diff options
Diffstat (limited to 'thirdparty')
| -rw-r--r-- | thirdparty/README.md | 11 | ||||
| -rw-r--r-- | thirdparty/rtaudio/RtAudio.cpp | 10232 | ||||
| -rw-r--r-- | thirdparty/rtaudio/RtAudio.h | 1183 |
3 files changed, 0 insertions, 11426 deletions
diff --git a/thirdparty/README.md b/thirdparty/README.md index 104d0d264d..c70e931c52 100644 --- a/thirdparty/README.md +++ b/thirdparty/README.md @@ -470,17 +470,6 @@ Files extracted from upstream source: - License.txt -## rtaudio - -- Upstream: http://www.music.mcgill.ca/~gary/rtaudio/ -- Version: 4.1.2 -- License: MIT-like - -Files extracted from upstream source: - -- `RtAudio.{cpp,h}` - - ## squish - Upstream: https://sourceforge.net/projects/libsquish diff --git a/thirdparty/rtaudio/RtAudio.cpp b/thirdparty/rtaudio/RtAudio.cpp deleted file mode 100644 index 04159776f7..0000000000 --- a/thirdparty/rtaudio/RtAudio.cpp +++ /dev/null @@ -1,10232 +0,0 @@ -#ifdef RTAUDIO_ENABLED // -GODOT- - -/************************************************************************/ -/*! \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) 2001-2016 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 4.1.2 - -#include "RtAudio.h" -#include <iostream> -#include <cstdlib> -#include <cstring> -#include <climits> -#include <algorithm> - -// 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__) - #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A) - #define MUTEX_DESTROY(A) DeleteCriticalSection(A) - #define MUTEX_LOCK(A) EnterCriticalSection(A) - #define MUTEX_UNLOCK(A) LeaveCriticalSection(A) - - #include "tchar.h" - - static std::string convertCharPointerToStdString(const char *text) - { - return std::string(text); - } - - static std::string convertCharPointerToStdString(const wchar_t *text) - { - int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL); - std::string s( length-1, '\0' ); - WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL); - return s; - } - -#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_ && 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 ) { - unsigned int tmpSr = (unsigned int) rangeList[i].mMinimum; - info.sampleRates.push_back( tmpSr ); - - if ( !info.preferredSampleRate || ( tmpSr <= 48000 && tmpSr > info.preferredSampleRate ) ) - info.preferredSampleRate = tmpSr; - - } 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] ); - - if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) - info.preferredSampleRate = 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 (handle) { - AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, - kAudioObjectPropertyScopeGlobal, - kAudioObjectPropertyElementMaster }; - - property.mSelector = kAudioDeviceProcessorOverload; - property.mScope = kAudioObjectPropertyScopeGlobal; - if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) { - errorText_ = "RtApiCore::closeStream(): error removing property listener!"; - error( RtAudioError::WARNING ); - } - } - 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 (handle) { - AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices, - kAudioObjectPropertyScopeGlobal, - kAudioObjectPropertyElementMaster }; - - property.mSelector = kAudioDeviceProcessorOverload; - property.mScope = kAudioObjectPropertyScopeGlobal; - if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) { - errorText_ = "RtApiCore::closeStream(): error removing property listener!"; - error( RtAudioError::WARNING ); - } - } - 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.preferredSampleRate = jack_get_sample_rate( client ); - info.sampleRates.push_back( info.preferredSampleRate ); - - // 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] ); - - if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) ) - info.preferredSampleRate = 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 ) -{//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// - - bool isDuplexInput = mode == INPUT && stream_.mode == OUTPUT; - - // For ASIO, a duplex stream MUST use the same driver. - if ( isDuplexInput && 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 ( !isDuplexInput ) { - // 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; - } - } - - // keep them before any "goto error", they are used for error cleanup + goto device boundary checks - bool buffersAllocated = false; - AsioHandle *handle = (AsioHandle *) stream_.apiHandle; - unsigned int nChannels; - - - // Check the device channel count. - long inputChannels, outputChannels; - result = ASIOGetChannels( &inputChannels, &outputChannels ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ")."; - errorText_ = errorStream_.str(); - goto error; - } - - if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) || - ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ")."; - errorText_ = errorStream_.str(); - goto error; - } - 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 ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ")."; - errorText_ = errorStream_.str(); - goto error; - } - - // Get the current sample rate - ASIOSampleRate currentRate; - result = ASIOGetSampleRate( ¤tRate ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate."; - errorText_ = errorStream_.str(); - goto error; - } - - // Set the sample rate only if necessary - if ( currentRate != sampleRate ) { - result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate ); - if ( result != ASE_OK ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ")."; - errorText_ = errorStream_.str(); - goto error; - } - } - - // 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 ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format."; - errorText_ = errorStream_.str(); - goto error; - } - - // 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 ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio."; - errorText_ = errorStream_.str(); - goto error; - } - - // 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 ) { - errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size."; - errorText_ = errorStream_.str(); - goto error; - } - - if ( isDuplexInput ) { - // When this is the duplex input (output was opened before), then we have to use the same - // buffersize as the output, because it might use the preferred buffer size, which most - // likely wasn't passed as input to this. The buffer sizes have to be identically anyway, - // So instead of throwing an error, make them equal. The caller uses the reference - // to the "bufferSize" param as usual to set up processing buffers. - - *bufferSize = stream_.bufferSize; - - } else { - if ( *bufferSize == 0 ) *bufferSize = preferSize; - else 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; - } - } - - /* - // we don't use it anymore, see above! - // Just left it here for the case... - if ( isDuplexInput && stream_.bufferSize != *bufferSize ) { - errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!"; - goto error; - } - */ - - 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. - if ( handle == 0 ) { - try { - handle = new AsioHandle; - } - catch ( std::bad_alloc& ) { - errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory."; - goto error; - } - 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. - if ( mode == INPUT && stream_.mode == OUTPUT ) { - ASIODisposeBuffers(); - if ( handle->bufferInfos ) free( handle->bufferInfos ); - } - - // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure. - 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; - } - - // prepare for callbacks - stream_.sampleRate = sampleRate; - stream_.device[mode] = device; - stream_.mode = isDuplexInput ? DUPLEX : mode; - - // store this class instance before registering callbacks, that are going to use it - asioCallbackInfo = &stream_.callbackInfo; - stream_.callbackInfo.object = (void *) this; - - // 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 ) { - // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges - // but only accept the preferred buffer size as parameter for ASIOCreateBuffers. eg. Creatives ASIO driver - // in that case, let's be naïve and try that instead - *bufferSize = preferSize; - stream_.bufferSize = *bufferSize; - 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; - stream_.state = STREAM_STOPPED; - - // 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 ( isDuplexInput && 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; - } - } - } - - // Determine device latencies - long inputLatency, outputLatency; - 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 ( !isDuplexInput ) { - // the cleanup for error in the duplex input, is done by RtApi::openStream - // So we clean up for single channel only - - if ( buffersAllocated ) - ASIODisposeBuffers(); - - drivers.removeCurrentDriver(); - - if ( handle ) { - CloseHandle( handle->condition ); - if ( handle->bufferInfos ) - free( handle->bufferInfos ); - - delete handle; - stream_.apiHandle = 0; - } - - - if ( stream_.userBuffer[mode] ) { - free( stream_.userBuffer[mode] ); - stream_.userBuffer[mode] = 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() { - free( buffer_ ); - } - - // sets the length of the internal ring buffer - void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) { - free( 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 ) roundf( 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::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; - } - - defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal); - - // 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; - } - - info.name =convertCharPointerToStdString(deviceNameProp.pwszVal); - - // 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] ); - } - info.preferredSampleRate = deviceFormat->nSamplesPerSec; - - // 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 (code update submitted by Eli Zehngut). - for ( unsigned int i=0; i<dsDevices.size(); ) { - if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i ); - else i++; - } - - 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] ); - - if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) - info.preferredSampleRate = 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(); - MUTEX_UNLOCK( &stream_.mutex ); - 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(); - MUTEX_UNLOCK( &stream_.mutex ); - 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(); - MUTEX_UNLOCK( &stream_.mutex ); - 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(); - MUTEX_UNLOCK( &stream_.mutex ); - 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(); - MUTEX_UNLOCK( &stream_.mutex ); - 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(); - MUTEX_UNLOCK( &stream_.mutex ); - 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(); - MUTEX_UNLOCK( &stream_.mutex ); - 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(); - MUTEX_UNLOCK( &stream_.mutex ); - 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(); - MUTEX_UNLOCK( &stream_.mutex ); - 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(); - MUTEX_UNLOCK( &stream_.mutex ); - 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(); - MUTEX_UNLOCK( &stream_.mutex ); - 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(); - MUTEX_UNLOCK( &stream_.mutex ); - 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; -} - -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 = convertCharPointerToStdString( 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; - subdevice = -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.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) ) - info.preferredSampleRate = 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 - errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun."; - } - 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.preferredSampleRate = 48000; - 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 && !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, streamName.c_str(), 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] ); - - if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) - info.preferredSampleRate = 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.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) ) - info.preferredSampleRate = 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 ) -{ - char val; - 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 // RTAUDIO_ENABLED -GODOT- diff --git a/thirdparty/rtaudio/RtAudio.h b/thirdparty/rtaudio/RtAudio.h deleted file mode 100644 index aab109d907..0000000000 --- a/thirdparty/rtaudio/RtAudio.h +++ /dev/null @@ -1,1183 +0,0 @@ -// -GODOT- Start - -#ifdef RTAUDIO_ENABLED - -#if defined(OSX_ENABLED) - #define __MACOSX_CORE__ -#elif defined(UNIX_ENABLED) - #define __LINUX_ALSA__ -#elif defined(WINDOWS_ENABLED) - #if defined(UWP_ENABLED) - #define __RTAUDIO_DUMMY__ - #else - #define __WINDOWS_DS__ - #endif -#endif - -// -GODOT- End - -/************************************************************************/ -/*! \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) 2001-2016 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. -*/ -/************************************************************************/ - -/*! - \file RtAudio.h - */ - -#ifndef __RTAUDIO_H -#define __RTAUDIO_H - -#define RTAUDIO_VERSION "4.1.2" - -#include <string> -#include <vector> -#include <exception> -#include <iostream> - -/*! \typedef typedef unsigned long RtAudioFormat; - \brief RtAudio data format type. - - Support for signed integers and floats. Audio data fed to/from an - RtAudio stream is assumed to ALWAYS be in host byte order. The - internal routines will automatically take care of any necessary - byte-swapping between the host