0] ); - if ( handle->ports[1] ) free( handle->ports[1] ); - pthread_cond_destroy( &handle->condition ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} - -void RtApiJack :: startStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiJack::startStream(): the stream is already running!"; - error( RtError::WARNING ); - return; - } - - MUTEX_LOCK(&stream_.mutex); - - JackHandle *handle = (JackHandle *) stream_.apiHandle; - int result = jack_activate( handle->client ); - if ( result ) { - errorText_ = "RtApiJack::startStream(): unable to activate JACK client!"; - goto unlock; - } - - const char **ports; - - // Get the list of available ports. - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - result = 1; - ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput); - if ( ports == NULL) { - errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!"; - goto unlock; - } - - // Now make the port connections. Since RtAudio wasn't designed to - // allow the user to select particular channels of a device, we'll - // just open the first "nChannels" ports with offset. - for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) { - result = 1; - if ( ports[ stream_.channelOffset[0] + i ] ) - result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] ); - if ( result ) { - free( ports ); - errorText_ = "RtApiJack::startStream(): error connecting output ports!"; - goto unlock; - } - } - free(ports); - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - result = 1; - ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput ); - if ( ports == NULL) { - errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!"; - goto unlock; - } - - // Now make the port connections. See note above. - for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) { - result = 1; - if ( ports[ stream_.channelOffset[1] + i ] ) - result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) ); - if ( result ) { - free( ports ); - errorText_ = "RtApiJack::startStream(): error connecting input ports!"; - goto unlock; - } - } - free(ports); - } - - handle->drainCounter = 0; - handle->internalDrain = false; - stream_.state = STREAM_RUNNING; - - unlock: - MUTEX_UNLOCK(&stream_.mutex); - - if ( result == 0 ) return; - error( RtError::SYSTEM_ERROR ); -} - -void RtApiJack :: stopStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiJack::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - - JackHandle *handle = (JackHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 1; - pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled - } - } - - jack_deactivate( handle->client ); - stream_.state = STREAM_STOPPED; - - MUTEX_UNLOCK( &stream_.mutex ); -} - -void RtApiJack :: abortStream( void ) -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiJack::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } - - JackHandle *handle = (JackHandle *) stream_.apiHandle; - handle->drainCounter = 1; - - stopStream(); -} - -// This function will be called by a spawned thread when the user -// callback function signals that the stream should be stopped or -// aborted. It is necessary to handle it this way because the -// callbackEvent() function must return before the jack_deactivate() -// function will return. -extern "C" void *jackStopStream( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiJack *object = (RtApiJack *) info->object; - - object->stopStream(); - - pthread_exit( NULL ); -} - -bool RtApiJack :: callbackEvent( unsigned long nframes ) -{ - if ( stream_.state == STREAM_STOPPED ) return SUCCESS; - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); - return FAILURE; - } - if ( stream_.bufferSize != nframes ) { - errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!"; - error( RtError::WARNING ); - return FAILURE; - } - - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - JackHandle *handle = (JackHandle *) stream_.apiHandle; - - // Check if we were draining the stream and signal is finished. - if ( handle->drainCounter > 3 ) { - if ( handle->internalDrain == true ) { - ThreadHandle id; - pthread_create( &id, NULL, jackStopStream, info ); - } - else - pthread_cond_signal( &handle->condition ); - return SUCCESS; - } - - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return SUCCESS; - } - - // Invoke user callback first, to get fresh output data. - if ( handle->drainCounter == 0 ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; - } - handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( handle->drainCounter == 2 ) { - MUTEX_UNLOCK( &stream_.mutex ); - ThreadHandle id; - pthread_create( &id, NULL, jackStopStream, info ); - return SUCCESS; - } - else if ( handle->drainCounter == 1 ) - handle->internalDrain = true; - } - - jack_default_audio_sample_t *jackbuffer; - unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t ); - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - if ( handle->drainCounter > 0 ) { // write zeros to the output stream - - for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) { - jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes ); - memset( jackbuffer, 0, bufferBytes ); - } - - } - else if ( stream_.doConvertBuffer[0] ) { - - convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - - for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) { - jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes ); - memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes ); - } - } - else { // no buffer conversion - for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) { - jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes ); - memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes ); - } - } - - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - if ( stream_.doConvertBuffer[1] ) { - for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) { - jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes ); - memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes ); - } - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - } - else { // no buffer conversion - for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) { - jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes ); - memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes ); - } - } - } - - unlock: - MUTEX_UNLOCK(&stream_.mutex); - - RtApi::tickStreamTime(); - return SUCCESS; -} - //******************** End of __UNIX_JACK__ *********************// -#endif - -#if defined(__WINDOWS_ASIO__) // ASIO API on Windows - -// The ASIO API is designed around a callback scheme, so this -// implementation is similar to that used for OS-X CoreAudio and Linux -// Jack. The primary constraint with ASIO is that it only allows -// access to a single driver at a time. Thus, it is not possible to -// have more than one simultaneous RtAudio stream. -// -// This implementation also requires a number of external ASIO files -// and a few global variables. The ASIO callback scheme does not -// allow for the passing of user data, so we must create a global -// pointer to our callbackInfo structure. -// -// On unix systems, we make use of a pthread condition variable. -// Since there is no equivalent in Windows, I hacked something based -// on information found in -// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html. - -#include "asiosys.h" -#include "asio.h" -#include "iasiothiscallresolver.h" -#include "asiodrivers.h" -#include <cmath> - -AsioDrivers drivers; -ASIOCallbacks asioCallbacks; -ASIODriverInfo driverInfo; -CallbackInfo *asioCallbackInfo; -bool asioXRun; - -struct AsioHandle { - int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - ASIOBufferInfo *bufferInfos; - HANDLE condition; - - AsioHandle() - :drainCounter(0), internalDrain(false), bufferInfos(0) {} -}; - -// Function declarations (definitions at end of section) -static const char* getAsioErrorString( ASIOError result ); -void sampleRateChanged( ASIOSampleRate sRate ); -long asioMessages( long selector, long value, void* message, double* opt ); - -RtApiAsio :: RtApiAsio() -{ - // ASIO cannot run on a multi-threaded appartment. You can call - // CoInitialize beforehand, but it must be for appartment threading - // (in which case, CoInitilialize will return S_FALSE here). - coInitialized_ = false; - HRESULT hr = CoInitialize( NULL ); - if ( FAILED(hr) ) { - errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"; - error( RtError::WARNING ); - } - coInitialized_ = true; - - drivers.removeCurrentDriver(); - driverInfo.asioVersion = 2; - - // See note in DirectSound implementation about GetDesktopWindow(). - driverInfo.sysRef = GetForegroundWindow(); -} - -RtApiAsio :: ~RtApiAsio() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); - if ( coInitialized_ ) CoUninitialize(); -} - -unsigned int RtApiAsio :: getDeviceCount( void ) -{ - return (unsigned int) drivers.asioGetNumDev(); -} - -RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; - - // Get device ID - unsigned int nDevices = getDeviceCount(); - if ( nDevices == 0 ) { - errorText_ = "RtApiAsio::getDeviceInfo: no devices found!"; - error( RtError::INVALID_USE ); - } - - if ( device >= nDevices ) { - errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - } - - // If a stream is already open, we cannot probe other devices. Thus, use the saved results. - if ( stream_.state != STREAM_CLOSED ) { - if ( device >= devices_.size() ) { - errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened."; - error( RtError::WARNING ); - return info; - } - return devices_[ device ]; - } - - char driverName[32]; - ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - info.name = driverName; - - if ( !drivers.loadDriver( driverName ) ) { - errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - result = ASIOInit( &driverInfo ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - // Determine the device channel information. - long inputChannels, outputChannels; - result = ASIOGetChannels( &inputChannels, &outputChannels ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - info.outputChannels = outputChannels; - info.inputChannels = inputChannels; - if ( info.outputChannels > 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - - // Determine the supported sample rates. - info.sampleRates.clear(); - for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) { - result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] ); - if ( result == ASE_OK ) - info.sampleRates.push_back( SAMPLE_RATES[i] ); - } - - // Determine supported data types ... just check first channel and assume rest are the same. - ASIOChannelInfo channelInfo; - channelInfo.channel = 0; - channelInfo.isInput = true; - if ( info.inputChannels <= 0 ) channelInfo.isInput = false; - result = ASIOGetChannelInfo( &channelInfo ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - info.nativeFormats = 0; - if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) - info.nativeFormats |= RTAUDIO_SINT16; - else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) - info.nativeFormats |= RTAUDIO_SINT32; - else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) - info.nativeFormats |= RTAUDIO_FLOAT32; - else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) - info.nativeFormats |= RTAUDIO_FLOAT64; - - if ( getDefaultOutputDevice() == device ) - info.isDefaultOutput = true; - if ( getDefaultInputDevice() == device ) - info.isDefaultInput = true; - - info.probed = true; - drivers.removeCurrentDriver(); - return info; -} - -void bufferSwitch( long index, ASIOBool processNow ) -{ - RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object; - object->callbackEvent( index ); -} - -void RtApiAsio :: saveDeviceInfo( void ) -{ - devices_.clear(); - - unsigned int nDevices = getDeviceCount(); - devices_.resize( nDevices ); - for ( unsigned int i=0; i<nDevices; i++ ) - devices_[i] = getDeviceInfo( i ); -} - -bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - // For ASIO, a duplex stream MUST use the same driver. - if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) { - errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!"; - return FAILURE; - } - - char driverName[32]; - ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // The getDeviceInfo() function will not work when a stream is open - // because ASIO does not allow multiple devices to run at the same - // time. Thus, we'll probe the system before opening a stream and - // save the results for use by getDeviceInfo(). - this->saveDeviceInfo(); - - // Only load the driver once for duplex stream. - if ( mode != INPUT || stream_.mode != OUTPUT ) { - if ( !drivers.loadDriver( driverName ) ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - result = ASIOInit( &driverInfo ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // Check the device channel count. - long inputChannels, outputChannels; - result = ASIOGetChannels( &inputChannels, &outputChannels ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) || - ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.nDeviceChannels[mode] = channels; - stream_.nUserChannels[mode] = channels; - stream_.channelOffset[mode] = firstChannel; - - // Verify the sample rate is supported. - result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Get the current sample rate - ASIOSampleRate currentRate; - result = ASIOGetSampleRate( ¤tRate ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the sample rate only if necessary - if ( currentRate != sampleRate ) { - result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // Determine the driver data type. - ASIOChannelInfo channelInfo; - channelInfo.channel = 0; - if ( mode == OUTPUT ) channelInfo.isInput = false; - else channelInfo.isInput = true; - result = ASIOGetChannelInfo( &channelInfo ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Assuming WINDOWS host is always little-endian. - stream_.doByteSwap[mode] = false; - stream_.userFormat = format; - stream_.deviceFormat[mode] = 0; - if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true; - } - else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true; - } - else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true; - } - else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; - if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true; - } - - if ( stream_.deviceFormat[mode] == 0 ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the buffer size. For a duplex stream, this will end up - // setting the buffer size based on the input constraints, which - // should be ok. - long minSize, maxSize, preferSize, granularity; - result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity ); - if ( result != ASE_OK ) { - drivers.removeCurrentDriver(); - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; - else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; - else if ( granularity == -1 ) { - // Make sure bufferSize is a power of two. - int log2_of_min_size = 0; - int log2_of_max_size = 0; - - for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) { - if ( minSize & ((long)1 << i) ) log2_of_min_size = i; - if ( maxSize & ((long)1 << i) ) log2_of_max_size = i; - } - - long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) ); - int min_delta_num = log2_of_min_size; - - for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) { - long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) ); - if (current_delta < min_delta) { - min_delta = current_delta; - min_delta_num = i; - } - } - - *bufferSize = ( (unsigned int)1 << min_delta_num ); - if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize; - else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize; - } - else if ( granularity != 0 ) { - // Set to an even multiple of granularity, rounding up. - *bufferSize = (*bufferSize + granularity-1) / granularity * granularity; - } - - if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) { - drivers.removeCurrentDriver(); - errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!"; - return FAILURE; - } - - stream_.bufferSize = *bufferSize; - stream_.nBuffers = 2; - - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; - - // ASIO always uses non-interleaved buffers. - stream_.deviceInterleaved[mode] = false; - - // Allocate, if necessary, our AsioHandle structure for the stream. - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - if ( handle == 0 ) { - try { - handle = new AsioHandle; - } - catch ( std::bad_alloc& ) { - //if ( handle == NULL ) { - drivers.removeCurrentDriver(); - errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory."; - return FAILURE; - } - handle->bufferInfos = 0; - - // Create a manual-reset event. - handle->condition = CreateEvent( NULL, // no security - TRUE, // manual-reset - FALSE, // non-signaled initially - NULL ); // unnamed - stream_.apiHandle = (void *) handle; - } - - // Create the ASIO internal buffers. Since RtAudio sets up input - // and output separately, we'll have to dispose of previously - // created output buffers for a duplex stream. - long inputLatency, outputLatency; - if ( mode == INPUT && stream_.mode == OUTPUT ) { - ASIODisposeBuffers(); - if ( handle->bufferInfos ) free( handle->bufferInfos ); - } - - // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. - bool buffersAllocated = false; - unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; - handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) ); - if ( handle->bufferInfos == NULL ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ")."; - errorText_ = errorStream_.str(); - goto error; - } - - ASIOBufferInfo *infos; - infos = handle->bufferInfos; - for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) { - infos->isInput = ASIOFalse; - infos->channelNum = i + stream_.channelOffset[0]; - infos->buffers[0] = infos->buffers[1] = 0; - } - for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) { - infos->isInput = ASIOTrue; - infos->channelNum = i + stream_.channelOffset[1]; - infos->buffers[0] = infos->buffers[1] = 0; - } - - // Set up the ASIO callback structure and create the ASIO data buffers. - asioCallbacks.bufferSwitch = &bufferSwitch; - asioCallbacks.sampleRateDidChange = &sampleRateChanged; - asioCallbacks.asioMessage = &asioMessages; - asioCallbacks.bufferSwitchTimeInfo = NULL; - result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers."; - errorText_ = errorStream_.str(); - goto error; - } - buffersAllocated = true; - - // Set flags for buffer conversion. - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - // Allocate necessary internal buffers - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - if ( stream_.doConvertBuffer[mode] ) { - - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - stream_.sampleRate = sampleRate; - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - asioCallbackInfo = &stream_.callbackInfo; - stream_.callbackInfo.object = (void *) this; - if ( stream_.mode == OUTPUT && mode == INPUT ) - // We had already set up an output stream. - stream_.mode = DUPLEX; - else - stream_.mode = mode; - - // Determine device latencies - result = ASIOGetLatencies( &inputLatency, &outputLatency ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency."; - errorText_ = errorStream_.str(); - error( RtError::WARNING); // warn but don't fail - } - else { - stream_.latency[0] = outputLatency; - stream_.latency[1] = inputLatency; - } - - // Setup the buffer conversion information structure. We don't use - // buffers to do channel offsets, so we override that parameter - // here. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 ); - - return SUCCESS; - - error: - if ( buffersAllocated ) - ASIODisposeBuffers(); - drivers.removeCurrentDriver(); - - if ( handle ) { - CloseHandle( handle->condition ); - if ( handle->bufferInfos ) - free( handle->bufferInfos ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - return FAILURE; -} - -void RtApiAsio :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAsio::closeStream(): no open stream to close!"; - error( RtError::WARNING ); - return; - } - - if ( stream_.state == STREAM_RUNNING ) { - stream_.state = STREAM_STOPPED; - ASIOStop(); - } - ASIODisposeBuffers(); - drivers.removeCurrentDriver(); - - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - if ( handle ) { - CloseHandle( handle->condition ); - if ( handle->bufferInfos ) - free( handle->bufferInfos ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} - -void RtApiAsio :: startStream() -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiAsio::startStream(): the stream is already running!"; - error( RtError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - ASIOError result = ASIOStart(); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device."; - errorText_ = errorStream_.str(); - goto unlock; - } - - handle->drainCounter = 0; - handle->internalDrain = false; - stream_.state = STREAM_RUNNING; - asioXRun = false; - - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result == ASE_OK ) return; - error( RtError::SYSTEM_ERROR ); -} - -void RtApiAsio :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 1; - MUTEX_UNLOCK( &stream_.mutex ); - WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled - ResetEvent( handle->condition ); - MUTEX_LOCK( &stream_.mutex ); - } - } - - ASIOError result = ASIOStop(); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device."; - errorText_ = errorStream_.str(); - } - - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result == ASE_OK ) return; - error( RtError::SYSTEM_ERROR ); -} - -void RtApiAsio :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } - - // The following lines were commented-out because some behavior was - // noted where the device buffers need to be zeroed to avoid - // continuing sound, even when the device buffers are completely - // disposed. So now, calling abort is the same as calling stop. - // AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - // handle->drainCounter = 1; - stopStream(); -} - -bool RtApiAsio :: callbackEvent( long bufferIndex ) -{ - if ( stream_.state == STREAM_STOPPED ) return SUCCESS; - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); - return FAILURE; - } - - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - - // Check if we were draining the stream and signal is finished. - if ( handle->drainCounter > 3 ) { - if ( handle->internalDrain == false ) - SetEvent( handle->condition ); - else - stopStream(); - return SUCCESS; - } - - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) goto unlock; - - // Invoke user callback to get fresh output data UNLESS we are - // draining stream. - if ( handle->drainCounter == 0 ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && asioXRun == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - asioXRun = false; - } - if ( stream_.mode != OUTPUT && asioXRun == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - asioXRun = false; - } - handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( handle->drainCounter == 2 ) { - MUTEX_UNLOCK( &stream_.mutex ); - abortStream(); - return SUCCESS; - } - else if ( handle->drainCounter == 1 ) - handle->internalDrain = true; - } - - unsigned int nChannels, bufferBytes, i, j; - nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1]; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] ); - - if ( handle->drainCounter > 1 ) { // write zeros to the output stream - - for ( i=0, j=0; i<nChannels; i++ ) { - if ( handle->bufferInfos[i].isInput != ASIOTrue ) - memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes ); - } - - } - else if ( stream_.doConvertBuffer[0] ) { - - convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - if ( stream_.doByteSwap[0] ) - byteSwapBuffer( stream_.deviceBuffer, - stream_.bufferSize * stream_.nDeviceChannels[0], - stream_.deviceFormat[0] ); - - for ( i=0, j=0; i<nChannels; i++ ) { - if ( handle->bufferInfos[i].isInput != ASIOTrue ) - memcpy( handle->bufferInfos[i].buffers[bufferIndex], - &stream_.deviceBuffer[j++*bufferBytes], bufferBytes ); - } - - } - else { - - if ( stream_.doByteSwap[0] ) - byteSwapBuffer( stream_.userBuffer[0], - stream_.bufferSize * stream_.nUserChannels[0], - stream_.userFormat ); - - for ( i=0, j=0; i<nChannels; i++ ) { - if ( handle->bufferInfos[i].isInput != ASIOTrue ) - memcpy( handle->bufferInfos[i].buffers[bufferIndex], - &stream_.userBuffer[0][bufferBytes*j++], bufferBytes ); - } - - } - - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]); - - if (stream_.doConvertBuffer[1]) { - - // Always interleave ASIO input data. - for ( i=0, j=0; i<nChannels; i++ ) { - if ( handle->bufferInfos[i].isInput == ASIOTrue ) - memcpy( &stream_.deviceBuffer[j++*bufferBytes], - handle->bufferInfos[i].buffers[bufferIndex], - bufferBytes ); - } - - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( stream_.deviceBuffer, - stream_.bufferSize * stream_.nDeviceChannels[1], - stream_.deviceFormat[1] ); - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - - } - else { - for ( i=0, j=0; i<nChannels; i++ ) { - if ( handle->bufferInfos[i].isInput == ASIOTrue ) { - memcpy( &stream_.userBuffer[1][bufferBytes*j++], - handle->bufferInfos[i].buffers[bufferIndex], - bufferBytes ); - } - } - - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( stream_.userBuffer[1], - stream_.bufferSize * stream_.nUserChannels[1], - stream_.userFormat ); - } - } - - unlock: - // The following call was suggested by Malte Clasen. While the API - // documentation indicates it should not be required, some device - // drivers apparently do not function correctly without it. - ASIOOutputReady(); - - MUTEX_UNLOCK( &stream_.mutex ); - - RtApi::tickStreamTime(); - return SUCCESS; -} - -void sampleRateChanged( ASIOSampleRate sRate ) -{ - // The ASIO documentation says that this usually only happens during - // external sync. Audio processing is not stopped by the driver, - // actual sample rate might not have even changed, maybe only the - // sample rate status of an AES/EBU or S/PDIF digital input at the - // audio device. - - RtApi *object = (RtApi *) asioCallbackInfo->object; - try { - object->stopStream(); - } - catch ( RtError &exception ) { - std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl; - return; - } - - std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl; -} - -long asioMessages( long selector, long value, void* message, double* opt ) -{ - long ret = 0; - - switch( selector ) { - case kAsioSelectorSupported: - if ( value == kAsioResetRequest - || value == kAsioEngineVersion - || value == kAsioResyncRequest - || value == kAsioLatenciesChanged - // The following three were added for ASIO 2.0, you don't - // necessarily have to support them. - || value == kAsioSupportsTimeInfo - || value == kAsioSupportsTimeCode - || value == kAsioSupportsInputMonitor) - ret = 1L; - break; - case kAsioResetRequest: - // Defer the task and perform the reset of the driver during the - // next "safe" situation. You cannot reset the driver right now, - // as this code is called from the driver. Reset the driver is - // done by completely destruct is. I.e. ASIOStop(), - // ASIODisposeBuffers(), Destruction Afterwards you initialize the - // driver again. - std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl; - ret = 1L; - break; - case kAsioResyncRequest: - // This informs the application that the driver encountered some - // non-fatal data loss. It is used for synchronization purposes - // of different media. Added mainly to work around the Win16Mutex - // problems in Windows 95/98 with the Windows Multimedia system, - // which could lose data because the Mutex was held too long by - // another thread. However a driver can issue it in other - // situations, too. - // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl; - asioXRun = true; - ret = 1L; - break; - case kAsioLatenciesChanged: - // This will inform the host application that the drivers were - // latencies changed. Beware, it this does not mean that the - // buffer sizes have changed! You might need to update internal - // delay data. - std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl; - ret = 1L; - break; - case kAsioEngineVersion: - // Return the supported ASIO version of the host application. If - // a host application does not implement this selector, ASIO 1.0 - // is assumed by the driver. - ret = 2L; - break; - case kAsioSupportsTimeInfo: - // Informs the driver whether the - // asioCallbacks.bufferSwitchTimeInfo() callback is supported. - // For compatibility with ASIO 1.0 drivers the host application - // should always support the "old" bufferSwitch method, too. - ret = 0; - break; - case kAsioSupportsTimeCode: - // Informs the driver whether application is interested in time - // code info. If an application does not need to know about time - // code, the driver has less work to do. - ret = 0; - break; - } - return ret; -} - -static const char* getAsioErrorString( ASIOError result ) -{ - struct Messages - { - ASIOError value; - const char*message; - }; - - static Messages m[] = - { - { ASE_NotPresent, "Hardware input or output is not present or available." }, - { ASE_HWMalfunction, "Hardware is malfunctioning." }, - { ASE_InvalidParameter, "Invalid input parameter." }, - { ASE_InvalidMode, "Invalid mode." }, - { ASE_SPNotAdvancing, "Sample position not advancing." }, - { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." }, - { ASE_NoMemory, "Not enough memory to complete the request." } - }; - - for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i ) - if ( m[i].value == result ) return m[i].message; - - return "Unknown error."; -} -//******************** End of __WINDOWS_ASIO__ *********************// -#endif - - -#if defined(__WINDOWS_DS__) // Windows DirectSound API - -// Modified by Robin Davies, October 2005 -// - Improvements to DirectX pointer chasing. -// - Backdoor RtDsStatistics hook provides DirectX performance information. -// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30. -// - Auto-call CoInitialize for DSOUND and ASIO platforms. -// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007 - -#include <dsound.h> -#include <assert.h> - -#if defined(__MINGW32__) - // missing from latest mingw winapi -#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */ -#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */ -#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */ -#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */ -#endif - -#define MINIMUM_DEVICE_BUFFER_SIZE 32768 - -#ifdef _MSC_VER // if Microsoft Visual C++ -#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually. -#endif - -static inline DWORD dsPointerDifference( DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) -{ - if ( laterPointer > earlierPointer ) - return laterPointer - earlierPointer; - else - return laterPointer - earlierPointer + bufferSize; -} - -static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize ) -{ - if ( pointer > bufferSize ) pointer -= bufferSize; - if ( laterPointer < earlierPointer ) laterPointer += bufferSize; - if ( pointer < earlierPointer ) pointer += bufferSize; - return pointer >= earlierPointer && pointer < laterPointer; -} - -// A structure to hold various information related to the DirectSound -// API implementation. -struct DsHandle { - unsigned int drainCounter; // Tracks callback counts when draining - bool internalDrain; // Indicates if stop is initiated from callback or not. - void *id[2]; - void *buffer[2]; - bool xrun[2]; - UINT bufferPointer[2]; - DWORD dsBufferSize[2]; - DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by. - HANDLE condition; - - DsHandle() - :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; } -}; - -/* -RtApiDs::RtDsStatistics RtApiDs::statistics; - -// Provides a backdoor hook to monitor for DirectSound read overruns and write underruns. -RtApiDs::RtDsStatistics RtApiDs::getDsStatistics() -{ - RtDsStatistics s = statistics; - - // update the calculated fields. - if ( s.inputFrameSize != 0 ) - s.latency += s.readDeviceSafeLeadBytes * 1.0 / s.inputFrameSize / s.sampleRate; - - if ( s.outputFrameSize != 0 ) - s.latency += (s.writeDeviceSafeLeadBytes + s.writeDeviceBufferLeadBytes) * 1.0 / s.outputFrameSize / s.sampleRate; - - return s; -} -*/ - -// Declarations for utility functions, callbacks, and structures -// specific to the DirectSound implementation. -static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, - LPCTSTR description, - LPCTSTR module, - LPVOID lpContext ); - -static char* getErrorString( int code ); - -extern "C" unsigned __stdcall callbackHandler( void *ptr ); - -struct EnumInfo { - bool isInput; - bool getDefault; - bool findIndex; - unsigned int counter; - unsigned int index; - LPGUID id; - std::string name; - - EnumInfo() - : isInput(false), getDefault(false), findIndex(false), counter(0), index(0) {} -}; - -RtApiDs :: RtApiDs() -{ - // Dsound will run both-threaded. If CoInitialize fails, then just - // accept whatever the mainline chose for a threading model. - coInitialized_ = false; - HRESULT hr = CoInitialize( NULL ); - if ( !FAILED( hr ) ) coInitialized_ = true; -} - -RtApiDs :: ~RtApiDs() -{ - if ( coInitialized_ ) CoUninitialize(); // balanced call. - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} - -unsigned int RtApiDs :: getDefaultInputDevice( void ) -{ - // Count output devices. - EnumInfo info; - HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") counting output devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return 0; - } - - // Now enumerate input devices until we find the id = NULL. - info.isInput = true; - info.getDefault = true; - result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDefaultInputDevice: error (" << getErrorString( result ) << ") enumerating input devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return 0; - } - - if ( info.counter > 0 ) return info.counter - 1; - return 0; -} - -unsigned int RtApiDs :: getDefaultOutputDevice( void ) -{ - // Enumerate output devices until we find the id = NULL. - EnumInfo info; - info.getDefault = true; - HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") enumerating output devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return 0; - } - - if ( info.counter > 0 ) return info.counter - 1; - return 0; -} - -unsigned int RtApiDs :: getDeviceCount( void ) -{ - // Count DirectSound devices. - EnumInfo info; - HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } - - // Count DirectSoundCapture devices. - info.isInput = true; - result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } - - return info.counter; -} - -RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device ) -{ - // Because DirectSound always enumerates input and output devices - // separately (and because we don't attempt to combine devices - // internally), none of our "devices" will ever be duplex. - - RtAudio::DeviceInfo info; - info.probed = false; - - // Enumerate through devices to find the id (if it exists). Note - // that we have to do the output enumeration first, even if this is - // an input device, in order for the device counter to be correct. - EnumInfo dsinfo; - dsinfo.findIndex = true; - dsinfo.index = device; - HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating output devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } - - if ( dsinfo.name.empty() ) goto probeInput; - - LPDIRECTSOUND output; - DSCAPS outCaps; - result = DirectSoundCreate( dsinfo.id, &output, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - outCaps.dwSize = sizeof( outCaps ); - result = output->GetCaps( &outCaps ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - // Get output channel information. - info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1; - - // Get sample rate information. - info.sampleRates.clear(); - for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) { - if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate && - SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) - info.sampleRates.push_back( SAMPLE_RATES[k] ); - } - - // Get format information. - if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16; - if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8; - - output->Release(); - - if ( getDefaultOutputDevice() == device ) - info.isDefaultOutput = true; - - // Copy name and return. - info.name = dsinfo.name; - - info.probed = true; - return info; - - probeInput: - - dsinfo.isInput = true; - result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating input devices!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } - - if ( dsinfo.name.empty() ) return info; - - LPDIRECTSOUNDCAPTURE input; - result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - DSCCAPS inCaps; - inCaps.dwSize = sizeof( inCaps ); - result = input->GetCaps( &inCaps ); - if ( FAILED( result ) ) { - input->Release(); - errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - // Get input channel information. - info.inputChannels = inCaps.dwChannels; - - // Get sample rate and format information. - if ( inCaps.dwChannels == 2 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8; - - if ( info.nativeFormats & RTAUDIO_SINT16 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.sampleRates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.sampleRates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.sampleRates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.sampleRates.push_back( 96000 ); - } - else if ( info.nativeFormats & RTAUDIO_SINT8 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.sampleRates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.sampleRates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.sampleRates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.sampleRates.push_back( 44100 ); - } - } - else if ( inCaps.dwChannels == 1 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16; - if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8; - if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8; - - if ( info.nativeFormats & RTAUDIO_SINT16 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.sampleRates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.sampleRates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.sampleRates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.sampleRates.push_back( 96000 ); - } - else if ( info.nativeFormats & RTAUDIO_SINT8 ) { - if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.sampleRates.push_back( 11025 ); - if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.sampleRates.push_back( 22050 ); - if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.sampleRates.push_back( 44100 ); - if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.sampleRates.push_back( 96000 ); - } - } - else info.inputChannels = 0; // technically, this would be an error - - input->Release(); - - if ( info.inputChannels == 0 ) return info; - - if ( getDefaultInputDevice() == device ) - info.isDefaultInput = true; - - // Copy name and return. - info.name = dsinfo.name; - info.probed = true; - return info; -} - -bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - if ( channels + firstChannel > 2 ) { - errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device."; - return FAILURE; - } - - // Enumerate through devices to find the id (if it exists). Note - // that we have to do the output enumeration first, even if this is - // an input device, in order for the device counter to be correct. - EnumInfo dsinfo; - dsinfo.findIndex = true; - dsinfo.index = device; - HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating output devices!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - if ( mode == OUTPUT ) { - if ( dsinfo.name.empty() ) { - errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - else { // mode == INPUT - dsinfo.isInput = true; - HRESULT result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating input devices!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - if ( dsinfo.name.empty() ) { - errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // According to a note in PortAudio, using GetDesktopWindow() - // instead of GetForegroundWindow() is supposed to avoid problems - // that occur when the application's window is not the foreground - // window. Also, if the application window closes before the - // DirectSound buffer, DirectSound can crash. However, for console - // applications, no sound was produced when using GetDesktopWindow(). - HWND hWnd = GetForegroundWindow(); - - // Check the numberOfBuffers parameter and limit the lowest value to - // two. This is a judgement call and a value of two is probably too - // low for capture, but it should work for playback. - int nBuffers = 0; - if ( options ) nBuffers = options->numberOfBuffers; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2; - if ( nBuffers < 2 ) nBuffers = 3; - - // Create the wave format structure. The data format setting will - // be determined later. - WAVEFORMATEX waveFormat; - ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) ); - waveFormat.wFormatTag = WAVE_FORMAT_PCM; - waveFormat.nChannels = channels + firstChannel; - waveFormat.nSamplesPerSec = (unsigned long) sampleRate; - - // Determine the device buffer size. By default, 32k, but we will - // grow it to make allowances for very large software buffer sizes. - DWORD dsBufferSize = 0; - DWORD dsPointerLeadTime = 0; - long bufferBytes = MINIMUM_DEVICE_BUFFER_SIZE; // sound cards will always *knock wood* support this - - void *ohandle = 0, *bhandle = 0; - if ( mode == OUTPUT ) { - - LPDIRECTSOUND output; - result = DirectSoundCreate( dsinfo.id, &output, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - DSCAPS outCaps; - outCaps.dwSize = sizeof( outCaps ); - result = output->GetCaps( &outCaps ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Check channel information. - if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) { - errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsinfo.name << ") does not support stereo playback."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Check format information. Use 16-bit format unless not - // supported or user requests 8-bit. - if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT && - !