-/************************************************************************/\r
-/*! \class RtAudio\r
- \brief Realtime audio i/o C++ classes.\r
-\r
- RtAudio provides a common API (Application Programming Interface)\r
- for realtime audio input/output across Linux (native ALSA, Jack,\r
- and OSS), Macintosh OS X (CoreAudio and Jack), and Windows\r
- (DirectSound and ASIO) operating systems.\r
-\r
- RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/\r
-\r
- RtAudio: realtime audio i/o C++ classes\r
- Copyright (c) 2001-2013 Gary P. Scavone\r
-\r
- Permission is hereby granted, free of charge, to any person\r
- obtaining a copy of this software and associated documentation files\r
- (the "Software"), to deal in the Software without restriction,\r
- including without limitation the rights to use, copy, modify, merge,\r
- publish, distribute, sublicense, and/or sell copies of the Software,\r
- and to permit persons to whom the Software is furnished to do so,\r
- subject to the following conditions:\r
-\r
- The above copyright notice and this permission notice shall be\r
- included in all copies or substantial portions of the Software.\r
-\r
- Any person wishing to distribute modifications to the Software is\r
- asked to send the modifications to the original developer so that\r
- they can be incorporated into the canonical version. This is,\r
- however, not a binding provision of this license.\r
-\r
- THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,\r
- EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF\r
- MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.\r
- IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR\r
- ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF\r
- CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION\r
- WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.\r
-*/\r
-/************************************************************************/\r
-\r
-// RtAudio: Version 4.0.12\r
-\r
-#include "RtAudio.h"\r
-#include <iostream>\r
-#include <cstdlib>\r
-#include <cstring>\r
-#include <climits>\r
-\r
-// Static variable definitions.\r
-const unsigned int RtApi::MAX_SAMPLE_RATES = 14;\r
-const unsigned int RtApi::SAMPLE_RATES[] = {\r
- 4000, 5512, 8000, 9600, 11025, 16000, 22050,\r
- 32000, 44100, 48000, 88200, 96000, 176400, 192000\r
-};\r
-\r
-#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)\r
- #define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)\r
- #define MUTEX_DESTROY(A) DeleteCriticalSection(A)\r
- #define MUTEX_LOCK(A) EnterCriticalSection(A)\r
- #define MUTEX_UNLOCK(A) LeaveCriticalSection(A)\r
-#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)\r
- // pthread API\r
- #define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)\r
- #define MUTEX_DESTROY(A) pthread_mutex_destroy(A)\r
- #define MUTEX_LOCK(A) pthread_mutex_lock(A)\r
- #define MUTEX_UNLOCK(A) pthread_mutex_unlock(A)\r
-#else\r
- #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions\r
- #define MUTEX_DESTROY(A) abs(*A) // dummy definitions\r
-#endif\r
-\r
-// *************************************************** //\r
-//\r
-// RtAudio definitions.\r
-//\r
-// *************************************************** //\r
-\r
-std::string RtAudio :: getVersion( void ) throw()\r
-{\r
- return RTAUDIO_VERSION;\r
-}\r
-\r
-void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()\r
-{\r
- apis.clear();\r
-\r
- // The order here will control the order of RtAudio's API search in\r
- // the constructor.\r
-#if defined(__UNIX_JACK__)\r
- apis.push_back( UNIX_JACK );\r
-#endif\r
-#if defined(__LINUX_ALSA__)\r
- apis.push_back( LINUX_ALSA );\r
-#endif\r
-#if defined(__LINUX_PULSE__)\r
- apis.push_back( LINUX_PULSE );\r
-#endif\r
-#if defined(__LINUX_OSS__)\r
- apis.push_back( LINUX_OSS );\r
-#endif\r
-#if defined(__WINDOWS_ASIO__)\r
- apis.push_back( WINDOWS_ASIO );\r
-#endif\r
-#if defined(__WINDOWS_WASAPI__)\r
- apis.push_back( WINDOWS_WASAPI );\r
-#endif\r
-#if defined(__WINDOWS_DS__)\r
- apis.push_back( WINDOWS_DS );\r
-#endif\r
-#if defined(__MACOSX_CORE__)\r
- apis.push_back( MACOSX_CORE );\r
-#endif\r
-#if defined(__RTAUDIO_DUMMY__)\r
- apis.push_back( RTAUDIO_DUMMY );\r
-#endif\r
-}\r
-\r
-void RtAudio :: openRtApi( RtAudio::Api api )\r
-{\r
- if ( rtapi_ )\r
- delete rtapi_;\r
- rtapi_ = 0;\r
-\r
-#if defined(__UNIX_JACK__)\r
- if ( api == UNIX_JACK )\r
- rtapi_ = new RtApiJack();\r
-#endif\r
-#if defined(__LINUX_ALSA__)\r
- if ( api == LINUX_ALSA )\r
- rtapi_ = new RtApiAlsa();\r
-#endif\r
-#if defined(__LINUX_PULSE__)\r
- if ( api == LINUX_PULSE )\r
- rtapi_ = new RtApiPulse();\r
-#endif\r
-#if defined(__LINUX_OSS__)\r
- if ( api == LINUX_OSS )\r
- rtapi_ = new RtApiOss();\r
-#endif\r
-#if defined(__WINDOWS_ASIO__)\r
- if ( api == WINDOWS_ASIO )\r
- rtapi_ = new RtApiAsio();\r
-#endif\r
-#if defined(__WINDOWS_WASAPI__)\r
- if ( api == WINDOWS_WASAPI )\r
- rtapi_ = new RtApiWasapi();\r
-#endif\r
-#if defined(__WINDOWS_DS__)\r
- if ( api == WINDOWS_DS )\r
- rtapi_ = new RtApiDs();\r
-#endif\r
-#if defined(__MACOSX_CORE__)\r
- if ( api == MACOSX_CORE )\r
- rtapi_ = new RtApiCore();\r
-#endif\r
-#if defined(__RTAUDIO_DUMMY__)\r
- if ( api == RTAUDIO_DUMMY )\r
- rtapi_ = new RtApiDummy();\r
-#endif\r
-}\r
-\r
-RtAudio :: RtAudio( RtAudio::Api api )\r
-{\r
- rtapi_ = 0;\r
-\r
- if ( api != UNSPECIFIED ) {\r
- // Attempt to open the specified API.\r
- openRtApi( api );\r
- if ( rtapi_ ) return;\r
-\r
- // No compiled support for specified API value. Issue a debug\r
- // warning and continue as if no API was specified.\r
- std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;\r
- }\r
-\r
- // Iterate through the compiled APIs and return as soon as we find\r
- // one with at least one device or we reach the end of the list.\r
- std::vector< RtAudio::Api > apis;\r
- getCompiledApi( apis );\r
- for ( unsigned int i=0; i<apis.size(); i++ ) {\r
- openRtApi( apis[i] );\r
- if ( rtapi_->getDeviceCount() ) break;\r
- }\r
-\r
- if ( rtapi_ ) return;\r
-\r
- // It should not be possible to get here because the preprocessor\r
- // definition __RTAUDIO_DUMMY__ is automatically defined if no\r
- // API-specific definitions are passed to the compiler. But just in\r
- // case something weird happens, we'll thow an error.\r
- std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";\r
- throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );\r
-}\r
-\r
-RtAudio :: ~RtAudio() throw()\r
-{\r
- if ( rtapi_ )\r
- delete rtapi_;\r
-}\r
-\r
-void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,\r
- RtAudio::StreamParameters *inputParameters,\r
- RtAudioFormat format, unsigned int sampleRate,\r
- unsigned int *bufferFrames,\r
- RtAudioCallback callback, void *userData,\r
- RtAudio::StreamOptions *options,\r
- RtAudioErrorCallback errorCallback )\r
-{\r
- return rtapi_->openStream( outputParameters, inputParameters, format,\r
- sampleRate, bufferFrames, callback,\r
- userData, options, errorCallback );\r