format and the soundcard. Thus, - endian-ness is not a concern in the following format definitions. - - - \e RTAUDIO_SINT8: 8-bit signed integer. - - \e RTAUDIO_SINT16: 16-bit signed integer. - - \e RTAUDIO_SINT24: 24-bit signed integer. - - \e RTAUDIO_SINT32: 32-bit signed integer. - - \e RTAUDIO_FLOAT32: Normalized between plus/minus 1.0. - - \e RTAUDIO_FLOAT64: Normalized between plus/minus 1.0. -*/ -typedef unsigned long RtAudioFormat; -static const RtAudioFormat RTAUDIO_SINT8 = 0x1; // 8-bit signed integer. -static const RtAudioFormat RTAUDIO_SINT16 = 0x2; // 16-bit signed integer. -static const RtAudioFormat RTAUDIO_SINT24 = 0x4; // 24-bit signed integer. -static const RtAudioFormat RTAUDIO_SINT32 = 0x8; // 32-bit signed integer. -static const RtAudioFormat RTAUDIO_FLOAT32 = 0x10; // Normalized between plus/minus 1.0. -static const RtAudioFormat RTAUDIO_FLOAT64 = 0x20; // Normalized between plus/minus 1.0. - -/*! \typedef typedef unsigned long RtAudioStreamFlags; - \brief RtAudio stream option flags. - - The following flags can be OR'ed together to allow a client to - make changes to the default stream behavior: - - - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved). - - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency. - - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use. - - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only). - - By default, RtAudio streams pass and receive audio data from the - client in an interleaved format. By passing the - RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio - data will instead be presented in non-interleaved buffers. In - this case, each buffer argument in the RtAudioCallback function - will point to a single array of data, with \c nFrames samples for - each channel concatenated back-to-back. For example, the first - sample of data for the second channel would be located at index \c - nFrames (assuming the \c buffer pointer was recast to the correct - data type for the stream). - - Certain audio APIs offer a number of parameters that influence the - I/O latency of a stream. By default, RtAudio will attempt to set - these parameters internally for robust (glitch-free) performance - (though some APIs, like Windows Direct Sound, make this difficult). - By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream() - function, internal stream settings will be influenced in an attempt - to minimize stream latency, though possibly at the expense of stream - performance. - - If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to - open the input and/or output stream device(s) for exclusive use. - Note that this is not possible with all supported audio APIs. - - If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt - to select realtime scheduling (round-robin) for the callback thread. - - If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to - open the "default" PCM device when using the ALSA API. Note that this - will override any specified input or output device id. -*/ -typedef unsigned int RtAudioStreamFlags; -static const RtAudioStreamFlags RTAUDIO_NONINTERLEAVED = 0x1; // Use non-interleaved buffers (default = interleaved). -static const RtAudioStreamFlags RTAUDIO_MINIMIZE_LATENCY = 0x2; // Attempt to set stream parameters for lowest possible latency. -static const RtAudioStreamFlags RTAUDIO_HOG_DEVICE = 0x4; // Attempt grab device and prevent use by others. -static const RtAudioStreamFlags RTAUDIO_SCHEDULE_REALTIME = 0x8; // Try to select realtime scheduling for callback thread. -static const RtAudioStreamFlags RTAUDIO_ALSA_USE_DEFAULT = 0x10; // Use the "default" PCM device (ALSA only). - -/*! \typedef typedef unsigned long RtAudioStreamStatus; - \brief RtAudio stream status (over- or underflow) flags. - - Notification of a stream over- or underflow is indicated by a - non-zero stream \c status argument in the RtAudioCallback function. - The stream status can be one of the following two options, - depending on whether the stream is open for output and/or input: - - - \e RTAUDIO_INPUT_OVERFLOW: Input data was discarded because of an overflow condition at the driver. - - \e RTAUDIO_OUTPUT_UNDERFLOW: The output buffer ran low, likely producing a break in the output sound. -*/ -typedef unsigned int RtAudioStreamStatus; -static const RtAudioStreamStatus RTAUDIO_INPUT_OVERFLOW = 0x1; // Input data was discarded because of an overflow condition at the driver. -static const RtAudioStreamStatus RTAUDIO_OUTPUT_UNDERFLOW = 0x2; // The output buffer ran low, likely causing a gap in the output sound. - -//! RtAudio callback function prototype. -/*! - All RtAudio clients must create a function of type RtAudioCallback - to read and/or write data from/to the audio stream. When the - underlying audio system is ready for new input or output data, this - function will be invoked. - - \param outputBuffer For output (or duplex) streams, the client - should write \c nFrames of audio sample frames into this - buffer. This argument should be recast to the datatype - specified when the stream was opened. For input-only - streams, this argument will be NULL. - - \param inputBuffer For input (or duplex) streams, this buffer will - hold \c nFrames of input audio sample frames. This - argument should be recast to the datatype specified when the - stream was opened. For output-only streams, this argument - will be NULL. - - \param nFrames The number of sample frames of input or output - data in the buffers. The actual buffer size in bytes is - dependent on the data type and number of channels in use. - - \param streamTime The number of seconds that have elapsed since the - stream was started. - - \param status If non-zero, this argument indicates a data overflow - or underflow condition for the stream. The particular - condition can be determined by comparison with the - RtAudioStreamStatus flags. - - \param userData A pointer to optional data provided by the client - when opening the stream (default = NULL). - - To continue normal stream operation, the RtAudioCallback function - should return a value of zero. To stop the stream and drain the - output buffer, the function should return a value of one. To abort - the stream immediately, the client should return a value of two. - */ -typedef int (*RtAudioCallback)( void *outputBuffer, void *inputBuffer, - unsigned int nFrames, - double streamTime, - RtAudioStreamStatus status, - void *userData ); - -/************************************************************************/ -/*! \class RtAudioError - \brief Exception handling class for RtAudio. - - The RtAudioError class is quite simple but it does allow errors to be - "caught" by RtAudioError::Type. See the RtAudio documentation to know - which methods can throw an RtAudioError. -*/ -/************************************************************************/ - -class RtAudioError : public std::exception -{ - public: - //! Defined RtAudioError types. - enum Type { - WARNING, /*!