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) { - waveFormat.wBitsPerSample = 16; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - else { - waveFormat.wBitsPerSample = 8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - stream_.userFormat = format; - - // Update wave format structure and buffer information. - waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; - waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; - dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; - - // If the user wants an even bigger buffer, increase the device buffer size accordingly. - while ( dsPointerLeadTime * 2U > (DWORD) bufferBytes ) - bufferBytes *= 2; - - // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes. - // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE ); - // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes. - result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Even though we will write to the secondary buffer, we need to - // access the primary buffer to set the correct output format - // (since the default is 8-bit, 22 kHz!). Setup the DS primary - // buffer description. - DSBUFFERDESC bufferDescription; - ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); - bufferDescription.dwSize = sizeof( DSBUFFERDESC ); - bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER; - - // Obtain the primary buffer - LPDIRECTSOUNDBUFFER buffer; - result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the primary DS buffer sound format. - result = buffer->SetFormat( &waveFormat ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Setup the secondary DS buffer description. - dsBufferSize = (DWORD) bufferBytes; - ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) ); - bufferDescription.dwSize = sizeof( DSBUFFERDESC ); - bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | - DSBCAPS_GLOBALFOCUS | - DSBCAPS_GETCURRENTPOSITION2 | - DSBCAPS_LOCHARDWARE ); // Force hardware mixing - bufferDescription.dwBufferBytes = bufferBytes; - bufferDescription.lpwfxFormat = &waveFormat; - - // Try to create the secondary DS buffer. If that doesn't work, - // try to use software mixing. Otherwise, there's a problem. - result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS | - DSBCAPS_GLOBALFOCUS | - DSBCAPS_GETCURRENTPOSITION2 | - DSBCAPS_LOCSOFTWARE ); // Force software mixing - result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - output->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // Get the buffer size ... might be different from what we specified. - DSBCAPS dsbcaps; - dsbcaps.dwSize = sizeof( DSBCAPS ); - result = buffer->GetCaps( &dsbcaps ); - if ( FAILED( result ) ) { - output->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - bufferBytes = dsbcaps.dwBufferBytes; - - // Lock the DS buffer - LPVOID audioPtr; - DWORD dataLen; - result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - output->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Zero the DS buffer - ZeroMemory( audioPtr, dataLen ); - - // Unlock the DS buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - output->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - dsBufferSize = bufferBytes; - ohandle = (void *) output; - bhandle = (void *) buffer; - } - - if ( mode == INPUT ) { - - LPDIRECTSOUNDCAPTURE input; - result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - DSCCAPS inCaps; - inCaps.dwSize = sizeof( inCaps ); - result = input->GetCaps( &inCaps ); - if ( FAILED( result ) ) { - input->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Check channel information. - if ( inCaps.dwChannels < channels + firstChannel ) { - errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels."; - return FAILURE; - } - - // Check format information. Use 16-bit format unless user - // requests 8-bit. - DWORD deviceFormats; - if ( channels + firstChannel == 2 ) { - deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08; - if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { - waveFormat.wBitsPerSample = 8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - else { // assume 16-bit is supported - waveFormat.wBitsPerSample = 16; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - } - else { // channel == 1 - deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08; - if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) { - waveFormat.wBitsPerSample = 8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - else { // assume 16-bit is supported - waveFormat.wBitsPerSample = 16; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - } - stream_.userFormat = format; - - // Update wave format structure and buffer information. - waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8; - waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; - dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels; - - // If the user wants an even bigger buffer, increase the device buffer size accordingly. - while ( dsPointerLeadTime * 2U > (DWORD) bufferBytes ) - bufferBytes *= 2; - - // Setup the secondary DS buffer description. - dsBufferSize = bufferBytes; - DSCBUFFERDESC bufferDescription; - ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) ); - bufferDescription.dwSize = sizeof( DSCBUFFERDESC ); - bufferDescription.dwFlags = 0; - bufferDescription.dwReserved = 0; - bufferDescription.dwBufferBytes = bufferBytes; - bufferDescription.lpwfxFormat = &waveFormat; - - // Create the capture buffer. - LPDIRECTSOUNDCAPTUREBUFFER buffer; - result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL ); - if ( FAILED( result ) ) { - input->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Get the buffer size ... might be different from what we specified. - DSCBCAPS dscbcaps; - dscbcaps.dwSize = sizeof( DSCBCAPS ); - result = buffer->GetCaps( &dscbcaps ); - if ( FAILED( result ) ) { - input->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - bufferBytes = dscbcaps.dwBufferBytes; - - // Lock the capture buffer - LPVOID audioPtr; - DWORD dataLen; - result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - input->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Zero the buffer - ZeroMemory( audioPtr, dataLen ); - - // Unlock the buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - input->Release(); - buffer->Release(); - errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsinfo.name << ")!"; - errorText_ = errorStream_.str(); - return FAILURE; - } - - dsBufferSize = bufferBytes; - ohandle = (void *) input; - bhandle = (void *) buffer; - } - - // Set various stream parameters - DsHandle *handle = 0; - stream_.nDeviceChannels[mode] = channels + firstChannel; - stream_.nUserChannels[mode] = channels; - stream_.bufferSize = *bufferSize; - stream_.channelOffset[mode] = firstChannel; - stream_.deviceInterleaved[mode] = true; - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false; - else stream_.userInterleaved = true; - - // Set flag for buffer conversion - stream_.doConvertBuffer[mode] = false; - if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode]) - stream_.doConvertBuffer[mode] = true; - if (stream_.userFormat != stream_.deviceFormat[mode]) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - // Allocate necessary internal buffers - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - if ( stream_.doConvertBuffer[mode] ) { - - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= (long) bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - // Allocate our DsHandle structures for the stream. - if ( stream_.apiHandle == 0 ) { - try { - handle = new DsHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory."; - goto error; - } - - // Create a manual-reset event. - handle->condition = CreateEvent( NULL, // no security - TRUE, // manual-reset - FALSE, // non-signaled initially - NULL ); // unnamed - stream_.apiHandle = (void *) handle; - } - else - handle = (DsHandle *) stream_.apiHandle; - handle->id[mode] = ohandle; - handle->buffer[mode] = bhandle; - handle->dsBufferSize[mode] = dsBufferSize; - handle->dsPointerLeadTime[mode] = dsPointerLeadTime; - - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - if ( stream_.mode == OUTPUT && mode == INPUT ) - // We had already set up an output stream. - stream_.mode = DUPLEX; - else - stream_.mode = mode; - stream_.nBuffers = nBuffers; - stream_.sampleRate = sampleRate; - - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - - // Setup the callback thread. - unsigned threadId; - stream_.callbackInfo.object = (void *) this; - stream_.callbackInfo.isRunning = true; - stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler, - &stream_.callbackInfo, 0, &threadId ); - if ( stream_.callbackInfo.thread == 0 ) { - errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!"; - goto error; - } - - // Boost DS thread priority - SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST ); - return SUCCESS; - - error: - if ( handle ) { - if ( handle->buffer[0] ) { // the object pointer can be NULL and valid - LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - if ( buffer ) buffer->Release(); - object->Release(); - } - if ( handle->buffer[1] ) { - LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - if ( buffer ) buffer->Release(); - object->Release(); - } - CloseHandle( handle->condition ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - return FAILURE; -} - -void RtApiDs :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiDs::closeStream(): no open stream to close!"; - error( RtError::WARNING ); - return; - } - - // Stop the callback thread. - stream_.callbackInfo.isRunning = false; - WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE ); - CloseHandle( (HANDLE) stream_.callbackInfo.thread ); - - DsHandle *handle = (DsHandle *) stream_.apiHandle; - if ( handle ) { - if ( handle->buffer[0] ) { // the object pointer can be NULL and valid - LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0]; - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - if ( buffer ) { - buffer->Stop(); - buffer->Release(); - } - object->Release(); - } - if ( handle->buffer[1] ) { - LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1]; - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - if ( buffer ) { - buffer->Stop(); - buffer->Release(); - } - object->Release(); - } - CloseHandle( handle->condition ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} - -void RtApiDs :: startStream() -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiDs::startStream(): the stream is already running!"; - error( RtError::WARNING ); - return; - } - - // Increase scheduler frequency on lesser windows (a side-effect of - // increasing timer accuracy). On greater windows (Win2K or later), - // this is already in effect. - - MUTEX_LOCK( &stream_.mutex ); - - DsHandle *handle = (DsHandle *) stream_.apiHandle; - - timeBeginPeriod( 1 ); - - /* - memset( &statistics, 0, sizeof( statistics ) ); - statistics.sampleRate = stream_.sampleRate; - statistics.writeDeviceBufferLeadBytes = handle->dsPointerLeadTime[0]; - */ - - buffersRolling = false; - duplexPrerollBytes = 0; - - if ( stream_.mode == DUPLEX ) { - // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize. - duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] ); - } - - HRESULT result = 0; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - //statistics.outputFrameSize = formatBytes( stream_.deviceFormat[0] ) * stream_.nDeviceChannels[0]; - - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - result = buffer->Play( 0, 0, DSBPLAY_LOOPING ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - //statistics.inputFrameSize = formatBytes( stream_.deviceFormat[1]) * stream_.nDeviceChannels[1]; - - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - result = buffer->Start( DSCBSTART_LOOPING ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - handle->drainCounter = 0; - handle->internalDrain = false; - stream_.state = STREAM_RUNNING; - - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - - if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); -} - -void RtApiDs :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - - HRESULT result = 0; - LPVOID audioPtr; - DWORD dataLen; - DsHandle *handle = (DsHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( handle->drainCounter == 0 ) { - handle->drainCounter = 1; - MUTEX_UNLOCK( &stream_.mutex ); - WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled - ResetEvent( handle->condition ); - MUTEX_LOCK( &stream_.mutex ); - } - - // Stop the buffer and clear memory - LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - result = buffer->Stop(); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // Lock the buffer and clear it so that if we start to play again, - // we won't have old data playing. - result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // Zero the DS buffer - ZeroMemory( audioPtr, dataLen ); - - // Unlock the DS buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // If we start playing again, we must begin at beginning of buffer. - handle->bufferPointer[0] = 0; - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - audioPtr = NULL; - dataLen = 0; - - result = buffer->Stop(); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // Lock the buffer and clear it so that if we start to play again, - // we won't have old data playing. - result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // Zero the DS buffer - ZeroMemory( audioPtr, dataLen ); - - // Unlock the DS buffer - result = buffer->Unlock( audioPtr, dataLen, NULL, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!"; - errorText_ = errorStream_.str(); - goto unlock; - } - - // If we start recording again, we must begin at beginning of buffer. - handle->bufferPointer[1] = 0; - } - - unlock: - timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows. - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); - - if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR ); -} - -void RtApiDs :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiDs::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } - - DsHandle *handle = (DsHandle *) stream_.apiHandle; - handle->drainCounter = 1; - - stopStream(); -} - -void RtApiDs :: callbackEvent() -{ - if ( stream_.state == STREAM_STOPPED ) { - Sleep(50); // sleep 50 milliseconds - return; - } - - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); - return; - } - - CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo; - DsHandle *handle = (DsHandle *) stream_.apiHandle; - - // Check if we were draining the stream and signal is finished. - if ( handle->drainCounter > stream_.nBuffers + 2 ) { - if ( handle->internalDrain == false ) - SetEvent( handle->condition ); - else - stopStream(); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - - // Invoke user callback to get fresh output data UNLESS we are - // draining stream. - if ( handle->drainCounter == 0 ) { - RtAudioCallback callback = (RtAudioCallback) info->callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; - } - handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, info->userData ); - if ( handle->drainCounter == 2 ) { - MUTEX_UNLOCK( &stream_.mutex ); - abortStream(); - return; - } - else if ( handle->drainCounter == 1 ) - handle->internalDrain = true; - } - - HRESULT result; - DWORD currentWritePos, safeWritePos; - DWORD currentReadPos, safeReadPos; - DWORD leadPos; - UINT nextWritePos; - -#ifdef GENERATE_DEBUG_LOG - DWORD writeTime, readTime; -#endif - - LPVOID buffer1 = NULL; - LPVOID buffer2 = NULL; - DWORD bufferSize1 = 0; - DWORD bufferSize2 = 0; - - char *buffer; - long bufferBytes; - - if ( stream_.mode == DUPLEX && !buffersRolling ) { - //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); - - // It takes a while for the devices to get rolling. As a result, - // there's no guarantee that the capture and write device pointers - // will move in lockstep. Wait here for both devices to start - // rolling, and then set our buffer pointers accordingly. - // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600 - // bytes later than the write buffer. - - // Stub: a serious risk of having a pre-emptive scheduling round - // take place between the two GetCurrentPosition calls... but I'm - // really not sure how to solve the problem. Temporarily boost to - // Realtime priority, maybe; but I'm not sure what priority the - // DirectSound service threads run at. We *should* be roughly - // within a ms or so of correct. - - LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - - DWORD initialWritePos, initialSafeWritePos; - DWORD initialReadPos, initialSafeReadPos; - - result = dsWriteBuffer->GetCurrentPosition( &initialWritePos, &initialSafeWritePos ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - result = dsCaptureBuffer->GetCurrentPosition( &initialReadPos, &initialSafeReadPos ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - while ( true ) { - result = dsWriteBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - result = dsCaptureBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - if ( safeWritePos != initialSafeWritePos && safeReadPos != initialSafeReadPos ) break; - Sleep( 1 ); - } - - //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] ); - - buffersRolling = true; - handle->bufferPointer[0] = ( safeWritePos + handle->dsPointerLeadTime[0] ); - handle->bufferPointer[1] = safeReadPos; - } - - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; - - if ( handle->drainCounter > 1 ) { // write zeros to the output stream - bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; - bufferBytes *= formatBytes( stream_.userFormat ); - memset( stream_.userBuffer[0], 0, bufferBytes ); - } - - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0]; - bufferBytes *= formatBytes( stream_.deviceFormat[0] ); - } - else { - buffer = stream_.userBuffer[0]; - bufferBytes = stream_.bufferSize * stream_.nUserChannels[0]; - bufferBytes *= formatBytes( stream_.userFormat ); - } - - // No byte swapping necessary in DirectSound implementation. - - // Ahhh ... windoze. 16-bit data is signed but 8-bit data is - // unsigned. So, we need to convert our signed 8-bit data here to - // unsigned. - if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 ) - for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 ); - - DWORD dsBufferSize = handle->dsBufferSize[0]; - nextWritePos = handle->bufferPointer[0]; - - DWORD endWrite; - while ( true ) { - // Find out where the read and "safe write" pointers are. - result = dsBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - - leadPos = safeWritePos + handle->dsPointerLeadTime[0]; - if ( leadPos > dsBufferSize ) leadPos -= dsBufferSize; - if ( leadPos < nextWritePos ) leadPos += dsBufferSize; // unwrap offset - endWrite = nextWritePos + bufferBytes; - - // Check whether the entire write region is behind the play pointer. - if ( leadPos >= endWrite ) break; - - // If we are here, then we must wait until the play pointer gets - // beyond the write region. The approach here is to use the - // Sleep() function to suspend operation until safePos catches - // up. Calculate number of milliseconds to wait as: - // time = distance * (milliseconds/second) * fudgefactor / - // ((bytes/sample) * (samples/second)) - // A "fudgefactor" less than 1 is used because it was found - // that sleeping too long was MUCH worse than sleeping for - // several shorter periods. - double millis = ( endWrite - leadPos ) * 900.0; - millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate); - if ( millis < 1.0 ) millis = 1.0; - if ( millis > 50.0 ) { - static int nOverruns = 0; - ++nOverruns; - } - Sleep( (DWORD) millis ); - } - - //if ( statistics.writeDeviceSafeLeadBytes < dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ) ) { - // statistics.writeDeviceSafeLeadBytes = dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ); - //} - - if ( dsPointerBetween( nextWritePos, safeWritePos, currentWritePos, dsBufferSize ) - || dsPointerBetween( endWrite, safeWritePos, currentWritePos, dsBufferSize ) ) { - // We've strayed into the forbidden zone ... resync the read pointer. - //++statistics.numberOfWriteUnderruns; - handle->xrun[0] = true; - nextWritePos = safeWritePos + handle->dsPointerLeadTime[0] - bufferBytes + dsBufferSize; - while ( nextWritePos >= dsBufferSize ) nextWritePos -= dsBufferSize; - handle->bufferPointer[0] = nextWritePos; - endWrite = nextWritePos + bufferBytes; - } - - // Lock free space in the buffer - result = dsBuffer->Lock( nextWritePos, bufferBytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - - // Copy our buffer into the DS buffer - CopyMemory( buffer1, buffer, bufferSize1 ); - if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 ); - - // Update our buffer offset and unlock sound buffer - dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - nextWritePos = ( nextWritePos + bufferSize1 + bufferSize2 ) % dsBufferSize; - handle->bufferPointer[0] = nextWritePos; - - if ( handle->drainCounter ) { - handle->drainCounter++; - goto unlock; - } - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1]; - bufferBytes *= formatBytes( stream_.deviceFormat[1] ); - } - else { - buffer = stream_.userBuffer[1]; - bufferBytes = stream_.bufferSize * stream_.nUserChannels[1]; - bufferBytes *= formatBytes( stream_.userFormat ); - } - - LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1]; - long nextReadPos = handle->bufferPointer[1]; - DWORD dsBufferSize = handle->dsBufferSize[1]; - - // Find out where the write and "safe read" pointers are. - result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - - if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset - DWORD endRead = nextReadPos + bufferBytes; - - // Handling depends on whether we are INPUT or DUPLEX. - // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode, - // then a wait here will drag the write pointers into the forbidden zone. - // - // In DUPLEX mode, rather than wait, we will back off the read pointer until - // it's in a safe position. This causes dropouts, but it seems to be the only - // practical way to sync up the read and write pointers reliably, given the - // the very complex relationship between phase and increment of the read and write - // pointers. - // - // In order to minimize audible dropouts in DUPLEX mode, we will - // provide a pre-roll period of 0.5 seconds in which we return - // zeros from the read buffer while the pointers sync up. - - if ( stream_.mode == DUPLEX ) { - if ( safeReadPos < endRead ) { - if ( duplexPrerollBytes <= 0 ) { - // Pre-roll time over. Be more agressive. - int adjustment = endRead-safeReadPos; - - handle->xrun[1] = true; - //++statistics.numberOfReadOverruns; - // Two cases: - // - large adjustments: we've probably run out of CPU cycles, so just resync exactly, - // and perform fine adjustments later. - // - small adjustments: back off by twice as much. - if ( adjustment >= 2*bufferBytes ) - nextReadPos = safeReadPos-2*bufferBytes; - else - nextReadPos = safeReadPos-bufferBytes-adjustment; - - //statistics.readDeviceSafeLeadBytes = currentReadPos-nextReadPos; - //if ( statistics.readDeviceSafeLeadBytes < 0) statistics.readDeviceSafeLeadBytes += dsBufferSize; - if ( nextReadPos < 0 ) nextReadPos += dsBufferSize; - - } - else { - // In pre=roll time. Just do it. - nextReadPos = safeReadPos-bufferBytes; - while ( nextReadPos < 0 ) nextReadPos += dsBufferSize; - } - endRead = nextReadPos + bufferBytes; - } - } - else { // mode == INPUT - while ( safeReadPos < endRead ) { - // See comments for playback. - double millis = (endRead - safeReadPos) * 900.0; - millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate); - if ( millis < 1.0 ) millis = 1.0; - Sleep( (DWORD) millis ); - - // Wake up, find out where we are now - result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - - if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset - } - } - - //if (statistics.readDeviceSafeLeadBytes < dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ) ) - // statistics.readDeviceSafeLeadBytes = dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ); - - // Lock free space in the buffer - result = dsBuffer->Lock( nextReadPos, bufferBytes, &buffer1, - &bufferSize1, &buffer2, &bufferSize2, 0 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - - if ( duplexPrerollBytes <= 0 ) { - // Copy our buffer into the DS buffer - CopyMemory( buffer, buffer1, bufferSize1 ); - if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 ); - } - else { - memset( buffer, 0, bufferSize1 ); - if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 ); - duplexPrerollBytes -= bufferSize1 + bufferSize2; - } - - // Update our buffer offset and unlock sound buffer - nextReadPos = ( nextReadPos + bufferSize1 + bufferSize2 ) % dsBufferSize; - dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 ); - if ( FAILED( result ) ) { - errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!"; - errorText_ = errorStream_.str(); - error( RtError::SYSTEM_ERROR ); - } - handle->bufferPointer[1] = nextReadPos; - - // No byte swapping necessary in DirectSound implementation. - - // If necessary, convert 8-bit data from unsigned to signed. - if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 ) - for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 ); - - // Do buffer conversion if necessary. - if ( stream_.doConvertBuffer[1] ) - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - } -#ifdef GENERATE_DEBUG_LOG - if ( currentDebugLogEntry < debugLog.size() ) - { - TTickRecord &r = debugLog[currentDebugLogEntry++]; - r.currentReadPointer = currentReadPos; - r.safeReadPointer = safeReadPos; - r.currentWritePointer = currentWritePos; - r.safeWritePointer = safeWritePos; - r.readTime = readTime; - r.writeTime = writeTime; - r.nextReadPointer = handles[1].bufferPointer; - r.nextWritePointer = handles[0].bufferPointer; - } -#endif - - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - - RtApi::tickStreamTime(); -} - -// Definitions for utility functions and callbacks -// specific to the DirectSound implementation. - -extern "C" unsigned __stdcall callbackHandler( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiDs *object = (RtApiDs *) info->object; - bool* isRunning = &info->isRunning; - - while ( *isRunning == true ) { - object->callbackEvent(); - } - - _endthreadex( 0 ); - return 0; -} - -#include "tchar.h" - -std::string convertTChar( LPCTSTR name ) -{ - std::string s; - -#if defined( UNICODE ) || defined( _UNICODE ) - // Yes, this conversion doesn't make sense for two-byte characters - // but RtAudio is currently written to return an std::string of - // one-byte chars for the device name. - for ( unsigned int i=0; i<wcslen( name ); i++ ) - s.push_back( name[i] ); -#else - s.append( std::string( name ) ); -#endif - - return s; -} - -static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid, - LPCTSTR description, - LPCTSTR module, - LPVOID lpContext ) -{ - EnumInfo *info = (EnumInfo *) lpContext; - - HRESULT hr; - if ( info->isInput == true ) { - DSCCAPS caps; - LPDIRECTSOUNDCAPTURE object; - - hr = DirectSoundCaptureCreate( lpguid, &object, NULL ); - if ( hr != DS_OK ) return TRUE; - - caps.dwSize = sizeof(caps); - hr = object->GetCaps( &caps ); - if ( hr == DS_OK ) { - if ( caps.dwChannels > 0 && caps.dwFormats > 0 ) - info->counter++; - } - object->Release(); - } - else { - DSCAPS caps; - LPDIRECTSOUND object; - hr = DirectSoundCreate( lpguid, &object, NULL ); - if ( hr != DS_OK ) return TRUE; - - caps.dwSize = sizeof(caps); - hr = object->GetCaps( &caps ); - if ( hr == DS_OK ) { - if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO ) - info->counter++; - } - object->Release(); - } - - if ( info->getDefault && lpguid == NULL ) return FALSE; - - if ( info->findIndex && info->counter > info->index ) { - info->id = lpguid; - info->name = convertTChar( description ); - return FALSE; - } - - return TRUE; -} - -static char* getErrorString( int code ) -{ - switch ( code ) { - - case DSERR_ALLOCATED: - return "Already allocated"; - - case DSERR_CONTROLUNAVAIL: - return "Control unavailable"; - - case DSERR_INVALIDPARAM: - return "Invalid parameter"; - - case DSERR_INVALIDCALL: - return "Invalid call"; - - case DSERR_GENERIC: - return "Generic error"; - - case DSERR_PRIOLEVELNEEDED: - return "Priority level needed"; - - case DSERR_OUTOFMEMORY: - return "Out of memory"; - - case DSERR_BADFORMAT: - return "The sample rate or the channel format is not supported"; - - case DSERR_UNSUPPORTED: - return "Not supported"; - - case DSERR_NODRIVER: - return "No driver"; - - case DSERR_ALREADYINITIALIZED: - return "Already initialized"; - - case DSERR_NOAGGREGATION: - return "No aggregation"; - - case DSERR_BUFFERLOST: - return "Buffer lost"; - - case DSERR_OTHERAPPHASPRIO: - return "Another application already has priority"; - - case DSERR_UNINITIALIZED: - return "Uninitialized"; - - default: - return "DirectSound unknown error"; - } -} -//******************** End of __WINDOWS_DS__ *********************// -#endif - - -#if defined(__LINUX_ALSA__) - -#include <alsa/asoundlib.h> -#include <unistd.h> - - // A structure to hold various information related to the ALSA API - // implementation. -struct AlsaHandle { - snd_pcm_t *handles[2]; - bool synchronized; - bool xrun[2]; - pthread_cond_t runnable; - - AlsaHandle() - :synchronized(false) { xrun[0] = false; xrun[1] = false; } -}; - -extern "C" void *alsaCallbackHandler( void * ptr ); - -RtApiAlsa :: RtApiAlsa() -{ - // Nothing to do here. -} - -RtApiAlsa :: ~RtApiAlsa() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} - -unsigned int RtApiAlsa :: getDeviceCount( void ) -{ - unsigned nDevices = 0; - int result, subdevice, card; - char name[64]; - snd_ctl_t *handle; - - // Count cards and devices - card = -1; - snd_card_next( &card ); - while ( card >= 0 ) { - sprintf( name, "hw:%d", card ); - result = snd_ctl_open( &handle, name, 0 ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto nextcard; - } - subdevice = -1; - while( 1 ) { - result = snd_ctl_pcm_next_device( handle, &subdevice ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - break; - } - if ( subdevice < 0 ) - break; - nDevices++; - } - nextcard: - snd_ctl_close( handle ); - snd_card_next( &card ); - } - - return nDevices; -} - -RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; - - unsigned nDevices = 0; - int result, subdevice, card; - char name[64]; - snd_ctl_t *chandle; - - // Count cards and devices - card = -1; - snd_card_next( &card ); - while ( card >= 0 ) { - sprintf( name, "hw:%d", card ); - result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto nextcard; - } - subdevice = -1; - while( 1 ) { - result = snd_ctl_pcm_next_device( chandle, &subdevice ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - break; - } - if ( subdevice < 0 ) break; - if ( nDevices == device ) { - sprintf( name, "hw:%d,%d", card, subdevice ); - goto foundDevice; - } - nDevices++; - } - nextcard: - snd_ctl_close( chandle ); - snd_card_next( &card ); - } - - if ( nDevices == 0 ) { - errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!"; - error( RtError::INVALID_USE ); - } - - if ( device >= nDevices ) { - errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - } - - foundDevice: - - // If a stream is already open, we cannot probe the stream devices. - // Thus, use the saved results. - if ( stream_.state != STREAM_CLOSED && - ( stream_.device[0] == device || stream_.device[1] == device ) ) { - if ( device >= devices_.size() ) { - errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened."; - error( RtError::WARNING ); - return info; - } - return devices_[ device ]; - } - - int openMode = SND_PCM_ASYNC; - snd_pcm_stream_t stream; - snd_pcm_info_t *pcminfo; - snd_pcm_info_alloca( &pcminfo ); - snd_pcm_t *phandle; - snd_pcm_hw_params_t *params; - snd_pcm_hw_params_alloca( ¶ms ); - - // First try for playback - stream = SND_PCM_STREAM_PLAYBACK; - snd_pcm_info_set_device( pcminfo, subdevice ); - snd_pcm_info_set_subdevice( pcminfo, 0 ); - snd_pcm_info_set_stream( pcminfo, stream ); - - result = snd_ctl_pcm_info( chandle, pcminfo ); - if ( result < 0 ) { - // Device probably doesn't support playback. - goto captureProbe; - } - - result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto captureProbe; - } - - // The device is open ... fill the parameter structure. - result = snd_pcm_hw_params_any( phandle, params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto captureProbe; - } - - // Get output channel information. - unsigned int value; - result = snd_pcm_hw_params_get_channels_max( params, &value ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - goto captureProbe; - } - info.outputChannels = value; - snd_pcm_close( phandle ); - - captureProbe: - // Now try for capture - stream = SND_PCM_STREAM_CAPTURE; - snd_pcm_info_set_stream( pcminfo, stream ); - - result = snd_ctl_pcm_info( chandle, pcminfo ); - snd_ctl_close( chandle ); - if ( result < 0 ) { - // Device probably doesn't support capture. - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } - - result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } - - // The device is open ... fill the parameter structure. - result = snd_pcm_hw_params_any( phandle, params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } - - result = snd_pcm_hw_params_get_channels_max( params, &value ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - if ( info.outputChannels == 0 ) return info; - goto probeParameters; - } - info.inputChannels = value; - snd_pcm_close( phandle ); - - // If device opens for both playback and capture, we determine the channels. - if ( info.outputChannels > 0 && info.inputChannels > 0 ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - - // ALSA doesn't provide default devices so we'll use the first available one. - if ( device == 0 && info.outputChannels > 0 ) - info.isDefaultOutput = true; - if ( device == 0 && info.inputChannels > 0 ) - info.isDefaultInput = true; - - probeParameters: - // At this point, we just need to figure out the supported data - // formats and sample rates. We'll proceed by opening the device in - // the direction with the maximum number of channels, or playback if - // they are equal. This might limit our sample rate options, but so - // be it. - - if ( info.outputChannels >= info.inputChannels ) - stream = SND_PCM_STREAM_PLAYBACK; - else - stream = SND_PCM_STREAM_CAPTURE; - snd_pcm_info_set_stream( pcminfo, stream ); - - result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - // The device is open ... fill the parameter structure. - result = snd_pcm_hw_params_any( phandle, params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - // Test our discrete set of sample rate values. - info.sampleRates.clear(); - for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) { - if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 ) - info.sampleRates.push_back( SAMPLE_RATES[i] ); - } - if ( info.sampleRates.size() == 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - // Probe the supported data formats ... we don't care about endian-ness just yet - snd_pcm_format_t format; - info.nativeFormats = 0; - format = SND_PCM_FORMAT_S8; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_SINT8; - format = SND_PCM_FORMAT_S16; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_SINT16; - format = SND_PCM_FORMAT_S24; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_SINT24; - format = SND_PCM_FORMAT_S32; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_SINT32; - format = SND_PCM_FORMAT_FLOAT; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_FLOAT32; - format = SND_PCM_FORMAT_FLOAT64; - if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 ) - info.nativeFormats |= RTAUDIO_FLOAT64; - - // Check that we have at least one supported format - if ( info.nativeFormats == 0 ) { - errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - // Get the device name - char *cardname; - result = snd_card_get_name( card, &cardname ); - if ( result >= 0 ) - sprintf( name, "hw:%s,%d", cardname, subdevice ); - info.name = name; - - // That's all ... close the device and return - snd_pcm_close( phandle ); - info.probed = true; - return info; -} - -void RtApiAlsa :: saveDeviceInfo( void ) -{ - devices_.clear(); - - unsigned int nDevices = getDeviceCount(); - devices_.resize( nDevices ); - for ( unsigned int i=0; i<nDevices; i++ ) - devices_[i] = getDeviceInfo( i ); -} - -bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) - -{ -#if defined(__RTAUDIO_DEBUG__) - snd_output_t *out; - snd_output_stdio_attach(&out, stderr, 0); -#endif - - // I'm not using the "plug" interface ... too much inconsistent behavior. - - unsigned nDevices = 0; - int result, subdevice, card; - char name[64]; - snd_ctl_t *chandle; - - // Count cards and devices - card = -1; - snd_card_next( &card ); - while ( card >= 0 ) { - sprintf( name, "hw:%d", card ); - result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - subdevice = -1; - while( 1 ) { - result = snd_ctl_pcm_next_device( chandle, &subdevice ); - if ( result < 0 ) break; - if ( subdevice < 0 ) break; - if ( nDevices == device ) { - sprintf( name, "hw:%d,%d", card, subdevice ); - snd_ctl_close( chandle ); - goto foundDevice; - } - nDevices++; - } - snd_ctl_close( chandle ); - snd_card_next( &card ); - } - - if ( nDevices == 0 ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!"; - return FAILURE; - } - - if ( device >= nDevices ) { - // This should not happen because a check is made before this function is called. - errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } - - foundDevice: - - // The getDeviceInfo() function will not work for a device that is - // already open. Thus, we'll probe the system before opening a - // stream and save the results for use by getDeviceInfo(). - if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once - this->saveDeviceInfo(); - - snd_pcm_stream_t stream; - if ( mode == OUTPUT ) - stream = SND_PCM_STREAM_PLAYBACK; - else - stream = SND_PCM_STREAM_CAPTURE; - - snd_pcm_t *phandle; - int openMode = SND_PCM_ASYNC; - result = snd_pcm_open( &phandle, name, stream, openMode ); - if ( result < 0 ) { - if ( mode == OUTPUT ) - errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output."; - else - errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Fill the parameter structure. - snd_pcm_hw_params_t *hw_params; - snd_pcm_hw_params_alloca( &hw_params ); - result = snd_pcm_hw_params_any( phandle, hw_params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - -#if defined(__RTAUDIO_DEBUG__) - fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" ); - snd_pcm_hw_params_dump( hw_params, out ); -#endif - - // Set access ... check user preference. - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) { - stream_.userInterleaved = false; - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); - if ( result < 0 ) { - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); - stream_.deviceInterleaved[mode] = true; - } - else - stream_.deviceInterleaved[mode] = false; - } - else { - stream_.userInterleaved = true; - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED ); - if ( result < 0 ) { - result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED ); - stream_.deviceInterleaved[mode] = false; - } - else - stream_.deviceInterleaved[mode] = true; - } - - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Determine how to set the device format. - stream_.userFormat = format; - snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN; - - if ( format == RTAUDIO_SINT8 ) - deviceFormat = SND_PCM_FORMAT_S8; - else if ( format == RTAUDIO_SINT16 ) - deviceFormat = SND_PCM_FORMAT_S16; - else if ( format == RTAUDIO_SINT24 ) - deviceFormat = SND_PCM_FORMAT_S24; - else if ( format == RTAUDIO_SINT32 ) - deviceFormat = SND_PCM_FORMAT_S32; - else if ( format == RTAUDIO_FLOAT32 ) - deviceFormat = SND_PCM_FORMAT_FLOAT; - else if ( format == RTAUDIO_FLOAT64 ) - deviceFormat = SND_PCM_FORMAT_FLOAT64; - - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) { - stream_.deviceFormat[mode] = format; - goto setFormat; - } - - // The user requested format is not natively supported by the device. - deviceFormat = SND_PCM_FORMAT_FLOAT64; - if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT64; - goto setFormat; - } - - deviceFormat = SND_PCM_FORMAT_FLOAT; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_FLOAT32; - goto setFormat; - } - - deviceFormat = SND_PCM_FORMAT_S32; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - goto setFormat; - } - - deviceFormat = SND_PCM_FORMAT_S24; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - goto setFormat; - } - - deviceFormat = SND_PCM_FORMAT_S16; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - goto setFormat; - } - - deviceFormat = SND_PCM_FORMAT_S8; - if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) { - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - goto setFormat; - } - - // If we get here, no supported format was found. - errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - return FAILURE; - - setFormat: - result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Determine whether byte-swaping is necessary. - stream_.doByteSwap[mode] = false; - if ( deviceFormat != SND_PCM_FORMAT_S8 ) { - result = snd_pcm_format_cpu_endian( deviceFormat ); - if ( result == 0 ) - stream_.doByteSwap[mode] = true; - else if (result < 0) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - - // Set the sample rate. - result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Determine the number of channels for this device. We support a possible - // minimum device channel number > than the value requested by the user. - stream_.nUserChannels[mode] = channels; - unsigned int value; - result = snd_pcm_hw_params_get_channels_max( hw_params, &value ); - unsigned int deviceChannels = value; - if ( result < 0 || deviceChannels < channels + firstChannel ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - result = snd_pcm_hw_params_get_channels_min( hw_params, &value ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - deviceChannels = value; - if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel; - stream_.nDeviceChannels[mode] = deviceChannels; - - // Set the device channels. - result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the buffer number, which in ALSA is referred to as the "period". - int totalSize, dir = 0; - unsigned int periods = 0; - if ( options ) periods = options->numberOfBuffers; - totalSize = *bufferSize * periods; - - // Set the buffer (or period) size. - snd_pcm_uframes_t periodSize = *bufferSize; - result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - *bufferSize = periodSize; - - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2; - else periods = totalSize / *bufferSize; - // Even though the hardware might allow 1 buffer, it won't work reliably. - if ( periods < 2 ) periods = 2; - result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // If attempting to setup a duplex stream, the bufferSize parameter - // MUST be the same in both directions! - if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) { - errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - stream_.bufferSize = *bufferSize; - - // Install the hardware configuration - result = snd_pcm_hw_params( phandle, hw_params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - -#if defined(__RTAUDIO_DEBUG__) - fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n"); - snd_pcm_hw_params_dump( hw_params, out ); -#endif - - // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns. - snd_pcm_sw_params_t *sw_params = NULL; - snd_pcm_sw_params_alloca( &sw_params ); - snd_pcm_sw_params_current( phandle, sw_params ); - snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize ); - snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX ); - snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 ); - - // The following two settings were suggested by Theo Veenker - //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize ); - //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 ); - - // here are two options for a fix - //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX ); - snd_pcm_uframes_t val; - snd_pcm_sw_params_get_boundary( sw_params, &val ); - snd_pcm_sw_params_set_silence_size( phandle, sw_params, val ); - - result = snd_pcm_sw_params( phandle, sw_params ); - if ( result < 0 ) { - snd_pcm_close( phandle ); - errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - return FAILURE; - } - -#if defined(__RTAUDIO_DEBUG__) - fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n"); - snd_pcm_sw_params_dump( sw_params, out ); -#endif - - // Set flags for buffer conversion - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - // Allocate the ApiHandle if necessary and then save. - AlsaHandle *apiInfo = 0; - if ( stream_.apiHandle == 0 ) { - try { - apiInfo = (AlsaHandle *) new AlsaHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory."; - goto error; - } - - if ( pthread_cond_init( &apiInfo->runnable, NULL ) ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable."; - goto error; - } - - stream_.apiHandle = (void *) apiInfo; - apiInfo->handles[0] = 0; - apiInfo->handles[1] = 0; - } - else { - apiInfo = (AlsaHandle *) stream_.apiHandle; - } - apiInfo->handles[mode] = phandle; - - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - if ( stream_.doConvertBuffer[mode] ) { - - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - stream_.sampleRate = sampleRate; - stream_.nBuffers = periods; - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - - // Setup thread if necessary. - if ( stream_.mode == OUTPUT && mode == INPUT ) { - // We had already set up an output stream. - stream_.mode = DUPLEX; - // Link the streams if possible. - apiInfo->synchronized = false; - if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 ) - apiInfo->synchronized = true; - else { - errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices."; - error( RtError::WARNING ); - } - } - else { - stream_.mode = mode; - - // Setup callback thread. - stream_.callbackInfo.object = (void *) this; - - // Set the thread attributes for joinable and realtime scheduling - // priority (optional). The higher priority will only take affect - // if the program is run as root or suid. Note, under Linux - // processes with CAP_SYS_NICE privilege, a user can change - // scheduling policy and priority (thus need not be root). See - // POSIX "capabilities". - pthread_attr_t attr; - pthread_attr_init( &attr ); - pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); -#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) - if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { - struct sched_param param; - int priority = options->priority; - int min = sched_get_priority_min( SCHED_RR ); - int max = sched_get_priority_max( SCHED_RR ); - if ( priority < min ) priority = min; - else if ( priority > max ) priority = max; - param.sched_priority = priority; - pthread_attr_setschedparam( &attr, ¶m ); - pthread_attr_setschedpolicy( &attr, SCHED_RR ); - } - else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#endif - - stream_.callbackInfo.isRunning = true; - result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo ); - pthread_attr_destroy( &attr ); - if ( result ) { - stream_.callbackInfo.isRunning = false; - errorText_ = "RtApiAlsa::error creating callback thread!"; - goto error; - } - } - - return SUCCESS; - - error: - if ( apiInfo ) { - pthread_cond_destroy( &apiInfo->runnable ); - if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); - if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); - delete apiInfo; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - return FAILURE; -} - -void RtApiAlsa :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAlsa::closeStream(): no open stream to close!"; - error( RtError::WARNING ); - return; - } - - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - stream_.callbackInfo.isRunning = false; - MUTEX_LOCK( &stream_.mutex ); - if ( stream_.state == STREAM_STOPPED ) - pthread_cond_signal( &apiInfo->runnable ); - MUTEX_UNLOCK( &stream_.mutex ); - pthread_join( stream_.callbackInfo.thread, NULL ); - - if ( stream_.state == STREAM_RUNNING ) { - stream_.state = STREAM_STOPPED; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) - snd_pcm_drop( apiInfo->handles[0] ); - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) - snd_pcm_drop( apiInfo->handles[1] ); - } - - if ( apiInfo ) { - pthread_cond_destroy( &apiInfo->runnable ); - if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] ); - if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] ); - delete apiInfo; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} - -void RtApiAlsa :: startStream() -{ - // This method calls snd_pcm_prepare if the device isn't already in that state. - - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiAlsa::startStream(): the stream is already running!"; - error( RtError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - int result = 0; - snd_pcm_state_t state; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - state = snd_pcm_state( handle[0] ); - if ( state != SND_PCM_STATE_PREPARED ) { - result = snd_pcm_prepare( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - } - - if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { - state = snd_pcm_state( handle[1] ); - if ( state != SND_PCM_STATE_PREPARED ) { - result = snd_pcm_prepare( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - } - - stream_.state = STREAM_RUNNING; - - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - - pthread_cond_signal( &apiInfo->runnable ); - - if ( result >= 0 ) return; - error( RtError::SYSTEM_ERROR ); -} - -void RtApiAlsa :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - - int result = 0; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( apiInfo->synchronized ) - result = snd_pcm_drop( handle[0] ); - else - result = snd_pcm_drain( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { - result = snd_pcm_drop( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - unlock: - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result >= 0 ) return; - error( RtError::SYSTEM_ERROR ); -} - -void RtApiAlsa :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - - int result = 0; - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - result = snd_pcm_drop( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) { - result = snd_pcm_drop( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - unlock: - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result >= 0 ) return; - error( RtError::SYSTEM_ERROR ); -} - -void RtApiAlsa :: callbackEvent() -{ - AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_LOCK( &stream_.mutex ); - pthread_cond_wait( &apiInfo->runnable, &stream_.mutex ); - if ( stream_.state != STREAM_RUNNING ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - MUTEX_UNLOCK( &stream_.mutex ); - } - - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); - return; - } - - int doStopStream = 0; - RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - apiInfo->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - apiInfo->xrun[1] = false; - } - doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); - - if ( doStopStream == 2 ) { - abortStream(); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) goto