-}\r
-\r
-// *************************************************** //\r
-//\r
-// Public RtApi definitions (see end of file for\r
-// private or protected utility functions).\r
-//\r
-// *************************************************** //\r
-\r
-RtApi :: RtApi()\r
-{\r
- stream_.state = STREAM_CLOSED;\r
- stream_.mode = UNINITIALIZED;\r
- stream_.apiHandle = 0;\r
- stream_.userBuffer[0] = 0;\r
- stream_.userBuffer[1] = 0;\r
- MUTEX_INITIALIZE( &stream_.mutex );\r
- showWarnings_ = true;\r
- firstErrorOccurred_ = false;\r
-}\r
-\r
-RtApi :: ~RtApi()\r
-{\r
- MUTEX_DESTROY( &stream_.mutex );\r
-}\r
-\r
-void RtApi :: openStream( RtAudio::StreamParameters *oParams,\r
- RtAudio::StreamParameters *iParams,\r
- RtAudioFormat format, unsigned int sampleRate,\r
- unsigned int *bufferFrames,\r
- RtAudioCallback callback, void *userData,\r
- RtAudio::StreamOptions *options,\r
- RtAudioErrorCallback errorCallback )\r
-{\r
- if ( stream_.state != STREAM_CLOSED ) {\r
- errorText_ = "RtApi::openStream: a stream is already open!";\r
- error( RtAudioError::INVALID_USE );\r
- return;\r
- }\r
-\r
- if ( oParams && oParams->nChannels < 1 ) {\r
- errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";\r
- error( RtAudioError::INVALID_USE );\r
- return;\r
- }\r
-\r
- if ( iParams && iParams->nChannels < 1 ) {\r
- errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";\r
- error( RtAudioError::INVALID_USE );\r
- return;\r
- }\r
-\r
- if ( oParams == NULL && iParams == NULL ) {\r
- errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";\r
- error( RtAudioError::INVALID_USE );\r
- return;\r
- }\r
-\r
- if ( formatBytes(format) == 0 ) {\r
- errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";\r
- error( RtAudioError::INVALID_USE );\r
- return;\r
- }\r
-\r
- unsigned int nDevices = getDeviceCount();\r
- unsigned int oChannels = 0;\r
- if ( oParams ) {\r
- oChannels = oParams->nChannels;\r
- if ( oParams->deviceId >= nDevices ) {\r
- errorText_ = "RtApi::openStream: output device parameter value is invalid.";\r
- error( RtAudioError::INVALID_USE );\r
- return;\r
- }\r
- }\r
-\r
- unsigned int iChannels = 0;\r
- if ( iParams ) {\r
- iChannels = iParams->nChannels;\r
- if ( iParams->deviceId >= nDevices ) {\r
- errorText_ = "RtApi::openStream: input device parameter value is invalid.";\r
- error( RtAudioError::INVALID_USE );\r
- return;\r
- }\r
- }\r
-\r
- clearStreamInfo();\r
- bool result;\r
-\r
- if ( oChannels > 0 ) {\r
-\r
- result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,\r
- sampleRate, format, bufferFrames, options );\r
- if ( result == false ) {\r
- error( RtAudioError::SYSTEM_ERROR );\r
- return;\r
- }\r
- }\r
-\r
- if ( iChannels > 0 ) {\r
-\r
- result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,\r
- sampleRate, format, bufferFrames, options );\r
- if ( result == false ) {\r
- if ( oChannels > 0 ) closeStream();\r
- error( RtAudioError::SYSTEM_ERROR );\r
- return;\r
- }\r
- }\r
-\r
- stream_.callbackInfo.callback = (void *) callback;\r
- stream_.callbackInfo.userData = userData;\r
- stream_.callbackInfo.errorCallback = (void *) errorCallback;\r
-\r
- if ( options ) options->numberOfBuffers = stream_.nBuffers;\r
- stream_.state = STREAM_STOPPED;\r
-}\r
-\r
-unsigned int RtApi :: getDefaultInputDevice( void )\r
-{\r
- // Should be implemented in subclasses if possible.\r
- return 0;\r
-}\r
-\r
-unsigned int RtApi :: getDefaultOutputDevice( void )\r
-{\r
- // Should be implemented in subclasses if possible.\r
- return 0;\r
-}\r
-\r
-void RtApi :: closeStream( void )\r
-{\r
- // MUST be implemented in subclasses!\r
- return;\r
-}\r
-\r
-bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,\r
- unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,\r
- RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,\r
- RtAudio::StreamOptions * /*options*/ )\r
-{\r
- // MUST be implemented in subclasses!\r
- return FAILURE;\r
-}\r
-\r
-void RtApi :: tickStreamTime( void )\r
-{\r
- // Subclasses that do not provide their own implementation of\r
- // getStreamTime should call this function once per buffer I/O to\r
- // provide basic stream time support.\r
-\r
- stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );\r
-\r
-#if defined( HAVE_GETTIMEOFDAY )\r
- gettimeofday( &stream_.lastTickTimestamp, NULL );\r
-#endif\r
-}\r
-\r
-long RtApi :: getStreamLatency( void )\r
-{\r
- verifyStream();\r
-\r
- long totalLatency = 0;\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )\r
- totalLatency = stream_.latency[0];\r
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX )\r
- totalLatency += stream_.latency[1];\r
-\r
- return totalLatency;\r
-}\r
-\r
-double RtApi :: getStreamTime( void )\r
-{\r
- verifyStream();\r
-\r
-#if defined( HAVE_GETTIMEOFDAY )\r
- // Return a very accurate estimate of the stream time by\r
- // adding in the elapsed time since the last tick.\r
- struct timeval then;\r
- struct timeval now;\r
-\r
- if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )\r
- return stream_.streamTime;\r
-\r
- gettimeofday( &now, NULL );\r
- then = stream_.lastTickTimestamp;\r
- return stream_.streamTime +\r
- ((now.tv_sec + 0.000001 * now.tv_usec) -\r
- (then.tv_sec + 0.000001 * then.tv_usec)); \r
-#else\r
- return stream_.streamTime;\r
-#endif\r
-}\r
-\r
-unsigned int RtApi :: getStreamSampleRate( void )\r
-{\r
- verifyStream();\r
-\r
- return stream_.sampleRate;\r
-}\r
-\r
-\r
-// *************************************************** //\r
-//\r
-// OS/API-specific methods.\r
-//\r
-// *************************************************** //\r
-\r
-#if defined(__MACOSX_CORE__)\r
-\r
-// The OS X CoreAudio API is designed to use a separate callback\r
-// procedure for each of its audio devices. A single RtAudio duplex\r
-// stream using two different devices is supported here, though it\r
-// cannot be guaranteed to always behave correctly because we cannot\r
-// synchronize these two callbacks.\r
-//\r
-// A property listener is installed for over/underrun information.\r
-// However, no functionality is currently provided to allow property\r
-// listeners to trigger user handlers because it is unclear what could\r
-// be done if a critical stream parameter (buffer size, sample rate,\r
-// device disconnect) notification arrived. The listeners entail\r
-// quite a bit of extra code and most likely, a user program wouldn't\r
-// be prepared for the result anyway. However, we do provide a flag\r
-// to the client callback function to inform of an over/underrun.\r
-\r
-// A structure to hold various information related to the CoreAudio API\r
-// implementation.\r
-struct CoreHandle {\r
- AudioDeviceID id[2]; // device ids\r
-#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )\r
- AudioDeviceIOProcID procId[2];\r
-#endif\r
- UInt32 iStream[2]; // device stream index (or first if using multiple)\r
- UInt32 nStreams[2]; // number of streams to use\r
- bool xrun[2];\r
- char *deviceBuffer;\r
- pthread_cond_t condition;\r