< A non-critical error. */ - DEBUG_WARNING, /*!< A non-critical error which might be useful for debugging. */ - UNSPECIFIED, /*!< The default, unspecified error type. */ - NO_DEVICES_FOUND, /*!< No devices found on system. */ - INVALID_DEVICE, /*!< An invalid device ID was specified. */ - MEMORY_ERROR, /*!< An error occured during memory allocation. */ - INVALID_PARAMETER, /*!< An invalid parameter was specified to a function. */ - INVALID_USE, /*!< The function was called incorrectly. */ - DRIVER_ERROR, /*!< A system driver error occured. */ - SYSTEM_ERROR, /*!< A system error occured. */ - THREAD_ERROR /*!< A thread error occured. */ - }; - - //! The constructor. - RtAudioError( const std::string& message, Type type = RtAudioError::UNSPECIFIED ) throw() : message_(message), type_(type) {} - - //! The destructor. - virtual ~RtAudioError( void ) throw() {} - - //! Prints thrown error message to stderr. - virtual void printMessage( void ) const throw() { std::cerr << '\n' << message_ << "\n\n"; } - - //! Returns the thrown error message type. - virtual const Type& getType(void) const throw() { return type_; } - - //! Returns the thrown error message string. - virtual const std::string& getMessage(void) const throw() { return message_; } - - //! Returns the thrown error message as a c-style string. - virtual const char* what( void ) const throw() { return message_.c_str(); } - - protected: - std::string message_; - Type type_; -}; - -//! RtAudio error callback function prototype. -/*! - \param type Type of error. - \param errorText Error description. - */ -typedef void (*RtAudioErrorCallback)( RtAudioError::Type type, const std::string &errorText ); - -// **************************************************************** // -// -// RtAudio class declaration. -// -// RtAudio is a "controller" used to select an available audio i/o -// interface. It presents a common API for the user to call but all -// functionality is implemented by the class RtApi and its -// subclasses. RtAudio creates an instance of an RtApi subclass -// based on the user's API choice. If no choice is made, RtAudio -// attempts to make a "logical" API selection. -// -// **************************************************************** // - -class RtApi; - -class RtAudio -{ - public: - - //! Audio API specifier arguments. - enum Api { - UNSPECIFIED, /*!< Search for a working compiled API. */ - LINUX_ALSA, /*!< The Advanced Linux Sound Architecture API. */ - LINUX_PULSE, /*!< The Linux PulseAudio API. */ - LINUX_OSS, /*!< The Linux Open Sound System API. */ - UNIX_JACK, /*!< The Jack Low-Latency Audio Server API. */ - MACOSX_CORE, /*!< Macintosh OS-X Core Audio API. */ - WINDOWS_WASAPI, /*!< The Microsoft WASAPI API. */ - WINDOWS_ASIO, /*!< The Steinberg Audio Stream I/O API. */ - WINDOWS_DS, /*!< The Microsoft Direct Sound API. */ - RTAUDIO_DUMMY /*!< A compilable but non-functional API. */ - }; - - //! The public device information structure for returning queried values. - struct DeviceInfo { - bool probed; /*!< true if the device capabilities were successfully probed. */ - std::string name; /*!< Character string device identifier. */ - unsigned int outputChannels; /*!< Maximum output channels supported by device. */ - unsigned int inputChannels; /*!< Maximum input channels supported by device. */ - unsigned int duplexChannels; /*!< Maximum simultaneous input/output channels supported by device. */ - bool isDefaultOutput; /*!< true if this is the default output device. */ - bool isDefaultInput; /*!< true if this is the default input device. */ - std::vector<unsigned int> sampleRates; /*!< Supported sample rates (queried from list of standard rates). */ - unsigned int preferredSampleRate; /*!< Preferred sample rate, eg. for WASAPI the system sample rate. */ - RtAudioFormat nativeFormats; /*!< Bit mask of supported data formats. */ - - // Default constructor. - DeviceInfo() - :probed(false), outputChannels(0), inputChannels(0), duplexChannels(0), - isDefaultOutput(false), isDefaultInput(false), preferredSampleRate(0), nativeFormats(0) {} - }; - - //! The structure for specifying input or ouput stream parameters. - struct StreamParameters { - unsigned int deviceId; /*!< Device index (0 to getDeviceCount() - 1). */ - unsigned int nChannels; /*!< Number of channels. */ - unsigned int firstChannel; /*!< First channel index on device (default = 0). */ - - // Default constructor. - StreamParameters() - : deviceId(0), nChannels(0), firstChannel(0) {} - }; - - //! The structure for specifying stream options. - /*! - The following flags can be OR'ed together to allow a client to - make changes to the default stream behavior: - - - \e RTAUDIO_NONINTERLEAVED: Use non-interleaved buffers (default = interleaved). - - \e RTAUDIO_MINIMIZE_LATENCY: Attempt to set stream parameters for lowest possible latency. - - \e RTAUDIO_HOG_DEVICE: Attempt grab device for exclusive use. - - \e RTAUDIO_SCHEDULE_REALTIME: Attempt to select realtime scheduling for callback thread. - - \e RTAUDIO_ALSA_USE_DEFAULT: Use the "default" PCM device (ALSA only). - - By default, RtAudio streams pass and receive audio data from the - client in an interleaved format. By passing the - RTAUDIO_NONINTERLEAVED flag to the openStream() function, audio - data will instead be presented in non-interleaved buffers. In - this case, each buffer argument in the RtAudioCallback function - will point to a single array of data, with \c nFrames samples for - each channel concatenated back-to-back. For example, the first - sample of data for the second channel would be located at index \c - nFrames (assuming the \c buffer pointer was recast to the correct - data type for the stream). - - Certain audio APIs offer a number of parameters that influence the - I/O latency of a stream. By default, RtAudio will attempt to set - these parameters internally for robust (glitch-free) performance - (though some APIs, like Windows Direct Sound, make this difficult). - By passing the RTAUDIO_MINIMIZE_LATENCY flag to the openStream() - function, internal stream settings will be influenced in an attempt - to minimize stream latency, though possibly at the expense of stream - performance. - - If the RTAUDIO_HOG_DEVICE flag is set, RtAudio will attempt to - open the input and/or output stream device(s) for exclusive use. - Note that this is not possible with all supported audio APIs. - - If the RTAUDIO_SCHEDULE_REALTIME flag is set, RtAudio will attempt - to select realtime scheduling (round-robin) for the callback thread. - The \c priority parameter will only be used if the RTAUDIO_SCHEDULE_REALTIME - flag is set. It defines the thread's realtime priority. - - If the RTAUDIO_ALSA_USE_DEFAULT flag is set, RtAudio will attempt to - open the "default" PCM device when using the ALSA API. Note that this - will override any specified input or output device id. - - The \c numberOfBuffers parameter can be used to control stream - latency in the Windows DirectSound, Linux OSS, and Linux Alsa APIs - only. A value of two is usually the smallest allowed. Larger - numbers can potentially result in more robust stream performance, - though likely at the cost of stream latency. The value set by the - user is replaced during execution of the RtAudio::openStream() - function by the value actually used by the system. - - The \c streamName parameter can be used to set the client name - when using the Jack API. By default, the client name is set to - RtApiJack. However, if you wish to create multiple instances of - RtAudio with Jack, each instance must have a unique client name. - */ - struct StreamOptions { - RtAudioStreamFlags flags; /*!