unlock; - - int result; - char *buffer; - int channels; - snd_pcm_t **handle; - snd_pcm_sframes_t frames; - RtAudioFormat format; - handle = (snd_pcm_t **) apiInfo->handles; - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - channels = stream_.nDeviceChannels[1]; - format = stream_.deviceFormat[1]; - } - else { - buffer = stream_.userBuffer[1]; - channels = stream_.nUserChannels[1]; - format = stream_.userFormat; - } - - // Read samples from device in interleaved/non-interleaved format. - if ( stream_.deviceInterleaved[1] ) - result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize ); - else { - void *bufs[channels]; - size_t offset = stream_.bufferSize * formatBytes( format ); - for ( int i=0; i<channels; i++ ) - bufs[i] = (void *) (buffer + (i * offset)); - result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize ); - } - - if ( result < (int) stream_.bufferSize ) { - // Either an error or overrun occured. - if ( result == -EPIPE ) { - snd_pcm_state_t state = snd_pcm_state( handle[1] ); - if ( state == SND_PCM_STATE_XRUN ) { - apiInfo->xrun[1] = true; - result = snd_pcm_prepare( handle[1] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - error( RtError::WARNING ); - goto tryOutput; - } - - // Do byte swapping if necessary. - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( buffer, stream_.bufferSize * channels, format ); - - // Do buffer conversion if necessary. - if ( stream_.doConvertBuffer[1] ) - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - - // Check stream latency - result = snd_pcm_delay( handle[1], &frames ); - if ( result == 0 && frames > 0 ) stream_.latency[1] = frames; - } - - tryOutput: - - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - channels = stream_.nDeviceChannels[0]; - format = stream_.deviceFormat[0]; - } - else { - buffer = stream_.userBuffer[0]; - channels = stream_.nUserChannels[0]; - format = stream_.userFormat; - } - - // Do byte swapping if necessary. - if ( stream_.doByteSwap[0] ) - byteSwapBuffer(buffer, stream_.bufferSize * channels, format); - - // Write samples to device in interleaved/non-interleaved format. - if ( stream_.deviceInterleaved[0] ) - result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize ); - else { - void *bufs[channels]; - size_t offset = stream_.bufferSize * formatBytes( format ); - for ( int i=0; i<channels; i++ ) - bufs[i] = (void *) (buffer + (i * offset)); - result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize ); - } - - if ( result < (int) stream_.bufferSize ) { - // Either an error or underrun occured. - if ( result == -EPIPE ) { - snd_pcm_state_t state = snd_pcm_state( handle[0] ); - if ( state == SND_PCM_STATE_XRUN ) { - apiInfo->xrun[0] = true; - result = snd_pcm_prepare( handle[0] ); - if ( result < 0 ) { - errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - } - else { - errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << "."; - errorText_ = errorStream_.str(); - } - error( RtError::WARNING ); - goto unlock; - } - - // Check stream latency - result = snd_pcm_delay( handle[0], &frames ); - if ( result == 0 && frames > 0 ) stream_.latency[0] = frames; - } - - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - - RtApi::tickStreamTime(); - if ( doStopStream == 1 ) this->stopStream(); -} - -extern "C" void *alsaCallbackHandler( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiAlsa *object = (RtApiAlsa *) info->object; - bool *isRunning = &info->isRunning; - - while ( *isRunning == true ) { - pthread_testcancel(); - object->callbackEvent(); - } - - pthread_exit( NULL ); -} - -//******************** End of __LINUX_ALSA__ *********************// -#endif - - -#if defined(__LINUX_OSS__) - -#include <unistd.h> -#include <sys/ioctl.h> -#include <unistd.h> -#include <fcntl.h> -#include "soundcard.h" -#include <errno.h> -#include <math.h> - -extern "C" void *ossCallbackHandler(void * ptr); - -// A structure to hold various information related to the OSS API -// implementation. -struct OssHandle { - int id[2]; // device ids - bool xrun[2]; - bool triggered; - pthread_cond_t runnable; - - OssHandle() - :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; } -}; - -RtApiOss :: RtApiOss() -{ - // Nothing to do here. -} - -RtApiOss :: ~RtApiOss() -{ - if ( stream_.state != STREAM_CLOSED ) closeStream(); -} - -unsigned int RtApiOss :: getDeviceCount( void ) -{ - int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); - if ( mixerfd == -1 ) { - errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'."; - error( RtError::WARNING ); - return 0; - } - - oss_sysinfo sysinfo; - if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required."; - error( RtError::WARNING ); - return 0; - } - - close( mixerfd ); - return sysinfo.numaudios; -} - -RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device ) -{ - RtAudio::DeviceInfo info; - info.probed = false; - - int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); - if ( mixerfd == -1 ) { - errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'."; - error( RtError::WARNING ); - return info; - } - - oss_sysinfo sysinfo; - int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); - if ( result == -1 ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required."; - error( RtError::WARNING ); - return info; - } - - unsigned nDevices = sysinfo.numaudios; - if ( nDevices == 0 ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceInfo: no devices found!"; - error( RtError::INVALID_USE ); - } - - if ( device >= nDevices ) { - close( mixerfd ); - errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!"; - error( RtError::INVALID_USE ); - } - - oss_audioinfo ainfo; - ainfo.dev = device; - result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); - close( mixerfd ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - // Probe channels - if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels; - if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels; - if ( ainfo.caps & PCM_CAP_DUPLEX ) { - if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX ) - info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels; - } - - // Probe data formats ... do for input - unsigned long mask = ainfo.iformats; - if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE ) - info.nativeFormats |= RTAUDIO_SINT16; - if ( mask & AFMT_S8 ) - info.nativeFormats |= RTAUDIO_SINT8; - if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE ) - info.nativeFormats |= RTAUDIO_SINT32; - if ( mask & AFMT_FLOAT ) - info.nativeFormats |= RTAUDIO_FLOAT32; - if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE ) - info.nativeFormats |= RTAUDIO_SINT24; - - // Check that we have at least one supported format - if ( info.nativeFormats == 0 ) { - errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - return info; - } - - // Probe the supported sample rates. - info.sampleRates.clear(); - if ( ainfo.nrates ) { - for ( unsigned int i=0; i<ainfo.nrates; i++ ) { - for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) { - if ( ainfo.rates[i] == SAMPLE_RATES[k] ) { - info.sampleRates.push_back( SAMPLE_RATES[k] ); - break; - } - } - } - } - else { - // Check min and max rate values; - for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) { - if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] ) - info.sampleRates.push_back( SAMPLE_RATES[k] ); - } - } - - if ( info.sampleRates.size() == 0 ) { - errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - error( RtError::WARNING ); - } - else { - info.probed = true; - info.name = ainfo.name; - } - - return info; -} - - -bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ) -{ - int mixerfd = open( "/dev/mixer", O_RDWR, 0 ); - if ( mixerfd == -1 ) { - errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'."; - return FAILURE; - } - - oss_sysinfo sysinfo; - int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ); - if ( result == -1 ) { - close( mixerfd ); - errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required."; - return FAILURE; - } - - unsigned nDevices = sysinfo.numaudios; - if ( nDevices == 0 ) { - // This should not happen because a check is made before this function is called. - close( mixerfd ); - errorText_ = "RtApiOss::probeDeviceOpen: no devices found!"; - return FAILURE; - } - - if ( device >= nDevices ) { - // This should not happen because a check is made before this function is called. - close( mixerfd ); - errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!"; - return FAILURE; - } - - oss_audioinfo ainfo; - ainfo.dev = device; - result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo ); - close( mixerfd ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Check if device supports input or output - if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) || - ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) { - if ( mode == OUTPUT ) - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output."; - else - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - int flags = 0; - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( mode == OUTPUT ) - flags |= O_WRONLY; - else { // mode == INPUT - if (stream_.mode == OUTPUT && stream_.device[0] == device) { - // We just set the same device for playback ... close and reopen for duplex (OSS only). - close( handle->id[0] ); - handle->id[0] = 0; - if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) { - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode."; - errorText_ = errorStream_.str(); - return FAILURE; - } - // Check that the number previously set channels is the same. - if ( stream_.nUserChannels[0] != channels ) { - errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - flags |= O_RDWR; - } - else - flags |= O_RDONLY; - } - - // Set exclusive access if specified. - if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL; - - // Try to open the device. - int fd; - fd = open( ainfo.devnode, flags, 0 ); - if ( fd == -1 ) { - if ( errno == EBUSY ) - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy."; - else - errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // For duplex operation, specifically set this mode (this doesn't seem to work). - /* - if ( flags | O_RDWR ) { - result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL ); - if ( result == -1) { - errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - } - */ - - // Check the device channel support. - stream_.nUserChannels[mode] = channels; - if ( ainfo.max_channels < (int)(channels + firstChannel) ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the number of channels. - int deviceChannels = channels + firstChannel; - result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels ); - if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.nDeviceChannels[mode] = deviceChannels; - - // Get the data format mask - int mask; - result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask ); - if ( result == -1 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Determine how to set the device format. - stream_.userFormat = format; - int deviceFormat = -1; - stream_.doByteSwap[mode] = false; - if ( format == RTAUDIO_SINT8 ) { - if ( mask & AFMT_S8 ) { - deviceFormat = AFMT_S8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - } - else if ( format == RTAUDIO_SINT16 ) { - if ( mask & AFMT_S16_NE ) { - deviceFormat = AFMT_S16_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - else if ( mask & AFMT_S16_OE ) { - deviceFormat = AFMT_S16_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - stream_.doByteSwap[mode] = true; - } - } - else if ( format == RTAUDIO_SINT24 ) { - if ( mask & AFMT_S24_NE ) { - deviceFormat = AFMT_S24_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - } - else if ( mask & AFMT_S24_OE ) { - deviceFormat = AFMT_S24_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - stream_.doByteSwap[mode] = true; - } - } - else if ( format == RTAUDIO_SINT32 ) { - if ( mask & AFMT_S32_NE ) { - deviceFormat = AFMT_S32_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - } - else if ( mask & AFMT_S32_OE ) { - deviceFormat = AFMT_S32_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - stream_.doByteSwap[mode] = true; - } - } - - if ( deviceFormat == -1 ) { - // The user requested format is not natively supported by the device. - if ( mask & AFMT_S16_NE ) { - deviceFormat = AFMT_S16_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - } - else if ( mask & AFMT_S32_NE ) { - deviceFormat = AFMT_S32_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - } - else if ( mask & AFMT_S24_NE ) { - deviceFormat = AFMT_S24_NE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - } - else if ( mask & AFMT_S16_OE ) { - deviceFormat = AFMT_S16_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT16; - stream_.doByteSwap[mode] = true; - } - else if ( mask & AFMT_S32_OE ) { - deviceFormat = AFMT_S32_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT32; - stream_.doByteSwap[mode] = true; - } - else if ( mask & AFMT_S24_OE ) { - deviceFormat = AFMT_S24_OE; - stream_.deviceFormat[mode] = RTAUDIO_SINT24; - stream_.doByteSwap[mode] = true; - } - else if ( mask & AFMT_S8) { - deviceFormat = AFMT_S8; - stream_.deviceFormat[mode] = RTAUDIO_SINT8; - } - } - - if ( stream_.deviceFormat[mode] == 0 ) { - // This really shouldn't happen ... - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Set the data format. - int temp = deviceFormat; - result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat ); - if ( result == -1 || deviceFormat != temp ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Attempt to set the buffer size. According to OSS, the minimum - // number of buffers is two. The supposed minimum buffer size is 16 - // bytes, so that will be our lower bound. The argument to this - // call is in the form 0xMMMMSSSS (hex), where the buffer size (in - // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM. - // We'll check the actual value used near the end of the setup - // procedure. - int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels; - if ( ossBufferBytes < 16 ) ossBufferBytes = 16; - int buffers = 0; - if ( options ) buffers = options->numberOfBuffers; - if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2; - if ( buffers < 2 ) buffers = 3; - temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) ); - result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp ); - if ( result == -1 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.nBuffers = buffers; - - // Save buffer size (in sample frames). - *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels ); - stream_.bufferSize = *bufferSize; - - // Set the sample rate. - int srate = sampleRate; - result = ioctl( fd, SNDCTL_DSP_SPEED, &srate ); - if ( result == -1 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - - // Verify the sample rate setup worked. - if ( abs( srate - sampleRate ) > 100 ) { - close( fd ); - errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ")."; - errorText_ = errorStream_.str(); - return FAILURE; - } - stream_.sampleRate = sampleRate; - - if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) { - // We're doing duplex setup here. - stream_.deviceFormat[0] = stream_.deviceFormat[1]; - stream_.nDeviceChannels[0] = deviceChannels; - } - - // Set interleaving parameters. - stream_.userInterleaved = true; - stream_.deviceInterleaved[mode] = true; - if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) - stream_.userInterleaved = false; - - // Set flags for buffer conversion - stream_.doConvertBuffer[mode] = false; - if ( stream_.userFormat != stream_.deviceFormat[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] ) - stream_.doConvertBuffer[mode] = true; - if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && - stream_.nUserChannels[mode] > 1 ) - stream_.doConvertBuffer[mode] = true; - - // Allocate the stream handles if necessary and then save. - if ( stream_.apiHandle == 0 ) { - try { - handle = new OssHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory."; - goto error; - } - - if ( pthread_cond_init( &handle->runnable, NULL ) ) { - errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable."; - goto error; - } - - stream_.apiHandle = (void *) handle; - } - else { - handle = (OssHandle *) stream_.apiHandle; - } - handle->id[mode] = fd; - - // Allocate necessary internal buffers. - unsigned long bufferBytes; - bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat ); - stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 ); - if ( stream_.userBuffer[mode] == NULL ) { - errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory."; - goto error; - } - - if ( stream_.doConvertBuffer[mode] ) { - - bool makeBuffer = true; - bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] ); - if ( mode == INPUT ) { - if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) { - unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] ); - if ( bufferBytes <= bytesOut ) makeBuffer = false; - } - } - - if ( makeBuffer ) { - bufferBytes *= *bufferSize; - if ( stream_.deviceBuffer ) free( stream_.deviceBuffer ); - stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 ); - if ( stream_.deviceBuffer == NULL ) { - errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory."; - goto error; - } - } - } - - stream_.device[mode] = device; - stream_.state = STREAM_STOPPED; - - // Setup the buffer conversion information structure. - if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel ); - - // Setup thread if necessary. - if ( stream_.mode == OUTPUT && mode == INPUT ) { - // We had already set up an output stream. - stream_.mode = DUPLEX; - if ( stream_.device[0] == device ) handle->id[0] = fd; - } - else { - stream_.mode = mode; - - // Setup callback thread. - stream_.callbackInfo.object = (void *) this; - - // Set the thread attributes for joinable and realtime scheduling - // priority. The higher priority will only take affect if the - // program is run as root or suid. - pthread_attr_t attr; - pthread_attr_init( &attr ); - pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE ); -#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) - if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { - struct sched_param param; - int priority = options->priority; - int min = sched_get_priority_min( SCHED_RR ); - int max = sched_get_priority_max( SCHED_RR ); - if ( priority < min ) priority = min; - else if ( priority > max ) priority = max; - param.sched_priority = priority; - pthread_attr_setschedparam( &attr, ¶m ); - pthread_attr_setschedpolicy( &attr, SCHED_RR ); - } - else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#else - pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); -#endif - - stream_.callbackInfo.isRunning = true; - result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo ); - pthread_attr_destroy( &attr ); - if ( result ) { - stream_.callbackInfo.isRunning = false; - errorText_ = "RtApiOss::error creating callback thread!"; - goto error; - } - } - - return SUCCESS; - - error: - if ( handle ) { - pthread_cond_destroy( &handle->runnable ); - if ( handle->id[0] ) close( handle->id[0] ); - if ( handle->id[1] ) close( handle->id[1] ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - return FAILURE; -} - -void RtApiOss :: closeStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiOss::closeStream(): no open stream to close!"; - error( RtError::WARNING ); - return; - } - - OssHandle *handle = (OssHandle *) stream_.apiHandle; - stream_.callbackInfo.isRunning = false; - MUTEX_LOCK( &stream_.mutex ); - if ( stream_.state == STREAM_STOPPED ) - pthread_cond_signal( &handle->runnable ); - MUTEX_UNLOCK( &stream_.mutex ); - pthread_join( stream_.callbackInfo.thread, NULL ); - - if ( stream_.state == STREAM_RUNNING ) { - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) - ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); - else - ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); - stream_.state = STREAM_STOPPED; - } - - if ( handle ) { - pthread_cond_destroy( &handle->runnable ); - if ( handle->id[0] ) close( handle->id[0] ); - if ( handle->id[1] ) close( handle->id[1] ); - delete handle; - stream_.apiHandle = 0; - } - - for ( int i=0; i<2; i++ ) { - if ( stream_.userBuffer[i] ) { - free( stream_.userBuffer[i] ); - stream_.userBuffer[i] = 0; - } - } - - if ( stream_.deviceBuffer ) { - free( stream_.deviceBuffer ); - stream_.deviceBuffer = 0; - } - - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; -} - -void RtApiOss :: startStream() -{ - verifyStream(); - if ( stream_.state == STREAM_RUNNING ) { - errorText_ = "RtApiOss::startStream(): the stream is already running!"; - error( RtError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - stream_.state = STREAM_RUNNING; - - // No need to do anything else here ... OSS automatically starts - // when fed samples. - - MUTEX_UNLOCK( &stream_.mutex ); - - OssHandle *handle = (OssHandle *) stream_.apiHandle; - pthread_cond_signal( &handle->runnable ); -} - -void RtApiOss :: stopStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiOss::stopStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - - int result = 0; - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - // Flush the output with zeros a few times. - char *buffer; - int samples; - RtAudioFormat format; - - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - samples = stream_.bufferSize * stream_.nDeviceChannels[0]; - format = stream_.deviceFormat[0]; - } - else { - buffer = stream_.userBuffer[0]; - samples = stream_.bufferSize * stream_.nUserChannels[0]; - format = stream_.userFormat; - } - - memset( buffer, 0, samples * formatBytes(format) ); - for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) { - result = write( handle->id[0], buffer, samples * formatBytes(format) ); - if ( result == -1 ) { - errorText_ = "RtApiOss::stopStream: audio write error."; - error( RtError::WARNING ); - } - } - - result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - handle->triggered = false; - } - - if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { - result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - unlock: - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result != -1 ) return; - error( RtError::SYSTEM_ERROR ); -} - -void RtApiOss :: abortStream() -{ - verifyStream(); - if ( stream_.state == STREAM_STOPPED ) { - errorText_ = "RtApiOss::abortStream(): the stream is already stopped!"; - error( RtError::WARNING ); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - - int result = 0; - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - handle->triggered = false; - } - - if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) { - result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 ); - if ( result == -1 ) { - errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ")."; - errorText_ = errorStream_.str(); - goto unlock; - } - } - - unlock: - stream_.state = STREAM_STOPPED; - MUTEX_UNLOCK( &stream_.mutex ); - - if ( result != -1 ) return; - error( RtError::SYSTEM_ERROR ); -} - -void RtApiOss :: callbackEvent() -{ - OssHandle *handle = (OssHandle *) stream_.apiHandle; - if ( stream_.state == STREAM_STOPPED ) { - MUTEX_LOCK( &stream_.mutex ); - pthread_cond_wait( &handle->runnable, &stream_.mutex ); - if ( stream_.state != STREAM_RUNNING ) { - MUTEX_UNLOCK( &stream_.mutex ); - return; - } - MUTEX_UNLOCK( &stream_.mutex ); - } - - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!"; - error( RtError::WARNING ); - return; - } - - // Invoke user callback to get fresh output data. - int doStopStream = 0; - RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback; - double streamTime = getStreamTime(); - RtAudioStreamStatus status = 0; - if ( stream_.mode != INPUT && handle->xrun[0] == true ) { - status |= RTAUDIO_OUTPUT_UNDERFLOW; - handle->xrun[0] = false; - } - if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) { - status |= RTAUDIO_INPUT_OVERFLOW; - handle->xrun[1] = false; - } - doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1], - stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData ); - if ( doStopStream == 2 ) { - this->abortStream(); - return; - } - - MUTEX_LOCK( &stream_.mutex ); - - // The state might change while waiting on a mutex. - if ( stream_.state == STREAM_STOPPED ) goto unlock; - - int result; - char *buffer; - int samples; - RtAudioFormat format; - - if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - - // Setup parameters and do buffer conversion if necessary. - if ( stream_.doConvertBuffer[0] ) { - buffer = stream_.deviceBuffer; - convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] ); - samples = stream_.bufferSize * stream_.nDeviceChannels[0]; - format = stream_.deviceFormat[0]; - } - else { - buffer = stream_.userBuffer[0]; - samples = stream_.bufferSize * stream_.nUserChannels[0]; - format = stream_.userFormat; - } - - // Do byte swapping if necessary. - if ( stream_.doByteSwap[0] ) - byteSwapBuffer( buffer, samples, format ); - - if ( stream_.mode == DUPLEX && handle->triggered == false ) { - int trig = 0; - ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); - result = write( handle->id[0], buffer, samples * formatBytes(format) ); - trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT; - ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig ); - handle->triggered = true; - } - else - // Write samples to device. - result = write( handle->id[0], buffer, samples * formatBytes(format) ); - - if ( result == -1 ) { - // We'll assume this is an underrun, though there isn't a - // specific means for determining that. - handle->xrun[0] = true; - errorText_ = "RtApiOss::callbackEvent: audio write error."; - error( RtError::WARNING ); - // Continue on to input section. - } - } - - if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) { - - // Setup parameters. - if ( stream_.doConvertBuffer[1] ) { - buffer = stream_.deviceBuffer; - samples = stream_.bufferSize * stream_.nDeviceChannels[1]; - format = stream_.deviceFormat[1]; - } - else { - buffer = stream_.userBuffer[1]; - samples = stream_.bufferSize * stream_.nUserChannels[1]; - format = stream_.userFormat; - } - - // Read samples from device. - result = read( handle->id[1], buffer, samples * formatBytes(format) ); - - if ( result == -1 ) { - // We'll assume this is an overrun, though there isn't a - // specific means for determining that. - handle->xrun[1] = true; - errorText_ = "RtApiOss::callbackEvent: audio read error."; - error( RtError::WARNING ); - goto unlock; - } - - // Do byte swapping if necessary. - if ( stream_.doByteSwap[1] ) - byteSwapBuffer( buffer, samples, format ); - - // Do buffer conversion if necessary. - if ( stream_.doConvertBuffer[1] ) - convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] ); - } - - unlock: - MUTEX_UNLOCK( &stream_.mutex ); - - RtApi::tickStreamTime(); - if ( doStopStream == 1 ) this->stopStream(); -} - -extern "C" void *ossCallbackHandler( void *ptr ) -{ - CallbackInfo *info = (CallbackInfo *) ptr; - RtApiOss *object = (RtApiOss *) info->object; - bool *isRunning = &info->isRunning; - - while ( *isRunning == true ) { - pthread_testcancel(); - object->callbackEvent(); - } - - pthread_exit( NULL ); -} - -//******************** End of __LINUX_OSS__ *********************// -#endif - - -// *************************************************** // -// -// Protected common (OS-independent) RtAudio methods. -// -// *************************************************** // - -// This method can be modified to control the behavior of error -// message printing. -void RtApi :: error( RtError::Type type ) -{ - errorStream_.str(""); // clear the ostringstream - if ( type == RtError::WARNING && showWarnings_ == true ) - std::cerr << '\n' << errorText_ << "\n\n"; - else - throw( RtError( errorText_, type ) ); -} - -void RtApi :: verifyStream() -{ - if ( stream_.state == STREAM_CLOSED ) { - errorText_ = "RtApi:: a stream is not open!"; - error( RtError::INVALID_USE ); - } -} - -void RtApi :: clearStreamInfo() -{ - stream_.mode = UNINITIALIZED; - stream_.state = STREAM_CLOSED; - stream_.sampleRate = 0; - stream_.bufferSize = 0; - stream_.nBuffers = 0; - stream_.userFormat = 0; - stream_.userInterleaved = true; - stream_.streamTime = 0.0; - stream_.apiHandle = 0; - stream_.deviceBuffer = 0; - stream_.callbackInfo.callback = 0; - stream_.callbackInfo.userData = 0; - stream_.callbackInfo.isRunning = false; - for ( int i=0; i<2; i++ ) { - stream_.device[i] = 11111; - stream_.doConvertBuffer[i] = false; - stream_.deviceInterleaved[i] = true; - stream_.doByteSwap[i] = false; - stream_.nUserChannels[i] = 0; - stream_.nDeviceChannels[i] = 0; - stream_.channelOffset[i] = 0; - stream_.deviceFormat[i] = 0; - stream_.latency[i] = 0; - stream_.userBuffer[i] = 0; - stream_.convertInfo[i].channels = 0; - stream_.convertInfo[i].inJump = 0; - stream_.convertInfo[i].outJump = 0; - stream_.convertInfo[i].inFormat = 0; - stream_.convertInfo[i].outFormat = 0; - stream_.convertInfo[i].inOffset.clear(); - stream_.convertInfo[i].outOffset.clear(); - } -} - -unsigned int RtApi :: formatBytes( RtAudioFormat format ) -{ - if ( format == RTAUDIO_SINT16 ) - return 2; - else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 || - format == RTAUDIO_FLOAT32 ) - return 4; - else if ( format == RTAUDIO_FLOAT64 ) - return 8; - else if ( format == RTAUDIO_SINT8 ) - return 1; - - errorText_ = "RtApi::formatBytes: undefined format."; - error( RtError::WARNING ); - - return 0; -} - -void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel ) -{ - if ( mode == INPUT ) { // convert device to user buffer - stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1]; - stream_.convertInfo[mode].outJump = stream_.nUserChannels[1]; - stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1]; - stream_.convertInfo[mode].outFormat = stream_.userFormat; - } - else { // convert user to device buffer - stream_.convertInfo[mode].inJump = stream_.nUserChannels[0]; - stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0]; - stream_.convertInfo[mode].inFormat = stream_.userFormat; - stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0]; - } - - if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump ) - stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump; - else - stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump; - - // Set up the interleave/deinterleave offsets. - if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) { - if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) || - ( mode == INPUT && stream_.userInterleaved ) ) { - for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) { - stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize ); - stream_.convertInfo[mode].outOffset.push_back( k ); - stream_.convertInfo[mode].inJump = 1; - } - } - else { - for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) { - stream_.convertInfo[mode].inOffset.push_back( k ); - stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize ); - stream_.convertInfo[mode].outJump = 1; - } - } - } - else { // no (de)interleaving - if ( stream_.userInterleaved ) { - for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) { - stream_.convertInfo[mode].inOffset.push_back( k ); - stream_.convertInfo[mode].outOffset.push_back( k ); - } - } - else { - for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) { - stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize ); - stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize ); - stream_.convertInfo[mode].inJump = 1; - stream_.convertInfo[mode].outJump = 1; - } - } - } - - // Add channel offset. - if ( firstChannel > 0 ) { - if ( stream_.deviceInterleaved[mode] ) { - if ( mode == OUTPUT ) { - for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) - stream_.convertInfo[mode].outOffset[k] += firstChannel; - } - else { - for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) - stream_.convertInfo[mode].inOffset[k] += firstChannel; - } - } - else { - if ( mode == OUTPUT ) { - for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) - stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize ); - } - else { - for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) - stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize ); - } - } - } -} - -void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info ) -{ - // This function does format conversion, input/output channel compensation, and - // data interleaving/deinterleaving. 24-bit integers are assumed to occupy - // the upper three bytes of a 32-bit integer. - - // Clear our device buffer when in/out duplex device channels are different - if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX && - ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) ) - memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) ); - - int j; - if (info.outFormat == RTAUDIO_FLOAT64) { - Float64 scale; - Float64 *out = (Float64 *)outBuffer; - - if (info.inFormat == RTAUDIO_SINT8) { - signed char *in = (signed char *)inBuffer; - scale = 1.0 / 127.5; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Float64) in[info.inOffset[j]]; - out[info.outOffset[j]] += 0.5; - out[info.outOffset[j]] *= scale; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT16) { - Int16 *in = (Int16 *)inBuffer; - scale = 1.0 / 32767.5; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Float64) in[info.inOffset[j]]; - out[info.outOffset[j]] += 0.5; - out[info.outOffset[j]] *= scale; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT24) { - Int32 *in = (Int32 *)inBuffer; - scale = 1.0 / 8388607.5; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]] & 0x00ffffff); - out[info.outOffset[j]] += 0.5; - out[info.outOffset[j]] *= scale; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT32) { - Int32 *in = (Int32 *)inBuffer; - scale = 1.0 / 2147483647.5; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Float64) in[info.inOffset[j]]; - out[info.outOffset[j]] += 0.5; - out[info.outOffset[j]] *= scale; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT32) { - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Float64) in[info.inOffset[j]]; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT64) { - // Channel compensation and/or (de)interleaving only. - Float64 *in = (Float64 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = in[info.inOffset[j]]; - } - in += info.inJump; - out += info.outJump; - } - } - } - else if (info.outFormat == RTAUDIO_FLOAT32) { - Float32 scale; - Float32 *out = (Float32 *)outBuffer; - - if (info.inFormat == RTAUDIO_SINT8) { - signed char *in = (signed char *)inBuffer; - scale = (Float32) ( 1.0 / 127.5 ); - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Float32) in[info.inOffset[j]]; - out[info.outOffset[j]] += 0.5; - out[info.outOffset[j]] *= scale; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT16) { - Int16 *in = (Int16 *)inBuffer; - scale = (Float32) ( 1.0 / 32767.5 ); - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Float32) in[info.inOffset[j]]; - out[info.outOffset[j]] += 0.5; - out[info.outOffset[j]] *= scale; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT24) { - Int32 *in = (Int32 *)inBuffer; - scale = (Float32) ( 1.0 / 8388607.5 ); - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]] & 0x00ffffff); - out[info.outOffset[j]] += 0.5; - out[info.outOffset[j]] *= scale; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT32) { - Int32 *in = (Int32 *)inBuffer; - scale = (Float32) ( 1.0 / 2147483647.5 ); - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Float32) in[info.inOffset[j]]; - out[info.outOffset[j]] += 0.5; - out[info.outOffset[j]] *= scale; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT32) { - // Channel compensation and/or (de)interleaving only. - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = in[info.inOffset[j]]; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT64) { - Float64 *in = (Float64 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Float32) in[info.inOffset[j]]; - } - in += info.inJump; - out += info.outJump; - } - } - } - else if (info.outFormat == RTAUDIO_SINT32) { - Int32 *out = (Int32 *)outBuffer; - if (info.inFormat == RTAUDIO_SINT8) { - signed char *in = (signed char *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int32) in[info.inOffset[j]]; - out[info.outOffset[j]] <<= 24; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT16) { - Int16 *in = (Int16 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int32) in[info.inOffset[j]]; - out[info.outOffset[j]] <<= 16; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT24) { - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int32) in[info.inOffset[j]]; - out[info.outOffset[j]] <<= 8; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT32) { - // Channel compensation and/or (de)interleaving only. - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = in[info.inOffset[j]]; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT32) { - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT64) { - Float64 *in = (Float64 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5); - } - in += info.inJump; - out += info.outJump; - } - } - } - else if (info.outFormat == RTAUDIO_SINT24) { - Int32 *out = (Int32 *)outBuffer; - if (info.inFormat == RTAUDIO_SINT8) { - signed char *in = (signed char *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int32) in[info.inOffset[j]]; - out[info.outOffset[j]] <<= 16; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT16) { - Int16 *in = (Int16 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int32) in[info.inOffset[j]]; - out[info.outOffset[j]] <<= 8; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT24) { - // Channel compensation and/or (de)interleaving only. - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = in[info.inOffset[j]]; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT32) { - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int32) in[info.inOffset[j]]; - out[info.outOffset[j]] >>= 8; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT32) { - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT64) { - Float64 *in = (Float64 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5); - } - in += info.inJump; - out += info.outJump; - } - } - } - else if (info.outFormat == RTAUDIO_SINT16) { - Int16 *out = (Int16 *)outBuffer; - if (info.inFormat == RTAUDIO_SINT8) { - signed char *in = (signed char *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int16) in[info.inOffset[j]]; - out[info.outOffset[j]] <<= 8; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT16) { - // Channel compensation and/or (de)interleaving only. - Int16 *in = (Int16 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = in[info.inOffset[j]]; - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT24) { - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 8) & 0x0000ffff); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT32) { - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT32) { - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT64) { - Float64 *in = (Float64 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5); - } - in += info.inJump; - out += info.outJump; - } - } - } - else if (info.outFormat == RTAUDIO_SINT8) { - signed char *out = (signed char *)outBuffer; - if (info.inFormat == RTAUDIO_SINT8) { - // Channel compensation and/or (de)interleaving only. - signed char *in = (signed char *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = in[info.inOffset[j]]; - } - in += info.inJump; - out += info.outJump; - } - } - if (info.inFormat == RTAUDIO_SINT16) { - Int16 *in = (Int16 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT24) { - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 16) & 0x000000ff); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_SINT32) { - Int32 *in = (Int32 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT32) { - Float32 *in = (Float32 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5); - } - in += info.inJump; - out += info.outJump; - } - } - else if (info.inFormat == RTAUDIO_FLOAT64) { - Float64 *in = (Float64 *)inBuffer; - for (unsigned int i=0; i<stream_.bufferSize; i++) { - for (j=0; j<info.channels; j++) { - out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5); - } - in += info.inJump; - out += info.outJump; - } - } - } -} - - //static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); } - //static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); } - //static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); } - -void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ) -{ - register char val; - register char *ptr; - - ptr = buffer; - if ( format == RTAUDIO_SINT16 ) { - for ( unsigned int i=0; i<samples; i++ ) { - // Swap 1st and 2nd bytes. - val = *(ptr); - *(ptr) = *(ptr+1); - *(ptr+1) = val; - - // Increment 2 bytes. - ptr += 2; - } - } - else if ( format == RTAUDIO_SINT24 || - format == RTAUDIO_SINT32 || - format == RTAUDIO_FLOAT32 ) { - for ( unsigned int i=0; i<samples; i++ ) { - // Swap 1st and 4th bytes. - val = *(ptr); - *(ptr) = *(ptr+3); - *(ptr+3) = val; - - // Swap 2nd and 3rd bytes. - ptr += 1; - val = *(ptr); - *(ptr) = *(ptr+1); - *(ptr+1) = val; - - // Increment 3 more bytes. - ptr += 3; - } - } - else if ( format == RTAUDIO_FLOAT64 ) { - for ( unsigned int i=0; i<samples; i++ ) { - // Swap 1st and 8th bytes - val = *(ptr); - *(ptr) = *(ptr+7); - *(ptr+7) = val; - - // Swap 2nd and 7th bytes - ptr += 1; - val = *(ptr); - *(ptr) = *(ptr+5); - *(ptr+5) = val; - - // Swap 3rd and 6th bytes - ptr += 1; - val = *(ptr); - *(ptr) = *(ptr+3); - *(ptr+3) = val; - - // Swap 4th and 5th bytes - ptr += 1; - val = *(ptr); - *(ptr) = *(ptr+1); - *(ptr+1) = val; - - // Increment 5 more bytes. - ptr += 5; - } - } -} - - // Indentation settings for Vim and Emacs - // - // Local Variables: - // c-basic-offset: 2 - // indent-tabs-mode: nil - // End: - // - // vim: et sts=2 sw=2 - -#endif +#ifdef RTAUDIO_ENABLED
+/************************************************************************/
+/*! \class RtAudio
+ \brief Realtime audio i/o C++ classes.
+
+ RtAudio provides a common API (Application Programming Interface)
+ for realtime audio input/output across Linux (native ALSA, Jack,
+ and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
+ (DirectSound, ASIO and WASAPI) operating systems.
+
+ RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
+
+ RtAudio: realtime audio i/o C++ classes
+ Copyright (c) 2001-2014 Gary P. Scavone
+
+ Permission is hereby granted, free of charge, to any person
+ obtaining a copy of this software and associated documentation files
+ (the "Software"), to deal in the Software without restriction,
+ including without limitation the rights to use, copy, modify, merge,
+ publish, distribute, sublicense, and/or sell copies of the Software,
+ and to permit persons to whom the Software is furnished to do so,
+ subject to the following conditions:
+
+ The above copyright notice and this permission notice shall be
+ included in all copies or substantial portions of the Software.
+
+ Any person wishing to distribute modifications to the Software is
+ asked to send the modifications to the original developer so that
+ they can be incorporated into the canonical version. This is,
+ however, not a binding provision of this license.
+
+ THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+*/
+/************************************************************************/
+
+// RtAudio: Version 4.1.1
+
+#include "RtAudio.h"
+#include <iostream>
+#include <cstdlib>
+#include <cstring>
+#include <climits>
+
+// Static variable definitions.
+const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
+const unsigned int RtApi::SAMPLE_RATES[] = {
+ 4000, 5512, 8000, 9600, 11025, 16000, 22050,
+ 32000, 44100, 48000, 88200, 96000, 176400, 192000
+};
+
+#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
+#ifdef WINRT_ENABLED
+ #define MUTEX_INITIALIZE(A) InitializeCriticalSectionEx(A, 0, 0)
+#else
+ #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
+#endif
+ #define MUTEX_DESTROY(A) DeleteCriticalSection(A)
+ #define MUTEX_LOCK(A) EnterCriticalSection(A)
+ #define MUTEX_UNLOCK(A) LeaveCriticalSection(A)
+#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
+ // pthread API
+ #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
+ #define MUTEX_DESTROY(A) pthread_mutex_destroy(A)
+ #define MUTEX_LOCK(A) pthread_mutex_lock(A)
+ #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A)
+#else
+ #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions
+ #define MUTEX_DESTROY(A) abs(*A) // dummy definitions
+#endif
+
+// *************************************************** //
+//
+// RtAudio definitions.
+//
+// *************************************************** //
+
+std::string RtAudio :: getVersion( void ) throw()
+{
+ return RTAUDIO_VERSION;
+}
+
+void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()
+{
+ apis.clear();
+
+ // The order here will control the order of RtAudio's API search in
+ // the constructor.
+#if defined(__UNIX_JACK__)
+ apis.push_back( UNIX_JACK );
+#endif
+#if defined(__LINUX_ALSA__)
+ apis.push_back( LINUX_ALSA );
+#endif
+#if defined(__LINUX_PULSE__)
+ apis.push_back( LINUX_PULSE );
+#endif
+#if defined(__LINUX_OSS__)
+ apis.push_back( LINUX_OSS );
+#endif
+#if defined(__WINDOWS_ASIO__)
+ apis.push_back( WINDOWS_ASIO );
+#endif
+#if defined(__WINDOWS_WASAPI__)
+ apis.push_back( WINDOWS_WASAPI );
+#endif
+#if defined(__WINDOWS_DS__)
+ apis.push_back( WINDOWS_DS );
+#endif
+#if defined(__MACOSX_CORE__)
+ apis.push_back( MACOSX_CORE );
+#endif
+#if defined(__RTAUDIO_DUMMY__)
+ apis.push_back( RTAUDIO_DUMMY );
+#endif
+}
+
+void RtAudio :: openRtApi( RtAudio::Api api )
+{
+ if ( rtapi_ )
+ delete rtapi_;
+ rtapi_ = 0;
+
+#if defined(__UNIX_JACK__)
+ if ( api == UNIX_JACK )
+ rtapi_ = new RtApiJack();
+#endif
+#if defined(__LINUX_ALSA__)
+ if ( api == LINUX_ALSA )
+ rtapi_ = new RtApiAlsa();
+#endif
+#if defined(__LINUX_PULSE__)
+ if ( api == LINUX_PULSE )
+ rtapi_ = new RtApiPulse();
+#endif
+#if defined(__LINUX_OSS__)
+ if ( api == LINUX_OSS )
+ rtapi_ = new RtApiOss();
+#endif
+#if defined(__WINDOWS_ASIO__)
+ if ( api == WINDOWS_ASIO )
+ rtapi_ = new RtApiAsio();
+#endif
+#if defined(__WINDOWS_WASAPI__)
+ if ( api == WINDOWS_WASAPI )
+ rtapi_ = new RtApiWasapi();
+#endif
+#if defined(__WINDOWS_DS__)
+ if ( api == WINDOWS_DS )
+ rtapi_ = new RtApiDs();
+#endif
+#if defined(__MACOSX_CORE__)
+ if ( api == MACOSX_CORE )
+ rtapi_ = new RtApiCore();
+#endif
+#if defined(__RTAUDIO_DUMMY__)
+ if ( api == RTAUDIO_DUMMY )
+ rtapi_ = new RtApiDummy();
+#endif
+}
+
+RtAudio :: RtAudio( RtAudio::Api api )
+{
+ rtapi_ = 0;
+
+ if ( api != UNSPECIFIED ) {
+ // Attempt to open the specified API.