- int drainCounter; // Tracks callback counts when draining\r
- bool internalDrain; // Indicates if stop is initiated from callback or not.\r
-\r
- CoreHandle()\r
- :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }\r
-};\r
-\r
-RtApiCore:: RtApiCore()\r
-{\r
-#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )\r
- // This is a largely undocumented but absolutely necessary\r
- // requirement starting with OS-X 10.6. If not called, queries and\r
- // updates to various audio device properties are not handled\r
- // correctly.\r
- CFRunLoopRef theRunLoop = NULL;\r
- AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,\r
- kAudioObjectPropertyScopeGlobal,\r
- kAudioObjectPropertyElementMaster };\r
- OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);\r
- if ( result != noErr ) {\r
- errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";\r
- error( RtAudioError::WARNING );\r
- }\r
-#endif\r
-}\r
-\r
-RtApiCore :: ~RtApiCore()\r
-{\r
- // The subclass destructor gets called before the base class\r
- // destructor, so close an existing stream before deallocating\r
- // apiDeviceId memory.\r
- if ( stream_.state != STREAM_CLOSED ) closeStream();\r
-}\r
-\r
-unsigned int RtApiCore :: getDeviceCount( void )\r
-{\r
- // Find out how many audio devices there are, if any.\r
- UInt32 dataSize;\r
- AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };\r
- OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );\r
- if ( result != noErr ) {\r
- errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";\r
- error( RtAudioError::WARNING );\r
- return 0;\r
- }\r
-\r
- return dataSize / sizeof( AudioDeviceID );\r
-}\r
-\r
-unsigned int RtApiCore :: getDefaultInputDevice( void )\r
-{\r
- unsigned int nDevices = getDeviceCount();\r
- if ( nDevices <= 1 ) return 0;\r
-\r
- AudioDeviceID id;\r
- UInt32 dataSize = sizeof( AudioDeviceID );\r
- AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };\r
- OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );\r
- if ( result != noErr ) {\r
- errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";\r
- error( RtAudioError::WARNING );\r
- return 0;\r
- }\r
-\r
- dataSize *= nDevices;\r
- AudioDeviceID deviceList[ nDevices ];\r
- property.mSelector = kAudioHardwarePropertyDevices;\r
- result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );\r
- if ( result != noErr ) {\r
- errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";\r
- error( RtAudioError::WARNING );\r
- return 0;\r
- }\r
-\r
- for ( unsigned int i=0; i<nDevices; i++ )\r
- if ( id == deviceList[i] ) return i;\r
-\r
- errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";\r
- error( RtAudioError::WARNING );\r
- return 0;\r
-}\r
-\r
-unsigned int RtApiCore :: getDefaultOutputDevice( void )\r
-{\r
- unsigned int nDevices = getDeviceCount();\r
- if ( nDevices <= 1 ) return 0;\r
-\r
- AudioDeviceID id;\r
- UInt32 dataSize = sizeof( AudioDeviceID );\r
- AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };\r
- OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );\r
- if ( result != noErr ) {\r
- errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";\r
- error( RtAudioError::WARNING );\r
- return 0;\r
- }\r
-\r
- dataSize = sizeof( AudioDeviceID ) * nDevices;\r
- AudioDeviceID deviceList[ nDevices ];\r
- property.mSelector = kAudioHardwarePropertyDevices;\r
- result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );\r
- if ( result != noErr ) {\r
- errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";\r
- error( RtAudioError::WARNING );\r
- return 0;\r
- }\r
-\r
- for ( unsigned int i=0; i<nDevices; i++ )\r
- if ( id == deviceList[i] ) return i;\r
-\r
- errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";\r
- error( RtAudioError::WARNING );\r
- return 0;\r
-}\r
-\r
-RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )\r
-{\r
- RtAudio::DeviceInfo info;\r
- info.probed = false;\r
-\r
- // Get device ID\r
- unsigned int nDevices = getDeviceCount();\r
- if ( nDevices == 0 ) {\r
- errorText_ = "RtApiCore::getDeviceInfo: no devices found!";\r
- error( RtAudioError::INVALID_USE );\r
- return info;\r
- }\r
-\r
- if ( device >= nDevices ) {\r
- errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";\r
- error( RtAudioError::INVALID_USE );\r
- return info;\r
- }\r
-\r
- AudioDeviceID deviceList[ nDevices ];\r
- UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;\r
- AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,\r
- kAudioObjectPropertyScopeGlobal,\r
- kAudioObjectPropertyElementMaster };\r
- OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,\r
- 0, NULL, &dataSize, (void *) &deviceList );\r
- if ( result != noErr ) {\r
- errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- AudioDeviceID id = deviceList[ device ];\r
-\r
- // Get the device name.\r
- info.name.erase();\r
- CFStringRef cfname;\r
- dataSize = sizeof( CFStringRef );\r
- property.mSelector = kAudioObjectPropertyManufacturer;\r
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );\r
- if ( result != noErr ) {\r
- errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );\r
- int length = CFStringGetLength(cfname);\r
- char *mname = (char *)malloc(length * 3 + 1);\r
-#if defined( UNICODE ) || defined( _UNICODE )\r
- CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8);\r
-#else\r
- CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());\r
-#endif\r
- info.name.append( (const char *)mname, strlen(mname) );\r
- info.name.append( ": " );\r
- CFRelease( cfname );\r
- free(mname);\r
-\r
- property.mSelector = kAudioObjectPropertyName;\r
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );\r
- if ( result != noErr ) {\r
- errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );\r
- length = CFStringGetLength(cfname);\r
- char *name = (char *)malloc(length * 3 + 1);\r
-#if defined( UNICODE ) || defined( _UNICODE )\r
- CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8);\r
-#else\r
- CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());\r
-#endif\r
- info.name.append( (const char *)name, strlen(name) );\r
- CFRelease( cfname );\r
- free(name);\r
-\r
- // Get the output stream "configuration".\r
- AudioBufferList *bufferList = nil;\r
- property.mSelector = kAudioDevicePropertyStreamConfiguration;\r
- property.mScope = kAudioDevicePropertyScopeOutput;\r
- // property.mElement = kAudioObjectPropertyElementWildcard;\r
- dataSize = 0;\r
- result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );\r
- if ( result != noErr || dataSize == 0 ) {\r
- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- // Allocate the AudioBufferList.\r
- bufferList = (AudioBufferList *) malloc( dataSize );\r
- if ( bufferList == NULL ) {\r
- errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );\r
- if ( result != noErr || dataSize == 0 ) {\r
- free( bufferList );\r
- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- // Get output channel information.\r
- unsigned int i, nStreams = bufferList->mNumberBuffers;\r