< A bit-mask of stream flags (RTAUDIO_NONINTERLEAVED, RTAUDIO_MINIMIZE_LATENCY, RTAUDIO_HOG_DEVICE, RTAUDIO_ALSA_USE_DEFAULT). */ - unsigned int numberOfBuffers; /*!< Number of stream buffers. */ - std::string streamName; /*!< A stream name (currently used only in Jack). */ - int priority; /*!< Scheduling priority of callback thread (only used with flag RTAUDIO_SCHEDULE_REALTIME). */ - - // Default constructor. - StreamOptions() - : flags(0), numberOfBuffers(0), priority(0) {} - }; - - //! A static function to determine the current RtAudio version. - static std::string getVersion( void ) throw(); - - //! A static function to determine the available compiled audio APIs. - /*! - The values returned in the std::vector can be compared against - the enumerated list values. Note that there can be more than one - API compiled for certain operating systems. - */ - static void getCompiledApi( std::vector<RtAudio::Api> &apis ) throw(); - - //! The class constructor. - /*! - The constructor performs minor initialization tasks. An exception - can be thrown if no API support is compiled. - - If no API argument is specified and multiple API support has been - compiled, the default order of use is JACK, ALSA, OSS (Linux - systems) and ASIO, DS (Windows systems). - */ - RtAudio( RtAudio::Api api=UNSPECIFIED ); - - //! The destructor. - /*! - If a stream is running or open, it will be stopped and closed - automatically. - */ - ~RtAudio() throw(); - - //! Returns the audio API specifier for the current instance of RtAudio. - RtAudio::Api getCurrentApi( void ) throw(); - - //! A public function that queries for the number of audio devices available. - /*! - This function performs a system query of available devices each time it - is called, thus supporting devices connected \e after instantiation. If - a system error occurs during processing, a warning will be issued. - */ - unsigned int getDeviceCount( void ) throw(); - - //! Return an RtAudio::DeviceInfo structure for a specified device number. - /*! - - Any device integer between 0 and getDeviceCount() - 1 is valid. - If an invalid argument is provided, an RtAudioError (type = INVALID_USE) - will be thrown. If a device is busy or otherwise unavailable, the - structure member "probed" will have a value of "false" and all - other members are undefined. If the specified device is the - current default input or output device, the corresponding - "isDefault" member will have a value of "true". - */ - RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); - - //! A function that returns the index of the default output device. - /*! - If the underlying audio API does not provide a "default - device", or if no devices are available, the return value will be - 0. Note that this is a valid device identifier and it is the - client's responsibility to verify that a device is available - before attempting to open a stream. - */ - unsigned int getDefaultOutputDevice( void ) throw(); - - //! A function that returns the index of the default input device. - /*! - If the underlying audio API does not provide a "default - device", or if no devices are available, the return value will be - 0. Note that this is a valid device identifier and it is the - client's responsibility to verify that a device is available - before attempting to open a stream. - */ - unsigned int getDefaultInputDevice( void ) throw(); - - //! A public function for opening a stream with the specified parameters. - /*! - An RtAudioError (type = SYSTEM_ERROR) is thrown if a stream cannot be - opened with the specified parameters or an error occurs during - processing. An RtAudioError (type = INVALID_USE) is thrown if any - invalid device ID or channel number parameters are specified. - - \param outputParameters Specifies output stream parameters to use - when opening a stream, including a device ID, number of channels, - and starting channel number. For input-only streams, this - argument should be NULL. The device ID is an index value between - 0 and getDeviceCount() - 1. - \param inputParameters Specifies input stream parameters to use - when opening a stream, including a device ID, number of channels, - and starting channel number. For output-only streams, this - argument should be NULL. The device ID is an index value between - 0 and getDeviceCount() - 1. - \param format An RtAudioFormat specifying the desired sample data format. - \param sampleRate The desired sample rate (sample frames per second). - \param *bufferFrames A pointer to a value indicating the desired - internal buffer size in sample frames. The actual value - used by the device is returned via the same pointer. A - value of zero can be specified, in which case the lowest - allowable value is determined. - \param callback A client-defined function that will be invoked - when input data is available and/or output data is needed. - \param userData An optional pointer to data that can be accessed - from within the callback function. - \param options An optional pointer to a structure containing various - global stream options, including a list of OR'ed RtAudioStreamFlags - and a suggested number of stream buffers that can be used to - control stream latency. More buffers typically result in more - robust performance, though at a cost of greater latency. If a - value of zero is specified, a system-specific median value is - chosen. If the RTAUDIO_MINIMIZE_LATENCY flag bit is set, the - lowest allowable value is used. The actual value used is - returned via the structure argument. The parameter is API dependent. - \param errorCallback A client-defined function that will be invoked - when an error has occured. - */ - void openStream( RtAudio::StreamParameters *outputParameters, - RtAudio::StreamParameters *inputParameters, - RtAudioFormat format, unsigned int