+ openRtApi( api );
+ if ( rtapi_ ) return;
+
+ // No compiled support for specified API value. Issue a debug
+ // warning and continue as if no API was specified.
+ std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
+ }
+
+ // Iterate through the compiled APIs and return as soon as we find
+ // one with at least one device or we reach the end of the list.
+ std::vector< RtAudio::Api > apis;
+ getCompiledApi( apis );
+ for ( unsigned int i=0; i<apis.size(); i++ ) {
+ openRtApi( apis[i] );
+ if ( rtapi_->getDeviceCount() ) break;
+ }
+
+ if ( rtapi_ ) return;
+
+ // It should not be possible to get here because the preprocessor
+ // definition __RTAUDIO_DUMMY__ is automatically defined if no
+ // API-specific definitions are passed to the compiler. But just in
+ // case something weird happens, we'll thow an error.
+ std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";
+ throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );
+}
+
+RtAudio :: ~RtAudio() throw()
+{
+ if ( rtapi_ )
+ delete rtapi_;
+}
+
+void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
+ RtAudio::StreamParameters *inputParameters,
+ RtAudioFormat format, unsigned int sampleRate,
+ unsigned int *bufferFrames,
+ RtAudioCallback callback, void *userData,
+ RtAudio::StreamOptions *options,
+ RtAudioErrorCallback errorCallback )
+{
+ return rtapi_->openStream( outputParameters, inputParameters, format,
+ sampleRate, bufferFrames, callback,
+ userData, options, errorCallback );
+}
+
+// *************************************************** //
+//
+// Public RtApi definitions (see end of file for
+// private or protected utility functions).
+//
+// *************************************************** //
+
+RtApi :: RtApi()
+{
+ stream_.state = STREAM_CLOSED;
+ stream_.mode = UNINITIALIZED;
+ stream_.apiHandle = 0;
+ stream_.userBuffer[0] = 0;
+ stream_.userBuffer[1] = 0;
+ MUTEX_INITIALIZE( &stream_.mutex );
+ showWarnings_ = true;
+ firstErrorOccurred_ = false;
+}
+
+RtApi :: ~RtApi()
+{
+ MUTEX_DESTROY( &stream_.mutex );
+}
+
+void RtApi :: openStream( RtAudio::StreamParameters *oParams,
+ RtAudio::StreamParameters *iParams,
+ RtAudioFormat format, unsigned int sampleRate,
+ unsigned int *bufferFrames,
+ RtAudioCallback callback, void *userData,
+ RtAudio::StreamOptions *options,
+ RtAudioErrorCallback errorCallback )
+{
+ if ( stream_.state != STREAM_CLOSED ) {
+ errorText_ = "RtApi::openStream: a stream is already open!";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+
+ // Clear stream information potentially left from a previously open stream.
+ clearStreamInfo();
+
+ if ( oParams && oParams->nChannels < 1 ) {
+ errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+
+ if ( iParams && iParams->nChannels < 1 ) {
+ errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+
+ if ( oParams == NULL && iParams == NULL ) {
+ errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+
+ if ( formatBytes(format) == 0 ) {
+ errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+
+ unsigned int nDevices = getDeviceCount();
+ unsigned int oChannels = 0;
+ if ( oParams ) {
+ oChannels = oParams->nChannels;
+ if ( oParams->deviceId >= nDevices ) {
+ errorText_ = "RtApi::openStream: output device parameter value is invalid.";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+ }
+
+ unsigned int iChannels = 0;
+ if ( iParams ) {
+ iChannels = iParams->nChannels;
+ if ( iParams->deviceId >= nDevices ) {
+ errorText_ = "RtApi::openStream: input device parameter value is invalid.";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+ }
+
+ bool result;
+
+ if ( oChannels > 0 ) {
+
+ result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
+ sampleRate, format, bufferFrames, options );
+ if ( result == false ) {
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ }
+
+ if ( iChannels > 0 ) {
+
+ result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
+ sampleRate, format, bufferFrames, options );
+ if ( result == false ) {
+ if ( oChannels > 0 ) closeStream();
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ }
+
+ stream_.callbackInfo.callback = (void *) callback;
+ stream_.callbackInfo.userData = userData;
+ stream_.callbackInfo.errorCallback = (void *) errorCallback;
+
+ if ( options ) options->numberOfBuffers = stream_.nBuffers;
+ stream_.state = STREAM_STOPPED;
+}
+
+unsigned int RtApi :: getDefaultInputDevice( void )
+{
+ // Should be implemented in subclasses if possible.
+ return 0;
+}
+
+unsigned int RtApi :: getDefaultOutputDevice( void )
+{
+ // Should be implemented in subclasses if possible.
+ return 0;
+}
+
+void RtApi :: closeStream( void )
+{
+ // MUST be implemented in subclasses!
+ return;
+}
+
+bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
+ unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
+ RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
+ RtAudio::StreamOptions * /*options*/ )
+{
+ // MUST be implemented in subclasses!
+ return FAILURE;
+}
+
+void RtApi :: tickStreamTime( void )
+{
+ // Subclasses that do not provide their own implementation of
+ // getStreamTime should call this function once per buffer I/O to
+ // provide basic stream time support.
+
+ stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
+
+#if defined( HAVE_GETTIMEOFDAY )
+ gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
+}
+
+long RtApi :: getStreamLatency( void )
+{
+ verifyStream();
+
+ long totalLatency = 0;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
+ totalLatency = stream_.latency[0];
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
+ totalLatency += stream_.latency[1];
+
+ return totalLatency;
+}
+
+double RtApi :: getStreamTime( void )
+{
+ verifyStream();
+
+#if defined( HAVE_GETTIMEOFDAY )
+ // Return a very accurate estimate of the stream time by
+ // adding in the elapsed time since the last tick.
+ struct timeval then;
+ struct timeval now;
+
+ if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
+ return stream_.streamTime;
+
+ gettimeofday( &now, NULL );
+ then = stream_.lastTickTimestamp;
+ return stream_.streamTime +
+ ((now.tv_sec + 0.000001 * now.tv_usec) -
+ (then.tv_sec + 0.000001 * then.tv_usec));
+#else
+ return stream_.streamTime;
+#endif
+}
+
+void RtApi :: setStreamTime( double time )
+{
+ verifyStream();
+
+ if ( time >= 0.0 )
+ stream_.streamTime = time;
+}
+
+unsigned int RtApi :: getStreamSampleRate( void )
+{
+ verifyStream();
+
+ return stream_.sampleRate;
+}
+
+
+// *************************************************** //
+//
+// OS/API-specific methods.
+//
+// *************************************************** //
+
+#if defined(__MACOSX_CORE__)
+
+// The OS X CoreAudio API is designed to use a separate callback
+// procedure for each of its audio devices. A single RtAudio duplex
+// stream using two different devices is supported here, though it
+// cannot be guaranteed to always behave correctly because we cannot
+// synchronize these two callbacks.
+//
+// A property listener is installed for over/underrun information.
+// However, no functionality is currently provided to allow property
+// listeners to trigger user handlers because it is unclear what could
+// be done if a critical stream parameter (buffer size, sample rate,
+// device disconnect) notification arrived. The listeners entail
+// quite a bit of extra code and most likely, a user program wouldn't
+// be prepared for the result anyway. However, we do provide a flag
+// to the client callback function to inform of an over/underrun.
+
+// A structure to hold various information related to the CoreAudio API
+// implementation.
+struct CoreHandle {
+ AudioDeviceID id[2]; // device ids
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ AudioDeviceIOProcID procId[2];
+#endif
+ UInt32 iStream[2]; // device stream index (or first if using multiple)
+ UInt32 nStreams[2]; // number of streams to use
+ bool xrun[2];
+ char *deviceBuffer;
+ pthread_cond_t condition;
+ int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
+
+ CoreHandle()
+ :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
+};
+
+RtApiCore:: RtApiCore()
+{
+#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )
+ // This is a largely undocumented but absolutely necessary
+ // requirement starting with OS-X 10.6. If not called, queries and
+ // updates to various audio device properties are not handled
+ // correctly.
+ CFRunLoopRef theRunLoop = NULL;
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";
+ error( RtAudioError::WARNING );
+ }
+#endif
+}
+
+RtApiCore :: ~RtApiCore()
+{
+ // The subclass destructor gets called before the base class
+ // destructor, so close an existing stream before deallocating
+ // apiDeviceId memory.
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
+
+unsigned int RtApiCore :: getDeviceCount( void )
+{
+ // Find out how many audio devices there are, if any.
+ UInt32 dataSize;
+ AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
+ error( RtAudioError::WARNING );
+ return 0;
+ }
+
+ return dataSize / sizeof( AudioDeviceID );
+}
+
+unsigned int RtApiCore :: getDefaultInputDevice( void )
+{
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices <= 1 ) return 0;
+
+ AudioDeviceID id;
+ UInt32 dataSize = sizeof( AudioDeviceID );
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
+ error( RtAudioError::WARNING );
+ return 0;
+ }
+
+ dataSize *= nDevices;
+ AudioDeviceID deviceList[ nDevices ];
+ property.mSelector = kAudioHardwarePropertyDevices;
+ result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
+ error( RtAudioError::WARNING );
+ return 0;
+ }
+
+ for ( unsigned int i=0; i<nDevices; i++ )
+ if ( id == deviceList[i] ) return i;
+
+ errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
+ error( RtAudioError::WARNING );
+ return 0;
+}
+
+unsigned int RtApiCore :: getDefaultOutputDevice( void )
+{
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices <= 1 ) return 0;
+
+ AudioDeviceID id;
+ UInt32 dataSize = sizeof( AudioDeviceID );
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
+ error( RtAudioError::WARNING );
+ return 0;
+ }
+
+ dataSize = sizeof( AudioDeviceID ) * nDevices;
+ AudioDeviceID deviceList[ nDevices ];
+ property.mSelector = kAudioHardwarePropertyDevices;
+ result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
+ error( RtAudioError::WARNING );
+ return 0;
+ }
+
+ for ( unsigned int i=0; i<nDevices; i++ )
+ if ( id == deviceList[i] ) return i;
+
+ errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
+ error( RtAudioError::WARNING );
+ return 0;
+}
+
+RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ // Get device ID
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices == 0 ) {
+ errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ AudioDeviceID deviceList[ nDevices ];
+ UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
+ 0, NULL, &dataSize, (void *) &deviceList );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ AudioDeviceID id = deviceList[ device ];
+
+ // Get the device name.
+ info.name.erase();
+ CFStringRef cfname;
+ dataSize = sizeof( CFStringRef );
+ property.mSelector = kAudioObjectPropertyManufacturer;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
+ int length = CFStringGetLength(cfname);
+ char *mname = (char *)malloc(length * 3 + 1);
+#if defined( UNICODE ) || defined( _UNICODE )
+ CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8);
+#else
+ CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());
+#endif
+ info.name.append( (const char *)mname, strlen(mname) );
+ info.name.append( ": " );
+ CFRelease( cfname );
+ free(mname);
+
+ property.mSelector = kAudioObjectPropertyName;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
+ length = CFStringGetLength(cfname);
+ char *name = (char *)malloc(length * 3 + 1);
+#if defined( UNICODE ) || defined( _UNICODE )
+ CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8);
+#else
+ CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());
+#endif
+ info.name.append( (const char *)name, strlen(name) );
+ CFRelease( cfname );
+ free(name);
+
+ // Get the output stream "configuration".
+ AudioBufferList *bufferList = nil;
+ property.mSelector = kAudioDevicePropertyStreamConfiguration;
+ property.mScope = kAudioDevicePropertyScopeOutput;
+ // property.mElement = kAudioObjectPropertyElementWildcard;
+ dataSize = 0;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+ if ( result != noErr || dataSize == 0 ) {
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Allocate the AudioBufferList.
+ bufferList = (AudioBufferList *) malloc( dataSize );
+ if ( bufferList == NULL ) {
+ errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+ if ( result != noErr || dataSize == 0 ) {
+ free( bufferList );
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Get output channel information.
+ unsigned int i, nStreams = bufferList->mNumberBuffers;
+ for ( i=0; i<nStreams; i++ )
+ info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
+ free( bufferList );
+
+ // Get the input stream "configuration".
+ property.mScope = kAudioDevicePropertyScopeInput;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+ if ( result != noErr || dataSize == 0 ) {
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Allocate the AudioBufferList.
+ bufferList = (AudioBufferList *) malloc( dataSize );
+ if ( bufferList == NULL ) {
+ errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+ if (result != noErr || dataSize == 0) {
+ free( bufferList );
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Get input channel information.
+ nStreams = bufferList->mNumberBuffers;
+ for ( i=0; i<nStreams; i++ )
+ info.inputChannels += bufferList->mBuffers[i].mNumberChannels;
+ free( bufferList );
+
+ // If device opens for both playback and capture, we determine the channels.
+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+
+ // Probe the device sample rates.
+ bool isInput = false;
+ if ( info.outputChannels == 0 ) isInput = true;
+
+ // Determine the supported sample rates.
+ property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
+ if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+ if ( result != kAudioHardwareNoError || dataSize == 0 ) {
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ UInt32 nRanges = dataSize / sizeof( AudioValueRange );
+ AudioValueRange rangeList[ nRanges ];
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );
+ if ( result != kAudioHardwareNoError ) {
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // The sample rate reporting mechanism is a bit of a mystery. It
+ // seems that it can either return individual rates or a range of
+ // rates. I assume that if the min / max range values are the same,
+ // then that represents a single supported rate and if the min / max
+ // range values are different, the device supports an arbitrary
+ // range of values (though there might be multiple ranges, so we'll
+ // use the most conservative range).
+ Float64 minimumRate = 1.0, maximumRate = 10000000000.0;
+ bool haveValueRange = false;
+ info.sampleRates.clear();
+ for ( UInt32 i=0; i<nRanges; i++ ) {
+ if ( rangeList[i].mMinimum == rangeList[i].mMaximum )
+ info.sampleRates.push_back( (unsigned int) rangeList[i].mMinimum );
+ else {
+ haveValueRange = true;
+ if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;
+ if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;
+ }
+ }
+
+ if ( haveValueRange ) {
+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+ if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate )
+ info.sampleRates.push_back( SAMPLE_RATES[k] );
+ }
+ }
+
+ // Sort and remove any redundant values
+ std::sort( info.sampleRates.begin(), info.sampleRates.end() );
+ info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );
+
+ if ( info.sampleRates.size() == 0 ) {
+ errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // CoreAudio always uses 32-bit floating point data for PCM streams.
+ // Thus, any other "physical" formats supported by the device are of
+ // no interest to the client.
+ info.nativeFormats = RTAUDIO_FLOAT32;
+
+ if ( info.outputChannels > 0 )
+ if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
+ if ( info.inputChannels > 0 )
+ if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
+
+ info.probed = true;
+ return info;
+}
+
+static OSStatus callbackHandler( AudioDeviceID inDevice,
+ const AudioTimeStamp* /*inNow*/,
+ const AudioBufferList* inInputData,
+ const AudioTimeStamp* /*inInputTime*/,
+ AudioBufferList* outOutputData,
+ const AudioTimeStamp* /*inOutputTime*/,
+ void* infoPointer )
+{
+ CallbackInfo *info = (CallbackInfo *) infoPointer;
+
+ RtApiCore *object = (RtApiCore *) info->object;
+ if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
+ return kAudioHardwareUnspecifiedError;
+ else
+ return kAudioHardwareNoError;
+}
+
+static OSStatus xrunListener( AudioObjectID /*inDevice*/,
+ UInt32 nAddresses,
+ const AudioObjectPropertyAddress properties[],
+ void* handlePointer )
+{
+ CoreHandle *handle = (CoreHandle *) handlePointer;
+ for ( UInt32 i=0; i<nAddresses; i++ ) {
+ if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) {
+ if ( properties[i].mScope == kAudioDevicePropertyScopeInput )
+ handle->xrun[1] = true;
+ else
+ handle->xrun[0] = true;
+ }
+ }
+
+ return kAudioHardwareNoError;
+}
+
+static OSStatus rateListener( AudioObjectID inDevice,
+ UInt32 /*nAddresses*/,
+ const AudioObjectPropertyAddress /*properties*/[],
+ void* ratePointer )
+{
+ Float64 *rate = (Float64 *) ratePointer;
+ UInt32 dataSize = sizeof( Float64 );
+ AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate );
+ return kAudioHardwareNoError;
+}
+
+bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ // Get device ID
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices == 0 ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";
+ return FAILURE;
+ }
+
+ if ( device >= nDevices ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
+
+ AudioDeviceID deviceList[ nDevices ];
+ UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
+ 0, NULL, &dataSize, (void *) &deviceList );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";
+ return FAILURE;
+ }
+
+ AudioDeviceID id = deviceList[ device ];
+
+ // Setup for stream mode.
+ bool isInput = false;
+ if ( mode == INPUT ) {
+ isInput = true;
+ property.mScope = kAudioDevicePropertyScopeInput;
+ }
+ else
+ property.mScope = kAudioDevicePropertyScopeOutput;
+
+ // Get the stream "configuration".
+ AudioBufferList *bufferList = nil;
+ dataSize = 0;
+ property.mSelector = kAudioDevicePropertyStreamConfiguration;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+ if ( result != noErr || dataSize == 0 ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Allocate the AudioBufferList.
+ bufferList = (AudioBufferList *) malloc( dataSize );
+ if ( bufferList == NULL ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";
+ return FAILURE;
+ }
+
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+ if (result != noErr || dataSize == 0) {
+ free( bufferList );
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Search for one or more streams that contain the desired number of
+ // channels. CoreAudio devices can have an arbitrary number of
+ // streams and each stream can have an arbitrary number of channels.
+ // For each stream, a single buffer of interleaved samples is
+ // provided. RtAudio prefers the use of one stream of interleaved
+ // data or multiple consecutive single-channel streams. However, we
+ // now support multiple consecutive multi-channel streams of
+ // interleaved data as well.
+ UInt32 iStream, offsetCounter = firstChannel;
+ UInt32 nStreams = bufferList->mNumberBuffers;
+ bool monoMode = false;
+ bool foundStream = false;
+
+ // First check that the device supports the requested number of
+ // channels.
+ UInt32 deviceChannels = 0;
+ for ( iStream=0; iStream<nStreams; iStream++ )
+ deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;
+
+ if ( deviceChannels < ( channels + firstChannel ) ) {
+ free( bufferList );
+ errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Look for a single stream meeting our needs.
+ UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;
+ for ( iStream=0; iStream<nStreams; iStream++ ) {
+ streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
+ if ( streamChannels >= channels + offsetCounter ) {
+ firstStream = iStream;
+ channelOffset = offsetCounter;
+ foundStream = true;
+ break;
+ }
+ if ( streamChannels > offsetCounter ) break;
+ offsetCounter -= streamChannels;
+ }
+
+ // If we didn't find a single stream above, then we should be able
+ // to meet the channel specification with multiple streams.
+ if ( foundStream == false ) {
+ monoMode = true;
+ offsetCounter = firstChannel;
+ for ( iStream=0; iStream<nStreams; iStream++ ) {
+ streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
+ if ( streamChannels > offsetCounter ) break;
+ offsetCounter -= streamChannels;
+ }
+
+ firstStream = iStream;
+ channelOffset = offsetCounter;
+ Int32 channelCounter = channels + offsetCounter - streamChannels;
+
+ if ( streamChannels > 1 ) monoMode = false;
+ while ( channelCounter > 0 ) {
+ streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;
+ if ( streamChannels > 1 ) monoMode = false;
+ channelCounter -= streamChannels;
+ streamCount++;
+ }
+ }
+
+ free( bufferList );
+
+ // Determine the buffer size.
+ AudioValueRange bufferRange;
+ dataSize = sizeof( AudioValueRange );
+ property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange );
+
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;
+ else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;
+
+ // Set the buffer size. For multiple streams, I'm assuming we only
+ // need to make this setting for the master channel.
+ UInt32 theSize = (UInt32) *bufferSize;
+ dataSize = sizeof( UInt32 );
+ property.mSelector = kAudioDevicePropertyBufferFrameSize;
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize );
+
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // If attempting to setup a duplex stream, the bufferSize parameter
+ // MUST be the same in both directions!
+ *bufferSize = theSize;
+ if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ stream_.bufferSize = *bufferSize;
+ stream_.nBuffers = 1;
+
+ // Try to set "hog" mode ... it's not clear to me this is working.
+ if ( options && options->flags & RTAUDIO_HOG_DEVICE ) {
+ pid_t hog_pid;
+ dataSize = sizeof( hog_pid );
+ property.mSelector = kAudioDevicePropertyHogMode;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ if ( hog_pid != getpid() ) {
+ hog_pid = getpid();
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+ }
+
+ // Check and if necessary, change the sample rate for the device.
+ Float64 nominalRate;
+ dataSize = sizeof( Float64 );
+ property.mSelector = kAudioDevicePropertyNominalSampleRate;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Only change the sample rate if off by more than 1 Hz.
+ if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {
+
+ // Set a property listener for the sample rate change
+ Float64 reportedRate = 0.0;
+ AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+ result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ nominalRate = (Float64) sampleRate;
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );
+ if ( result != noErr ) {
+ AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Now wait until the reported nominal rate is what we just set.
+ UInt32 microCounter = 0;
+ while ( reportedRate != nominalRate ) {
+ microCounter += 5000;
+ if ( microCounter > 5000000 ) break;
+ usleep( 5000 );
+ }
+
+ // Remove the property listener.
+ AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
+
+ if ( microCounter > 5000000 ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // Now set the stream format for all streams. Also, check the
+ // physical format of the device and change that if necessary.
+ AudioStreamBasicDescription description;
+ dataSize = sizeof( AudioStreamBasicDescription );
+ property.mSelector = kAudioStreamPropertyVirtualFormat;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Set the sample rate and data format id. However, only make the
+ // change if the sample rate is not within 1.0 of the desired
+ // rate and the format is not linear pcm.
+ bool updateFormat = false;
+ if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) {
+ description.mSampleRate = (Float64) sampleRate;
+ updateFormat = true;
+ }
+
+ if ( description.mFormatID != kAudioFormatLinearPCM ) {
+ description.mFormatID = kAudioFormatLinearPCM;
+ updateFormat = true;
+ }
+
+ if ( updateFormat ) {
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // Now check the physical format.
+ property.mSelector = kAudioStreamPropertyPhysicalFormat;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ //std::cout << "Current physical stream format:" << std::endl;
+ //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl;
+ //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
+ //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl;
+ //std::cout << " sample rate = " << description.mSampleRate << std::endl;
+
+ if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) {
+ description.mFormatID = kAudioFormatLinearPCM;
+ //description.mSampleRate = (Float64) sampleRate;
+ AudioStreamBasicDescription testDescription = description;
+ UInt32 formatFlags;
+
+ // We'll try higher bit rates first and then work our way down.
+ std::vector< std::pair<UInt32, UInt32> > physicalFormats;
+ formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
+ formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) ); // 24-bit packed
+ formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh );
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low
+ formatFlags |= kAudioFormatFlagIsAlignedHigh;
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high
+ formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) );
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) );
+
+ bool setPhysicalFormat = false;
+ for( unsigned int i=0; i<physicalFormats.size(); i++ ) {
+ testDescription = description;
+ testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first;
+ testDescription.mFormatFlags = physicalFormats[i].second;
+ if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) )
+ testDescription.mBytesPerFrame = 4 * testDescription.mChannelsPerFrame;
+ else
+ testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
+ testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription );
+ if ( result == noErr ) {
+ setPhysicalFormat = true;
+ //std::cout << "Updated physical stream format:" << std::endl;
+ //std::cout << " mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;
+ //std::cout << " aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
+ //std::cout << " bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;
+ //std::cout << " sample rate = " << testDescription.mSampleRate << std::endl;
+ break;
+ }
+ }
+
+ if ( !setPhysicalFormat ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ } // done setting virtual/physical formats.
+
+ // Get the stream / device latency.
+ UInt32 latency;
+ dataSize = sizeof( UInt32 );
+ property.mSelector = kAudioDevicePropertyLatency;
+ if ( AudioObjectHasProperty( id, &property ) == true ) {
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &latency );
+ if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
+ else {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ }
+ }
+
+ // Byte-swapping: According to AudioHardware.h, the stream data will
+ // always be presented in native-endian format, so we should never
+ // need to byte swap.
+ stream_.doByteSwap[mode] = false;
+
+ // From the CoreAudio documentation, PCM data must be supplied as
+ // 32-bit floats.
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+
+ if ( streamCount == 1 )
+ stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
+ else // multiple streams
+ stream_.nDeviceChannels[mode] = channels;
+ stream_.nUserChannels[mode] = channels;
+ stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
+ stream_.deviceInterleaved[mode] = true;
+ if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( streamCount == 1 ) {
+ if ( stream_.nUserChannels[mode] > 1 &&
+ stream_.userInterleaved != stream_.deviceInterleaved[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ }
+ else if ( monoMode && stream_.userInterleaved )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate our CoreHandle structure for the stream.
+ CoreHandle *handle = 0;
+ if ( stream_.apiHandle == 0 ) {
+ try {
+ handle = new CoreHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";
+ goto error;
+ }
+
+ if ( pthread_cond_init( &handle->condition, NULL ) ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";
+ goto error;
+ }
+ stream_.apiHandle = (void *) handle;
+ }
+ else
+ handle = (CoreHandle *) stream_.apiHandle;
+ handle->iStream[mode] = firstStream;
+ handle->nStreams[mode] = streamCount;
+ handle->id[mode] = id;
+
+ // Allocate necessary internal buffers.
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) );
+ memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+
+ // If possible, we will make use of the CoreAudio stream buffers as
+ // "device buffers". However, we can't do this if using multiple
+ // streams.
+ if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {
+
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
+ }
+ }
+
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
+
+ stream_.sampleRate = sampleRate;
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ stream_.callbackInfo.object = (void *) this;
+
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) {
+ if ( streamCount > 1 ) setConvertInfo( mode, 0 );
+ else setConvertInfo( mode, channelOffset );
+ }
+
+ if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
+ // Only one callback procedure per device.
+ stream_.mode = DUPLEX;
+ else {
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );
+#else
+ // deprecated in favor of AudioDeviceCreateIOProcID()
+ result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
+#endif
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ stream_.mode = DUPLEX;
+ else
+ stream_.mode = mode;
+ }
+
+ // Setup the device property listener for over/underload.
+ property.mSelector = kAudioDeviceProcessorOverload;
+ property.mScope = kAudioObjectPropertyScopeGlobal;
+ result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle );
+
+ return SUCCESS;
+
+ error:
+ if ( handle ) {
+ pthread_cond_destroy( &handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.state = STREAM_CLOSED;
+ return FAILURE;
+}
+
+void RtApiCore :: closeStream( void )
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiCore::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( stream_.state == STREAM_RUNNING )
+ AudioDeviceStop( handle->id[0], callbackHandler );
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
+#else
+ // deprecated in favor of AudioDeviceDestroyIOProcID()
+ AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
+#endif
+ }
+
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+ if ( stream_.state == STREAM_RUNNING )
+ AudioDeviceStop( handle->id[1], callbackHandler );
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
+#else
+ // deprecated in favor of AudioDeviceDestroyIOProcID()
+ AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
+#endif
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ // Destroy pthread condition variable.
+ pthread_cond_destroy( &handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
+
+void RtApiCore :: startStream( void )
+{
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiCore::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ OSStatus result = noErr;
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ result = AudioDeviceStart( handle->id[0], callbackHandler );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ if ( stream_.mode == INPUT ||
+ ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+
+ result = AudioDeviceStart( handle->id[1], callbackHandler );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ stream_.state = STREAM_RUNNING;
+
+ unlock:
+ if ( result == noErr ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiCore :: stopStream( void )
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ OSStatus result = noErr;
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 2;
+ pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
+ }
+
+ result = AudioDeviceStop( handle->id[0], callbackHandler );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+
+ result = AudioDeviceStop( handle->id[1], callbackHandler );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ stream_.state = STREAM_STOPPED;
+
+ unlock:
+ if ( result == noErr ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiCore :: abortStream( void )
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ handle->drainCounter = 2;
+
+ stopStream();
+}
+
+// This function will be called by a spawned thread when the user
+// callback function signals that the stream should be stopped or
+// aborted. It is better to handle it this way because the
+// callbackEvent() function probably should return before the AudioDeviceStop()
+// function is called.
+static void *coreStopStream( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiCore *object = (RtApiCore *) info->object;
+
+ object->stopStream();
+ pthread_exit( NULL );
+}
+
+bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
+ const AudioBufferList *inBufferList,
+ const AudioBufferList *outBufferList )
+{
+ if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return FAILURE;
+ }
+
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+
+ // Check if we were draining the stream and signal is finished.
+ if ( handle->drainCounter > 3 ) {
+ ThreadHandle threadId;
+
+ stream_.state = STREAM_STOPPING;
+ if ( handle->internalDrain == true )
+ pthread_create( &threadId, NULL, coreStopStream, info );
+ else // external call to stopStream()
+ pthread_cond_signal( &handle->condition );
+ return SUCCESS;
+ }
+
+ AudioDeviceID outputDevice = handle->id[0];
+
+ // Invoke user callback to get fresh output data UNLESS we are
+ // draining stream or duplex mode AND the input/output devices are
+ // different AND this function is called for the input device.
+ if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
+ }
+
+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( cbReturnValue == 2 ) {
+ stream_.state = STREAM_STOPPING;
+ handle->drainCounter = 2;
+ abortStream();
+ return SUCCESS;
+ }
+ else if ( cbReturnValue == 1 ) {
+ handle->drainCounter = 1;
+ handle->internalDrain = true;
+ }
+ }
+
+ if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
+
+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+
+ if ( handle->nStreams[0] == 1 ) {
+ memset( outBufferList->mBuffers[handle->iStream[0]].mData,
+ 0,
+ outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
+ }
+ else { // fill multiple streams with zeros
+ for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
+ memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
+ 0,
+ outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
+ }
+ }
+ }
+ else if ( handle->nStreams[0] == 1 ) {
+ if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
+ convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
+ stream_.userBuffer[0], stream_.convertInfo[0] );
+ }
+ else { // copy from user buffer
+ memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
+ stream_.userBuffer[0],
+ outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
+ }
+ }
+ else { // fill multiple streams
+ Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
+ if ( stream_.doConvertBuffer[0] ) {
+ convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ inBuffer = (Float32 *) stream_.deviceBuffer;
+ }
+
+ if ( stream_.deviceInterleaved[0] == false ) { // mono mode
+ UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
+ (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
+ }
+ }
+ else { // fill multiple multi-channel streams with interleaved data
+ UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
+ Float32 *out, *in;
+
+ bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
+ UInt32 inChannels = stream_.nUserChannels[0];
+ if ( stream_.doConvertBuffer[0] ) {
+ inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
+ inChannels = stream_.nDeviceChannels[0];
+ }
+
+ if ( inInterleaved ) inOffset = 1;
+ else inOffset = stream_.bufferSize;
+
+ channelsLeft = inChannels;
+ for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
+ in = inBuffer;
+ out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
+ streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;
+
+ outJump = 0;
+ // Account for possible channel offset in first stream
+ if ( i == 0 && stream_.channelOffset[0] > 0 ) {
+ streamChannels -= stream_.channelOffset[0];
+ outJump = stream_.channelOffset[0];
+ out += outJump;
+ }
+
+ // Account for possible unfilled channels at end of the last stream
+ if ( streamChannels > channelsLeft ) {
+ outJump = streamChannels - channelsLeft;
+ streamChannels = channelsLeft;
+ }
+
+ // Determine input buffer offsets and skips
+ if ( inInterleaved ) {
+ inJump = inChannels;
+ in += inChannels - channelsLeft;
+ }
+ else {
+ inJump = 1;
+ in += (inChannels - channelsLeft) * inOffset;
+ }
+
+ for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
+ for ( unsigned int j=0; j<streamChannels; j++ ) {
+ *out++ = in[j*inOffset];
+ }
+ out += outJump;
+ in += inJump;
+ }
+ channelsLeft -= streamChannels;
+ }
+ }
+ }
+ }
+
+ // Don't bother draining input
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
+ }
+
+ AudioDeviceID inputDevice;
+ inputDevice = handle->id[1];
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
+
+ if ( handle->nStreams[1] == 1 ) {
+ if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer
+ convertBuffer( stream_.userBuffer[1],
+ (char *) inBufferList->mBuffers[handle->iStream[1]].mData,
+ stream_.convertInfo[1] );
+ }
+ else { // copy to user buffer
+ memcpy( stream_.userBuffer[1],
+ inBufferList->mBuffers[handle->iStream[1]].mData,
+ inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
+ }
+ }
+ else { // read from multiple streams
+ Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];
+ if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;
+
+ if ( stream_.deviceInterleaved[1] == false ) { // mono mode
+ UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ memcpy( (void *)&outBuffer[i*stream_.bufferSize],
+ inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
+ }
+ }
+ else { // read from multiple multi-channel streams
+ UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;
+ Float32 *out, *in;
+
+ bool outInterleaved = ( stream_.userInterleaved ) ? true : false;
+ UInt32 outChannels = stream_.nUserChannels[1];
+ if ( stream_.doConvertBuffer[1] ) {
+ outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
+ outChannels = stream_.nDeviceChannels[1];
+ }
+
+ if ( outInterleaved ) outOffset = 1;
+ else outOffset = stream_.bufferSize;
+
+ channelsLeft = outChannels;
+ for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {
+ out = outBuffer;
+ in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;
+ streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;
+
+ inJump = 0;
+ // Account for possible channel offset in first stream
+ if ( i == 0 && stream_.channelOffset[1] > 0 ) {
+ streamChannels -= stream_.channelOffset[1];
+ inJump = stream_.channelOffset[1];
+ in += inJump;
+ }
+
+ // Account for possible unread channels at end of the last stream
+ if ( streamChannels > channelsLeft ) {
+ inJump = streamChannels - channelsLeft;
+ streamChannels = channelsLeft;
+ }
+
+ // Determine output buffer offsets and skips
+ if ( outInterleaved ) {
+ outJump = outChannels;
+ out += outChannels - channelsLeft;
+ }
+ else {
+ outJump = 1;
+ out += (outChannels - channelsLeft) * outOffset;
+ }
+
+ for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
+ for ( unsigned int j=0; j<streamChannels; j++ ) {
+ out[j*outOffset] = *in++;
+ }
+ out += outJump;
+ in += inJump;
+ }
+ channelsLeft -= streamChannels;
+ }
+ }
+
+ if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
+ convertBuffer( stream_.userBuffer[1],
+ stream_.deviceBuffer,
+ stream_.convertInfo[1] );
+ }
+ }
+ }
+
+ unlock:
+ //MUTEX_UNLOCK( &stream_.mutex );
+
+ RtApi::tickStreamTime();
+ return SUCCESS;
+}
+
+const char* RtApiCore :: getErrorCode( OSStatus code )
+{
+ switch( code ) {
+
+ case kAudioHardwareNotRunningError:
+ return "kAudioHardwareNotRunningError";
+
+ case kAudioHardwareUnspecifiedError:
+ return "kAudioHardwareUnspecifiedError";
+
+ case kAudioHardwareUnknownPropertyError:
+ return "kAudioHardwareUnknownPropertyError";
+
+ case kAudioHardwareBadPropertySizeError:
+ return "kAudioHardwareBadPropertySizeError";
+
+ case kAudioHardwareIllegalOperationError:
+ return "kAudioHardwareIllegalOperationError";
+
+ case kAudioHardwareBadObjectError:
+ return "kAudioHardwareBadObjectError";
+
+ case kAudioHardwareBadDeviceError:
+ return "kAudioHardwareBadDeviceError";
+
+ case kAudioHardwareBadStreamError:
+ return "kAudioHardwareBadStreamError";
+
+ case kAudioHardwareUnsupportedOperationError:
+ return "kAudioHardwareUnsupportedOperationError";
+
+ case kAudioDeviceUnsupportedFormatError:
+ return "kAudioDeviceUnsupportedFormatError";
+
+ case kAudioDevicePermissionsError:
+ return "kAudioDevicePermissionsError";
+
+ default:
+ return "CoreAudio unknown error";
+ }
+}
+
+ //******************** End of __MACOSX_CORE__ *********************//
+#endif
+
+#if defined(__UNIX_JACK__)
+
+// JACK is a low-latency audio server, originally written for the
+// GNU/Linux operating system and now also ported to OS-X. It can
+// connect a number of different applications to an audio device, as
+// well as allowing them to share audio between themselves.
+//
+// When using JACK with RtAudio, "devices" refer to JACK clients that
+// have ports connected to the server. The JACK server is typically
+// started in a terminal as follows:
+//
+// .jackd -d alsa -d hw:0
+//
+// or through an interface program such as qjackctl. Many of the
+// parameters normally set for a stream are fixed by the JACK server
+// and can be specified when the JACK server is started. In
+// particular,
+//
+// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
+//
+// specifies a sample rate of 44100 Hz, a buffer size of 512 sample
+// frames, and number of buffers = 4. Once the server is running, it
+// is not possible to override these values. If the values are not
+// specified in the command-line, the JACK server uses default values.
+//
+// The JACK server does not have to be running when an instance of
+// RtApiJack is created, though the function getDeviceCount() will
+// report 0 devices found until JACK has been started. When no
+// devices are available (i.e., the JACK server is not running), a
+// stream cannot be opened.
+
+#include <jack/jack.h>
+#include <unistd.h>
+#include <cstdio>
+
+// A structure to hold various information related to the Jack API
+// implementation.
+struct JackHandle {
+ jack_client_t *client;
+ jack_port_t **ports[2];
+ std::string deviceName[2];
+ bool xrun[2];
+ pthread_cond_t condition;
+ int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
+
+ JackHandle()
+ :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
+};
+
+static void jackSilentError( const char * ) {};
+
+RtApiJack :: RtApiJack()
+{
+ // Nothing to do here.
+#if !defined(__RTAUDIO_DEBUG__)
+ // Turn off Jack's internal error reporting.
+ jack_set_error_function( &jackSilentError );
+#endif
+}
+
+RtApiJack :: ~RtApiJack()
+{
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
+
+unsigned int RtApiJack :: getDeviceCount( void )
+{
+ // See if we can become a jack client.
+ jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
+ jack_status_t *status = NULL;
+ jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
+ if ( client == 0 ) return 0;
+
+ const char **ports;
+ std::string port, previousPort;
+ unsigned int nChannels = 0, nDevices = 0;
+ ports = jack_get_ports( client, NULL, NULL, 0 );
+ if ( ports ) {
+ // Parse the port names up to the first colon (:).