- for ( i=0; i<nStreams; i++ )\r
- info.outputChannels += bufferList->mBuffers[i].mNumberChannels;\r
- free( bufferList );\r
-\r
- // Get the input stream "configuration".\r
- property.mScope = kAudioDevicePropertyScopeInput;\r
- result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );\r
- if ( result != noErr || dataSize == 0 ) {\r
- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- // Allocate the AudioBufferList.\r
- bufferList = (AudioBufferList *) malloc( dataSize );\r
- if ( bufferList == NULL ) {\r
- errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );\r
- if (result != noErr || dataSize == 0) {\r
- free( bufferList );\r
- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- // Get input channel information.\r
- nStreams = bufferList->mNumberBuffers;\r
- for ( i=0; i<nStreams; i++ )\r
- info.inputChannels += bufferList->mBuffers[i].mNumberChannels;\r
- free( bufferList );\r
-\r
- // If device opens for both playback and capture, we determine the channels.\r
- if ( info.outputChannels > 0 && info.inputChannels > 0 )\r
- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;\r
-\r
- // Probe the device sample rates.\r
- bool isInput = false;\r
- if ( info.outputChannels == 0 ) isInput = true;\r
-\r
- // Determine the supported sample rates.\r
- property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;\r
- if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;\r
- result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );\r
- if ( result != kAudioHardwareNoError || dataSize == 0 ) {\r
- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- UInt32 nRanges = dataSize / sizeof( AudioValueRange );\r
- AudioValueRange rangeList[ nRanges ];\r
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );\r
- if ( result != kAudioHardwareNoError ) {\r
- errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- // The sample rate reporting mechanism is a bit of a mystery. It\r
- // seems that it can either return individual rates or a range of\r
- // rates. I assume that if the min / max range values are the same,\r
- // then that represents a single supported rate and if the min / max\r
- // range values are different, the device supports an arbitrary\r
- // range of values (though there might be multiple ranges, so we'll\r
- // use the most conservative range).\r
- Float64 minimumRate = 1.0, maximumRate = 10000000000.0;\r
- bool haveValueRange = false;\r
- info.sampleRates.clear();\r
- for ( UInt32 i=0; i<nRanges; i++ ) {\r
- if ( rangeList[i].mMinimum == rangeList[i].mMaximum )\r
- info.sampleRates.push_back( (unsigned int) rangeList[i].mMinimum );\r
- else {\r
- haveValueRange = true;\r
- if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;\r
- if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;\r
- }\r
- }\r
-\r
- if ( haveValueRange ) {\r
- for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {\r
- if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate )\r
- info.sampleRates.push_back( SAMPLE_RATES[k] );\r
- }\r
- }\r
-\r
- // Sort and remove any redundant values\r
- std::sort( info.sampleRates.begin(), info.sampleRates.end() );\r
- info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );\r
-\r
- if ( info.sampleRates.size() == 0 ) {\r
- errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- // CoreAudio always uses 32-bit floating point data for PCM streams.\r
- // Thus, any other "physical" formats supported by the device are of\r
- // no interest to the client.\r
- info.nativeFormats = RTAUDIO_FLOAT32;\r
-\r
- if ( info.outputChannels > 0 )\r
- if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;\r
- if ( info.inputChannels > 0 )\r
- if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;\r
-\r
- info.probed = true;\r
- return info;\r
-}\r
-\r
-static OSStatus callbackHandler( AudioDeviceID inDevice,\r
- const AudioTimeStamp* /*inNow*/,\r
- const AudioBufferList* inInputData,\r
- const AudioTimeStamp* /*inInputTime*/,\r
- AudioBufferList* outOutputData,\r
- const AudioTimeStamp* /*inOutputTime*/,\r
- void* infoPointer )\r
-{\r
- CallbackInfo *info = (CallbackInfo *) infoPointer;\r
-\r
- RtApiCore *object = (RtApiCore *) info->object;\r
- if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )\r
- return kAudioHardwareUnspecifiedError;\r
- else\r
- return kAudioHardwareNoError;\r
-}\r
-\r
-static OSStatus xrunListener( AudioObjectID /*inDevice*/,\r
- UInt32 nAddresses,\r
- const AudioObjectPropertyAddress properties[],\r
- void* handlePointer )\r
-{\r
- CoreHandle *handle = (CoreHandle *) handlePointer;\r
- for ( UInt32 i=0; i<nAddresses; i++ ) {\r
- if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) {\r
- if ( properties[i].mScope == kAudioDevicePropertyScopeInput )\r
- handle->xrun[1] = true;\r
- else\r
- handle->xrun[0] = true;\r
- }\r
- }\r
-\r
- return kAudioHardwareNoError;\r
-}\r
-\r
-static OSStatus rateListener( AudioObjectID inDevice,\r
- UInt32 /*nAddresses*/,\r
- const AudioObjectPropertyAddress /*properties*/[],\r
- void* ratePointer )\r
-{\r
- Float64 *rate = (Float64 *) ratePointer;\r
- UInt32 dataSize = sizeof( Float64 );\r
- AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,\r
- kAudioObjectPropertyScopeGlobal,\r
- kAudioObjectPropertyElementMaster };\r
- AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate );\r
- return kAudioHardwareNoError;\r
-}\r
-\r
-bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,\r
- unsigned int firstChannel, unsigned int sampleRate,\r
- RtAudioFormat format, unsigned int *bufferSize,\r
- RtAudio::StreamOptions *options )\r
-{\r
- // Get device ID\r
- unsigned int nDevices = getDeviceCount();\r
- if ( nDevices == 0 ) {\r
- // This should not happen because a check is made before this function is called.\r
- errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";\r
- return FAILURE;\r
- }\r
-\r
- if ( device >= nDevices ) {\r
- // This should not happen because a check is made before this function is called.\r
- errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!";\r
- return FAILURE;\r
- }\r
-\r
- AudioDeviceID deviceList[ nDevices ];\r
- UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;\r
- AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,\r
- kAudioObjectPropertyScopeGlobal,\r
- kAudioObjectPropertyElementMaster };\r
- OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,\r
- 0, NULL, &dataSize, (void *) &deviceList );\r
- if ( result != noErr ) {\r
- errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";\r
- return FAILURE;\r
- }\r
-\r
- AudioDeviceID id = deviceList[ device ];\r
-\r
- // Setup for stream mode.\r
- bool isInput = false;\r
- if ( mode == INPUT ) {\r
- isInput = true;\r
- property.mScope = kAudioDevicePropertyScopeInput;\r
- }\r
- else\r
- property.mScope = kAudioDevicePropertyScopeOutput;\r
-\r
- // Get the stream "configuration".\r
- AudioBufferList *bufferList = nil;\r
- dataSize = 0;\r
- property.mSelector = kAudioDevicePropertyStreamConfiguration;\r