sampleRate, - unsigned int *bufferFrames, RtAudioCallback callback, - void *userData = NULL, RtAudio::StreamOptions *options = NULL, RtAudioErrorCallback errorCallback = NULL ); - - //! A function that closes a stream and frees any associated stream memory. - /*! - If a stream is not open, this function issues a warning and - returns (no exception is thrown). - */ - void closeStream( void ) throw(); - - //! A function that starts a stream. - /*! - An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs - during processing. An RtAudioError (type = INVALID_USE) is thrown if a - stream is not open. A warning is issued if the stream is already - running. - */ - void startStream( void ); - - //! Stop a stream, allowing any samples remaining in the output queue to be played. - /*! - An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs - during processing. An RtAudioError (type = INVALID_USE) is thrown if a - stream is not open. A warning is issued if the stream is already - stopped. - */ - void stopStream( void ); - - //! Stop a stream, discarding any samples remaining in the input/output queue. - /*! - An RtAudioError (type = SYSTEM_ERROR) is thrown if an error occurs - during processing. An RtAudioError (type = INVALID_USE) is thrown if a - stream is not open. A warning is issued if the stream is already - stopped. - */ - void abortStream( void ); - - //! Returns true if a stream is open and false if not. - bool isStreamOpen( void ) const throw(); - - //! Returns true if the stream is running and false if it is stopped or not open. - bool isStreamRunning( void ) const throw(); - - //! Returns the number of elapsed seconds since the stream was started. - /*! - If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown. - */ - double getStreamTime( void ); - - //! Set the stream time to a time in seconds greater than or equal to 0.0. - /*! - If a stream is not open, an RtAudioError (type = INVALID_USE) will be thrown. - */ - void setStreamTime( double time ); - - //! Returns the internal stream latency in sample frames. - /*! - The stream latency refers to delay in audio input and/or output - caused by internal buffering by the audio system and/or hardware. - For duplex streams, the returned value will represent the sum of - the input and output latencies. If a stream is not open, an - RtAudioError (type = INVALID_USE) will be thrown. If the API does not - report latency, the return value will be zero. - */ - long getStreamLatency( void ); - - //! Returns actual sample rate in use by the stream. - /*! - On some systems, the sample rate used may be slightly different - than that specified in the stream parameters. If a stream is not - open, an RtAudioError (type = INVALID_USE) will be thrown. - */ - unsigned int getStreamSampleRate( void ); - - //! Specify whether warning messages should be printed to stderr. - void showWarnings( bool value = true ) throw(); - - protected: - - void openRtApi( RtAudio::Api api ); - RtApi *rtapi_; -}; - -// Operating system dependent thread functionality. -#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__) - - #ifndef NOMINMAX - #define NOMINMAX - #endif - #include <windows.h> - #include <process.h> - - typedef uintptr_t ThreadHandle; - typedef CRITICAL_SECTION StreamMutex; - -#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__) - // Using pthread library for various flavors of unix. - #include <pthread.h> - - typedef pthread_t ThreadHandle; - typedef pthread_mutex_t StreamMutex; - -#else // Setup for "dummy" behavior - - #define __RTAUDIO_DUMMY__ - typedef int ThreadHandle; - typedef int StreamMutex; - -#endif - -// This global structure type is used to pass callback information -// between the private RtAudio stream structure and global callback -// handling functions. -struct CallbackInfo { - void *object; // Used as a "this" pointer. - ThreadHandle thread; - void *callback; - void *userData; - void *errorCallback; - void *apiInfo; // void pointer for API specific callback information - bool isRunning; - bool doRealtime; - int priority; - - // Default constructor. - CallbackInfo() - :object(0), callback(0), userData(0), errorCallback(0), apiInfo(0), isRunning(false), doRealtime(false) {} -}; - -// **************************************************************** // -// -// RtApi class declaration. -// -// Subclasses of RtApi contain all API- and OS-specific code necessary -// to fully implement the RtAudio API. -// -// Note that RtApi is an abstract base class and cannot be -// explicitly instantiated. The class RtAudio will create an -// instance of an RtApi subclass (RtApiOss, RtApiAlsa, -// RtApiJack, RtApiCore, RtApiDs, or RtApiAsio). -// -// **************************************************************** // - -#pragma pack(push, 1) -class S24 { - - protected: - unsigned char c3[3]; - - public: - S24() {} - - S24& operator = ( const int& i ) { - c3[0] = (i & 0x000000ff); - c3[1] = (i & 0x0000ff00) >> 8; - c3[2] = (i & 0x00ff0000) >> 16; - return *this; - } - - S24( const S24& v ) { *this = v; } - S24( const double& d ) { *this = (int) d; } - S24( const float& f ) { *this = (int) f; } - S24( const signed short& s ) { *this = (int) s; } - S24( const char& c ) { *this = (int) c; } - - int asInt() { - int i = c3[0] | (c3[1] << 8) | (c3[2] << 16); - if (i & 0x800000) i |= ~0xffffff; - return i; - } -}; -#pragma pack(pop) - -#if defined( HAVE_GETTIMEOFDAY ) - #include <sys/time.h> -#endif - -#include <sstream> - -class RtApi -{ -public: - - RtApi(); - virtual ~RtApi(); - virtual RtAudio::Api getCurrentApi( void ) = 0; - virtual unsigned int getDeviceCount( void ) = 0; - virtual RtAudio::DeviceInfo getDeviceInfo( unsigned int device ) = 0; - virtual unsigned int getDefaultInputDevice( void ); - virtual unsigned int getDefaultOutputDevice( void ); - void openStream( RtAudio::StreamParameters *outputParameters, - RtAudio::StreamParameters *inputParameters, - RtAudioFormat format, unsigned int sampleRate, - unsigned int *bufferFrames, RtAudioCallback callback, - void *userData, RtAudio::StreamOptions *options, - RtAudioErrorCallback errorCallback ); - virtual void closeStream( void ); - virtual void startStream( void ) = 0; - virtual void stopStream( void ) = 0; - virtual void abortStream( void ) = 0; - long getStreamLatency( void ); - unsigned int getStreamSampleRate( void ); - virtual double getStreamTime( void ); - virtual void setStreamTime( double time ); - bool isStreamOpen( void ) const { return stream_.state != STREAM_CLOSED; } - bool isStreamRunning( void ) const { return stream_.state == STREAM_RUNNING; } - void showWarnings( bool value ) { showWarnings_ = value; } - - -protected: - - static const unsigned int MAX_SAMPLE_RATES; - static const unsigned int SAMPLE_RATES[]; - - enum { FAILURE, SUCCESS }; - - enum StreamState { - STREAM_STOPPED, - STREAM_STOPPING, - STREAM_RUNNING, - STREAM_CLOSED = -50 - }; - - enum StreamMode { - OUTPUT, - INPUT, - DUPLEX, - UNINITIALIZED = -75 - }; - - // A protected structure used for buffer conversion. - struct ConvertInfo { - int channels; - int inJump, outJump; - RtAudioFormat inFormat, outFormat; - std::vector<int> inOffset; - std::vector<int> outOffset; - }; - - // A protected structure for audio streams. - struct RtApiStream { - unsigned int device[2]; // Playback and record, respectively. - void *apiHandle; // void pointer for API specific stream handle information - StreamMode mode; // OUTPUT, INPUT, or DUPLEX. - StreamState state; // STOPPED, RUNNING, or CLOSED - char *userBuffer[2]; // Playback and record, respectively. - char *deviceBuffer; - bool doConvertBuffer[2]; // Playback and record, respectively. - bool userInterleaved; - bool deviceInterleaved[2]; // Playback and record, respectively. - bool doByteSwap[2]; // Playback and record, respectively. - unsigned int sampleRate; - unsigned int bufferSize; - unsigned int nBuffers; - unsigned int nUserChannels[2]; // Playback and record, respectively. - unsigned int nDeviceChannels[2]; // Playback and record channels, respectively. - unsigned int channelOffset[2]; // Playback and record, respectively. - unsigned long latency[2]; // Playback and record, respectively. - RtAudioFormat userFormat; - RtAudioFormat deviceFormat[2]; // Playback and record, respectively. - StreamMutex mutex; - CallbackInfo callbackInfo; - ConvertInfo convertInfo[2]; - double streamTime; // Number of elapsed seconds since the stream started. - -#if defined(HAVE_GETTIMEOFDAY) - struct timeval lastTickTimestamp; -#endif - - RtApiStream() - :apiHandle(0), deviceBuffer(0) { device[0] = 11111; device[1] = 11111; } - }; - - typedef S24 Int24; - typedef signed short Int16; - typedef signed int Int32; - typedef float Float32; - typedef double Float64; - - std::ostringstream errorStream_; - std::string errorText_; - bool showWarnings_; - RtApiStream stream_; - bool firstErrorOccurred_; - - /*! - Protected, api-specific method that attempts to open a device - with the given parameters. This function MUST be implemented by - all subclasses. If an error is encountered during the probe, a - "warning" message is reported and FAILURE is returned. A - successful probe is indicated by a return value of SUCCESS. - */ - virtual bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ); - - //! A protected function used to increment the stream time. - void tickStreamTime( void ); - - //! Protected common method to clear an RtApiStream structure. - void clearStreamInfo(); - - /*! - Protected common method that throws an RtAudioError (type = - INVALID_USE) if a stream is not open. - */ - void verifyStream( void ); - - //! Protected common error method to allow global control over error handling. - void error( RtAudioError::Type type ); - - /*! - Protected method used to perform format, channel number, and/or interleaving - conversions between the user and device buffers. - */ - void convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info ); - - //! Protected common method used to perform byte-swapping on buffers. - void byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format ); - - //! Protected common method that returns the number of bytes for a given format. - unsigned int formatBytes( RtAudioFormat format ); - - //! Protected common method that sets up the parameters for buffer conversion. - void setConvertInfo( StreamMode mode, unsigned int firstChannel ); -}; - -// **************************************************************** // -// -// Inline RtAudio definitions. -// -// **************************************************************** // - -inline RtAudio::Api RtAudio :: getCurrentApi( void ) throw() { return rtapi_->getCurrentApi(); } -inline unsigned int RtAudio :: getDeviceCount( void ) throw() { return rtapi_->getDeviceCount(); } -inline RtAudio::DeviceInfo RtAudio :: getDeviceInfo( unsigned int device ) { return rtapi_->getDeviceInfo( device ); } -inline unsigned int RtAudio :: getDefaultInputDevice( void ) throw() { return rtapi_->getDefaultInputDevice(); } -inline unsigned int RtAudio :: getDefaultOutputDevice( void ) throw() { return rtapi_->getDefaultOutputDevice(); } -inline void RtAudio :: closeStream( void ) throw() { return rtapi_->closeStream(); } -inline void RtAudio :: startStream( void ) { return rtapi_->startStream(); } -inline void RtAudio :: stopStream( void ) { return rtapi_->stopStream(); } -inline void RtAudio :: abortStream( void ) { return rtapi_->abortStream(); } -inline bool RtAudio :: isStreamOpen( void ) const throw() { return rtapi_->isStreamOpen(); } -inline bool RtAudio :: isStreamRunning( void ) const throw() { return rtapi_->isStreamRunning(); } -inline long RtAudio :: getStreamLatency( void ) { return rtapi_->getStreamLatency(); } -inline unsigned int RtAudio :: getStreamSampleRate( void ) { return rtapi_->getStreamSampleRate(); } -inline double RtAudio :: getStreamTime( void ) { return rtapi_->getStreamTime(); } -inline void RtAudio :: setStreamTime( double time ) { return rtapi_->setStreamTime( time ); } -inline void RtAudio :: showWarnings( bool value ) throw() { rtapi_->showWarnings( value ); } - -// RtApi Subclass prototypes. - -#if defined(__MACOSX_CORE__) - -#include <CoreAudio/AudioHardware.h> - -class RtApiCore: public RtApi -{ -public: - - RtApiCore(); - ~RtApiCore(); - RtAudio::Api getCurrentApi( void ) { return RtAudio::MACOSX_CORE; } - unsigned int getDeviceCount( void ); - RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); - unsigned int getDefaultOutputDevice( void ); - unsigned int getDefaultInputDevice( void ); - void closeStream( void ); - void startStream( void ); - void stopStream( void ); - void abortStream( void ); - long getStreamLatency( void ); - - // This function is intended for internal use only. It must be - // public because it is called by the internal callback handler, - // which is not a member of RtAudio. External use of this function - // will most likely produce highly undesireable results! - bool callbackEvent( AudioDeviceID deviceId, - const AudioBufferList *inBufferList, - const AudioBufferList *outBufferList ); - - private: - - bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ); - static const char* getErrorCode( OSStatus code ); -}; - -#endif - -#if defined(__UNIX_JACK__) - -class RtApiJack: public RtApi -{ -public: - - RtApiJack(); - ~RtApiJack(); - RtAudio::Api getCurrentApi( void ) { return RtAudio::UNIX_JACK; } - unsigned int getDeviceCount( void ); - RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); - void closeStream( void ); - void startStream( void ); - void stopStream( void ); - void abortStream( void ); - long getStreamLatency( void ); - - // This