+ size_t iColon = 0;
+ do {
+ port = (char *) ports[ nChannels ];
+ iColon = port.find(":");
+ if ( iColon != std::string::npos ) {
+ port = port.substr( 0, iColon + 1 );
+ if ( port != previousPort ) {
+ nDevices++;
+ previousPort = port;
+ }
+ }
+ } while ( ports[++nChannels] );
+ free( ports );
+ }
+
+ jack_client_close( client );
+ return nDevices;
+}
+
+RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption
+ jack_status_t *status = NULL;
+ jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
+ if ( client == 0 ) {
+ errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ const char **ports;
+ std::string port, previousPort;
+ unsigned int nPorts = 0, nDevices = 0;
+ ports = jack_get_ports( client, NULL, NULL, 0 );
+ if ( ports ) {
+ // Parse the port names up to the first colon (:).
+ size_t iColon = 0;
+ do {
+ port = (char *) ports[ nPorts ];
+ iColon = port.find(":");
+ if ( iColon != std::string::npos ) {
+ port = port.substr( 0, iColon );
+ if ( port != previousPort ) {
+ if ( nDevices == device ) info.name = port;
+ nDevices++;
+ previousPort = port;
+ }
+ }
+ } while ( ports[++nPorts] );
+ free( ports );
+ }
+
+ if ( device >= nDevices ) {
+ jack_client_close( client );
+ errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ // Get the current jack server sample rate.
+ info.sampleRates.clear();
+ info.sampleRates.push_back( jack_get_sample_rate( client ) );
+
+ // Count the available ports containing the client name as device
+ // channels. Jack "input ports" equal RtAudio output channels.
+ unsigned int nChannels = 0;
+ ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
+ if ( ports ) {
+ while ( ports[ nChannels ] ) nChannels++;
+ free( ports );
+ info.outputChannels = nChannels;
+ }
+
+ // Jack "output ports" equal RtAudio input channels.
+ nChannels = 0;
+ ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
+ if ( ports ) {
+ while ( ports[ nChannels ] ) nChannels++;
+ free( ports );
+ info.inputChannels = nChannels;
+ }
+
+ if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
+ jack_client_close(client);
+ errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // If device opens for both playback and capture, we determine the channels.
+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+
+ // Jack always uses 32-bit floats.
+ info.nativeFormats = RTAUDIO_FLOAT32;
+
+ // Jack doesn't provide default devices so we'll use the first available one.
+ if ( device == 0 && info.outputChannels > 0 )
+ info.isDefaultOutput = true;
+ if ( device == 0 && info.inputChannels > 0 )
+ info.isDefaultInput = true;
+
+ jack_client_close(client);
+ info.probed = true;
+ return info;
+}
+
+static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
+{
+ CallbackInfo *info = (CallbackInfo *) infoPointer;
+
+ RtApiJack *object = (RtApiJack *) info->object;
+ if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
+
+ return 0;
+}
+
+// This function will be called by a spawned thread when the Jack
+// server signals that it is shutting down. It is necessary to handle
+// it this way because the jackShutdown() function must return before
+// the jack_deactivate() function (in closeStream()) will return.
+static void *jackCloseStream( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiJack *object = (RtApiJack *) info->object;
+
+ object->closeStream();
+
+ pthread_exit( NULL );
+}
+static void jackShutdown( void *infoPointer )
+{
+ CallbackInfo *info = (CallbackInfo *) infoPointer;
+ RtApiJack *object = (RtApiJack *) info->object;
+
+ // Check current stream state. If stopped, then we'll assume this
+ // was called as a result of a call to RtApiJack::stopStream (the
+ // deactivation of a client handle causes this function to be called).
+ // If not, we'll assume the Jack server is shutting down or some
+ // other problem occurred and we should close the stream.
+ if ( object->isStreamRunning() == false ) return;
+
+ ThreadHandle threadId;
+ pthread_create( &threadId, NULL, jackCloseStream, info );
+ std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
+}
+
+static int jackXrun( void *infoPointer )
+{
+ JackHandle *handle = (JackHandle *) infoPointer;
+
+ if ( handle->ports[0] ) handle->xrun[0] = true;
+ if ( handle->ports[1] ) handle->xrun[1] = true;
+
+ return 0;
+}
+
+bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+
+ // Look for jack server and try to become a client (only do once per stream).
+ jack_client_t *client = 0;
+ if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
+ jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
+ jack_status_t *status = NULL;
+ if ( options && !options->streamName.empty() )
+ client = jack_client_open( options->streamName.c_str(), jackoptions, status );
+ else
+ client = jack_client_open( "RtApiJack", jackoptions, status );
+ if ( client == 0 ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
+ error( RtAudioError::WARNING );
+ return FAILURE;
+ }
+ }
+ else {
+ // The handle must have been created on an earlier pass.
+ client = handle->client;
+ }
+
+ const char **ports;
+ std::string port, previousPort, deviceName;
+ unsigned int nPorts = 0, nDevices = 0;
+ ports = jack_get_ports( client, NULL, NULL, 0 );
+ if ( ports ) {
+ // Parse the port names up to the first colon (:).
+ size_t iColon = 0;
+ do {
+ port = (char *) ports[ nPorts ];
+ iColon = port.find(":");
+ if ( iColon != std::string::npos ) {
+ port = port.substr( 0, iColon );
+ if ( port != previousPort ) {
+ if ( nDevices == device ) deviceName = port;
+ nDevices++;
+ previousPort = port;
+ }
+ }
+ } while ( ports[++nPorts] );
+ free( ports );
+ }
+
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
+
+ // Count the available ports containing the client name as device
+ // channels. Jack "input ports" equal RtAudio output channels.
+ unsigned int nChannels = 0;
+ unsigned long flag = JackPortIsInput;
+ if ( mode == INPUT ) flag = JackPortIsOutput;
+ ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
+ if ( ports ) {
+ while ( ports[ nChannels ] ) nChannels++;
+ free( ports );
+ }
+
+ // Compare the jack ports for specified client to the requested number of channels.
+ if ( nChannels < (channels + firstChannel) ) {
+ errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Check the jack server sample rate.
+ unsigned int jackRate = jack_get_sample_rate( client );
+ if ( sampleRate != jackRate ) {
+ jack_client_close( client );
+ errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.sampleRate = jackRate;
+
+ // Get the latency of the JACK port.
+ ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
+ if ( ports[ firstChannel ] ) {
+ // Added by Ge Wang
+ jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);
+ // the range (usually the min and max are equal)
+ jack_latency_range_t latrange; latrange.min = latrange.max = 0;
+ // get the latency range
+ jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );
+ // be optimistic, use the min!
+ stream_.latency[mode] = latrange.min;
+ //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
+ }
+ free( ports );
+
+ // The jack server always uses 32-bit floating-point data.
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ stream_.userFormat = format;
+
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
+
+ // Jack always uses non-interleaved buffers.
+ stream_.deviceInterleaved[mode] = false;
+
+ // Jack always provides host byte-ordered data.
+ stream_.doByteSwap[mode] = false;
+
+ // Get the buffer size. The buffer size and number of buffers
+ // (periods) is set when the jack server is started.
+ stream_.bufferSize = (int) jack_get_buffer_size( client );
+ *bufferSize = stream_.bufferSize;
+
+ stream_.nDeviceChannels[mode] = channels;
+ stream_.nUserChannels[mode] = channels;
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate our JackHandle structure for the stream.
+ if ( handle == 0 ) {
+ try {
+ handle = new JackHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
+ goto error;
+ }
+
+ if ( pthread_cond_init(&handle->condition, NULL) ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
+ goto error;
+ }
+ stream_.apiHandle = (void *) handle;
+ handle->client = client;
+ }
+ handle->deviceName[mode] = deviceName;
+
+ // Allocate necessary internal buffers.
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ if ( mode == OUTPUT )
+ bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ else { // mode == INPUT
+ bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
+ if ( bufferBytes < bytesOut ) makeBuffer = false;
+ }
+ }
+
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
+
+ // Allocate memory for the Jack ports (channels) identifiers.
+ handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
+ if ( handle->ports[mode] == NULL ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
+ goto error;
+ }
+
+ stream_.device[mode] = device;
+ stream_.channelOffset[mode] = firstChannel;
+ stream_.state = STREAM_STOPPED;
+ stream_.callbackInfo.object = (void *) this;
+
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ // We had already set up the stream for output.
+ stream_.mode = DUPLEX;
+ else {
+ stream_.mode = mode;
+ jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
+ jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );
+ jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
+ }
+
+ // Register our ports.
+ char label[64];
+ if ( mode == OUTPUT ) {
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ snprintf( label, 64, "outport %d", i );
+ handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
+ JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
+ }
+ }
+ else {
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ snprintf( label, 64, "inport %d", i );
+ handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
+ JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
+ }
+ }
+
+ // Setup the buffer conversion information structure. We don't use
+ // buffers to do channel offsets, so we override that parameter
+ // here.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
+
+ return SUCCESS;
+
+ error:
+ if ( handle ) {
+ pthread_cond_destroy( &handle->condition );
+ jack_client_close( handle->client );
+
+ if ( handle->ports[0] ) free( handle->ports[0] );
+ if ( handle->ports[1] ) free( handle->ports[1] );
+
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ return FAILURE;
+}
+
+void RtApiJack :: closeStream( void )
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiJack::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ if ( handle ) {
+
+ if ( stream_.state == STREAM_RUNNING )
+ jack_deactivate( handle->client );
+
+ jack_client_close( handle->client );
+ }
+
+ if ( handle ) {
+ if ( handle->ports[0] ) free( handle->ports[0] );
+ if ( handle->ports[1] ) free( handle->ports[1] );
+ pthread_cond_destroy( &handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
+
+void RtApiJack :: startStream( void )
+{
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiJack::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ int result = jack_activate( handle->client );
+ if ( result ) {
+ errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
+ goto unlock;
+ }
+
+ const char **ports;
+
+ // Get the list of available ports.
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ result = 1;
+ ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);
+ if ( ports == NULL) {
+ errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
+ goto unlock;
+ }
+
+ // Now make the port connections. Since RtAudio wasn't designed to
+ // allow the user to select particular channels of a device, we'll
+ // just open the first "nChannels" ports with offset.
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ result = 1;
+ if ( ports[ stream_.channelOffset[0] + i ] )
+ result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
+ if ( result ) {
+ free( ports );
+ errorText_ = "RtApiJack::startStream(): error connecting output ports!";
+ goto unlock;
+ }
+ }
+ free(ports);
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ result = 1;
+ ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );
+ if ( ports == NULL) {
+ errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
+ goto unlock;
+ }
+
+ // Now make the port connections. See note above.
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ result = 1;
+ if ( ports[ stream_.channelOffset[1] + i ] )
+ result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
+ if ( result ) {
+ free( ports );
+ errorText_ = "RtApiJack::startStream(): error connecting input ports!";
+ goto unlock;
+ }
+ }
+ free(ports);
+ }
+
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ stream_.state = STREAM_RUNNING;
+
+ unlock:
+ if ( result == 0 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiJack :: stopStream( void )
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 2;
+ pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
+ }
+ }
+
+ jack_deactivate( handle->client );
+ stream_.state = STREAM_STOPPED;
+}
+
+void RtApiJack :: abortStream( void )
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ handle->drainCounter = 2;
+
+ stopStream();
+}
+
+// This function will be called by a spawned thread when the user
+// callback function signals that the stream should be stopped or
+// aborted. It is necessary to handle it this way because the
+// callbackEvent() function must return before the jack_deactivate()
+// function will return.
+static void *jackStopStream( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiJack *object = (RtApiJack *) info->object;
+
+ object->stopStream();
+ pthread_exit( NULL );
+}
+
+bool RtApiJack :: callbackEvent( unsigned long nframes )
+{
+ if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return FAILURE;
+ }
+ if ( stream_.bufferSize != nframes ) {
+ errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
+ error( RtAudioError::WARNING );
+ return FAILURE;
+ }
+
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+
+ // Check if we were draining the stream and signal is finished.
+ if ( handle->drainCounter > 3 ) {
+ ThreadHandle threadId;
+
+ stream_.state = STREAM_STOPPING;
+ if ( handle->internalDrain == true )
+ pthread_create( &threadId, NULL, jackStopStream, info );
+ else
+ pthread_cond_signal( &handle->condition );
+ return SUCCESS;
+ }
+
+ // Invoke user callback first, to get fresh output data.
+ if ( handle->drainCounter == 0 ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
+ }
+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( cbReturnValue == 2 ) {
+ stream_.state = STREAM_STOPPING;
+ handle->drainCounter = 2;
+ ThreadHandle id;
+ pthread_create( &id, NULL, jackStopStream, info );
+ return SUCCESS;
+ }
+ else if ( cbReturnValue == 1 ) {
+ handle->drainCounter = 1;
+ handle->internalDrain = true;
+ }
+ }
+
+ jack_default_audio_sample_t *jackbuffer;
+ unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+
+ for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+ memset( jackbuffer, 0, bufferBytes );
+ }
+
+ }
+ else if ( stream_.doConvertBuffer[0] ) {
+
+ convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+
+ for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+ memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
+ }
+ }
+ else { // no buffer conversion
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+ memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
+ }
+ }
+ }
+
+ // Don't bother draining input
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ if ( stream_.doConvertBuffer[1] ) {
+ for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
+ memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
+ }
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+ }
+ else { // no buffer conversion
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
+ memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
+ }
+ }
+ }
+
+ unlock:
+ RtApi::tickStreamTime();
+ return SUCCESS;
+}
+ //******************** End of __UNIX_JACK__ *********************//
+#endif
+
+#if defined(__WINDOWS_ASIO__) // ASIO API on Windows
+
+// The ASIO API is designed around a callback scheme, so this
+// implementation is similar to that used for OS-X CoreAudio and Linux
+// Jack. The primary constraint with ASIO is that it only allows
+// access to a single driver at a time. Thus, it is not possible to
+// have more than one simultaneous RtAudio stream.
+//
+// This implementation also requires a number of external ASIO files
+// and a few global variables. The ASIO callback scheme does not
+// allow for the passing of user data, so we must create a global
+// pointer to our callbackInfo structure.
+//
+// On unix systems, we make use of a pthread condition variable.
+// Since there is no equivalent in Windows, I hacked something based
+// on information found in
+// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
+
+#include "asiosys.h"
+#include "asio.h"
+#include "iasiothiscallresolver.h"
+#include "asiodrivers.h"
+#include <cmath>
+
+static AsioDrivers drivers;
+static ASIOCallbacks asioCallbacks;
+static ASIODriverInfo driverInfo;
+static CallbackInfo *asioCallbackInfo;
+static bool asioXRun;
+
+struct AsioHandle {
+ int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
+ ASIOBufferInfo *bufferInfos;
+ HANDLE condition;
+
+ AsioHandle()
+ :drainCounter(0), internalDrain(false), bufferInfos(0) {}
+};
+
+// Function declarations (definitions at end of section)
+static const char* getAsioErrorString( ASIOError result );
+static void sampleRateChanged( ASIOSampleRate sRate );
+static long asioMessages( long selector, long value, void* message, double* opt );
+
+RtApiAsio :: RtApiAsio()
+{
+ // ASIO cannot run on a multi-threaded appartment. You can call
+ // CoInitialize beforehand, but it must be for appartment threading
+ // (in which case, CoInitilialize will return S_FALSE here).
+ coInitialized_ = false;
+ HRESULT hr = CoInitialize( NULL );
+ if ( FAILED(hr) ) {
+ errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
+ error( RtAudioError::WARNING );
+ }
+ coInitialized_ = true;
+
+ drivers.removeCurrentDriver();
+ driverInfo.asioVersion = 2;
+
+ // See note in DirectSound implementation about GetDesktopWindow().
+ driverInfo.sysRef = GetForegroundWindow();
+}
+
+RtApiAsio :: ~RtApiAsio()
+{
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+ if ( coInitialized_ ) CoUninitialize();
+}
+
+unsigned int RtApiAsio :: getDeviceCount( void )
+{
+ return (unsigned int) drivers.asioGetNumDev();
+}
+
+RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ // Get device ID
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices == 0 ) {
+ errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ // If a stream is already open, we cannot probe other devices. Thus, use the saved results.
+ if ( stream_.state != STREAM_CLOSED ) {
+ if ( device >= devices_.size() ) {
+ errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+ return devices_[ device ];
+ }
+
+ char driverName[32];
+ ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ info.name = driverName;
+
+ if ( !drivers.loadDriver( driverName ) ) {
+ errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ result = ASIOInit( &driverInfo );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Determine the device channel information.
+ long inputChannels, outputChannels;
+ result = ASIOGetChannels( &inputChannels, &outputChannels );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ info.outputChannels = outputChannels;
+ info.inputChannels = inputChannels;
+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+
+ // Determine the supported sample rates.
+ info.sampleRates.clear();
+ for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
+ result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
+ if ( result == ASE_OK )
+ info.sampleRates.push_back( SAMPLE_RATES[i] );
+ }
+
+ // Determine supported data types ... just check first channel and assume rest are the same.
+ ASIOChannelInfo channelInfo;
+ channelInfo.channel = 0;
+ channelInfo.isInput = true;
+ if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
+ result = ASIOGetChannelInfo( &channelInfo );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ info.nativeFormats = 0;
+ if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
+ info.nativeFormats |= RTAUDIO_SINT16;
+ else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
+ info.nativeFormats |= RTAUDIO_SINT32;
+ else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+ else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
+ info.nativeFormats |= RTAUDIO_FLOAT64;
+ else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )
+ info.nativeFormats |= RTAUDIO_SINT24;
+
+ if ( info.outputChannels > 0 )
+ if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
+ if ( info.inputChannels > 0 )
+ if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
+
+ info.probed = true;
+ drivers.removeCurrentDriver();
+ return info;
+}
+
+static void bufferSwitch( long index, ASIOBool /*processNow*/ )
+{
+ RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
+ object->callbackEvent( index );
+}
+
+void RtApiAsio :: saveDeviceInfo( void )
+{
+ devices_.clear();
+
+ unsigned int nDevices = getDeviceCount();
+ devices_.resize( nDevices );
+ for ( unsigned int i=0; i<nDevices; i++ )
+ devices_[i] = getDeviceInfo( i );
+}
+
+bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ // For ASIO, a duplex stream MUST use the same driver.
+ if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
+ return FAILURE;
+ }
+
+ char driverName[32];
+ ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Only load the driver once for duplex stream.
+ if ( mode != INPUT || stream_.mode != OUTPUT ) {
+ // The getDeviceInfo() function will not work when a stream is open
+ // because ASIO does not allow multiple devices to run at the same
+ // time. Thus, we'll probe the system before opening a stream and
+ // save the results for use by getDeviceInfo().
+ this->saveDeviceInfo();
+
+ if ( !drivers.loadDriver( driverName ) ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ result = ASIOInit( &driverInfo );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // Check the device channel count.
+ long inputChannels, outputChannels;
+ result = ASIOGetChannels( &inputChannels, &outputChannels );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
+ ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.nDeviceChannels[mode] = channels;
+ stream_.nUserChannels[mode] = channels;
+ stream_.channelOffset[mode] = firstChannel;
+
+ // Verify the sample rate is supported.
+ result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Get the current sample rate
+ ASIOSampleRate currentRate;
+ result = ASIOGetSampleRate( ¤tRate );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Set the sample rate only if necessary
+ if ( currentRate != sampleRate ) {
+ result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // Determine the driver data type.
+ ASIOChannelInfo channelInfo;
+ channelInfo.channel = 0;
+ if ( mode == OUTPUT ) channelInfo.isInput = false;
+ else channelInfo.isInput = true;
+ result = ASIOGetChannelInfo( &channelInfo );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Assuming WINDOWS host is always little-endian.
+ stream_.doByteSwap[mode] = false;
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = 0;
+ if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
+ }
+ else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
+ }
+ else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
+ }
+ else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
+ if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
+ }
+ else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;
+ }
+
+ if ( stream_.deviceFormat[mode] == 0 ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Set the buffer size. For a duplex stream, this will end up
+ // setting the buffer size based on the input constraints, which
+ // should be ok.
+ long minSize, maxSize, preferSize, granularity;
+ result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
+ else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
+ else if ( granularity == -1 ) {
+ // Make sure bufferSize is a power of two.
+ int log2_of_min_size = 0;
+ int log2_of_max_size = 0;
+
+ for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
+ if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
+ if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
+ }
+
+ long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
+ int min_delta_num = log2_of_min_size;
+
+ for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
+ long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
+ if (current_delta < min_delta) {
+ min_delta = current_delta;
+ min_delta_num = i;
+ }
+ }
+
+ *bufferSize = ( (unsigned int)1 << min_delta_num );
+ if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
+ else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
+ }
+ else if ( granularity != 0 ) {
+ // Set to an even multiple of granularity, rounding up.
+ *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
+ }
+
+ if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) {
+ drivers.removeCurrentDriver();
+ errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
+ return FAILURE;
+ }
+
+ stream_.bufferSize = *bufferSize;
+ stream_.nBuffers = 2;
+
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
+
+ // ASIO always uses non-interleaved buffers.
+ stream_.deviceInterleaved[mode] = false;
+
+ // Allocate, if necessary, our AsioHandle structure for the stream.
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ if ( handle == 0 ) {
+ try {
+ handle = new AsioHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ //if ( handle == NULL ) {
+ drivers.removeCurrentDriver();
+ errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
+ return FAILURE;
+ }
+ handle->bufferInfos = 0;
+
+ // Create a manual-reset event.
+ handle->condition = CreateEvent( NULL, // no security
+ TRUE, // manual-reset
+ FALSE, // non-signaled initially
+ NULL ); // unnamed
+ stream_.apiHandle = (void *) handle;
+ }
+
+ // Create the ASIO internal buffers. Since RtAudio sets up input
+ // and output separately, we'll have to dispose of previously
+ // created output buffers for a duplex stream.
+ long inputLatency, outputLatency;
+ if ( mode == INPUT && stream_.mode == OUTPUT ) {
+ ASIODisposeBuffers();
+ if ( handle->bufferInfos ) free( handle->bufferInfos );
+ }
+
+ // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
+ bool buffersAllocated = false;
+ unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+ handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
+ if ( handle->bufferInfos == NULL ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+
+ ASIOBufferInfo *infos;
+ infos = handle->bufferInfos;
+ for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
+ infos->isInput = ASIOFalse;
+ infos->channelNum = i + stream_.channelOffset[0];
+ infos->buffers[0] = infos->buffers[1] = 0;
+ }
+ for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
+ infos->isInput = ASIOTrue;
+ infos->channelNum = i + stream_.channelOffset[1];
+ infos->buffers[0] = infos->buffers[1] = 0;
+ }
+
+ // Set up the ASIO callback structure and create the ASIO data buffers.
+ asioCallbacks.bufferSwitch = &bufferSwitch;
+ asioCallbacks.sampleRateDidChange = &sampleRateChanged;
+ asioCallbacks.asioMessage = &asioMessages;
+ asioCallbacks.bufferSwitchTimeInfo = NULL;
+ result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+ buffersAllocated = true;
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate necessary internal buffers
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
+ }
+ }
+
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
+
+ stream_.sampleRate = sampleRate;
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ asioCallbackInfo = &stream_.callbackInfo;
+ stream_.callbackInfo.object = (void *) this;
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ // We had already set up an output stream.
+ stream_.mode = DUPLEX;
+ else
+ stream_.mode = mode;
+
+ // Determine device latencies
+ result = ASIOGetLatencies( &inputLatency, &outputLatency );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING); // warn but don't fail
+ }
+ else {
+ stream_.latency[0] = outputLatency;
+ stream_.latency[1] = inputLatency;
+ }
+
+ // Setup the buffer conversion information structure. We don't use
+ // buffers to do channel offsets, so we override that parameter
+ // here.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
+
+ return SUCCESS;
+
+ error:
+ if ( buffersAllocated )
+ ASIODisposeBuffers();
+ drivers.removeCurrentDriver();
+
+ if ( handle ) {
+ CloseHandle( handle->condition );
+ if ( handle->bufferInfos )
+ free( handle->bufferInfos );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ return FAILURE;
+}
+
+void RtApiAsio :: closeStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ if ( stream_.state == STREAM_RUNNING ) {
+ stream_.state = STREAM_STOPPED;
+ ASIOStop();
+ }
+ ASIODisposeBuffers();
+ drivers.removeCurrentDriver();
+
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ if ( handle ) {
+ CloseHandle( handle->condition );
+ if ( handle->bufferInfos )
+ free( handle->bufferInfos );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
+
+bool stopThreadCalled = false;
+
+void RtApiAsio :: startStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiAsio::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ ASIOError result = ASIOStart();
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ ResetEvent( handle->condition );
+ stream_.state = STREAM_RUNNING;
+ asioXRun = false;
+
+ unlock:
+ stopThreadCalled = false;
+
+ if ( result == ASE_OK ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiAsio :: stopStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 2;
+ WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
+ }
+ }
+
+ stream_.state = STREAM_STOPPED;
+
+ ASIOError result = ASIOStop();
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
+ errorText_ = errorStream_.str();
+ }
+
+ if ( result == ASE_OK ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiAsio :: abortStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ // The following lines were commented-out because some behavior was
+ // noted where the device buffers need to be zeroed to avoid
+ // continuing sound, even when the device buffers are completely
+ // disposed. So now, calling abort is the same as calling stop.
+ // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ // handle->drainCounter = 2;
+ stopStream();
+}
+
+// This function will be called by a spawned thread when the user
+// callback function signals that the stream should be stopped or
+// aborted. It is necessary to handle it this way because the
+// callbackEvent() function must return before the ASIOStop()
+// function will return.
+static unsigned __stdcall asioStopStream( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiAsio *object = (RtApiAsio *) info->object;
+
+ object->stopStream();
+ _endthreadex( 0 );
+ return 0;
+}
+
+bool RtApiAsio :: callbackEvent( long bufferIndex )
+{
+ if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return FAILURE;
+ }
+
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+
+ // Check if we were draining the stream and signal if finished.
+ if ( handle->drainCounter > 3 ) {
+
+ stream_.state = STREAM_STOPPING;
+ if ( handle->internalDrain == false )
+ SetEvent( handle->condition );
+ else { // spawn a thread to stop the stream
+ unsigned threadId;
+ stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
+ &stream_.callbackInfo, 0, &threadId );
+ }
+ return SUCCESS;
+ }
+
+ // Invoke user callback to get fresh output data UNLESS we are
+ // draining stream.
+ if ( handle->drainCounter == 0 ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && asioXRun == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ asioXRun = false;
+ }
+ if ( stream_.mode != OUTPUT && asioXRun == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ asioXRun = false;
+ }
+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( cbReturnValue == 2 ) {
+ stream_.state = STREAM_STOPPING;
+ handle->drainCounter = 2;
+ unsigned threadId;
+ stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
+ &stream_.callbackInfo, 0, &threadId );
+ return SUCCESS;
+ }
+ else if ( cbReturnValue == 1 ) {
+ handle->drainCounter = 1;
+ handle->internalDrain = true;
+ }
+ }
+
+ unsigned int nChannels, bufferBytes, i, j;
+ nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
+
+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput != ASIOTrue )
+ memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
+ }
+
+ }
+ else if ( stream_.doConvertBuffer[0] ) {
+
+ convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer( stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nDeviceChannels[0],
+ stream_.deviceFormat[0] );
+
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput != ASIOTrue )
+ memcpy( handle->bufferInfos[i].buffers[bufferIndex],
+ &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
+ }
+
+ }
+ else {
+
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer( stream_.userBuffer[0],
+ stream_.bufferSize * stream_.nUserChannels[0],
+ stream_.userFormat );
+
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput != ASIOTrue )
+ memcpy( handle->bufferInfos[i].buffers[bufferIndex],
+ &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
+ }
+
+ }
+ }
+
+ // Don't bother draining input
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
+
+ if (stream_.doConvertBuffer[1]) {
+
+ // Always interleave ASIO input data.
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput == ASIOTrue )
+ memcpy( &stream_.deviceBuffer[j++*bufferBytes],
+ handle->bufferInfos[i].buffers[bufferIndex],
+ bufferBytes );
+ }
+
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nDeviceChannels[1],
+ stream_.deviceFormat[1] );
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+
+ }
+ else {
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
+ memcpy( &stream_.userBuffer[1][bufferBytes*j++],
+ handle->bufferInfos[i].buffers[bufferIndex],
+ bufferBytes );
+ }
+ }
+
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( stream_.userBuffer[1],
+ stream_.bufferSize * stream_.nUserChannels[1],
+ stream_.userFormat );
+ }
+ }
+
+ unlock:
+ // The following call was suggested by Malte Clasen. While the API
+ // documentation indicates it should not be required, some device
+ // drivers apparently do not function correctly without it.
+ ASIOOutputReady();
+
+ RtApi::tickStreamTime();
+ return SUCCESS;
+}
+
+static void sampleRateChanged( ASIOSampleRate sRate )
+{
+ // The ASIO documentation says that this usually only happens during
+ // external sync. Audio processing is not stopped by the driver,
+ // actual sample rate might not have even changed, maybe only the
+ // sample rate status of an AES/EBU or S/PDIF digital input at the
+ // audio device.
+
+ RtApi *object = (RtApi *) asioCallbackInfo->object;
+ try {
+ object->stopStream();
+ }
+ catch ( RtAudioError &exception ) {
+ std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
+ return;
+ }
+
+ std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
+}
+
+static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ )
+{
+ long ret = 0;
+
+ switch( selector ) {
+ case kAsioSelectorSupported:
+ if ( value == kAsioResetRequest
+ || value == kAsioEngineVersion
+ || value == kAsioResyncRequest
+ || value == kAsioLatenciesChanged
+ // The following three were added for ASIO 2.0, you don't
+ // necessarily have to support them.
+ || value == kAsioSupportsTimeInfo
+ || value == kAsioSupportsTimeCode
+ || value == kAsioSupportsInputMonitor)
+ ret = 1L;
+ break;
+ case kAsioResetRequest:
+ // Defer the task and perform the reset of the driver during the
+ // next "safe" situation. You cannot reset the driver right now,
+ // as this code is called from the driver. Reset the driver is
+ // done by completely destruct is. I.e. ASIOStop(),
+ // ASIODisposeBuffers(), Destruction Afterwards you initialize the
+ // driver again.
+ std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
+ ret = 1L;
+ break;
+ case kAsioResyncRequest:
+ // This informs the application that the driver encountered some
+ // non-fatal data loss. It is used for synchronization purposes
+ // of different media. Added mainly to work around the Win16Mutex
+ // problems in Windows 95/98 with the Windows Multimedia system,
+ // which could lose data because the Mutex was held too long by
+ // another thread. However a driver can issue it in other
+ // situations, too.
+ // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
+ asioXRun = true;
+ ret = 1L;
+ break;
+ case kAsioLatenciesChanged:
+ // This will inform the host application that the drivers were
+ // latencies changed. Beware, it this does not mean that the
+ // buffer sizes have changed! You might need to update internal
+ // delay data.
+ std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
+ ret = 1L;
+ break;
+ case kAsioEngineVersion:
+ // Return the supported ASIO version of the host application. If
+ // a host application does not implement this selector, ASIO 1.0
+ // is assumed by the driver.
+ ret = 2L;
+ break;
+ case kAsioSupportsTimeInfo:
+ // Informs the driver whether the
+ // asioCallbacks.bufferSwitchTimeInfo() callback is supported.
+ // For compatibility with ASIO 1.0 drivers the host application
+ // should always support the "old" bufferSwitch method, too.
+ ret = 0;
+ break;
+ case kAsioSupportsTimeCode:
+ // Informs the driver whether application is interested in time
+ // code info. If an application does not need to know about time
+ // code, the driver has less work to do.
+ ret = 0;
+ break;
+ }
+ return ret;
+}
+
+static const char* getAsioErrorString( ASIOError result )
+{
+ struct Messages
+ {
+ ASIOError value;
+ const char*message;
+ };
+
+ static const Messages m[] =
+ {
+ { ASE_NotPresent, "Hardware input or output is not present or available." },
+ { ASE_HWMalfunction, "Hardware is malfunctioning." },
+ { ASE_InvalidParameter, "Invalid input parameter." },
+ { ASE_InvalidMode, "Invalid mode." },
+ { ASE_SPNotAdvancing, "Sample position not advancing." },
+ { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },
+ { ASE_NoMemory, "Not enough memory to complete the request." }
+ };
+
+ for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
+ if ( m[i].value == result ) return m[i].message;
+
+ return "Unknown error.";
+}
+
+//******************** End of __WINDOWS_ASIO__ *********************//
+#endif
+
+
+#if defined(__WINDOWS_WASAPI__) // Windows WASAPI API
+
+// Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014
+// - Introduces support for the Windows WASAPI API
+// - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required
+// - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface
+// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
+
+#ifndef INITGUID
+ #define INITGUID
+#endif
+#include <audioclient.h>
+#include <avrt.h>
+#include <mmdeviceapi.h>
+#include <functiondiscoverykeys_devpkey.h>
+
+//=============================================================================
+
+#define SAFE_RELEASE( objectPtr )\
+if ( objectPtr )\
+{\
+ objectPtr->Release();\
+ objectPtr = NULL;\
+}
+
+typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
+
+//-----------------------------------------------------------------------------
+
+// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
+// Therefore we must perform all necessary conversions to user buffers in order to satisfy these
+// requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
+// provide intermediate storage for read / write synchronization.
+class WasapiBuffer
+{
+public:
+ WasapiBuffer()
+ : buffer_( NULL ),
+ bufferSize_( 0 ),
+ inIndex_( 0 ),
+ outIndex_( 0 ) {}
+
+ ~WasapiBuffer() {
+ delete buffer_;
+ }
+
+ // sets the length of the internal ring buffer
+ void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
+ delete buffer_;
+
+ buffer_ = ( char* ) calloc( bufferSize, formatBytes );
+
+ bufferSize_ = bufferSize;
+ inIndex_ = 0;
+ outIndex_ = 0;
+ }
+
+ // attempt to push a buffer into the ring buffer at the current "in" index
+ bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
+ {
+ if ( !buffer || // incoming buffer is NULL
+ bufferSize == 0 || // incoming buffer has no data
+ bufferSize > bufferSize_ ) // incoming buffer too large
+ {
+ return false;
+ }
+
+ unsigned int relOutIndex = outIndex_;
+ unsigned int inIndexEnd = inIndex_ + bufferSize;
+ if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
+ relOutIndex += bufferSize_;
+ }
+
+ // "in" index can end on the "out" index but cannot begin at it
+ if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
+ return false; // not enough space between "in" index and "out" index
+ }
+
+ // copy buffer from external to internal
+ int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
+ fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
+ int fromInSize = bufferSize - fromZeroSize;
+
+ switch( format )
+ {
+ case RTAUDIO_SINT8:
+ memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
+ memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
+ break;
+ case RTAUDIO_SINT16:
+ memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
+ memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
+ break;
+ case RTAUDIO_SINT24:
+ memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
+ memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
+ break;
+ case RTAUDIO_SINT32:
+ memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
+ memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
+ break;
+ case RTAUDIO_FLOAT32:
+ memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
+ memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
+ break;
+ case RTAUDIO_FLOAT64:
+ memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
+ memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
+ break;
+ }
+
+ // update "in" index
+ inIndex_ += bufferSize;
+ inIndex_ %= bufferSize_;
+
+ return true;
+ }
+
+ // attempt to pull a buffer from the ring buffer from the current "out" index
+ bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
+ {
+ if ( !buffer || // incoming buffer is NULL
+ bufferSize == 0 || // incoming buffer has no data
+ bufferSize > bufferSize_ ) // incoming buffer too large
+ {
+ return false;
+ }
+
+ unsigned int relInIndex = inIndex_;
+ unsigned int outIndexEnd = outIndex_ + bufferSize;
+ if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
+ relInIndex += bufferSize_;
+ }
+
+ // "out" index can begin at and end on the "in" index
+ if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
+ return false; // not enough space between "out" index and "in" index
+ }
+
+ // copy buffer from internal to external
+ int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
+ fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
+ int fromOutSize = bufferSize - fromZeroSize;
+
+ switch( format )
+ {
+ case RTAUDIO_SINT8:
+ memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
+ memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
+ break;
+ case RTAUDIO_SINT16:
+ memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
+ memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
+ break;
+ case RTAUDIO_SINT24:
+ memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
+ memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
+ break;
+ case RTAUDIO_SINT32:
+ memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
+ memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
+ break;
+ case RTAUDIO_FLOAT32:
+ memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
+ memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
+ break;
+ case RTAUDIO_FLOAT64:
+ memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
+ memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
+ break;
+ }
+
+ // update "out" index
+ outIndex_ += bufferSize;
+ outIndex_ %= bufferSize_;
+
+ return true;
+ }
+
+private:
+ char* buffer_;
+ unsigned int bufferSize_;
+ unsigned int inIndex_;
+ unsigned int outIndex_;
+};
+
+//-----------------------------------------------------------------------------
+
+// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate
+// between HW and the user. The convertBufferWasapi function is used to perform this conversion
+// between HwIn->UserIn and UserOut->HwOut during the stream callback loop.
+// This sample rate converter favors speed over quality, and works best with conversions between
+// one rate and its multiple.
+void convertBufferWasapi( char* outBuffer,
+ const char* inBuffer,
+ const unsigned int& channelCount,
+ const unsigned int& inSampleRate,
+ const unsigned int& outSampleRate,
+ const unsigned int& inSampleCount,
+ unsigned int& outSampleCount,
+ const RtAudioFormat& format )
+{
+ // calculate the new outSampleCount and relative sampleStep
+ float sampleRatio = ( float ) outSampleRate / inSampleRate;
+ float sampleStep = 1.0f / sampleRatio;
+ float inSampleFraction = 0.0f;
+
+ outSampleCount = ( unsigned int ) ( inSampleCount * sampleRatio );
+
+ // frame-by-frame, copy each relative input sample into it's corresponding output sample
+ for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )
+ {
+ unsigned int inSample = ( unsigned int ) inSampleFraction;
+
+ switch ( format )
+ {
+ case RTAUDIO_SINT8:
+ memcpy( &( ( char* ) outBuffer )[ outSample * channelCount ], &( ( char* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( char ) );
+ break;
+ case RTAUDIO_SINT16:
+ memcpy( &( ( short* ) outBuffer )[ outSample * channelCount ], &( ( short* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( short ) );
+ break;
+ case RTAUDIO_SINT24:
+ memcpy( &( ( S24* ) outBuffer )[ outSample * channelCount ], &( ( S24* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( S24 ) );
+ break;
+ case RTAUDIO_SINT32:
+ memcpy( &( ( int* ) outBuffer )[ outSample * channelCount ], &( ( int* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( int ) );
+ break;
+ case RTAUDIO_FLOAT32:
+ memcpy( &( ( float* ) outBuffer )[ outSample * channelCount ], &( ( float* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( float ) );
+ break;
+ case RTAUDIO_FLOAT64:
+ memcpy( &( ( double* ) outBuffer )[ outSample * channelCount ], &( ( double* ) inBuffer )[ inSample * channelCount ], channelCount * sizeof( double ) );
+ break;
+ }
+
+ // jump to next in sample
+ inSampleFraction += sampleStep;
+ }
+}
+
+//-----------------------------------------------------------------------------
+
+// A structure to hold various information related to the WASAPI implementation.