- result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );\r
- if ( result != noErr || dataSize == 0 ) {\r
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Allocate the AudioBufferList.\r
- bufferList = (AudioBufferList *) malloc( dataSize );\r
- if ( bufferList == NULL ) {\r
- errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";\r
- return FAILURE;\r
- }\r
-\r
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );\r
- if (result != noErr || dataSize == 0) {\r
- free( bufferList );\r
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Search for one or more streams that contain the desired number of\r
- // channels. CoreAudio devices can have an arbitrary number of\r
- // streams and each stream can have an arbitrary number of channels.\r
- // For each stream, a single buffer of interleaved samples is\r
- // provided. RtAudio prefers the use of one stream of interleaved\r
- // data or multiple consecutive single-channel streams. However, we\r
- // now support multiple consecutive multi-channel streams of\r
- // interleaved data as well.\r
- UInt32 iStream, offsetCounter = firstChannel;\r
- UInt32 nStreams = bufferList->mNumberBuffers;\r
- bool monoMode = false;\r
- bool foundStream = false;\r
-\r
- // First check that the device supports the requested number of\r
- // channels.\r
- UInt32 deviceChannels = 0;\r
- for ( iStream=0; iStream<nStreams; iStream++ )\r
- deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;\r
-\r
- if ( deviceChannels < ( channels + firstChannel ) ) {\r
- free( bufferList );\r
- errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Look for a single stream meeting our needs.\r
- UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;\r
- for ( iStream=0; iStream<nStreams; iStream++ ) {\r
- streamChannels = bufferList->mBuffers[iStream].mNumberChannels;\r
- if ( streamChannels >= channels + offsetCounter ) {\r
- firstStream = iStream;\r
- channelOffset = offsetCounter;\r
- foundStream = true;\r
- break;\r
- }\r
- if ( streamChannels > offsetCounter ) break;\r
- offsetCounter -= streamChannels;\r
- }\r
-\r
- // If we didn't find a single stream above, then we should be able\r
- // to meet the channel specification with multiple streams.\r
- if ( foundStream == false ) {\r
- monoMode = true;\r
- offsetCounter = firstChannel;\r
- for ( iStream=0; iStream<nStreams; iStream++ ) {\r
- streamChannels = bufferList->mBuffers[iStream].mNumberChannels;\r
- if ( streamChannels > offsetCounter ) break;\r
- offsetCounter -= streamChannels;\r
- }\r
-\r
- firstStream = iStream;\r
- channelOffset = offsetCounter;\r
- Int32 channelCounter = channels + offsetCounter - streamChannels;\r
-\r
- if ( streamChannels > 1 ) monoMode = false;\r
- while ( channelCounter > 0 ) {\r
- streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;\r
- if ( streamChannels > 1 ) monoMode = false;\r
- channelCounter -= streamChannels;\r
- streamCount++;\r
- }\r
- }\r
-\r
- free( bufferList );\r
-\r
- // Determine the buffer size.\r
- AudioValueRange bufferRange;\r
- dataSize = sizeof( AudioValueRange );\r
- property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;\r
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange );\r
-\r
- if ( result != noErr ) {\r
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;\r
- else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;\r
- if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;\r
-\r
- // Set the buffer size. For multiple streams, I'm assuming we only\r
- // need to make this setting for the master channel.\r
- UInt32 theSize = (UInt32) *bufferSize;\r
- dataSize = sizeof( UInt32 );\r
- property.mSelector = kAudioDevicePropertyBufferFrameSize;\r
- result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize );\r
-\r
- if ( result != noErr ) {\r
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // If attempting to setup a duplex stream, the bufferSize parameter\r
- // MUST be the same in both directions!\r
- *bufferSize = theSize;\r
- if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {\r
- errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- stream_.bufferSize = *bufferSize;\r
- stream_.nBuffers = 1;\r
-\r
- // Try to set "hog" mode ... it's not clear to me this is working.\r
- if ( options && options->flags & RTAUDIO_HOG_DEVICE ) {\r
- pid_t hog_pid;\r
- dataSize = sizeof( hog_pid );\r
- property.mSelector = kAudioDevicePropertyHogMode;\r
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid );\r
- if ( result != noErr ) {\r
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- if ( hog_pid != getpid() ) {\r
- hog_pid = getpid();\r
- result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid );\r
- if ( result != noErr ) {\r
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- }\r
- }\r
-\r
- // Check and if necessary, change the sample rate for the device.\r
- Float64 nominalRate;\r
- dataSize = sizeof( Float64 );\r
- property.mSelector = kAudioDevicePropertyNominalSampleRate;\r
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );\r
- if ( result != noErr ) {\r
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Only change the sample rate if off by more than 1 Hz.\r
- if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {\r
-\r
- // Set a property listener for the sample rate change\r
- Float64 reportedRate = 0.0;\r
- AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };\r
- result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate );\r
- if ( result != noErr ) {\r
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- nominalRate = (Float64) sampleRate;\r
- result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );\r
- if ( result != noErr ) {\r
- AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );\r
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Now wait until the reported nominal rate is what we just set.\r
- UInt32 microCounter = 0;\r
- while ( reportedRate != nominalRate ) {\r
- microCounter += 5000;\r
- if ( microCounter > 5000000 ) break;\r
- usleep( 5000 );\r
- }\r
-\r
- // Remove the property listener.\r
- AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );\r
-\r
- if ( microCounter > 5000000 ) {\r
- errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- }\r
-\r
- // Now set the stream format for all streams. Also, check the\r
- // physical format of the device and change that if necessary.\r
- AudioStreamBasicDescription description;\r
- dataSize = sizeof( AudioStreamBasicDescription );\r
- property.mSelector = kAudioStreamPropertyVirtualFormat;\r
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );\r
- if ( result != noErr ) {\r
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Set the sample rate and data format id. However, only make the\r
- // change if the sample rate is not within 1.0 of the desired\r
- // rate and the format is not linear pcm.\r
- bool updateFormat = false;\r
- if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) {\r