function is intended for internal use only. It must be - // public because it is called by the internal callback handler, - // which is not a member of RtAudio. External use of this function - // will most likely produce highly undesireable results! - bool callbackEvent( unsigned long nframes ); - - private: - - bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ); -}; - -#endif - -#if defined(__WINDOWS_ASIO__) - -class RtApiAsio: public RtApi -{ -public: - - RtApiAsio(); - ~RtApiAsio(); - RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_ASIO; } - unsigned int getDeviceCount( void ); - RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); - void closeStream( void ); - void startStream( void ); - void stopStream( void ); - void abortStream( void ); - long getStreamLatency( void ); - - // This function is intended for internal use only. It must be - // public because it is called by the internal callback handler, - // which is not a member of RtAudio. External use of this function - // will most likely produce highly undesireable results! - bool callbackEvent( long bufferIndex ); - - private: - - std::vector<RtAudio::DeviceInfo> devices_; - void saveDeviceInfo( void ); - bool coInitialized_; - bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ); -}; - -#endif - -#if defined(__WINDOWS_DS__) - -class RtApiDs: public RtApi -{ -public: - - RtApiDs(); - ~RtApiDs(); - RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_DS; } - unsigned int getDeviceCount( void ); - unsigned int getDefaultOutputDevice( void ); - unsigned int getDefaultInputDevice( void ); - RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); - void closeStream( void ); - void startStream( void ); - void stopStream( void ); - void abortStream( void ); - long getStreamLatency( void ); - - // This function is intended for internal use only. It must be - // public because it is called by the internal callback handler, - // which is not a member of RtAudio. External use of this function - // will most likely produce highly undesireable results! - void callbackEvent( void ); - - private: - - bool coInitialized_; - bool buffersRolling; - long duplexPrerollBytes; - std::vector<struct DsDevice> dsDevices; - bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ); -}; - -#endif - -#if defined(__WINDOWS_WASAPI__) - -struct IMMDeviceEnumerator; - -class RtApiWasapi : public RtApi -{ -public: - RtApiWasapi(); - ~RtApiWasapi(); - - RtAudio::Api getCurrentApi( void ) { return RtAudio::WINDOWS_WASAPI; } - unsigned int getDeviceCount( void ); - RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); - unsigned int getDefaultOutputDevice( void ); - unsigned int getDefaultInputDevice( void ); - void closeStream( void ); - void startStream( void ); - void stopStream( void ); - void abortStream( void ); - -private: - bool coInitialized_; - IMMDeviceEnumerator* deviceEnumerator_; - - bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int* bufferSize, - RtAudio::StreamOptions* options ); - - static DWORD WINAPI runWasapiThread( void* wasapiPtr ); - static DWORD WINAPI stopWasapiThread( void* wasapiPtr ); - static DWORD WINAPI abortWasapiThread( void* wasapiPtr ); - void wasapiThread(); -}; - -#endif - -#if defined(__LINUX_ALSA__) - -class RtApiAlsa: public RtApi -{ -public: - - RtApiAlsa(); - ~RtApiAlsa(); - RtAudio::Api getCurrentApi() { return RtAudio::LINUX_ALSA; } - unsigned int getDeviceCount( void ); - RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); - void closeStream( void ); - void startStream( void ); - void stopStream( void ); - void abortStream( void ); - - // This function is intended for internal use only. It must be - // public because it is called by the internal callback handler, - // which is not a member of RtAudio. External use of this function - // will most likely produce highly undesireable results! - void callbackEvent( void ); - - private: - - std::vector<RtAudio::DeviceInfo> devices_; - void saveDeviceInfo( void ); - bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ); -}; - -#endif - -#if defined(__LINUX_PULSE__) - -class RtApiPulse: public RtApi -{ -public: - ~RtApiPulse(); - RtAudio::Api getCurrentApi() { return RtAudio::LINUX_PULSE; } - unsigned int getDeviceCount( void ); - RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); - void closeStream( void ); - void startStream( void ); - void stopStream( void ); - void abortStream( void ); - - // This function is intended for internal use only. It must be - // public because it is called by the internal callback handler, - // which is not a member of RtAudio. External use of this function - // will most likely produce highly undesireable results! - void callbackEvent( void ); - - private: - - std::vector<RtAudio::DeviceInfo> devices_; - void saveDeviceInfo( void ); - bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ); -}; - -#endif - -#if defined(__LINUX_OSS__) - -class RtApiOss: public RtApi -{ -public: - - RtApiOss(); - ~RtApiOss(); - RtAudio::Api getCurrentApi() { return RtAudio::LINUX_OSS; } - unsigned int getDeviceCount( void ); - RtAudio::DeviceInfo getDeviceInfo( unsigned int device ); - void closeStream( void ); - void startStream( void ); - void stopStream( void ); - void abortStream( void ); - - // This function is intended for internal use only. It must be - // public because it is called by the internal callback handler, - // which is not a member of RtAudio. External use of this function - // will most likely produce highly undesireable results! - void callbackEvent( void ); - - private: - - bool probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels, - unsigned int firstChannel, unsigned int sampleRate, - RtAudioFormat format, unsigned int *bufferSize, - RtAudio::StreamOptions *options ); -}; - -#endif - -#if defined(__RTAUDIO_DUMMY__) - -class RtApiDummy: public RtApi -{ -public: - - RtApiDummy() { errorText_ = "RtApiDummy: This class provides no functionality."; error( RtAudioError::WARNING ); } - RtAudio::Api getCurrentApi( void ) { return RtAudio::RTAUDIO_DUMMY; } - unsigned int getDeviceCount( void ) { return 0; } - RtAudio::DeviceInfo getDeviceInfo( unsigned int /*device*/ ) { RtAudio::DeviceInfo info; return info; } - void closeStream( void ) {} - void startStream( void ) {} - void stopStream( void ) {} - void abortStream( void ) {} - - private: - - bool probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/, - unsigned int /*firstChannel*/, unsigned int /*sampleRate*/, - RtAudioFormat /*format*/, unsigned int * /*bufferSize*/, - RtAudio::StreamOptions * /*options*/ ) { return false; } -}; - -#endif - -#endif - -// Indentation settings for Vim and Emacs -// -// Local Variables: -// c-basic-offset: 2 -// indent-tabs-mode: nil -// End: -// -// vim: et sts=2 sw=2 - -#endif // RTAUDIO_ENABLED -GODOT- |