+struct WasapiHandle
+{
+ IAudioClient* captureAudioClient;
+ IAudioClient* renderAudioClient;
+ IAudioCaptureClient* captureClient;
+ IAudioRenderClient* renderClient;
+ HANDLE captureEvent;
+ HANDLE renderEvent;
+
+ WasapiHandle()
+ : captureAudioClient( NULL ),
+ renderAudioClient( NULL ),
+ captureClient( NULL ),
+ renderClient( NULL ),
+ captureEvent( NULL ),
+ renderEvent( NULL ) {}
+};
+
+//=============================================================================
+
+RtApiWasapi::RtApiWasapi()
+ : coInitialized_( false ), deviceEnumerator_( NULL )
+{
+ // WASAPI can run either apartment or multi-threaded
+ HRESULT hr = CoInitialize( NULL );
+ if ( !FAILED( hr ) )
+ coInitialized_ = true;
+
+ // Instantiate device enumerator
+ hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
+ CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
+ ( void** ) &deviceEnumerator_ );
+
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator";
+ error( RtAudioError::DRIVER_ERROR );
+ }
+}
+
+//-----------------------------------------------------------------------------
+
+RtApiWasapi::~RtApiWasapi()
+{
+ if ( stream_.state != STREAM_CLOSED )
+ closeStream();
+
+ SAFE_RELEASE( deviceEnumerator_ );
+
+ // If this object previously called CoInitialize()
+ if ( coInitialized_ )
+ CoUninitialize();
+}
+
+//=============================================================================
+
+unsigned int RtApiWasapi::getDeviceCount( void )
+{
+ unsigned int captureDeviceCount = 0;
+ unsigned int renderDeviceCount = 0;
+
+ IMMDeviceCollection* captureDevices = NULL;
+ IMMDeviceCollection* renderDevices = NULL;
+
+ // Count capture devices
+ errorText_.clear();
+ HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
+ goto Exit;
+ }
+
+ hr = captureDevices->GetCount( &captureDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
+ goto Exit;
+ }
+
+ // Count render devices
+ hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
+ goto Exit;
+ }
+
+ hr = renderDevices->GetCount( &renderDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
+ goto Exit;
+ }
+
+Exit:
+ // release all references
+ SAFE_RELEASE( captureDevices );
+ SAFE_RELEASE( renderDevices );
+
+ if ( errorText_.empty() )
+ return captureDeviceCount + renderDeviceCount;
+
+ error( RtAudioError::DRIVER_ERROR );
+ return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ unsigned int captureDeviceCount = 0;
+ unsigned int renderDeviceCount = 0;
+ std::wstring deviceName;
+ std::string defaultDeviceName;
+ bool isCaptureDevice = false;
+
+ PROPVARIANT deviceNameProp;
+ PROPVARIANT defaultDeviceNameProp;
+
+ IMMDeviceCollection* captureDevices = NULL;
+ IMMDeviceCollection* renderDevices = NULL;
+ IMMDevice* devicePtr = NULL;
+ IMMDevice* defaultDevicePtr = NULL;
+ IAudioClient* audioClient = NULL;
+ IPropertyStore* devicePropStore = NULL;
+ IPropertyStore* defaultDevicePropStore = NULL;
+
+ WAVEFORMATEX* deviceFormat = NULL;
+ WAVEFORMATEX* closestMatchFormat = NULL;
+
+ // probed
+ info.probed = false;
+
+ // Count capture devices
+ errorText_.clear();
+ RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
+ HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
+ goto Exit;
+ }
+
+ hr = captureDevices->GetCount( &captureDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
+ goto Exit;
+ }
+
+ // Count render devices
+ hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
+ goto Exit;
+ }
+
+ hr = renderDevices->GetCount( &renderDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
+ goto Exit;
+ }
+
+ // validate device index
+ if ( device >= captureDeviceCount + renderDeviceCount ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
+ errorType = RtAudioError::INVALID_USE;
+ goto Exit;
+ }
+
+ // determine whether index falls within capture or render devices
+ if ( device >= renderDeviceCount ) {
+ hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
+ goto Exit;
+ }
+ isCaptureDevice = true;
+ }
+ else {
+ hr = renderDevices->Item( device, &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
+ goto Exit;
+ }
+ isCaptureDevice = false;
+ }
+
+ // get default device name
+ if ( isCaptureDevice ) {
+ hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
+ goto Exit;
+ }
+ }
+ else {
+ hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
+ goto Exit;
+ }
+ }
+
+ hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
+ goto Exit;
+ }
+ PropVariantInit( &defaultDeviceNameProp );
+
+ hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
+ goto Exit;
+ }
+
+ deviceName = defaultDeviceNameProp.pwszVal;
+ defaultDeviceName = std::string( deviceName.begin(), deviceName.end() );
+
+ // name
+ hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
+ goto Exit;
+ }
+
+ PropVariantInit( &deviceNameProp );
+
+ hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
+ goto Exit;
+ }
+
+ deviceName = deviceNameProp.pwszVal;
+ info.name = std::string( deviceName.begin(), deviceName.end() );
+
+ // is default
+ if ( isCaptureDevice ) {
+ info.isDefaultInput = info.name == defaultDeviceName;
+ info.isDefaultOutput = false;
+ }
+ else {
+ info.isDefaultInput = false;
+ info.isDefaultOutput = info.name == defaultDeviceName;
+ }
+
+ // channel count
+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
+ goto Exit;
+ }
+
+ hr = audioClient->GetMixFormat( &deviceFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
+ goto Exit;
+ }
+
+ if ( isCaptureDevice ) {
+ info.inputChannels = deviceFormat->nChannels;
+ info.outputChannels = 0;
+ info.duplexChannels = 0;
+ }
+ else {
+ info.inputChannels = 0;
+ info.outputChannels = deviceFormat->nChannels;
+ info.duplexChannels = 0;
+ }
+
+ // sample rates
+ info.sampleRates.clear();
+
+ // allow support for all sample rates as we have a built-in sample rate converter
+ for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {
+ info.sampleRates.push_back( SAMPLE_RATES[i] );
+ }
+
+ // native format
+ info.nativeFormats = 0;
+
+ if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
+ ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
+ ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
+ {
+ if ( deviceFormat->wBitsPerSample == 32 ) {
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+ }
+ else if ( deviceFormat->wBitsPerSample == 64 ) {
+ info.nativeFormats |= RTAUDIO_FLOAT64;
+ }
+ }
+ else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
+ ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
+ ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
+ {
+ if ( deviceFormat->wBitsPerSample == 8 ) {
+ info.nativeFormats |= RTAUDIO_SINT8;
+ }
+ else if ( deviceFormat->wBitsPerSample == 16 ) {
+ info.nativeFormats |= RTAUDIO_SINT16;
+ }
+ else if ( deviceFormat->wBitsPerSample == 24 ) {
+ info.nativeFormats |= RTAUDIO_SINT24;
+ }
+ else if ( deviceFormat->wBitsPerSample == 32 ) {
+ info.nativeFormats |= RTAUDIO_SINT32;
+ }
+ }
+
+ // probed
+ info.probed = true;
+
+Exit:
+ // release all references
+ PropVariantClear( &deviceNameProp );
+ PropVariantClear( &defaultDeviceNameProp );
+
+ SAFE_RELEASE( captureDevices );
+ SAFE_RELEASE( renderDevices );
+ SAFE_RELEASE( devicePtr );
+ SAFE_RELEASE( defaultDevicePtr );
+ SAFE_RELEASE( audioClient );
+ SAFE_RELEASE( devicePropStore );
+ SAFE_RELEASE( defaultDevicePropStore );
+
+ CoTaskMemFree( deviceFormat );
+ CoTaskMemFree( closestMatchFormat );
+
+ if ( !errorText_.empty() )
+ error( errorType );
+ return info;
+}
+
+//-----------------------------------------------------------------------------
+
+unsigned int RtApiWasapi::getDefaultOutputDevice( void )
+{
+ for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
+ if ( getDeviceInfo( i ).isDefaultOutput ) {
+ return i;
+ }
+ }
+
+ return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+unsigned int RtApiWasapi::getDefaultInputDevice( void )
+{
+ for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
+ if ( getDeviceInfo( i ).isDefaultInput ) {
+ return i;
+ }
+ }
+
+ return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::closeStream( void )
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ if ( stream_.state != STREAM_STOPPED )
+ stopStream();
+
+ // clean up stream memory
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
+
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
+
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
+ CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
+
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
+ CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
+
+ delete ( WasapiHandle* ) stream_.apiHandle;
+ stream_.apiHandle = NULL;
+
+ for ( int i = 0; i < 2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ // update stream state
+ stream_.state = STREAM_CLOSED;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::startStream( void )
+{
+ verifyStream();
+
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiWasapi::startStream: The stream is already running.";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ // update stream state
+ stream_.state = STREAM_RUNNING;
+
+ // create WASAPI stream thread
+ stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
+
+ if ( !stream_.callbackInfo.thread ) {
+ errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
+ error( RtAudioError::THREAD_ERROR );
+ }
+ else {
+ SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
+ ResumeThread( ( void* ) stream_.callbackInfo.thread );
+ }
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::stopStream( void )
+{
+ verifyStream();
+
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ // inform stream thread by setting stream state to STREAM_STOPPING
+ stream_.state = STREAM_STOPPING;
+
+ // wait until stream thread is stopped
+ while( stream_.state != STREAM_STOPPED ) {
+ Sleep( 1 );
+ }
+
+ // Wait for the last buffer to play before stopping.
+ Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
+
+ // stop capture client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
+ error( RtAudioError::DRIVER_ERROR );
+ return;
+ }
+ }
+
+ // stop render client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
+ error( RtAudioError::DRIVER_ERROR );
+ return;
+ }
+ }
+
+ // close thread handle
+ if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
+ errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
+ error( RtAudioError::THREAD_ERROR );
+ return;
+ }
+
+ stream_.callbackInfo.thread = (ThreadHandle) NULL;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::abortStream( void )
+{
+ verifyStream();
+
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ // inform stream thread by setting stream state to STREAM_STOPPING
+ stream_.state = STREAM_STOPPING;
+
+ // wait until stream thread is stopped
+ while ( stream_.state != STREAM_STOPPED ) {
+ Sleep( 1 );
+ }
+
+ // stop capture client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
+ error( RtAudioError::DRIVER_ERROR );
+ return;
+ }
+ }
+
+ // stop render client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
+ error( RtAudioError::DRIVER_ERROR );
+ return;
+ }
+ }
+
+ // close thread handle
+ if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
+ errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
+ error( RtAudioError::THREAD_ERROR );
+ return;
+ }
+
+ stream_.callbackInfo.thread = (ThreadHandle) NULL;
+}
+
+//-----------------------------------------------------------------------------
+
+bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int* bufferSize,
+ RtAudio::StreamOptions* options )
+{
+ bool methodResult = FAILURE;
+ unsigned int captureDeviceCount = 0;
+ unsigned int renderDeviceCount = 0;
+
+ IMMDeviceCollection* captureDevices = NULL;
+ IMMDeviceCollection* renderDevices = NULL;
+ IMMDevice* devicePtr = NULL;
+ WAVEFORMATEX* deviceFormat = NULL;
+ unsigned int bufferBytes;
+ stream_.state = STREAM_STOPPED;
+
+ // create API Handle if not already created
+ if ( !stream_.apiHandle )
+ stream_.apiHandle = ( void* ) new WasapiHandle();
+
+ // Count capture devices
+ errorText_.clear();
+ RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
+ HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
+ goto Exit;
+ }
+
+ hr = captureDevices->GetCount( &captureDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
+ goto Exit;
+ }
+
+ // Count render devices
+ hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
+ goto Exit;
+ }
+
+ hr = renderDevices->GetCount( &renderDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
+ goto Exit;
+ }
+
+ // validate device index
+ if ( device >= captureDeviceCount + renderDeviceCount ) {
+ errorType = RtAudioError::INVALID_USE;
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
+ goto Exit;
+ }
+
+ // determine whether index falls within capture or render devices
+ if ( device >= renderDeviceCount ) {
+ if ( mode != INPUT ) {
+ errorType = RtAudioError::INVALID_USE;
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
+ goto Exit;
+ }
+
+ // retrieve captureAudioClient from devicePtr
+ IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
+
+ hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
+ goto Exit;
+ }
+
+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+ NULL, ( void** ) &captureAudioClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
+ goto Exit;
+ }
+
+ hr = captureAudioClient->GetMixFormat( &deviceFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
+ goto Exit;
+ }
+
+ stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+ captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
+ }
+ else {
+ if ( mode != OUTPUT ) {
+ errorType = RtAudioError::INVALID_USE;
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device.";
+ goto Exit;
+ }
+
+ // retrieve renderAudioClient from devicePtr
+ IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
+
+ hr = renderDevices->Item( device, &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
+ goto Exit;
+ }
+
+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+ NULL, ( void** ) &renderAudioClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
+ goto Exit;
+ }
+
+ hr = renderAudioClient->GetMixFormat( &deviceFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
+ goto Exit;
+ }
+
+ stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+ renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
+ }
+
+ // fill stream data
+ if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
+ ( stream_.mode == INPUT && mode == OUTPUT ) ) {
+ stream_.mode = DUPLEX;
+ }
+ else {
+ stream_.mode = mode;
+ }
+
+ stream_.device[mode] = device;
+ stream_.doByteSwap[mode] = false;
+ stream_.sampleRate = sampleRate;
+ stream_.bufferSize = *bufferSize;
+ stream_.nBuffers = 1;
+ stream_.nUserChannels[mode] = channels;
+ stream_.channelOffset[mode] = firstChannel;
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;
+
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
+ stream_.userInterleaved = false;
+ else
+ stream_.userInterleaved = true;
+ stream_.deviceInterleaved[mode] = true;
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] ||
+ stream_.nUserChannels != stream_.nDeviceChannels )
+ stream_.doConvertBuffer[mode] = true;
+ else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ if ( stream_.doConvertBuffer[mode] )
+ setConvertInfo( mode, 0 );
+
+ // Allocate necessary internal buffers
+ bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
+
+ stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
+ if ( !stream_.userBuffer[mode] ) {
+ errorType = RtAudioError::MEMORY_ERROR;
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
+ goto Exit;
+ }
+
+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
+ stream_.callbackInfo.priority = 15;
+ else
+ stream_.callbackInfo.priority = 0;
+
+ ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
+ ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
+
+ methodResult = SUCCESS;
+
+Exit:
+ //clean up
+ SAFE_RELEASE( captureDevices );
+ SAFE_RELEASE( renderDevices );
+ SAFE_RELEASE( devicePtr );
+ CoTaskMemFree( deviceFormat );
+
+ // if method failed, close the stream
+ if ( methodResult == FAILURE )
+ closeStream();
+
+ if ( !errorText_.empty() )
+ error( errorType );
+ return methodResult;
+}
+
+//=============================================================================
+
+DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
+{
+ if ( wasapiPtr )
+ ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
+
+ return 0;
+}
+
+DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
+{
+ if ( wasapiPtr )
+ ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
+
+ return 0;
+}
+
+DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
+{
+ if ( wasapiPtr )
+ ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
+
+ return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::wasapiThread()
+{
+ // as this is a new thread, we must CoInitialize it
+ CoInitialize( NULL );
+
+ HRESULT hr;
+
+ IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
+ IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
+ IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
+ IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
+ HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
+ HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
+
+ WAVEFORMATEX* captureFormat = NULL;
+ WAVEFORMATEX* renderFormat = NULL;
+ float captureSrRatio = 0.0f;
+ float renderSrRatio = 0.0f;
+ WasapiBuffer captureBuffer;
+ WasapiBuffer renderBuffer;
+
+ // declare local stream variables
+ RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
+ BYTE* streamBuffer = NULL;
+ unsigned long captureFlags = 0;
+ unsigned int bufferFrameCount = 0;
+ unsigned int numFramesPadding = 0;
+ unsigned int convBufferSize = 0;
+ bool callbackPushed = false;
+ bool callbackPulled = false;
+ bool callbackStopped = false;
+ int callbackResult = 0;
+
+ // convBuffer is used to store converted buffers between WASAPI and the user
+ char* convBuffer = NULL;
+ unsigned int convBuffSize = 0;
+ unsigned int deviceBuffSize = 0;
+
+ errorText_.clear();
+ RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
+
+ // Attempt to assign "Pro Audio" characteristic to thread
+ HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
+ if ( AvrtDll ) {
+ DWORD taskIndex = 0;
+ TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
+ AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
+ FreeLibrary( AvrtDll );
+ }
+
+ // start capture stream if applicable
+ if ( captureAudioClient ) {
+ hr = captureAudioClient->GetMixFormat( &captureFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
+ goto Exit;
+ }
+
+ captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );
+
+ // initialize capture stream according to desire buffer size
+ float desiredBufferSize = stream_.bufferSize * captureSrRatio;
+ REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );
+
+ if ( !captureClient ) {
+ hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ desiredBufferPeriod,
+ desiredBufferPeriod,
+ captureFormat,
+ NULL );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
+ goto Exit;
+ }
+
+ hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
+ ( void** ) &captureClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
+ goto Exit;
+ }
+
+ // configure captureEvent to trigger on every available capture buffer
+ captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
+ if ( !captureEvent ) {
+ errorType = RtAudioError::SYSTEM_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
+ goto Exit;
+ }
+
+ hr = captureAudioClient->SetEventHandle( captureEvent );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
+ goto Exit;
+ }
+
+ ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
+ ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
+ }
+
+ unsigned int inBufferSize = 0;
+ hr = captureAudioClient->GetBufferSize( &inBufferSize );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
+ goto Exit;
+ }
+
+ // scale outBufferSize according to stream->user sample rate ratio
+ unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];
+ inBufferSize *= stream_.nDeviceChannels[INPUT];
+
+ // set captureBuffer size
+ captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
+
+ // reset the capture stream
+ hr = captureAudioClient->Reset();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
+ goto Exit;
+ }
+
+ // start the capture stream
+ hr = captureAudioClient->Start();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
+ goto Exit;
+ }
+ }
+
+ // start render stream if applicable
+ if ( renderAudioClient ) {
+ hr = renderAudioClient->GetMixFormat( &renderFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
+ goto Exit;
+ }
+
+ renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );
+
+ // initialize render stream according to desire buffer size
+ float desiredBufferSize = stream_.bufferSize * renderSrRatio;
+ REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );
+
+ if ( !renderClient ) {
+ hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ desiredBufferPeriod,
+ desiredBufferPeriod,
+ renderFormat,
+ NULL );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
+ goto Exit;
+ }
+
+ hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
+ ( void** ) &renderClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
+ goto Exit;
+ }
+
+ // configure renderEvent to trigger on every available render buffer
+ renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
+ if ( !renderEvent ) {
+ errorType = RtAudioError::SYSTEM_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
+ goto Exit;
+ }
+
+ hr = renderAudioClient->SetEventHandle( renderEvent );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
+ goto Exit;
+ }
+
+ ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
+ ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
+ }
+
+ unsigned int outBufferSize = 0;
+ hr = renderAudioClient->GetBufferSize( &outBufferSize );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
+ goto Exit;
+ }
+
+ // scale inBufferSize according to user->stream sample rate ratio
+ unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];
+ outBufferSize *= stream_.nDeviceChannels[OUTPUT];
+
+ // set renderBuffer size
+ renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
+
+ // reset the render stream
+ hr = renderAudioClient->Reset();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
+ goto Exit;
+ }
+
+ // start the render stream
+ hr = renderAudioClient->Start();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
+ goto Exit;
+ }
+ }
+
+ if ( stream_.mode == INPUT ) {
+ convBuffSize = ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
+ deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
+ }
+ else if ( stream_.mode == OUTPUT ) {
+ convBuffSize = ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
+ deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
+ }
+ else if ( stream_.mode == DUPLEX ) {
+ convBuffSize = std::max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
+ ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
+ deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
+ stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
+ }
+
+ convBuffer = ( char* ) malloc( convBuffSize );
+ stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
+ if ( !convBuffer || !stream_.deviceBuffer ) {
+ errorType = RtAudioError::MEMORY_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
+ goto Exit;
+ }
+
+ // stream process loop
+ while ( stream_.state != STREAM_STOPPING ) {
+ if ( !callbackPulled ) {
+ // Callback Input
+ // ==============
+ // 1. Pull callback buffer from inputBuffer
+ // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count
+ // Convert callback buffer to user format
+
+ if ( captureAudioClient ) {
+ // Pull callback buffer from inputBuffer
+ callbackPulled = captureBuffer.pullBuffer( convBuffer,
+ ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT],
+ stream_.deviceFormat[INPUT] );
+
+ if ( callbackPulled ) {
+ // Convert callback buffer to user sample rate
+ convertBufferWasapi( stream_.deviceBuffer,
+ convBuffer,
+ stream_.nDeviceChannels[INPUT],
+ captureFormat->nSamplesPerSec,
+ stream_.sampleRate,
+ ( unsigned int ) ( stream_.bufferSize * captureSrRatio ),
+ convBufferSize,
+ stream_.deviceFormat[INPUT] );
+
+ if ( stream_.doConvertBuffer[INPUT] ) {
+ // Convert callback buffer to user format
+ convertBuffer( stream_.userBuffer[INPUT],
+ stream_.deviceBuffer,
+ stream_.convertInfo[INPUT] );
+ }
+ else {
+ // no further conversion, simple copy deviceBuffer to userBuffer
+ memcpy( stream_.userBuffer[INPUT],
+ stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
+ }
+ }
+ }
+ else {
+ // if there is no capture stream, set callbackPulled flag
+ callbackPulled = true;
+ }
+
+ // Execute Callback
+ // ================
+ // 1. Execute user callback method
+ // 2. Handle return value from callback
+
+ // if callback has not requested the stream to stop
+ if ( callbackPulled && !callbackStopped ) {
+ // Execute user callback method
+ callbackResult = callback( stream_.userBuffer[OUTPUT],
+ stream_.userBuffer[INPUT],
+ stream_.bufferSize,
+ getStreamTime(),
+ captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
+ stream_.callbackInfo.userData );
+
+ // Handle return value from callback
+ if ( callbackResult == 1 ) {
+ // instantiate a thread to stop this thread
+ HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
+ if ( !threadHandle ) {
+ errorType = RtAudioError::THREAD_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
+ goto Exit;
+ }
+ else if ( !CloseHandle( threadHandle ) ) {
+ errorType = RtAudioError::THREAD_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
+ goto Exit;
+ }
+
+ callbackStopped = true;
+ }
+ else if ( callbackResult == 2 ) {
+ // instantiate a thread to stop this thread
+ HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
+ if ( !threadHandle ) {
+ errorType = RtAudioError::THREAD_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
+ goto Exit;
+ }
+ else if ( !CloseHandle( threadHandle ) ) {
+ errorType = RtAudioError::THREAD_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
+ goto Exit;
+ }
+
+ callbackStopped = true;
+ }
+ }
+ }
+
+ // Callback Output
+ // ===============
+ // 1. Convert callback buffer to stream format
+ // 2. Convert callback buffer to stream sample rate and channel count
+ // 3. Push callback buffer into outputBuffer
+
+ if ( renderAudioClient && callbackPulled ) {
+ if ( stream_.doConvertBuffer[OUTPUT] ) {
+ // Convert callback buffer to stream format
+ convertBuffer( stream_.deviceBuffer,
+ stream_.userBuffer[OUTPUT],
+ stream_.convertInfo[OUTPUT] );
+
+ }
+
+ // Convert callback buffer to stream sample rate
+ convertBufferWasapi( convBuffer,
+ stream_.deviceBuffer,
+ stream_.nDeviceChannels[OUTPUT],
+ stream_.sampleRate,
+ renderFormat->nSamplesPerSec,
+ stream_.bufferSize,
+ convBufferSize,
+ stream_.deviceFormat[OUTPUT] );
+
+ // Push callback buffer into outputBuffer
+ callbackPushed = renderBuffer.pushBuffer( convBuffer,
+ convBufferSize * stream_.nDeviceChannels[OUTPUT],
+ stream_.deviceFormat[OUTPUT] );
+ }
+ else {
+ // if there is no render stream, set callbackPushed flag
+ callbackPushed = true;
+ }
+
+ // Stream Capture
+ // ==============
+ // 1. Get capture buffer from stream
+ // 2. Push capture buffer into inputBuffer
+ // 3. If 2. was successful: Release capture buffer
+
+ if ( captureAudioClient ) {
+ // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
+ if ( !callbackPulled ) {
+ WaitForSingleObject( captureEvent, INFINITE );
+ }
+
+ // Get capture buffer from stream
+ hr = captureClient->GetBuffer( &streamBuffer,
+ &bufferFrameCount,
+ &captureFlags, NULL, NULL );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
+ goto Exit;
+ }
+
+ if ( bufferFrameCount != 0 ) {
+ // Push capture buffer into inputBuffer
+ if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
+ bufferFrameCount * stream_.nDeviceChannels[INPUT],
+ stream_.deviceFormat[INPUT] ) )
+ {
+ // Release capture buffer
+ hr = captureClient->ReleaseBuffer( bufferFrameCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
+ goto Exit;
+ }
+ }
+ else
+ {
+ // Inform WASAPI that capture was unsuccessful
+ hr = captureClient->ReleaseBuffer( 0 );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
+ goto Exit;
+ }
+ }
+ }
+ else
+ {
+ // Inform WASAPI that capture was unsuccessful
+ hr = captureClient->ReleaseBuffer( 0 );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
+ goto Exit;
+ }
+ }
+ }
+
+ // Stream Render
+ // =============
+ // 1. Get render buffer from stream
+ // 2. Pull next buffer from outputBuffer
+ // 3. If 2. was successful: Fill render buffer with next buffer
+ // Release render buffer
+
+ if ( renderAudioClient ) {
+ // if the callback output buffer was not pushed to renderBuffer, wait for next render event
+ if ( callbackPulled && !callbackPushed ) {
+ WaitForSingleObject( renderEvent, INFINITE );
+ }
+
+ // Get render buffer from stream
+ hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
+ goto Exit;
+ }
+
+ hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
+ goto Exit;
+ }
+
+ bufferFrameCount -= numFramesPadding;
+
+ if ( bufferFrameCount != 0 ) {
+ hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
+ goto Exit;
+ }
+
+ // Pull next buffer from outputBuffer
+ // Fill render buffer with next buffer
+ if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
+ bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
+ stream_.deviceFormat[OUTPUT] ) )
+ {
+ // Release render buffer
+ hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
+ goto Exit;
+ }
+ }
+ else
+ {
+ // Inform WASAPI that render was unsuccessful
+ hr = renderClient->ReleaseBuffer( 0, 0 );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
+ goto Exit;
+ }
+ }
+ }
+ else
+ {
+ // Inform WASAPI that render was unsuccessful
+ hr = renderClient->ReleaseBuffer( 0, 0 );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
+ goto Exit;
+ }
+ }
+ }
+
+ // if the callback buffer was pushed renderBuffer reset callbackPulled flag
+ if ( callbackPushed ) {
+ callbackPulled = false;
+ }
+
+ // tick stream time
+ RtApi::tickStreamTime();
+ }
+
+Exit:
+ // clean up
+ CoTaskMemFree( captureFormat );
+ CoTaskMemFree( renderFormat );
+
+ free ( convBuffer );
+
+ CoUninitialize();
+
+ // update stream state
+ stream_.state = STREAM_STOPPED;
+
+ if ( errorText_.empty() )
+ return;
+ else
+ error( errorType );
+}
+
+//******************** End of __WINDOWS_WASAPI__ *********************//
+#endif
+
+
+#if defined(__WINDOWS_DS__) // Windows DirectSound API
+
+// Modified by Robin Davies, October 2005
+// - Improvements to DirectX pointer chasing.
+// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
+// - Auto-call CoInitialize for DSOUND and ASIO platforms.
+// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
+// Changed device query structure for RtAudio 4.0.7, January 2010
+
+#include <dsound.h>
+#include <assert.h>
+#include <algorithm>
+
+#if defined(__MINGW32__)
+ // missing from latest mingw winapi
+#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
+#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
+#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
+#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
+#endif
+
+#define MINIMUM_DEVICE_BUFFER_SIZE 32768
+
+#ifdef _MSC_VER // if Microsoft Visual C++
+#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
+#endif
+
+static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
+{
+ if ( pointer > bufferSize ) pointer -= bufferSize;
+ if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
+ if ( pointer < earlierPointer ) pointer += bufferSize;
+ return pointer >= earlierPointer && pointer < laterPointer;
+}
+
+// A structure to hold various information related to the DirectSound
+// API implementation.
+struct DsHandle {
+ unsigned int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
+ void *id[2];
+ void *buffer[2];
+ bool xrun[2];
+ UINT bufferPointer[2];
+ DWORD dsBufferSize[2];
+ DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
+ HANDLE condition;
+
+ DsHandle()
+ :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; }
+};
+
+// Declarations for utility functions, callbacks, and structures
+// specific to the DirectSound implementation.
+static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
+ LPCTSTR description,
+ LPCTSTR module,
+ LPVOID lpContext );
+
+static const char* getErrorString( int code );
+
+static unsigned __stdcall callbackHandler( void *ptr );
+
+struct DsDevice {
+ LPGUID id[2];
+ bool validId[2];
+ bool found;
+ std::string name;
+
+ DsDevice()
+ : found(false) { validId[0] = false; validId[1] = false; }
+};
+
+struct DsProbeData {
+ bool isInput;
+ std::vector<struct DsDevice>* dsDevices;
+};
+
+RtApiDs :: RtApiDs()
+{
+ // Dsound will run both-threaded. If CoInitialize fails, then just
+ // accept whatever the mainline chose for a threading model.
+ coInitialized_ = false;
+ HRESULT hr = CoInitialize( NULL );
+ if ( !FAILED( hr ) ) coInitialized_ = true;
+}
+
+RtApiDs :: ~RtApiDs()
+{
+ if ( coInitialized_ ) CoUninitialize(); // balanced call.
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
+
+// The DirectSound default output is always the first device.
+unsigned int RtApiDs :: getDefaultOutputDevice( void )
+{
+ return 0;
+}
+
+// The DirectSound default input is always the first input device,
+// which is the first capture device enumerated.
+unsigned int RtApiDs :: getDefaultInputDevice( void )
+{
+ return 0;
+}
+
+unsigned int RtApiDs :: getDeviceCount( void )
+{
+ // Set query flag for previously found devices to false, so that we
+ // can check for any devices that have disappeared.
+ for ( unsigned int i=0; i<dsDevices.size(); i++ )
+ dsDevices[i].found = false;
+
+ // Query DirectSound devices.
+ struct DsProbeData probeInfo;
+ probeInfo.isInput = false;
+ probeInfo.dsDevices = &dsDevices;
+ HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ }
+
+ // Query DirectSoundCapture devices.
+ probeInfo.isInput = true;
+ result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ }
+
+ // Clean out any devices that may have disappeared.
+ std::vector< int > indices;
+ for ( unsigned int i=0; i<dsDevices.size(); i++ )
+ if ( dsDevices[i].found == false ) indices.push_back( i );
+ //unsigned int nErased = 0;
+ for ( unsigned int i=0; i<indices.size(); i++ )
+ dsDevices.erase( dsDevices.begin()+indices[i] );
+ //dsDevices.erase( dsDevices.begin()-nErased++ );
+
+ return static_cast<unsigned int>(dsDevices.size());
+}
+
+RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ if ( dsDevices.size() == 0 ) {
+ // Force a query of all devices
+ getDeviceCount();
+ if ( dsDevices.size() == 0 ) {
+ errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+ }
+
+ if ( device >= dsDevices.size() ) {
+ errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ HRESULT result;
+ if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
+
+ LPDIRECTSOUND output;
+ DSCAPS outCaps;
+ result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto probeInput;
+ }
+
+ outCaps.dwSize = sizeof( outCaps );
+ result = output->GetCaps( &outCaps );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto probeInput;
+ }
+
+ // Get output channel information.
+ info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
+
+ // Get sample rate information.
+ info.sampleRates.clear();
+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+ if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
+ SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate )
+ info.sampleRates.push_back( SAMPLE_RATES[k] );
+ }
+
+ // Get format information.
+ if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
+
+ output->Release();
+
+ if ( getDefaultOutputDevice() == device )
+ info.isDefaultOutput = true;
+
+ if ( dsDevices[ device ].validId[1] == false ) {
+ info.name = dsDevices[ device ].name;
+ info.probed = true;
+ return info;
+ }
+
+ probeInput:
+
+ LPDIRECTSOUNDCAPTURE input;
+ result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ DSCCAPS inCaps;
+ inCaps.dwSize = sizeof( inCaps );
+ result = input->GetCaps( &inCaps );
+ if ( FAILED( result ) ) {
+ input->Release();
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Get input channel information.
+ info.inputChannels = inCaps.dwChannels;
+
+ // Get sample rate and format information.
+ std::vector<unsigned int> rates;
+ if ( inCaps.dwChannels >= 2 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+
+ if ( info.nativeFormats & RTAUDIO_SINT16 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
+ }
+ else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
+ }
+ }
+ else if ( inCaps.dwChannels == 1 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+
+ if ( info.nativeFormats & RTAUDIO_SINT16 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
+ }
+ else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
+ }
+ }
+ else info.inputChannels = 0; // technically, this would be an error
+
+ input->Release();
+
+ if ( info.inputChannels == 0 ) return info;
+
+ // Copy the supported rates to the info structure but avoid duplication.
+ bool found;
+ for ( unsigned int i=0; i<rates.size(); i++ ) {
+ found = false;
+ for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
+ if ( rates[i] == info.sampleRates[j] ) {
+ found = true;
+ break;
+ }
+ }
+ if ( found == false ) info.sampleRates.push_back( rates[i] );
+ }
+ std::sort( info.sampleRates.begin(), info.sampleRates.end() );
+
+ // If device opens for both playback and capture, we determine the channels.
+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+
+ if ( device == 0 ) info.isDefaultInput = true;
+
+ // Copy name and return.
+ info.name = dsDevices[ device ].name;
+ info.probed = true;
+ return info;
+}
+
+bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ if ( channels + firstChannel > 2 ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
+ return FAILURE;
+ }
+
+ size_t nDevices = dsDevices.size();
+ if ( nDevices == 0 ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
+ return FAILURE;
+ }
+
+ if ( device >= nDevices ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
+
+ if ( mode == OUTPUT ) {
+ if ( dsDevices[ device ].validId[0] == false ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+ else { // mode == INPUT
+ if ( dsDevices[ device ].validId[1] == false ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // According to a note in PortAudio, using GetDesktopWindow()
+ // instead of GetForegroundWindow() is supposed to avoid problems
+ // that occur when the application's window is not the foreground
+ // window. Also, if the application window closes before the
+ // DirectSound buffer, DirectSound can crash. In the past, I had
+ // problems when using GetDesktopWindow() but it seems fine now
+ // (January 2010). I'll leave it commented here.
+ // HWND hWnd = GetForegroundWindow();
+ HWND hWnd = GetDesktopWindow();
+
+ // Check the numberOfBuffers parameter and limit the lowest value to
+ // two. This is a judgement call and a value of two is probably too
+ // low for capture, but it should work for playback.
+ int nBuffers = 0;
+ if ( options ) nBuffers = options->numberOfBuffers;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
+ if ( nBuffers < 2 ) nBuffers = 3;
+
+ // Check the lower range of the user-specified buffer size and set
+ // (arbitrarily) to a lower bound of 32.
+ if ( *bufferSize < 32 ) *bufferSize = 32;
+
+ // Create the wave format structure. The data format setting will
+ // be determined later.
+ WAVEFORMATEX waveFormat;
+ ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
+ waveFormat.wFormatTag = WAVE_FORMAT_PCM;
+ waveFormat.nChannels = channels + firstChannel;
+ waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
+
+ // Determine the device buffer size. By default, we'll use the value
+ // defined above (32K), but we will grow it to make allowances for
+ // very large software buffer sizes.
+ DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
+ DWORD dsPointerLeadTime = 0;
+
+ void *ohandle = 0, *bhandle = 0;
+ HRESULT result;
+ if ( mode == OUTPUT ) {
+
+ LPDIRECTSOUND output;
+ result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ DSCAPS outCaps;
+ outCaps.dwSize = sizeof( outCaps );
+ result = output->GetCaps( &outCaps );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Check channel information.
+ if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
+ errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Check format information. Use 16-bit format unless not
+ // supported or user requests 8-bit.
+ if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
+ !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
+ waveFormat.wBitsPerSample = 16;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ else {
+ waveFormat.wBitsPerSample = 8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ stream_.userFormat = format;
+
+ // Update wave format structure and buffer information.
+ waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+ waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
+
+ // If the user wants an even bigger buffer, increase the device buffer size accordingly.
+ while ( dsPointerLeadTime * 2U > dsBufferSize )
+ dsBufferSize *= 2;
+
+ // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
+ // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
+ // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
+ result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Even though we will write to the secondary buffer, we need to
+ // access the primary buffer to set the correct output format
+ // (since the default is 8-bit, 22 kHz!). Setup the DS primary
+ // buffer description.
+ DSBUFFERDESC bufferDescription;
+ ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
+ bufferDescription.dwSize = sizeof( DSBUFFERDESC );
+ bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
+
+ // Obtain the primary buffer
+ LPDIRECTSOUNDBUFFER buffer;
+ result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Set the primary DS buffer sound format.
+ result = buffer->SetFormat( &waveFormat );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Setup the secondary DS buffer description.
+ ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
+ bufferDescription.dwSize = sizeof( DSBUFFERDESC );
+ bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
+ DSBCAPS_GLOBALFOCUS |
+ DSBCAPS_GETCURRENTPOSITION2 |
+ DSBCAPS_LOCHARDWARE ); // Force hardware mixing
+ bufferDescription.dwBufferBytes = dsBufferSize;
+ bufferDescription.lpwfxFormat = &waveFormat;
+
+ // Try to create the secondary DS buffer. If that doesn't work,
+ // try to use software mixing. Otherwise, there's a problem.
+ result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
+ DSBCAPS_GLOBALFOCUS |
+ DSBCAPS_GETCURRENTPOSITION2 |
+ DSBCAPS_LOCSOFTWARE ); // Force software mixing
+ result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // Get the buffer size ... might be different from what we specified.
+ DSBCAPS dsbcaps;
+ dsbcaps.dwSize = sizeof( DSBCAPS );
+ result = buffer->GetCaps( &dsbcaps );
+ if ( FAILED( result ) ) {
+ output->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ dsBufferSize = dsbcaps.dwBufferBytes;
+
+ // Lock the DS buffer
+ LPVOID audioPtr;
+ DWORD dataLen;
+ result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ output->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Zero the DS buffer
+ ZeroMemory( audioPtr, dataLen );
+
+ // Unlock the DS buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ output->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ ohandle = (void *) output;
+ bhandle = (void *) buffer;
+ }
+
+ if ( mode == INPUT ) {
+
+ LPDIRECTSOUNDCAPTURE input;
+ result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ DSCCAPS inCaps;
+ inCaps.dwSize = sizeof( inCaps );
+ result = input->GetCaps( &inCaps );
+ if ( FAILED( result ) ) {
+ input->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Check channel information.