- description.mSampleRate = (Float64) sampleRate;\r
- updateFormat = true;\r
- }\r
-\r
- if ( description.mFormatID != kAudioFormatLinearPCM ) {\r
- description.mFormatID = kAudioFormatLinearPCM;\r
- updateFormat = true;\r
- }\r
-\r
- if ( updateFormat ) {\r
- result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description );\r
- if ( result != noErr ) {\r
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- }\r
-\r
- // Now check the physical format.\r
- property.mSelector = kAudioStreamPropertyPhysicalFormat;\r
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );\r
- if ( result != noErr ) {\r
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- //std::cout << "Current physical stream format:" << std::endl;\r
- //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl;\r
- //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;\r
- //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl;\r
- //std::cout << " sample rate = " << description.mSampleRate << std::endl;\r
-\r
- if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) {\r
- description.mFormatID = kAudioFormatLinearPCM;\r
- //description.mSampleRate = (Float64) sampleRate;\r
- AudioStreamBasicDescription testDescription = description;\r
- UInt32 formatFlags;\r
-\r
- // We'll try higher bit rates first and then work our way down.\r
- std::vector< std::pair<UInt32, UInt32> > physicalFormats;\r
- formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;\r
- physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );\r
- formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;\r
- physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );\r
- physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) ); // 24-bit packed\r
- formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh );\r
- physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low\r
- formatFlags |= kAudioFormatFlagIsAlignedHigh;\r
- physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high\r
- formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;\r
- physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) );\r
- physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) );\r
-\r
- bool setPhysicalFormat = false;\r
- for( unsigned int i=0; i<physicalFormats.size(); i++ ) {\r
- testDescription = description;\r
- testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first;\r
- testDescription.mFormatFlags = physicalFormats[i].second;\r
- if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) )\r
- testDescription.mBytesPerFrame = 4 * testDescription.mChannelsPerFrame;\r
- else\r
- testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;\r
- testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;\r
- result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription );\r
- if ( result == noErr ) {\r
- setPhysicalFormat = true;\r
- //std::cout << "Updated physical stream format:" << std::endl;\r
- //std::cout << " mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;\r
- //std::cout << " aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;\r
- //std::cout << " bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;\r
- //std::cout << " sample rate = " << testDescription.mSampleRate << std::endl;\r
- break;\r
- }\r
- }\r
-\r
- if ( !setPhysicalFormat ) {\r
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- } // done setting virtual/physical formats.\r
-\r
- // Get the stream / device latency.\r
- UInt32 latency;\r
- dataSize = sizeof( UInt32 );\r
- property.mSelector = kAudioDevicePropertyLatency;\r
- if ( AudioObjectHasProperty( id, &property ) == true ) {\r
- result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &latency );\r
- if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;\r
- else {\r
- errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- }\r
- }\r
-\r
- // Byte-swapping: According to AudioHardware.h, the stream data will\r
- // always be presented in native-endian format, so we should never\r
- // need to byte swap.\r
- stream_.doByteSwap[mode] = false;\r
-\r
- // From the CoreAudio documentation, PCM data must be supplied as\r
- // 32-bit floats.\r
- stream_.userFormat = format;\r
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;\r
-\r
- if ( streamCount == 1 )\r
- stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;\r
- else // multiple streams\r
- stream_.nDeviceChannels[mode] = channels;\r
- stream_.nUserChannels[mode] = channels;\r
- stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream\r
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;\r
- else stream_.userInterleaved = true;\r
- stream_.deviceInterleaved[mode] = true;\r
- if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;\r
-\r
- // Set flags for buffer conversion.\r
- stream_.doConvertBuffer[mode] = false;\r
- if ( stream_.userFormat != stream_.deviceFormat[mode] )\r
- stream_.doConvertBuffer[mode] = true;\r
- if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )\r
- stream_.doConvertBuffer[mode] = true;\r
- if ( streamCount == 1 ) {\r
- if ( stream_.nUserChannels[mode] > 1 &&\r
- stream_.userInterleaved != stream_.deviceInterleaved[mode] )\r
- stream_.doConvertBuffer[mode] = true;\r
- }\r
- else if ( monoMode && stream_.userInterleaved )\r
- stream_.doConvertBuffer[mode] = true;\r
-\r
- // Allocate our CoreHandle structure for the stream.\r
- CoreHandle *handle = 0;\r
- if ( stream_.apiHandle == 0 ) {\r
- try {\r
- handle = new CoreHandle;\r
- }\r
- catch ( std::bad_alloc& ) {\r
- errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";\r
- goto error;\r
- }\r
-\r
- if ( pthread_cond_init( &handle->condition, NULL ) ) {\r
- errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";\r
- goto error;\r
- }\r
- stream_.apiHandle = (void *) handle;\r
- }\r
- else\r
- handle = (CoreHandle *) stream_.apiHandle;\r
- handle->iStream[mode] = firstStream;\r
- handle->nStreams[mode] = streamCount;\r
- handle->id[mode] = id;\r
-\r
- // Allocate necessary internal buffers.\r
- unsigned long bufferBytes;\r
- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );\r
- // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );\r
- stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) );\r
- memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) );\r
- if ( stream_.userBuffer[mode] == NULL ) {\r
- errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";\r
- goto error;\r
- }\r
-\r
- // If possible, we will make use of the CoreAudio stream buffers as\r
- // "device buffers". However, we can't do this if using multiple\r
- // streams.\r
- if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {\r
-\r
- bool makeBuffer = true;\r
- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );\r
- if ( mode == INPUT ) {\r
- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {\r
- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );\r
- if ( bufferBytes <= bytesOut ) makeBuffer = false;\r
- }\r
- }\r
-\r
- if ( makeBuffer ) {\r
- bufferBytes *= *bufferSize;\r
- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );\r
- if ( stream_.deviceBuffer == NULL ) {\r
- errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";\r
- goto error;\r
- }\r
- }\r
- }\r
-\r
- stream_.sampleRate = sampleRate;\r
- stream_.device[mode] = device;\r
- stream_.state = STREAM_STOPPED;\r
- stream_.callbackInfo.object = (void *) this;\r
-\r
- // Setup the buffer conversion information structure.\r
- if ( stream_.doConvertBuffer[mode] ) {\r
- if ( streamCount > 1 ) setConvertInfo( mode, 0 );\r
- else setConvertInfo( mode, channelOffset );\r
- }\r
-\r
- if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )\r
- // Only one callback procedure per device.\r
- stream_.mode = DUPLEX;\r
- else {\r
-#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )\r
- result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );\r
-#else\r
- // deprecated in favor of AudioDeviceCreateIOProcID()\r
- result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );\r
-#endif\r
- if ( result != noErr ) {\r
- errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";\r
- errorText_ = errorStream_.str();\r
- goto error;\r
- }\r
- if ( stream_.mode == OUTPUT && mode == INPUT )\r
- stream_.mode = DUPLEX;\r
- else\r
- stream_.mode = mode;\r
- }\r
-\r
- // Setup the device property listener for over/underload.\r
- property.mSelector = kAudioDeviceProcessorOverload;\r
- property.mScope = kAudioObjectPropertyScopeGlobal;\r
- result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle );\r
-\r
- return SUCCESS;\r
-\r
- error:\r
- if ( handle ) {\r
- pthread_cond_destroy( &handle->condition );\r
- delete handle;\r
- stream_.apiHandle = 0;\r
- }\r
-\r
- for ( int i=0; i<2; i++ ) {\r
- if ( stream_.userBuffer[i] ) {\r
- free( stream_.userBuffer[i] );\r
- stream_.userBuffer[i] = 0;\r
- }\r
- }\r
-\r
- if ( stream_.deviceBuffer ) {\r
- free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = 0;\r
- }\r
-\r
- stream_.state = STREAM_CLOSED;\r
- return FAILURE;\r
-}\r
-\r
-void RtApiCore :: closeStream( void )\r
-{\r
- if ( stream_.state == STREAM_CLOSED ) {\r
- errorText_ = "RtApiCore::closeStream(): no open stream to close!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
- if ( stream_.state == STREAM_RUNNING )\r
- AudioDeviceStop( handle->id[0], callbackHandler );\r
-#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )\r
- AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );\r
-#else\r
- // deprecated in favor of AudioDeviceDestroyIOProcID()\r
- AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );\r
-#endif\r
- }\r
-\r
- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {\r
- if ( stream_.state == STREAM_RUNNING )\r
- AudioDeviceStop( handle->id[1], callbackHandler );\r
-#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )\r
- AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );\r
-#else\r
- // deprecated in favor of AudioDeviceDestroyIOProcID()\r
- AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );\r
-#endif\r
- }\r
-\r
- for ( int i=0; i<2; i++ ) {\r
- if ( stream_.userBuffer[i] ) {\r
- free( stream_.userBuffer[i] );\r
- stream_.userBuffer[i] = 0;\r
- }\r
- }\r
-\r
- if ( stream_.deviceBuffer ) {\r
- free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = 0;\r
- }\r
-\r
- // Destroy pthread condition variable.\r
- pthread_cond_destroy( &handle->condition );\r
- delete handle;\r
- stream_.apiHandle = 0;\r
-\r
- stream_.mode = UNINITIALIZED;\r
- stream_.state = STREAM_CLOSED;\r
-}\r
-\r
-void RtApiCore :: startStream( void )\r
-{\r
- verifyStream();\r
- if ( stream_.state == STREAM_RUNNING ) {\r
- errorText_ = "RtApiCore::startStream(): the stream is already running!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- OSStatus result = noErr;\r
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
-\r
- result = AudioDeviceStart( handle->id[0], callbackHandler );\r
- if ( result != noErr ) {\r
- errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
- }\r
-\r
- if ( stream_.mode == INPUT ||\r
- ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {\r
-\r
- result = AudioDeviceStart( handle->id[1], callbackHandler );\r
- if ( result != noErr ) {\r
- errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
- }\r
-\r
- handle->drainCounter = 0;\r
- handle->internalDrain = false;\r
- stream_.state = STREAM_RUNNING;\r
-\r
- unlock:\r
- if ( result == noErr ) return;\r
- error( RtAudioError::SYSTEM_ERROR );\r
-}\r
-\r
-void RtApiCore :: stopStream( void )\r
-{\r
- verifyStream();\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- OSStatus result = noErr;\r
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
-\r
- if ( handle->drainCounter == 0 ) {\r
- handle->drainCounter = 2;\r
- pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled\r
- }\r
-\r
- result = AudioDeviceStop( handle->id[0], callbackHandler );\r
- if ( result != noErr ) {\r
- errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
- }\r
-\r
- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {\r
-\r
- result = AudioDeviceStop( handle->id[1], callbackHandler );\r
- if ( result != noErr ) {\r
- errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
- }\r
-\r
- stream_.state = STREAM_STOPPED;\r
-\r
- unlock:\r
- if ( result == noErr ) return;\r
- error( RtAudioError::SYSTEM_ERROR );\r
-}\r
-\r
-void RtApiCore :: abortStream( void )\r
-{\r
- verifyStream();\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;\r
- handle->drainCounter = 2;\r
-\r
- stopStream();\r
-}\r
-\r
-// This function will be called by a spawned thread when the user\r
-// callback function signals that the stream should be stopped or\r
-// aborted. It is better to handle it this way because the\r
-// callbackEvent() function probably should return before the AudioDeviceStop()\r
-// function is called.\r
-static void *coreStopStream( void *ptr )\r
-{\r
- CallbackInfo *info = (CallbackInfo *) ptr;\r
- RtApiCore *object = (RtApiCore *) info->object;\r
-\r
- object->stopStream();\r
- pthread_exit( NULL );\r
-}\r
-\r
-bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,\r
- const AudioBufferList *inBufferList,\r
- const AudioBufferList *outBufferList )\r
-{\r
- if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;\r
- if ( stream_.state == STREAM_CLOSED ) {\r
- errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";\r
- error( RtAudioError::WARNING );\r
- return FAILURE;\r
- }\r
-\r
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;\r
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;\r
-\r
- // Check if we were draining the stream and signal is finished.\r
- if ( handle->drainCounter > 3 ) {\r
- ThreadHandle threadId;\r
-\r
- stream_.state = STREAM_STOPPING;\r
- if ( handle->internalDrain == true )\r
- pthread_create( &threadId, NULL, coreStopStream, info );\r
- else // external call to stopStream()\r
- pthread_cond_signal( &handle->condition );\r
- return SUCCESS;\r
- }\r
-\r
- AudioDeviceID outputDevice = handle->id[0];\r
-\r
- // Invoke user callback to get fresh output data UNLESS we are\r