+ if ( inCaps.dwChannels < channels + firstChannel ) {
+ errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
+ return FAILURE;
+ }
+
+ // Check format information. Use 16-bit format unless user
+ // requests 8-bit.
+ DWORD deviceFormats;
+ if ( channels + firstChannel == 2 ) {
+ deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
+ if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
+ waveFormat.wBitsPerSample = 8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ else { // assume 16-bit is supported
+ waveFormat.wBitsPerSample = 16;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ }
+ else { // channel == 1
+ deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
+ if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
+ waveFormat.wBitsPerSample = 8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ else { // assume 16-bit is supported
+ waveFormat.wBitsPerSample = 16;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ }
+ stream_.userFormat = format;
+
+ // Update wave format structure and buffer information.
+ waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+ waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
+
+ // If the user wants an even bigger buffer, increase the device buffer size accordingly.
+ while ( dsPointerLeadTime * 2U > dsBufferSize )
+ dsBufferSize *= 2;
+
+ // Setup the secondary DS buffer description.
+ DSCBUFFERDESC bufferDescription;
+ ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
+ bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
+ bufferDescription.dwFlags = 0;
+ bufferDescription.dwReserved = 0;
+ bufferDescription.dwBufferBytes = dsBufferSize;
+ bufferDescription.lpwfxFormat = &waveFormat;
+
+ // Create the capture buffer.
+ LPDIRECTSOUNDCAPTUREBUFFER buffer;
+ result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ input->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Get the buffer size ... might be different from what we specified.
+ DSCBCAPS dscbcaps;
+ dscbcaps.dwSize = sizeof( DSCBCAPS );
+ result = buffer->GetCaps( &dscbcaps );
+ if ( FAILED( result ) ) {
+ input->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ dsBufferSize = dscbcaps.dwBufferBytes;
+
+ // NOTE: We could have a problem here if this is a duplex stream
+ // and the play and capture hardware buffer sizes are different
+ // (I'm actually not sure if that is a problem or not).
+ // Currently, we are not verifying that.
+
+ // Lock the capture buffer
+ LPVOID audioPtr;
+ DWORD dataLen;
+ result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ input->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Zero the buffer
+ ZeroMemory( audioPtr, dataLen );
+
+ // Unlock the buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ input->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ ohandle = (void *) input;
+ bhandle = (void *) buffer;
+ }
+
+ // Set various stream parameters
+ DsHandle *handle = 0;
+ stream_.nDeviceChannels[mode] = channels + firstChannel;
+ stream_.nUserChannels[mode] = channels;
+ stream_.bufferSize = *bufferSize;
+ stream_.channelOffset[mode] = firstChannel;
+ stream_.deviceInterleaved[mode] = true;
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
+
+ // Set flag for buffer conversion
+ stream_.doConvertBuffer[mode] = false;
+ if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if (stream_.userFormat != stream_.deviceFormat[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate necessary internal buffers
+ long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
+ }
+ }
+
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
+
+ // Allocate our DsHandle structures for the stream.
+ if ( stream_.apiHandle == 0 ) {
+ try {
+ handle = new DsHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
+ goto error;
+ }
+
+ // Create a manual-reset event.
+ handle->condition = CreateEvent( NULL, // no security
+ TRUE, // manual-reset
+ FALSE, // non-signaled initially
+ NULL ); // unnamed
+ stream_.apiHandle = (void *) handle;
+ }
+ else
+ handle = (DsHandle *) stream_.apiHandle;
+ handle->id[mode] = ohandle;
+ handle->buffer[mode] = bhandle;
+ handle->dsBufferSize[mode] = dsBufferSize;
+ handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
+
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ // We had already set up an output stream.
+ stream_.mode = DUPLEX;
+ else
+ stream_.mode = mode;
+ stream_.nBuffers = nBuffers;
+ stream_.sampleRate = sampleRate;
+
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+
+ // Setup the callback thread.
+ if ( stream_.callbackInfo.isRunning == false ) {
+ unsigned threadId;
+ stream_.callbackInfo.isRunning = true;
+ stream_.callbackInfo.object = (void *) this;
+ stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
+ &stream_.callbackInfo, 0, &threadId );
+ if ( stream_.callbackInfo.thread == 0 ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
+ goto error;
+ }
+
+ // Boost DS thread priority
+ SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
+ }
+ return SUCCESS;
+
+ error:
+ if ( handle ) {
+ if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
+ LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ if ( buffer ) buffer->Release();
+ object->Release();
+ }
+ if ( handle->buffer[1] ) {
+ LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ if ( buffer ) buffer->Release();
+ object->Release();
+ }
+ CloseHandle( handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.state = STREAM_CLOSED;
+ return FAILURE;
+}
+
+void RtApiDs :: closeStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiDs::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ // Stop the callback thread.
+ stream_.callbackInfo.isRunning = false;
+ WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
+ CloseHandle( (HANDLE) stream_.callbackInfo.thread );
+
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ if ( handle ) {
+ if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
+ LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ if ( buffer ) {
+ buffer->Stop();
+ buffer->Release();
+ }
+ object->Release();
+ }
+ if ( handle->buffer[1] ) {
+ LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ if ( buffer ) {
+ buffer->Stop();
+ buffer->Release();
+ }
+ object->Release();
+ }
+ CloseHandle( handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
+
+void RtApiDs :: startStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiDs::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+
+ // Increase scheduler frequency on lesser windows (a side-effect of
+ // increasing timer accuracy). On greater windows (Win2K or later),
+ // this is already in effect.
+ timeBeginPeriod( 1 );
+
+ buffersRolling = false;
+ duplexPrerollBytes = 0;
+
+ if ( stream_.mode == DUPLEX ) {
+ // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
+ duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
+ }
+
+ HRESULT result = 0;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ result = buffer->Start( DSCBSTART_LOOPING );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ ResetEvent( handle->condition );
+ stream_.state = STREAM_RUNNING;
+
+ unlock:
+ if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiDs :: stopStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ HRESULT result = 0;
+ LPVOID audioPtr;
+ DWORD dataLen;
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 2;
+ WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
+ }
+
+ stream_.state = STREAM_STOPPED;
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ // Stop the buffer and clear memory
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ result = buffer->Stop();
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // Lock the buffer and clear it so that if we start to play again,
+ // we won't have old data playing.
+ result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // Zero the DS buffer
+ ZeroMemory( audioPtr, dataLen );
+
+ // Unlock the DS buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // If we start playing again, we must begin at beginning of buffer.
+ handle->bufferPointer[0] = 0;
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ audioPtr = NULL;
+ dataLen = 0;
+
+ stream_.state = STREAM_STOPPED;
+
+ if ( stream_.mode != DUPLEX )
+ MUTEX_LOCK( &stream_.mutex );
+
+ result = buffer->Stop();
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // Lock the buffer and clear it so that if we start to play again,
+ // we won't have old data playing.
+ result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // Zero the DS buffer
+ ZeroMemory( audioPtr, dataLen );
+
+ // Unlock the DS buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // If we start recording again, we must begin at beginning of buffer.
+ handle->bufferPointer[1] = 0;
+ }
+
+ unlock:
+ timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiDs :: abortStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ handle->drainCounter = 2;
+
+ stopStream();
+}
+
+void RtApiDs :: callbackEvent()
+{
+ if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
+ Sleep( 50 ); // sleep 50 milliseconds
+ return;
+ }
+
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+
+ // Check if we were draining the stream and signal is finished.
+ if ( handle->drainCounter > stream_.nBuffers + 2 ) {
+
+ stream_.state = STREAM_STOPPING;
+ if ( handle->internalDrain == false )
+ SetEvent( handle->condition );
+ else
+ stopStream();
+ return;
+ }
+
+ // Invoke user callback to get fresh output data UNLESS we are
+ // draining stream.
+ if ( handle->drainCounter == 0 ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
+ }
+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( cbReturnValue == 2 ) {
+ stream_.state = STREAM_STOPPING;
+ handle->drainCounter = 2;
+ abortStream();
+ return;
+ }
+ else if ( cbReturnValue == 1 ) {
+ handle->drainCounter = 1;
+ handle->internalDrain = true;
+ }
+ }
+
+ HRESULT result;
+ DWORD currentWritePointer, safeWritePointer;
+ DWORD currentReadPointer, safeReadPointer;
+ UINT nextWritePointer;
+
+ LPVOID buffer1 = NULL;
+ LPVOID buffer2 = NULL;
+ DWORD bufferSize1 = 0;
+ DWORD bufferSize2 = 0;
+
+ char *buffer;
+ long bufferBytes;
+
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+
+ if ( buffersRolling == false ) {
+ if ( stream_.mode == DUPLEX ) {
+ //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
+
+ // It takes a while for the devices to get rolling. As a result,
+ // there's no guarantee that the capture and write device pointers
+ // will move in lockstep. Wait here for both devices to start
+ // rolling, and then set our buffer pointers accordingly.
+ // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
+ // bytes later than the write buffer.
+
+ // Stub: a serious risk of having a pre-emptive scheduling round
+ // take place between the two GetCurrentPosition calls... but I'm
+ // really not sure how to solve the problem. Temporarily boost to
+ // Realtime priority, maybe; but I'm not sure what priority the
+ // DirectSound service threads run at. We *should* be roughly
+ // within a ms or so of correct.
+
+ LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+
+ DWORD startSafeWritePointer, startSafeReadPointer;
+
+ result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ while ( true ) {
+ result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
+ Sleep( 1 );
+ }
+
+ //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
+
+ handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
+ if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
+ handle->bufferPointer[1] = safeReadPointer;
+ }
+ else if ( stream_.mode == OUTPUT ) {
+
+ // Set the proper nextWritePosition after initial startup.
+ LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
+ if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
+ }
+
+ buffersRolling = true;
+ }
+
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+
+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+ bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
+ bufferBytes *= formatBytes( stream_.userFormat );
+ memset( stream_.userBuffer[0], 0, bufferBytes );
+ }
+
+ // Setup parameters and do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
+ bufferBytes *= formatBytes( stream_.deviceFormat[0] );
+ }
+ else {
+ buffer = stream_.userBuffer[0];
+ bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
+ bufferBytes *= formatBytes( stream_.userFormat );
+ }
+
+ // No byte swapping necessary in DirectSound implementation.
+
+ // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
+ // unsigned. So, we need to convert our signed 8-bit data here to
+ // unsigned.
+ if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
+ for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
+
+ DWORD dsBufferSize = handle->dsBufferSize[0];
+ nextWritePointer = handle->bufferPointer[0];
+
+ DWORD endWrite, leadPointer;
+ while ( true ) {
+ // Find out where the read and "safe write" pointers are.
+ result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+
+ // We will copy our output buffer into the region between
+ // safeWritePointer and leadPointer. If leadPointer is not
+ // beyond the next endWrite position, wait until it is.
+ leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
+ //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
+ if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
+ if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
+ endWrite = nextWritePointer + bufferBytes;
+
+ // Check whether the entire write region is behind the play pointer.
+ if ( leadPointer >= endWrite ) break;
+
+ // If we are here, then we must wait until the leadPointer advances
+ // beyond the end of our next write region. We use the
+ // Sleep() function to suspend operation until that happens.
+ double millis = ( endWrite - leadPointer ) * 1000.0;
+ millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
+ if ( millis < 1.0 ) millis = 1.0;
+ Sleep( (DWORD) millis );
+ }
+
+ if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
+ || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
+ // We've strayed into the forbidden zone ... resync the read pointer.
+ handle->xrun[0] = true;
+ nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
+ if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
+ handle->bufferPointer[0] = nextWritePointer;
+ endWrite = nextWritePointer + bufferBytes;
+ }
+
+ // Lock free space in the buffer
+ result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
+ &bufferSize1, &buffer2, &bufferSize2, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+
+ // Copy our buffer into the DS buffer
+ CopyMemory( buffer1, buffer, bufferSize1 );
+ if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
+
+ // Update our buffer offset and unlock sound buffer
+ dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
+ handle->bufferPointer[0] = nextWritePointer;
+ }
+
+ // Don't bother draining input
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ // Setup parameters.
+ if ( stream_.doConvertBuffer[1] ) {
+ buffer = stream_.deviceBuffer;
+ bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
+ bufferBytes *= formatBytes( stream_.deviceFormat[1] );
+ }
+ else {
+ buffer = stream_.userBuffer[1];
+ bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
+ bufferBytes *= formatBytes( stream_.userFormat );
+ }
+
+ LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ long nextReadPointer = handle->bufferPointer[1];
+ DWORD dsBufferSize = handle->dsBufferSize[1];
+
+ // Find out where the write and "safe read" pointers are.
+ result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+
+ if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
+ DWORD endRead = nextReadPointer + bufferBytes;
+
+ // Handling depends on whether we are INPUT or DUPLEX.
+ // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
+ // then a wait here will drag the write pointers into the forbidden zone.
+ //
+ // In DUPLEX mode, rather than wait, we will back off the read pointer until
+ // it's in a safe position. This causes dropouts, but it seems to be the only
+ // practical way to sync up the read and write pointers reliably, given the
+ // the very complex relationship between phase and increment of the read and write
+ // pointers.
+ //
+ // In order to minimize audible dropouts in DUPLEX mode, we will
+ // provide a pre-roll period of 0.5 seconds in which we return
+ // zeros from the read buffer while the pointers sync up.
+
+ if ( stream_.mode == DUPLEX ) {
+ if ( safeReadPointer < endRead ) {
+ if ( duplexPrerollBytes <= 0 ) {
+ // Pre-roll time over. Be more agressive.
+ int adjustment = endRead-safeReadPointer;
+
+ handle->xrun[1] = true;
+ // Two cases:
+ // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
+ // and perform fine adjustments later.
+ // - small adjustments: back off by twice as much.
+ if ( adjustment >= 2*bufferBytes )
+ nextReadPointer = safeReadPointer-2*bufferBytes;
+ else
+ nextReadPointer = safeReadPointer-bufferBytes-adjustment;
+
+ if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
+
+ }
+ else {
+ // In pre=roll time. Just do it.
+ nextReadPointer = safeReadPointer - bufferBytes;
+ while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
+ }
+ endRead = nextReadPointer + bufferBytes;
+ }
+ }
+ else { // mode == INPUT
+ while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
+ // See comments for playback.
+ double millis = (endRead - safeReadPointer) * 1000.0;
+ millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
+ if ( millis < 1.0 ) millis = 1.0;
+ Sleep( (DWORD) millis );
+
+ // Wake up and find out where we are now.
+ result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+
+ if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
+ }
+ }
+
+ // Lock free space in the buffer
+ result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
+ &bufferSize1, &buffer2, &bufferSize2, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+
+ if ( duplexPrerollBytes <= 0 ) {
+ // Copy our buffer into the DS buffer
+ CopyMemory( buffer, buffer1, bufferSize1 );
+ if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
+ }
+ else {
+ memset( buffer, 0, bufferSize1 );
+ if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
+ duplexPrerollBytes -= bufferSize1 + bufferSize2;
+ }
+
+ // Update our buffer offset and unlock sound buffer
+ nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
+ dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ handle->bufferPointer[1] = nextReadPointer;
+
+ // No byte swapping necessary in DirectSound implementation.
+
+ // If necessary, convert 8-bit data from unsigned to signed.
+ if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
+ for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
+
+ // Do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[1] )
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+ }
+
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
+ RtApi::tickStreamTime();
+}
+
+// Definitions for utility functions and callbacks
+// specific to the DirectSound implementation.
+
+static unsigned __stdcall callbackHandler( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiDs *object = (RtApiDs *) info->object;
+ bool* isRunning = &info->isRunning;
+
+ while ( *isRunning == true ) {
+ object->callbackEvent();
+ }
+
+ _endthreadex( 0 );
+ return 0;
+}
+
+#include "tchar.h"
+
+static std::string convertTChar( LPCTSTR name )
+{
+#if defined( UNICODE ) || defined( _UNICODE )
+ int length = WideCharToMultiByte(CP_UTF8, 0, name, -1, NULL, 0, NULL, NULL);
+ std::string s( length-1, '\0' );
+ WideCharToMultiByte(CP_UTF8, 0, name, -1, &s[0], length, NULL, NULL);
+#else
+ std::string s( name );
+#endif
+
+ return s;
+}
+
+static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
+ LPCTSTR description,
+ LPCTSTR /*module*/,
+ LPVOID lpContext )
+{
+ struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
+ std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
+
+ HRESULT hr;
+ bool validDevice = false;
+ if ( probeInfo.isInput == true ) {
+ DSCCAPS caps;
+ LPDIRECTSOUNDCAPTURE object;
+
+ hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
+ if ( hr != DS_OK ) return TRUE;
+
+ caps.dwSize = sizeof(caps);
+ hr = object->GetCaps( &caps );
+ if ( hr == DS_OK ) {
+ if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
+ validDevice = true;
+ }
+ object->Release();
+ }
+ else {
+ DSCAPS caps;
+ LPDIRECTSOUND object;
+ hr = DirectSoundCreate( lpguid, &object, NULL );
+ if ( hr != DS_OK ) return TRUE;
+
+ caps.dwSize = sizeof(caps);
+ hr = object->GetCaps( &caps );
+ if ( hr == DS_OK ) {
+ if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
+ validDevice = true;
+ }
+ object->Release();
+ }
+
+ // If good device, then save its name and guid.
+ std::string name = convertTChar( description );
+ //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
+ if ( lpguid == NULL )
+ name = "Default Device";
+ if ( validDevice ) {
+ for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
+ if ( dsDevices[i].name == name ) {
+ dsDevices[i].found = true;
+ if ( probeInfo.isInput ) {
+ dsDevices[i].id[1] = lpguid;
+ dsDevices[i].validId[1] = true;
+ }
+ else {
+ dsDevices[i].id[0] = lpguid;
+ dsDevices[i].validId[0] = true;
+ }
+ return TRUE;
+ }
+ }
+
+ DsDevice device;
+ device.name = name;
+ device.found = true;
+ if ( probeInfo.isInput ) {
+ device.id[1] = lpguid;
+ device.validId[1] = true;
+ }
+ else {
+ device.id[0] = lpguid;
+ device.validId[0] = true;
+ }
+ dsDevices.push_back( device );
+ }
+
+ return TRUE;
+}
+
+static const char* getErrorString( int code )
+{
+ switch ( code ) {
+
+ case DSERR_ALLOCATED:
+ return "Already allocated";
+
+ case DSERR_CONTROLUNAVAIL:
+ return "Control unavailable";
+
+ case DSERR_INVALIDPARAM:
+ return "Invalid parameter";
+
+ case DSERR_INVALIDCALL:
+ return "Invalid call";
+
+ case DSERR_GENERIC:
+ return "Generic error";
+
+ case DSERR_PRIOLEVELNEEDED:
+ return "Priority level needed";
+
+ case DSERR_OUTOFMEMORY:
+ return "Out of memory";
+
+ case DSERR_BADFORMAT:
+ return "The sample rate or the channel format is not supported";
+
+ case DSERR_UNSUPPORTED:
+ return "Not supported";
+
+ case DSERR_NODRIVER:
+ return "No driver";
+
+ case DSERR_ALREADYINITIALIZED:
+ return "Already initialized";
+
+ case DSERR_NOAGGREGATION:
+ return "No aggregation";
+
+ case DSERR_BUFFERLOST:
+ return "Buffer lost";
+
+ case DSERR_OTHERAPPHASPRIO:
+ return "Another application already has priority";
+
+ case DSERR_UNINITIALIZED:
+ return "Uninitialized";
+
+ default:
+ return "DirectSound unknown error";
+ }
+}
+//******************** End of __WINDOWS_DS__ *********************//
+#endif
+
+
+#if defined(__LINUX_ALSA__)
+
+#include <alsa/asoundlib.h>
+#include <unistd.h>
+
+ // A structure to hold various information related to the ALSA API
+ // implementation.
+struct AlsaHandle {
+ snd_pcm_t *handles[2];
+ bool synchronized;
+ bool xrun[2];
+ pthread_cond_t runnable_cv;
+ bool runnable;
+
+ AlsaHandle()
+ :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
+};
+
+static void *alsaCallbackHandler( void * ptr );
+
+RtApiAlsa :: RtApiAlsa()
+{
+ // Nothing to do here.
+}
+
+RtApiAlsa :: ~RtApiAlsa()
+{
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
+
+unsigned int RtApiAlsa :: getDeviceCount( void )
+{
+ unsigned nDevices = 0;
+ int result, subdevice, card;
+ char name[64];
+ snd_ctl_t *handle;
+
+ // Count cards and devices
+ card = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &handle, name, 0 );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto nextcard;
+ }
+ subdevice = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( handle, &subdevice );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ break;
+ }
+ if ( subdevice < 0 )
+ break;
+ nDevices++;
+ }
+ nextcard:
+ snd_ctl_close( handle );
+ snd_card_next( &card );
+ }
+
+ result = snd_ctl_open( &handle, "default", 0 );
+ if (result == 0) {
+ nDevices++;
+ snd_ctl_close( handle );
+ }
+
+ return nDevices;
+}
+
+RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ unsigned nDevices = 0;
+ int result, subdevice, card;
+ char name[64];
+ snd_ctl_t *chandle;
+
+ // Count cards and devices
+ card = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto nextcard;
+ }
+ subdevice = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( chandle, &subdevice );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ break;
+ }
+ if ( subdevice < 0 ) break;
+ if ( nDevices == device ) {
+ sprintf( name, "hw:%d,%d", card, subdevice );
+ goto foundDevice;
+ }
+ nDevices++;
+ }
+ nextcard:
+ snd_ctl_close( chandle );
+ snd_card_next( &card );
+ }
+
+ result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
+ if ( result == 0 ) {
+ if ( nDevices == device ) {
+ strcpy( name, "default" );
+ goto foundDevice;
+ }
+ nDevices++;
+ }
+
+ if ( nDevices == 0 ) {
+ errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ foundDevice:
+
+ // If a stream is already open, we cannot probe the stream devices.
+ // Thus, use the saved results.
+ if ( stream_.state != STREAM_CLOSED &&
+ ( stream_.device[0] == device || stream_.device[1] == device ) ) {
+ snd_ctl_close( chandle );
+ if ( device >= devices_.size() ) {
+ errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+ return devices_[ device ];
+ }
+
+ int openMode = SND_PCM_ASYNC;
+ snd_pcm_stream_t stream;
+ snd_pcm_info_t *pcminfo;
+ snd_pcm_info_alloca( &pcminfo );
+ snd_pcm_t *phandle;
+ snd_pcm_hw_params_t *params;
+ snd_pcm_hw_params_alloca( ¶ms );
+
+ // First try for playback unless default device (which has subdev -1)
+ stream = SND_PCM_STREAM_PLAYBACK;
+ snd_pcm_info_set_stream( pcminfo, stream );
+ if ( subdevice != -1 ) {
+ snd_pcm_info_set_device( pcminfo, subdevice );
+ snd_pcm_info_set_subdevice( pcminfo, 0 );
+
+ result = snd_ctl_pcm_info( chandle, pcminfo );
+ if ( result < 0 ) {
+ // Device probably doesn't support playback.
+ goto captureProbe;
+ }
+ }
+
+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto captureProbe;
+ }
+
+ // The device is open ... fill the parameter structure.
+ result = snd_pcm_hw_params_any( phandle, params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto captureProbe;
+ }
+
+ // Get output channel information.
+ unsigned int value;
+ result = snd_pcm_hw_params_get_channels_max( params, &value );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto captureProbe;
+ }
+ info.outputChannels = value;
+ snd_pcm_close( phandle );
+
+ captureProbe:
+ stream = SND_PCM_STREAM_CAPTURE;
+ snd_pcm_info_set_stream( pcminfo, stream );
+
+ // Now try for capture unless default device (with subdev = -1)
+ if ( subdevice != -1 ) {
+ result = snd_ctl_pcm_info( chandle, pcminfo );
+ snd_ctl_close( chandle );
+ if ( result < 0 ) {
+ // Device probably doesn't support capture.
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
+ }
+ else
+ snd_ctl_close( chandle );
+
+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
+
+ // The device is open ... fill the parameter structure.
+ result = snd_pcm_hw_params_any( phandle, params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
+
+ result = snd_pcm_hw_params_get_channels_max( params, &value );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
+ info.inputChannels = value;
+ snd_pcm_close( phandle );
+
+ // If device opens for both playback and capture, we determine the channels.
+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+
+ // ALSA doesn't provide default devices so we'll use the first available one.
+ if ( device == 0 && info.outputChannels > 0 )
+ info.isDefaultOutput = true;
+ if ( device == 0 && info.inputChannels > 0 )
+ info.isDefaultInput = true;
+
+ probeParameters:
+ // At this point, we just need to figure out the supported data
+ // formats and sample rates. We'll proceed by opening the device in
+ // the direction with the maximum number of channels, or playback if
+ // they are equal. This might limit our sample rate options, but so
+ // be it.
+
+ if ( info.outputChannels >= info.inputChannels )
+ stream = SND_PCM_STREAM_PLAYBACK;
+ else
+ stream = SND_PCM_STREAM_CAPTURE;
+ snd_pcm_info_set_stream( pcminfo, stream );
+
+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // The device is open ... fill the parameter structure.
+ result = snd_pcm_hw_params_any( phandle, params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Test our discrete set of sample rate values.
+ info.sampleRates.clear();
+ for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
+ if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 )
+ info.sampleRates.push_back( SAMPLE_RATES[i] );
+ }
+ if ( info.sampleRates.size() == 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Probe the supported data formats ... we don't care about endian-ness just yet
+ snd_pcm_format_t format;
+ info.nativeFormats = 0;
+ format = SND_PCM_FORMAT_S8;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_SINT8;
+ format = SND_PCM_FORMAT_S16;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_SINT16;
+ format = SND_PCM_FORMAT_S24;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_SINT24;
+ format = SND_PCM_FORMAT_S32;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_SINT32;
+ format = SND_PCM_FORMAT_FLOAT;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+ format = SND_PCM_FORMAT_FLOAT64;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_FLOAT64;
+
+ // Check that we have at least one supported format
+ if ( info.nativeFormats == 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Get the device name
+ char *cardname;
+ result = snd_card_get_name( card, &cardname );
+ if ( result >= 0 ) {
+ sprintf( name, "hw:%s,%d", cardname, subdevice );
+ free( cardname );
+ }
+ info.name = name;
+
+ // That's all ... close the device and return
+ snd_pcm_close( phandle );
+ info.probed = true;
+ return info;
+}
+
+void RtApiAlsa :: saveDeviceInfo( void )
+{
+ devices_.clear();
+
+ unsigned int nDevices = getDeviceCount();
+ devices_.resize( nDevices );
+ for ( unsigned int i=0; i<nDevices; i++ )
+ devices_[i] = getDeviceInfo( i );
+}
+
+bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+
+{
+#if defined(__RTAUDIO_DEBUG__)
+ snd_output_t *out;
+ snd_output_stdio_attach(&out, stderr, 0);
+#endif
+
+ // I'm not using the "plug" interface ... too much inconsistent behavior.
+
+ unsigned nDevices = 0;
+ int result, subdevice, card;
+ char name[64];
+ snd_ctl_t *chandle;
+
+ if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
+ snprintf(name, sizeof(name), "%s", "default");
+ else {
+ // Count cards and devices
+ card = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ subdevice = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( chandle, &subdevice );
+ if ( result < 0 ) break;
+ if ( subdevice < 0 ) break;
+ if ( nDevices == device ) {
+ sprintf( name, "hw:%d,%d", card, subdevice );
+ snd_ctl_close( chandle );
+ goto foundDevice;
+ }
+ nDevices++;
+ }
+ snd_ctl_close( chandle );
+ snd_card_next( &card );
+ }
+
+ result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
+ if ( result == 0 ) {
+ if ( nDevices == device ) {
+ strcpy( name, "default" );
+ goto foundDevice;
+ }
+ nDevices++;
+ }
+
+ if ( nDevices == 0 ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
+ return FAILURE;
+ }
+
+ if ( device >= nDevices ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
+ }
+
+ foundDevice:
+
+ // The getDeviceInfo() function will not work for a device that is
+ // already open. Thus, we'll probe the system before opening a
+ // stream and save the results for use by getDeviceInfo().
+ if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
+ this->saveDeviceInfo();
+
+ snd_pcm_stream_t stream;
+ if ( mode == OUTPUT )
+ stream = SND_PCM_STREAM_PLAYBACK;
+ else
+ stream = SND_PCM_STREAM_CAPTURE;
+
+ snd_pcm_t *phandle;
+ int openMode = SND_PCM_ASYNC;
+ result = snd_pcm_open( &phandle, name, stream, openMode );
+ if ( result < 0 ) {
+ if ( mode == OUTPUT )
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
+ else
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Fill the parameter structure.
+ snd_pcm_hw_params_t *hw_params;
+ snd_pcm_hw_params_alloca( &hw_params );
+ result = snd_pcm_hw_params_any( phandle, hw_params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+#if defined(__RTAUDIO_DEBUG__)
+ fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
+ snd_pcm_hw_params_dump( hw_params, out );
+#endif
+
+ // Set access ... check user preference.
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
+ stream_.userInterleaved = false;
+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
+ if ( result < 0 ) {
+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
+ stream_.deviceInterleaved[mode] = true;
+ }
+ else
+ stream_.deviceInterleaved[mode] = false;
+ }
+ else {
+ stream_.userInterleaved = true;
+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
+ if ( result < 0 ) {
+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
+ stream_.deviceInterleaved[mode] = false;
+ }
+ else
+ stream_.deviceInterleaved[mode] = true;
+ }
+
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Determine how to set the device format.
+ stream_.userFormat = format;
+ snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
+
+ if ( format == RTAUDIO_SINT8 )
+ deviceFormat = SND_PCM_FORMAT_S8;
+ else if ( format == RTAUDIO_SINT16 )
+ deviceFormat = SND_PCM_FORMAT_S16;
+ else if ( format == RTAUDIO_SINT24 )
+ deviceFormat = SND_PCM_FORMAT_S24;
+ else if ( format == RTAUDIO_SINT32 )
+ deviceFormat = SND_PCM_FORMAT_S32;
+ else if ( format == RTAUDIO_FLOAT32 )
+ deviceFormat = SND_PCM_FORMAT_FLOAT;
+ else if ( format == RTAUDIO_FLOAT64 )
+ deviceFormat = SND_PCM_FORMAT_FLOAT64;
+
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
+ stream_.deviceFormat[mode] = format;
+ goto setFormat;
+ }
+
+ // The user requested format is not natively supported by the device.
+ deviceFormat = SND_PCM_FORMAT_FLOAT64;
+ if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
+ goto setFormat;
+ }
+
+ deviceFormat = SND_PCM_FORMAT_FLOAT;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ goto setFormat;
+ }
+
+ deviceFormat = SND_PCM_FORMAT_S32;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ goto setFormat;
+ }
+
+ deviceFormat = SND_PCM_FORMAT_S24;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ goto setFormat;
+ }
+
+ deviceFormat = SND_PCM_FORMAT_S16;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ goto setFormat;
+ }
+
+ deviceFormat = SND_PCM_FORMAT_S8;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ goto setFormat;
+ }
+
+ // If we get here, no supported format was found.
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+
+ setFormat:
+ result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Determine whether byte-swaping is necessary.
+ stream_.doByteSwap[mode] = false;
+ if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
+ result = snd_pcm_format_cpu_endian( deviceFormat );
+ if ( result == 0 )
+ stream_.doByteSwap[mode] = true;
+ else if (result < 0) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // Set the sample rate.
+ result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Determine the number of channels for this device. We support a possible
+ // minimum device channel number > than the value requested by the user.
+ stream_.nUserChannels[mode] = channels;
+ unsigned int value;
+ result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
+ unsigned int deviceChannels = value;
+ if ( result < 0 || deviceChannels < channels + firstChannel ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ deviceChannels = value;
+ if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
+ stream_.nDeviceChannels[mode] = deviceChannels;
+
+ // Set the device channels.
+ result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Set the buffer (or period) size.
+ int dir = 0;
+ snd_pcm_uframes_t periodSize = *bufferSize;
+ result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ *bufferSize = periodSize;
+
+ // Set the buffer number, which in ALSA is referred to as the "period".
+ unsigned int periods = 0;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
+ if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
+ if ( periods < 2 ) periods = 4; // a fairly safe default value
+ result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // If attempting to setup a duplex stream, the bufferSize parameter
+ // MUST be the same in both directions!
+ if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ stream_.bufferSize = *bufferSize;
+
+ // Install the hardware configuration
+ result = snd_pcm_hw_params( phandle, hw_params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+#if defined(__RTAUDIO_DEBUG__)
+ fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
+ snd_pcm_hw_params_dump( hw_params, out );
+#endif
+
+ // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
+ snd_pcm_sw_params_t *sw_params = NULL;
+ snd_pcm_sw_params_alloca( &sw_params );
+ snd_pcm_sw_params_current( phandle, sw_params );
+ snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
+ snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
+ snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
+
+ // The following two settings were suggested by Theo Veenker
+ //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
+ //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
+
+ // here are two options for a fix
+ //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
+ snd_pcm_uframes_t val;
+ snd_pcm_sw_params_get_boundary( sw_params, &val );
+ snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
+
+ result = snd_pcm_sw_params( phandle, sw_params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+#if defined(__RTAUDIO_DEBUG__)
+ fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
+ snd_pcm_sw_params_dump( sw_params, out );
+#endif
+
+ // Set flags for buffer conversion
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate the ApiHandle if necessary and then save.
+ AlsaHandle *apiInfo = 0;
+ if ( stream_.apiHandle == 0 ) {
+ try {
+ apiInfo = (AlsaHandle *) new AlsaHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
+ goto error;
+ }
+
+ if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
+ goto error;
+ }
+
+ stream_.apiHandle = (void *) apiInfo;
+ apiInfo->handles[0] = 0;
+ apiInfo->handles[1] = 0;
+ }
+ else {
+ apiInfo = (AlsaHandle *) stream_.apiHandle;
+ }
+ apiInfo->handles[mode] = phandle;
+ phandle = 0;
+
+ // Allocate necessary internal buffers.
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
+ }
+ }
+
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
+
+ stream_.sampleRate = sampleRate;
+ stream_.nBuffers = periods;
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+
+ // Setup thread if necessary.
+ if ( stream_.mode == OUTPUT && mode == INPUT ) {
+ // We had already set up an output stream.
+ stream_.mode = DUPLEX;
+ // Link the streams if possible.
+ apiInfo->synchronized = false;
+ if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
+ apiInfo->synchronized = true;
+ else {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
+ error( RtAudioError::WARNING );
+ }
+ }
+ else {
+ stream_.mode = mode;
+
+ // Setup callback thread.
+ stream_.callbackInfo.object = (void *) this;
+
+ // Set the thread attributes for joinable and realtime scheduling
+ // priority (optional). The higher priority will only take affect
+ // if the program is run as root or suid. Note, under Linux
+ // processes with CAP_SYS_NICE privilege, a user can change
+ // scheduling policy and priority (thus need not be root). See
+ // POSIX "capabilities".
+ pthread_attr_t attr;
+ pthread_attr_init( &attr );
+ pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
+
+#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+ // We previously attempted to increase the audio callback priority
+ // to SCHED_RR here via the attributes. However, while no errors
+ // were reported in doing so, it did not work. So, now this is
+ // done in the alsaCallbackHandler function.
+ stream_.callbackInfo.doRealtime = true;
+ int priority = options->priority;
+ int min = sched_get_priority_min( SCHED_RR );
+ int max = sched_get_priority_max( SCHED_RR );
+ if ( priority < min ) priority = min;
+ else if ( priority > max ) priority = max;
+ stream_.callbackInfo.priority = priority;
+ }
+#endif
+
+ stream_.callbackInfo.isRunning = true;
+ result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
+ pthread_attr_destroy( &attr );
+ if ( result ) {
+ stream_.callbackInfo.isRunning = false;
+ errorText_ = "RtApiAlsa::error creating callback thread!";
+ goto error;
+ }
+ }
+
+ return SUCCESS;
+
+ error:
+ if ( apiInfo ) {
+ pthread_cond_destroy( &apiInfo->runnable_cv );
+ if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
+ if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
+ delete apiInfo;
+ stream_.apiHandle = 0;
+ }
+
+ if ( phandle) snd_pcm_close( phandle );
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.state = STREAM_CLOSED;
+ return FAILURE;
+}
+
+void RtApiAlsa :: closeStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ stream_.callbackInfo.isRunning = false;
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_STOPPED ) {
+ apiInfo->runnable = true;
+ pthread_cond_signal( &apiInfo->runnable_cv );
+ }
+ MUTEX_UNLOCK( &stream_.mutex );
+ pthread_join( stream_.callbackInfo.thread, NULL );
+
+ if ( stream_.state == STREAM_RUNNING ) {
+ stream_.state = STREAM_STOPPED;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
+ snd_pcm_drop( apiInfo->handles[0] );
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
+ snd_pcm_drop( apiInfo->handles[1] );
+ }
+
+ if ( apiInfo ) {
+ pthread_cond_destroy( &apiInfo->runnable_cv );
+ if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
+ if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
+ delete apiInfo;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
+
+void RtApiAlsa :: startStream()
+{
+ // This method calls snd_pcm_prepare if the device isn't already in that state.
+
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ int result = 0;
+ snd_pcm_state_t state;
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ state = snd_pcm_state( handle[0] );
+ if ( state != SND_PCM_STATE_PREPARED ) {
+ result = snd_pcm_prepare( handle[0] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+ }
+
+ if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+ result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
+ state = snd_pcm_state( handle[1] );
+ if ( state != SND_PCM_STATE_PREPARED ) {
+ result = snd_pcm_prepare( handle[1] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+ }
+
+ stream_.state = STREAM_RUNNING;
+
+ unlock:
+ apiInfo->runnable = true;
+ pthread_cond_signal( &apiInfo->runnable_cv );
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result >= 0 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiAlsa :: stopStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK( &stream_.mutex );
+
+ int result = 0;
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( apiInfo->synchronized )
+ result = snd_pcm_drop( handle[0] );
+ else
+ result = snd_pcm_drain( handle[0] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+ result = snd_pcm_drop( handle[1] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ unlock:
+ apiInfo->runnable = false; // fixes high CPU usage when stopped
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result >= 0 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiAlsa :: abortStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK( &stream_.mutex );
+
+ int result = 0;
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ result = snd_pcm_drop( handle[0] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+ result = snd_pcm_drop( handle[1] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ unlock:
+ apiInfo->runnable = false; // fixes high CPU usage when stopped
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result >= 0 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiAlsa :: callbackEvent()
+{
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_LOCK( &stream_.mutex );
+ while ( !apiInfo->runnable )
+ pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
+
+ if ( stream_.state != STREAM_RUNNING ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+ MUTEX_UNLOCK( &stream_.mutex );
+ }
+
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ int doStopStream = 0;
+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ apiInfo->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ apiInfo->xrun[1] = false;
+ }
+ doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
+
+ if ( doStopStream == 2 ) {
+ abortStream();
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) goto unlock;
+
+ int result;
+ char *buffer;
+ int channels;
+ snd_pcm_t **handle;
+ snd_pcm_sframes_t frames;
+ RtAudioFormat format;
+ handle = (snd_pcm_t **) apiInfo->handles;
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ // Setup parameters.
+ if ( stream_.doConvertBuffer[1] ) {
+ buffer = stream_.deviceBuffer;
+ channels = stream_.nDeviceChannels[1];
+ format = stream_.deviceFormat[1];
+ }
+ else {
+ buffer = stream_.userBuffer[1];
+ channels = stream_.nUserChannels[1];
+ format = stream_.userFormat;
+ }
+
+ // Read samples from device in interleaved/non-interleaved format.
+ if ( stream_.deviceInterleaved[1] )
+ result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
+ else {
+ void *bufs[channels];
+ size_t offset = stream_.bufferSize * formatBytes( format );
+ for ( int i=0; i<channels; i++ )
+ bufs[i] = (void *) (buffer + (i * offset));
+ result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
+ }
+
+ if ( result < (int) stream_.bufferSize ) {
+ // Either an error or overrun occured.
+ if ( result == -EPIPE ) {
+ snd_pcm_state_t state = snd_pcm_state( handle[1] );
+ if ( state == SND_PCM_STATE_XRUN ) {
+ apiInfo->xrun[1] = true;
+ result = snd_pcm_prepare( handle[1] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ }
+ else {
+ errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ }
+ else {
+ errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ error( RtAudioError::WARNING );
+ goto tryOutput;
+ }
+
+ // Do byte swapping if necessary.
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
+
+ // Do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[1] )
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+
+ // Check stream latency
+ result = snd_pcm_delay( handle[1], &frames );
+ if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
+ }
+
+ tryOutput:
+
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ // Setup parameters and do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ channels = stream_.nDeviceChannels[0];
+ format = stream_.deviceFormat[0];
+ }
+ else {
+ buffer = stream_.userBuffer[0];
+ channels = stream_.nUserChannels[0];
+ format = stream_.userFormat;
+ }
+
+ // Do byte swapping if necessary.
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
+
+ // Write samples to device in interleaved/non-interleaved format.
+ if ( stream_.deviceInterleaved[0] )
+ result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
+ else {
+ void *bufs[channels];
+ size_t offset = stream_.bufferSize * formatBytes( format );
+ for ( int i=0; i<channels; i++ )
+ bufs[i] = (void *) (buffer + (i * offset));
+ result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
+ }
+
+ if ( result < (int) stream_.bufferSize ) {
+ // Either an error or underrun occured.
+ if ( result == -EPIPE ) {
+ snd_pcm_state_t state = snd_pcm_state( handle[0] );
+ if ( state == SND_PCM_STATE_XRUN ) {
+ apiInfo->xrun[0] = true;
+ result = snd_pcm_prepare( handle[0] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ }
+ else {
+ errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ }
+ else {
+ errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ error( RtAudioError::WARNING );
+ goto unlock;
+ }
+
+ // Check stream latency
+ result = snd_pcm_delay( handle[0], &frames );
+ if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
+ }
+
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ RtApi::tickStreamTime();
+ if ( doStopStream == 1 ) this->stopStream();
+}
+
+static void *alsaCallbackHandler( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiAlsa *object = (RtApiAlsa *) info->object;
+ bool *isRunning = &info->isRunning;
+
+#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
+ if ( &info->doRealtime ) {
+ pthread_t tID = pthread_self(); // ID of this thread
+ sched_param prio = { info->priority }; // scheduling priority of thread
+ pthread_setschedparam( tID, SCHED_RR, &prio );
+ }
+#endif
+
+ while ( *isRunning == true ) {
+ pthread_testcancel();
+ object->callbackEvent();
+ }
+
+ pthread_exit( NULL );
+}
+
+//******************** End of __LINUX_ALSA__ *********************//
+#endif
+
+#if defined(__LINUX_PULSE__)
+
+// Code written by Peter Meerwald, pmeerw@pmeerw.net
+// and Tristan Matthews.
+
+#include <pulse/error.h>
+#include <pulse/simple.h>
+#include <cstdio>
+
+static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
+ 44100, 48000, 96000, 0};
+
+struct rtaudio_pa_format_mapping_t {
+ RtAudioFormat rtaudio_format;
+ pa_sample_format_t pa_format;
+};
+
+static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
+ {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
+ {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
+ {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
+ {0, PA_SAMPLE_INVALID}};
+
+struct PulseAudioHandle {
+ pa_simple *s_play;
+ pa_simple *s_rec;
+ pthread_t thread;
+ pthread_cond_t runnable_cv;
+ bool runnable;
+ PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
+};
+
+RtApiPulse::~RtApiPulse()
+{
+ if ( stream_.state != STREAM_CLOSED )
+ closeStream();
+}
+
+unsigned int RtApiPulse::getDeviceCount( void )
+{
+ return 1;
+}
+
+RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = true;
+ info.name = "PulseAudio";
+ info.outputChannels = 2;
+ info.inputChannels = 2;
+ info.duplexChannels = 2;
+ info.isDefaultOutput = true;
+ info.isDefaultInput = true;
+
+ for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
+ info.sampleRates.push_back( *sr );
+
+ info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
+
+ return info;
+}
+
+static void *pulseaudio_callback( void * user )
+{
+ CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
+ RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
+ volatile bool *isRunning = &cbi->isRunning;
+
+ while ( *isRunning ) {
+ pthread_testcancel();
+ context->callbackEvent();
+ }
+
+ pthread_exit( NULL );
+}
+
+void RtApiPulse::closeStream( void )
+{
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+ stream_.callbackInfo.isRunning = false;
+ if ( pah ) {
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_STOPPED ) {
+ pah->runnable = true;
+ pthread_cond_signal( &pah->runnable_cv );
+ }
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ pthread_join( pah->thread, 0 );
+ if ( pah->s_play ) {
+ pa_simple_flush( pah->s_play, NULL );
+ pa_simple_free( pah->s_play );
+ }
+ if ( pah->s_rec )
+ pa_simple_free( pah->s_rec );
+
+ pthread_cond_destroy( &pah->runnable_cv );
+ delete pah;
+ stream_.apiHandle = 0;
+ }
+
+ if ( stream_.userBuffer[0] ) {
+ free( stream_.userBuffer[0] );
+ stream_.userBuffer[0] = 0;
+ }
+ if ( stream_.userBuffer[1] ) {
+ free( stream_.userBuffer[1] );
+ stream_.userBuffer[1] = 0;
+ }
+
+ stream_.state = STREAM_CLOSED;
+ stream_.mode = UNINITIALIZED;
+}
+
+void RtApiPulse::callbackEvent( void )
+{
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_LOCK( &stream_.mutex );
+ while ( !pah->runnable )
+ pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
+
+ if ( stream_.state != STREAM_RUNNING ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+ MUTEX_UNLOCK( &stream_.mutex );
+ }
+
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
+ "this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
+ stream_.bufferSize, streamTime, status,
+ stream_.callbackInfo.userData );
+
+ if ( doStopStream == 2 ) {
+ abortStream();
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+ void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
+ void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
+
+ if ( stream_.state != STREAM_RUNNING )
+ goto unlock;
+
+ int pa_error;
+ size_t bytes;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( stream_.doConvertBuffer[OUTPUT] ) {
+ convertBuffer( stream_.deviceBuffer,
+ stream_.userBuffer[OUTPUT],
+ stream_.convertInfo[OUTPUT] );
+ bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
+ formatBytes( stream_.deviceFormat[OUTPUT] );
+ } else
+ bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
+ formatBytes( stream_.userFormat );
+
+ if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
+ errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
+ pa_strerror( pa_error ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ }
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
+ if ( stream_.doConvertBuffer[INPUT] )
+ bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
+ formatBytes( stream_.deviceFormat[INPUT] );
+ else
+ bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
+ formatBytes( stream_.userFormat );
+
+ if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
+ errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
+ pa_strerror( pa_error ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ }
+ if ( stream_.doConvertBuffer[INPUT] ) {
+ convertBuffer( stream_.userBuffer[INPUT],
+ stream_.deviceBuffer,
+ stream_.convertInfo[INPUT] );
+ }
+ }
+
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
+ RtApi::tickStreamTime();
+
+ if ( doStopStream == 1 )
+ stopStream();
+}
+
+void RtApiPulse::startStream( void )
+{
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiPulse::startStream(): the stream is not open!";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiPulse::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ stream_.state = STREAM_RUNNING;
+
+ pah->runnable = true;
+ pthread_cond_signal( &pah->runnable_cv );
+ MUTEX_UNLOCK( &stream_.mutex );
+}
+
+void RtApiPulse::stopStream( void )
+{
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK( &stream_.mutex );
+
+ if ( pah && pah->s_play ) {
+ int pa_error;
+ if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
+ errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
+ pa_strerror( pa_error ) << ".";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ }
+
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
+}
+
+void RtApiPulse::abortStream( void )
+{
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
+
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK( &stream_.mutex );
+
+ if ( pah && pah->s_play ) {
+ int pa_error;
+ if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
+ errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
+ pa_strerror( pa_error ) << ".";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ }
+
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
+}
+
+bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
+ unsigned int channels, unsigned int firstChannel,
+ unsigned int sampleRate, RtAudioFormat format,
+ unsigned int *bufferSize, RtAudio::StreamOptions *options )
+{
+ PulseAudioHandle *pah = 0;
+ unsigned long bufferBytes = 0;
+ pa_sample_spec ss;
+
+ if ( device != 0 ) return false;
+ if ( mode != INPUT && mode != OUTPUT ) return false;
+ if ( channels != 1 && channels != 2 ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
+ return false;
+ }
+ ss.channels = channels;
+
+ if ( firstChannel != 0 ) return false;
+
+ bool sr_found = false;
+ for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
+ if ( sampleRate == *sr ) {
+ sr_found = true;
+ stream_.sampleRate = sampleRate;
+ ss.rate = sampleRate;
+ break;
+ }
+ }
+ if ( !sr_found ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
+ return false;
+ }
+
+ bool sf_found = 0;
+ for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
+ sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
+ if ( format == sf->rtaudio_format ) {
+ sf_found = true;
+ stream_.userFormat = sf->rtaudio_format;
+ stream_.deviceFormat[mode] = stream_.userFormat;
+ ss.format = sf->pa_format;
+ break;
+ }
+ }
+ if ( !sf_found ) { // Use internal data format conversion.
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ ss.format = PA_SAMPLE_FLOAT32LE;
+ }
+
+ // Set other stream parameters.
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
+ stream_.deviceInterleaved[mode] = true;
+ stream_.nBuffers = 1;
+ stream_.doByteSwap[mode] = false;
+ stream_.nUserChannels[mode] = channels;
+ stream_.nDeviceChannels[mode] = channels + firstChannel;
+ stream_.channelOffset[mode] = 0;
+ std::string streamName = "RtAudio";
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate necessary internal buffers.
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+ stream_.bufferSize = *bufferSize;
+
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
+ }
+ }
+
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
+
+ stream_.device[mode] = device;
+
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+
+ if ( !stream_.apiHandle ) {
+ PulseAudioHandle *pah = new PulseAudioHandle;
+ if ( !pah ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
+ goto error;
+ }
+
+ stream_.apiHandle = pah;
+ if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
+ goto error;
+ }
+ }
+ pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
+
+ int error;
+ if ( !options->streamName.empty() ) streamName = options->streamName;
+ switch ( mode ) {
+ case INPUT:
+ pa_buffer_attr buffer_attr;
+ buffer_attr.fragsize = bufferBytes;
+ buffer_attr.maxlength = -1;
+
+ pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
+ if ( !pah->s_rec ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
+ goto error;
+ }
+ break;
+ case OUTPUT:
+ pah->s_play = pa_simple_new( NULL, "RtAudio", PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
+ if ( !pah->s_play ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
+ goto error;
+ }
+ break;
+ default:
+ goto error;
+ }
+
+ if ( stream_.mode == UNINITIALIZED )
+ stream_.mode = mode;
+ else if ( stream_.mode == mode )
+ goto error;
+ else
+ stream_.mode = DUPLEX;
+
+ if ( !stream_.callbackInfo.isRunning ) {
+ stream_.callbackInfo.object = this;
+ stream_.callbackInfo.isRunning = true;
+ if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
+ goto error;
+ }
+ }
+
+ stream_.state = STREAM_STOPPED;
+ return true;
+
+ error:
+ if ( pah && stream_.callbackInfo.isRunning ) {
+ pthread_cond_destroy( &pah->runnable_cv );
+ delete pah;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ return FAILURE;
+}
+
+//******************** End of __LINUX_PULSE__ *********************//
+#endif
+
+#if defined(__LINUX_OSS__)
+
+#include <unistd.h>
+#include <sys/ioctl.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/soundcard.h>
+#include <errno.h>
+#include <math.h>
+
+static void *ossCallbackHandler(void * ptr);
+
+// A structure to hold various information related to the OSS API
+// implementation.
+struct OssHandle {
+ int id[2]; // device ids
+ bool xrun[2];
+ bool triggered;
+ pthread_cond_t runnable;
+
+ OssHandle()
+ :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
+};
+
+RtApiOss :: RtApiOss()
+{
+ // Nothing to do here.
+}
+
+RtApiOss :: ~RtApiOss()
+{
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
+
+unsigned int RtApiOss :: getDeviceCount( void )
+{
+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+ if ( mixerfd == -1 ) {
+ errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
+ error( RtAudioError::WARNING );
+ return 0;
+ }
+
+ oss_sysinfo sysinfo;
+ if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
+ error( RtAudioError::WARNING );
+ return 0;
+ }
+
+ close( mixerfd );
+ return sysinfo.numaudios;
+}
+
+RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+ if ( mixerfd == -1 ) {
+ errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ oss_sysinfo sysinfo;
+ int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
+ if ( result == -1 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ unsigned nDevices = sysinfo.numaudios;
+ if ( nDevices == 0 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ if ( device >= nDevices ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ oss_audioinfo ainfo;
+ ainfo.dev = device;
+ result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
+ close( mixerfd );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Probe channels
+ if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
+ if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
+ if ( ainfo.caps & PCM_CAP_DUPLEX ) {
+ if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+ }
+
+ // Probe data formats ... do for input
+ unsigned long mask = ainfo.iformats;
+ if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
+ info.nativeFormats |= RTAUDIO_SINT16;
+ if ( mask & AFMT_S8 )
+ info.nativeFormats |= RTAUDIO_SINT8;
+ if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
+ info.nativeFormats |= RTAUDIO_SINT32;
+ if ( mask & AFMT_FLOAT )
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+ if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
+ info.nativeFormats |= RTAUDIO_SINT24;
+
+ // Check that we have at least one supported format
+ if ( info.nativeFormats == 0 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Probe the supported sample rates.
+ info.sampleRates.clear();
+ if ( ainfo.nrates ) {
+ for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+ if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
+ info.sampleRates.push_back( SAMPLE_RATES[k] );
+ break;
+ }
+ }
+ }
+ }
+ else {
+ // Check min and max rate values;
+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+ if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] )
+ info.sampleRates.push_back( SAMPLE_RATES[k] );
+ }
+ }
+
+ if ( info.sampleRates.size() == 0 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ }
+ else {
+ info.probed = true;
+ info.name = ainfo.name;
+ }
+
+ return info;
+}
+
+
+bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+ if ( mixerfd == -1 ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
+ return FAILURE;
+ }
+
+ oss_sysinfo sysinfo;
+ int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
+ if ( result == -1 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
+ return FAILURE;
+ }
+
+ unsigned nDevices = sysinfo.numaudios;
+ if ( nDevices == 0 ) {
+ // This should not happen because a check is made before this function is called.
+ close( mixerfd );
+ errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
+ return FAILURE;
+ }
+
+ if ( device >= nDevices ) {
+ // This should not happen because a check is made before this function is called.
+ close( mixerfd );
+ errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
+
+ oss_audioinfo ainfo;
+ ainfo.dev = device;
+ result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
+ close( mixerfd );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Check if device supports input or output
+ if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
+ ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
+ if ( mode == OUTPUT )
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
+ else
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ int flags = 0;
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( mode == OUTPUT )
+ flags |= O_WRONLY;
+ else { // mode == INPUT
+ if (stream_.mode == OUTPUT && stream_.device[0] == device) {
+ // We just set the same device for playback ... close and reopen for duplex (OSS only).
+ close( handle->id[0] );
+ handle->id[0] = 0;
+ if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ // Check that the number previously set channels is the same.
+ if ( stream_.nUserChannels[0] != channels ) {
+ errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ flags |= O_RDWR;
+ }
+ else
+ flags |= O_RDONLY;
+ }
+
+ // Set exclusive access if specified.
+ if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
+
+ // Try to open the device.
+ int fd;
+ fd = open( ainfo.devnode, flags, 0 );
+ if ( fd == -1 ) {
+ if ( errno == EBUSY )
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
+ else
+ errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // For duplex operation, specifically set this mode (this doesn't seem to work).
+ /*
+ if ( flags | O_RDWR ) {
+ result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
+ if ( result == -1) {
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+ */
+
+ // Check the device channel support.
+ stream_.nUserChannels[mode] = channels;
+ if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Set the number of channels.
+ int deviceChannels = channels + firstChannel;
+ result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
+ if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.nDeviceChannels[mode] = deviceChannels;
+
+ // Get the data format mask
+ int mask;
+ result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
+ if ( result == -1 ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Determine how to set the device format.
+ stream_.userFormat = format;
+ int deviceFormat = -1;
+ stream_.doByteSwap[mode] = false;
+ if ( format == RTAUDIO_SINT8 ) {
+ if ( mask & AFMT_S8 ) {
+ deviceFormat = AFMT_S8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ }
+ else if ( format == RTAUDIO_SINT16 ) {
+ if ( mask & AFMT_S16_NE ) {
+ deviceFormat = AFMT_S16_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ else if ( mask & AFMT_S16_OE ) {
+ deviceFormat = AFMT_S16_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ stream_.doByteSwap[mode] = true;
+ }
+ }
+ else if ( format == RTAUDIO_SINT24 ) {
+ if ( mask & AFMT_S24_NE ) {
+ deviceFormat = AFMT_S24_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ }
+ else if ( mask & AFMT_S24_OE ) {
+ deviceFormat = AFMT_S24_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ stream_.doByteSwap[mode] = true;
+ }
+ }
+ else if ( format == RTAUDIO_SINT32 ) {
+ if ( mask & AFMT_S32_NE ) {
+ deviceFormat = AFMT_S32_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ }
+ else if ( mask & AFMT_S32_OE ) {
+ deviceFormat = AFMT_S32_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ stream_.doByteSwap[mode] = true;
+ }
+ }
+
+ if ( deviceFormat == -1 ) {
+ // The user requested format is not natively supported by the device.
+ if ( mask & AFMT_S16_NE ) {
+ deviceFormat = AFMT_S16_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ else if ( mask & AFMT_S32_NE ) {
+ deviceFormat = AFMT_S32_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ }
+ else if ( mask & AFMT_S24_NE ) {
+ deviceFormat = AFMT_S24_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ }
+ else if ( mask & AFMT_S16_OE ) {
+ deviceFormat = AFMT_S16_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ stream_.doByteSwap[mode] = true;
+ }
+ else if ( mask & AFMT_S32_OE ) {
+ deviceFormat = AFMT_S32_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ stream_.doByteSwap[mode] = true;
+ }
+ else if ( mask & AFMT_S24_OE ) {
+ deviceFormat = AFMT_S24_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ stream_.doByteSwap[mode] = true;
+ }
+ else if ( mask & AFMT_S8) {
+ deviceFormat = AFMT_S8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ }
+
+ if ( stream_.deviceFormat[mode] == 0 ) {
+ // This really shouldn't happen ...
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Set the data format.
+ int temp = deviceFormat;
+ result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
+ if ( result == -1 || deviceFormat != temp ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Attempt to set the buffer size. According to OSS, the minimum
+ // number of buffers is two. The supposed minimum buffer size is 16
+ // bytes, so that will be our lower bound. The argument to this
+ // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
+ // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
+ // We'll check the actual value used near the end of the setup
+ // procedure.
+ int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
+ if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
+ int buffers = 0;
+ if ( options ) buffers = options->numberOfBuffers;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
+ if ( buffers < 2 ) buffers = 3;
+ temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
+ result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
+ if ( result == -1 ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.nBuffers = buffers;
+
+ // Save buffer size (in sample frames).
+ *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
+ stream_.bufferSize = *bufferSize;
+
+ // Set the sample rate.
+ int srate = sampleRate;
+ result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
+ if ( result == -1 ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Verify the sample rate setup worked.
+ if ( abs( srate - sampleRate ) > 100 ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.sampleRate = sampleRate;
+
+ if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
+ // We're doing duplex setup here.
+ stream_.deviceFormat[0] = stream_.deviceFormat[1];
+ stream_.nDeviceChannels[0] = deviceChannels;
+ }
+
+ // Set interleaving parameters.
+ stream_.userInterleaved = true;
+ stream_.deviceInterleaved[mode] = true;
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
+ stream_.userInterleaved = false;
+
+ // Set flags for buffer conversion
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate the stream handles if necessary and then save.
+ if ( stream_.apiHandle == 0 ) {
+ try {
+ handle = new OssHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
+ goto error;
+ }
+
+ if ( pthread_cond_init( &handle->runnable, NULL ) ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
+ goto error;
+ }
+
+ stream_.apiHandle = (void *) handle;
+ }
+ else {
+ handle = (OssHandle *) stream_.apiHandle;
+ }
+ handle->id[mode] = fd;
+
+ // Allocate necessary internal buffers.
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
+ }
+ }
+
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
+
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+
+ // Setup thread if necessary.
+ if ( stream_.mode == OUTPUT && mode == INPUT ) {
+ // We had already set up an output stream.
+ stream_.mode = DUPLEX;
+ if ( stream_.device[0] == device ) handle->id[0] = fd;
+ }
+ else {
+ stream_.mode = mode;
+
+ // Setup callback thread.
+ stream_.callbackInfo.object = (void *) this;
+
+ // Set the thread attributes for joinable and realtime scheduling
+ // priority. The higher priority will only take affect if the
+ // program is run as root or suid.
+ pthread_attr_t attr;
+ pthread_attr_init( &attr );
+ pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
+#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+ struct sched_param param;
+ int priority = options->priority;
+ int min = sched_get_priority_min( SCHED_RR );
+ int max = sched_get_priority_max( SCHED_RR );
+ if ( priority < min ) priority = min;
+ else if ( priority > max ) priority = max;
+ param.sched_priority = priority;
+ pthread_attr_setschedparam( &attr, ¶m );
+ pthread_attr_setschedpolicy( &attr, SCHED_RR );
+ }
+ else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#endif
+
+ stream_.callbackInfo.isRunning = true;
+ result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
+ pthread_attr_destroy( &attr );
+ if ( result ) {
+ stream_.callbackInfo.isRunning = false;
+ errorText_ = "RtApiOss::error creating callback thread!";
+ goto error;
+ }
+ }
+
+ return SUCCESS;
+
+ error:
+ if ( handle ) {
+ pthread_cond_destroy( &handle->runnable );
+ if ( handle->id[0] ) close( handle->id[0] );
+ if ( handle->id[1] ) close( handle->id[1] );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ return FAILURE;
+}
+
+void RtApiOss :: closeStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiOss::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ stream_.callbackInfo.isRunning = false;
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_STOPPED )
+ pthread_cond_signal( &handle->runnable );
+ MUTEX_UNLOCK( &stream_.mutex );
+ pthread_join( stream_.callbackInfo.thread, NULL );
+
+ if ( stream_.state == STREAM_RUNNING ) {
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
+ ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+ else
+ ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+ stream_.state = STREAM_STOPPED;
+ }
+
+ if ( handle ) {
+ pthread_cond_destroy( &handle->runnable );
+ if ( handle->id[0] ) close( handle->id[0] );
+ if ( handle->id[1] ) close( handle->id[1] );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
+
+void RtApiOss :: startStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiOss::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ stream_.state = STREAM_RUNNING;
+
+ // No need to do anything else here ... OSS automatically starts
+ // when fed samples.
+
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ pthread_cond_signal( &handle->runnable );
+}
+
+void RtApiOss :: stopStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+
+ int result = 0;
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ // Flush the output with zeros a few times.
+ char *buffer;
+ int samples;
+ RtAudioFormat format;
+
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ samples = stream_.bufferSize * stream_.nDeviceChannels[0];
+ format = stream_.deviceFormat[0];
+ }
+ else {
+ buffer = stream_.userBuffer[0];
+ samples = stream_.bufferSize * stream_.nUserChannels[0];
+ format = stream_.userFormat;
+ }
+
+ memset( buffer, 0, samples * formatBytes(format) );
+ for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
+ result = write( handle->id[0], buffer, samples * formatBytes(format) );
+ if ( result == -1 ) {
+ errorText_ = "RtApiOss::stopStream: audio write error.";
+ error( RtAudioError::WARNING );
+ }
+ }
+
+ result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ handle->triggered = false;
+ }
+
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
+ result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ unlock:
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result != -1 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiOss :: abortStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+
+ int result = 0;
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ handle->triggered = false;
+ }
+
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
+ result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ unlock:
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result != -1 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiOss :: callbackEvent()
+{
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_LOCK( &stream_.mutex );
+ pthread_cond_wait( &handle->runnable, &stream_.mutex );
+ if ( stream_.state != STREAM_RUNNING ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+ MUTEX_UNLOCK( &stream_.mutex );
+ }
+
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ // Invoke user callback to get fresh output data.
+ int doStopStream = 0;
+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
+ }
+ doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
+ if ( doStopStream == 2 ) {
+ this->abortStream();
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) goto unlock;
+
+ int result;
+ char *buffer;
+ int samples;
+ RtAudioFormat format;
+
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ // Setup parameters and do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ samples = stream_.bufferSize * stream_.nDeviceChannels[0];
+ format = stream_.deviceFormat[0];
+ }
+ else {
+ buffer = stream_.userBuffer[0];
+ samples = stream_.bufferSize * stream_.nUserChannels[0];
+ format = stream_.userFormat;
+ }
+
+ // Do byte swapping if necessary.
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer( buffer, samples, format );
+
+ if ( stream_.mode == DUPLEX && handle->triggered == false ) {
+ int trig = 0;
+ ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
+ result = write( handle->id[0], buffer, samples * formatBytes(format) );
+ trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
+ ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
+ handle->triggered = true;
+ }
+ else
+ // Write samples to device.
+ result = write( handle->id[0], buffer, samples * formatBytes(format) );
+
+ if ( result == -1 ) {
+ // We'll assume this is an underrun, though there isn't a
+ // specific means for determining that.
+ handle->xrun[0] = true;
+ errorText_ = "RtApiOss::callbackEvent: audio write error.";
+ error( RtAudioError::WARNING );
+ // Continue on to input section.
+ }
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ // Setup parameters.
+ if ( stream_.doConvertBuffer[1] ) {
+ buffer = stream_.deviceBuffer;
+ samples = stream_.bufferSize * stream_.nDeviceChannels[1];
+ format = stream_.deviceFormat[1];
+ }
+ else {
+ buffer = stream_.userBuffer[1];
+ samples = stream_.bufferSize * stream_.nUserChannels[1];
+ format = stream_.userFormat;
+ }
+
+ // Read samples from device.
+ result = read( handle->id[1], buffer, samples * formatBytes(format) );
+
+ if ( result == -1 ) {
+ // We'll assume this is an overrun, though there isn't a
+ // specific means for determining that.
+ handle->xrun[1] = true;
+ errorText_ = "RtApiOss::callbackEvent: audio read error.";
+ error( RtAudioError::WARNING );
+ goto unlock;
+ }
+
+ // Do byte swapping if necessary.
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( buffer, samples, format );
+
+ // Do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[1] )
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+ }
+
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ RtApi::tickStreamTime();
+ if ( doStopStream == 1 ) this->stopStream();
+}
+
+static void *ossCallbackHandler( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiOss *object = (RtApiOss *) info->object;
+ bool *isRunning = &info->isRunning;
+
+ while ( *isRunning == true ) {
+ pthread_testcancel();
+ object->callbackEvent();
+ }
+
+ pthread_exit( NULL );
+}
+
+//******************** End of __LINUX_OSS__ *********************//
+#endif
+
+
+// *************************************************** //
+//
+// Protected common (OS-independent) RtAudio methods.
+//
+// *************************************************** //
+
+// This method can be modified to control the behavior of error
+// message printing.
+void RtApi :: error( RtAudioError::Type type )
+{
+ errorStream_.str(""); // clear the ostringstream
+
+ RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
+ if ( errorCallback ) {
+ // abortStream() can generate new error messages. Ignore them. Just keep original one.
+
+ if ( firstErrorOccurred_ )
+ return;
+
+ firstErrorOccurred_ = true;
+ const std::string errorMessage = errorText_;
+
+ if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
+ stream_.callbackInfo.isRunning = false; // exit from the thread
+ abortStream();
+ }
+
+ errorCallback( type, errorMessage );
+ firstErrorOccurred_ = false;
+ return;
+ }
+
+ if ( type == RtAudioError::WARNING && showWarnings_ == true )
+ std::cerr << '\n' << errorText_ << "\n\n";
+ else if ( type != RtAudioError::WARNING )
+ throw( RtAudioError( errorText_, type ) );
+}
+
+void RtApi :: verifyStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApi:: a stream is not open!";
+ error( RtAudioError::INVALID_USE );
+ }
+}
+
+void RtApi :: clearStreamInfo()
+{
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+ stream_.sampleRate = 0;
+ stream_.bufferSize = 0;
+ stream_.nBuffers = 0;
+ stream_.userFormat = 0;
+ stream_.userInterleaved = true;
+ stream_.streamTime = 0.0;
+ stream_.apiHandle = 0;
+ stream_.deviceBuffer = 0;
+ stream_.callbackInfo.callback = 0;
+ stream_.callbackInfo.userData = 0;
+ stream_.callbackInfo.isRunning = false;
+ stream_.callbackInfo.errorCallback = 0;
+ for ( int i=0; i<2; i++ ) {
+ stream_.device[i] = 11111;
+ stream_.doConvertBuffer[i] = false;
+ stream_.deviceInterleaved[i] = true;
+ stream_.doByteSwap[i] = false;
+ stream_.nUserChannels[i] = 0;
+ stream_.nDeviceChannels[i] = 0;
+ stream_.channelOffset[i] = 0;
+ stream_.deviceFormat[i] = 0;
+ stream_.latency[i] = 0;
+ stream_.userBuffer[i] = 0;
+ stream_.convertInfo[i].channels = 0;
+ stream_.convertInfo[i].inJump = 0;
+ stream_.convertInfo[i].outJump = 0;
+ stream_.convertInfo[i].inFormat = 0;
+ stream_.convertInfo[i].outFormat = 0;
+ stream_.convertInfo[i].inOffset.clear();
+ stream_.convertInfo[i].outOffset.clear();
+ }
+}
+
+unsigned int RtApi :: formatBytes( RtAudioFormat format )
+{
+ if ( format == RTAUDIO_SINT16 )
+ return 2;
+ else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )
+ return 4;
+ else if ( format == RTAUDIO_FLOAT64 )
+ return 8;
+ else if ( format == RTAUDIO_SINT24 )
+ return 3;
+ else if ( format == RTAUDIO_SINT8 )
+ return 1;
+
+ errorText_ = "RtApi::formatBytes: undefined format.";
+ error( RtAudioError::WARNING );
+
+ return 0;
+}
+
+void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
+{
+ if ( mode == INPUT ) { // convert device to user buffer
+ stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
+ stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
+ stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
+ stream_.convertInfo[mode].outFormat = stream_.userFormat;
+ }
+ else { // convert user to device buffer
+ stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
+ stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
+ stream_.convertInfo[mode].inFormat = stream_.userFormat;
+ stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
+ }
+
+ if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
+ stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
+ else
+ stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
+
+ // Set up the interleave/deinterleave offsets.
+ if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
+ if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
+ ( mode == INPUT && stream_.userInterleaved ) ) {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+ stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
+ stream_.convertInfo[mode].outOffset.push_back( k );
+ stream_.convertInfo[mode].inJump = 1;
+ }
+ }
+ else {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+ stream_.convertInfo[mode].inOffset.push_back( k );
+ stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
+ stream_.convertInfo[mode].outJump = 1;
+ }
+ }
+ }
+ else { // no (de)interleaving
+ if ( stream_.userInterleaved ) {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+ stream_.convertInfo[mode].inOffset.push_back( k );
+ stream_.convertInfo[mode].outOffset.push_back( k );
+ }
+ }
+ else {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+ stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
+ stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
+ stream_.convertInfo[mode].inJump = 1;
+ stream_.convertInfo[mode].outJump = 1;
+ }
+ }
+ }
+
+ // Add channel offset.
+ if ( firstChannel > 0 ) {
+ if ( stream_.deviceInterleaved[mode] ) {
+ if ( mode == OUTPUT ) {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+ stream_.convertInfo[mode].outOffset[k] += firstChannel;
+ }
+ else {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+ stream_.convertInfo[mode].inOffset[k] += firstChannel;
+ }
+ }
+ else {
+ if ( mode == OUTPUT ) {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+ stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
+ }
+ else {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+ stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
+ }
+ }
+ }
+}
+
+void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
+{
+ // This function does format conversion, input/output channel compensation, and
+ // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
+ // the lower three bytes of a 32-bit integer.
+
+ // Clear our device buffer when in/out duplex device channels are different
+ if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
+ ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
+ memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
+
+ int j;
+ if (info.outFormat == RTAUDIO_FLOAT64) {
+ Float64 scale;
+ Float64 *out = (Float64 *)outBuffer;
+
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ scale = 1.0 / 127.5;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ scale = 1.0 / 32767.5;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int24 *in = (Int24 *)inBuffer;
+ scale = 1.0 / 8388607.5;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ scale = 1.0 / 2147483647.5;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ // Channel compensation and/or (de)interleaving only.
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ }
+ else if (info.outFormat == RTAUDIO_FLOAT32) {
+ Float32 scale;
+ Float32 *out = (Float32 *)outBuffer;
+
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ scale = (Float32) ( 1.0 / 127.5 );
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ scale = (Float32) ( 1.0 / 32767.5 );
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int24 *in = (Int24 *)inBuffer;
+ scale = (Float32) ( 1.0 / 8388607.5 );
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ scale = (Float32) ( 1.0 / 2147483647.5 );
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ // Channel compensation and/or (de)interleaving only.
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ }
+ else if (info.outFormat == RTAUDIO_SINT32) {
+ Int32 *out = (Int32 *)outBuffer;
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+ out[info.outOffset[j]] <<= 24;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+ out[info.outOffset[j]] <<= 16;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int24 *in = (Int24 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
+ out[info.outOffset[j]] <<= 8;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ // Channel compensation and/or (de)interleaving only.
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ }
+ else if (info.outFormat == RTAUDIO_SINT24) {
+ Int24 *out = (Int24 *)outBuffer;
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
+ //out[info.outOffset[j]] <<= 16;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);
+ //out[info.outOffset[j]] <<= 8;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ // Channel compensation and/or (de)interleaving only.
+ Int24 *in = (Int24 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);
+ //out[info.outOffset[j]] >>= 8;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ }
+ else if (info.outFormat == RTAUDIO_SINT16) {
+ Int16 *out = (Int16 *)outBuffer;
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
+ out[info.outOffset[j]] <<= 8;
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ // Channel compensation and/or (de)interleaving only.
+ Int16 *in = (Int16 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int24 *in = (Int24 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ }
+ else if (info.outFormat == RTAUDIO_SINT8) {
+ signed char *out = (signed char *)outBuffer;
+ if (info.inFormat == RTAUDIO_SINT8) {
+ // Channel compensation and/or (de)interleaving only.
+ signed char *in = (signed char *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int24 *in = (Int24 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);
+ }
+ in += info.inJump;
+ out += info.outJump;
+ }
+ }
+ }
+}
+
+//static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
+//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
+//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
+
+void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
+{
+ register char val;
+ register char *ptr;
+
+ ptr = buffer;
+ if ( format == RTAUDIO_SINT16 ) {
+ for ( unsigned int i=0; i<samples; i++ ) {
+ // Swap 1st and 2nd bytes.
+ val = *(ptr);
+ *(ptr) = *(ptr+1);
+ *(ptr+1) = val;
+
+ // Increment 2 bytes.
+ ptr += 2;
+ }
+ }
+ else if ( format == RTAUDIO_SINT32 ||
+ format == RTAUDIO_FLOAT32 ) {
+ for ( unsigned int i=0; i<samples; i++ ) {
+ // Swap 1st and 4th bytes.
+ val = *(ptr);
+ *(ptr) = *(ptr+3);
+ *(ptr+3) = val;
+
+ // Swap 2nd and 3rd bytes.
+ ptr += 1;
+ val = *(ptr);
+ *(ptr) = *(ptr+1);
+ *(ptr+1) = val;
+
+ // Increment 3 more bytes.
+ ptr += 3;
+ }
+ }
+ else if ( format == RTAUDIO_SINT24 ) {
+ for ( unsigned int i=0; i<samples; i++ ) {
+ // Swap 1st and 3rd bytes.
+ val = *(ptr);
+ *(ptr) = *(ptr+2);
+ *(ptr+2) = val;
+
+ // Increment 2 more bytes.
+ ptr += 2;
+ }
+ }
+ else if ( format == RTAUDIO_FLOAT64 ) {
+ for ( unsigned int i=0; i<samples; i++ ) {
+ // Swap 1st and 8th bytes
+ val = *(ptr);
+ *(ptr) = *(ptr+7);
+ *(ptr+7) = val;
+
+ // Swap 2nd and 7th bytes
+ ptr += 1;
+ val = *(ptr);
+ *(ptr) = *(ptr+5);
+ *(ptr+5) = val;
+
+ // Swap 3rd and 6th bytes
+ ptr += 1;
+ val = *(ptr);
+ *(ptr) = *(ptr+3);
+ *(ptr+3) = val;
+
+ // Swap 4th and 5th bytes
+ ptr += 1;
+ val = *(ptr);
+ *(ptr) = *(ptr+1);
+ *(ptr+1) = val;
+
+ // Increment 5 more bytes.
+ ptr += 5;
+ }
+ }
+}
+
+ // Indentation settings for Vim and Emacs
+ //
+ // Local Variables:
+ // c-basic-offset: 2
+ // indent-tabs-mode: nil
+ // End:
+ //
+ // vim: et sts=2 sw=2
+
+#endif
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