- // draining stream or duplex mode AND the input/output devices are\r
- // different AND this function is called for the input device.\r
- if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {\r
- RtAudioCallback callback = (RtAudioCallback) info->callback;\r
- double streamTime = getStreamTime();\r
- RtAudioStreamStatus status = 0;\r
- if ( stream_.mode != INPUT && handle->xrun[0] == true ) {\r
- status |= RTAUDIO_OUTPUT_UNDERFLOW;\r
- handle->xrun[0] = false;\r
- }\r
- if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {\r
- status |= RTAUDIO_INPUT_OVERFLOW;\r
- handle->xrun[1] = false;\r
- }\r
-\r
- int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
- stream_.bufferSize, streamTime, status, info->userData );\r
- if ( cbReturnValue == 2 ) {\r
- stream_.state = STREAM_STOPPING;\r
- handle->drainCounter = 2;\r
- abortStream();\r
- return SUCCESS;\r
- }\r
- else if ( cbReturnValue == 1 ) {\r
- handle->drainCounter = 1;\r
- handle->internalDrain = true;\r
- }\r
- }\r
-\r
- if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {\r
-\r
- if ( handle->drainCounter > 1 ) { // write zeros to the output stream\r
-\r
- if ( handle->nStreams[0] == 1 ) {\r
- memset( outBufferList->mBuffers[handle->iStream[0]].mData,\r
- 0,\r
- outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );\r
- }\r
- else { // fill multiple streams with zeros\r
- for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {\r
- memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,\r
- 0,\r
- outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );\r
- }\r
- }\r
- }\r
- else if ( handle->nStreams[0] == 1 ) {\r
- if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer\r
- convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,\r
- stream_.userBuffer[0], stream_.convertInfo[0] );\r
- }\r
- else { // copy from user buffer\r
- memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,\r
- stream_.userBuffer[0],\r
- outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );\r
- }\r
- }\r
- else { // fill multiple streams\r
- Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];\r
- if ( stream_.doConvertBuffer[0] ) {\r
- convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );\r
- inBuffer = (Float32 *) stream_.deviceBuffer;\r
- }\r
-\r
- if ( stream_.deviceInterleaved[0] == false ) { // mono mode\r
- UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;\r
- for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {\r
- memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,\r
- (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );\r
- }\r
- }\r
- else { // fill multiple multi-channel streams with interleaved data\r
- UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;\r
- Float32 *out, *in;\r
-\r
- bool inInterleaved = ( stream_.userInterleaved ) ? true : false;\r
- UInt32 inChannels = stream_.nUserChannels[0];\r
- if ( stream_.doConvertBuffer[0] ) {\r
- inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode\r
- inChannels = stream_.nDeviceChannels[0];\r
- }\r
-\r
- if ( inInterleaved ) inOffset = 1;\r
- else inOffset = stream_.bufferSize;\r
-\r
- channelsLeft = inChannels;\r
- for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {\r
- in = inBuffer;\r
- out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;\r
- streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;\r
-\r
- outJump = 0;\r
- // Account for possible channel offset in first stream\r
- if ( i == 0 && stream_.channelOffset[0] > 0 ) {\r
- streamChannels -= stream_.channelOffset[0];\r
- outJump = stream_.channelOffset[0];\r
- out += outJump;\r
- }\r
-\r
- // Account for possible unfilled channels at end of the last stream\r
- if ( streamChannels > channelsLeft ) {\r
- outJump = streamChannels - channelsLeft;\r
- streamChannels = channelsLeft;\r
- }\r
-\r
- // Determine input buffer offsets and skips\r
- if ( inInterleaved ) {\r
- inJump = inChannels;\r
- in += inChannels - channelsLeft;\r
- }\r
- else {\r
- inJump = 1;\r
- in += (inChannels - channelsLeft) * inOffset;\r
- }\r
-\r
- for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {\r
- for ( unsigned int j=0; j<streamChannels; j++ ) {\r
- *out++ = in[j*inOffset];\r
- }\r
- out += outJump;\r
- in += inJump;\r
- }\r
- channelsLeft -= streamChannels;\r
- }\r
- }\r
- }\r
-\r
- if ( handle->drainCounter ) {\r
- handle->drainCounter++;\r
- goto unlock;\r
- }\r
- }\r
-\r
- AudioDeviceID inputDevice;\r
- inputDevice = handle->id[1];\r
- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {\r
-\r
- if ( handle->nStreams[1] == 1 ) {\r
- if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer\r
- convertBuffer( stream_.userBuffer[1],\r
- (char *) inBufferList->mBuffers[handle->iStream[1]].mData,\r
- stream_.convertInfo[1] );\r
- }\r
- else { // copy to user buffer\r
- memcpy( stream_.userBuffer[1],\r
- inBufferList->mBuffers[handle->iStream[1]].mData,\r
- inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );\r
- }\r
- }\r
- else { // read from multiple streams\r
- Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];\r
- if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;\r
-\r
- if ( stream_.deviceInterleaved[1] == false ) { // mono mode\r
- UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;\r
- for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {\r
- memcpy( (void *)&outBuffer[i*stream_.bufferSize],\r
- inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );\r
- }\r
- }\r
- else { // read from multiple multi-channel streams\r
- UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;\r
- Float32 *out, *in;\r
-\r
- bool outInterleaved = ( stream_.userInterleaved ) ? true : false;\r
- UInt32 outChannels = stream_.nUserChannels[1];\r
- if ( stream_.doConvertBuffer[1] ) {\r
- outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode\r
- outChannels = stream_.nDeviceChannels[1];\r
- }\r
-\r
- if ( outInterleaved ) outOffset = 1;\r
- else outOffset = stream_.bufferSize;\r
-\r
- channelsLeft = outChannels;\r
- for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {\r
- out = outBuffer;\r
- in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;\r
- streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;\r
-\r
- inJump = 0;\r
- // Account for possible channel offset in first stream\r
- if ( i == 0 && stream_.channelOffset[1] > 0 ) {\r
- streamChannels -= stream_.channelOffset[1];\r
- inJump = stream_.channelOffset[1];\r
- in += inJump;\r
- }\r
-\r
- // Account for possible unread channels at end of the last stream\r
- if ( streamChannels > channelsLeft ) {\r
- inJump = streamChannels - channelsLeft;\r
- streamChannels = channelsLeft;\r
- }\r
-\r
- // Determine output buffer offsets and skips\r
- if ( outInterleaved ) {\r
- outJump = outChannels;\r
- out += outChannels - channelsLeft;\r
- }\r
- else {\r
- outJump = 1;\r
- out += (outChannels - channelsLeft) * outOffset;\r
- }\r
-\r
- for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {\r
- for ( unsigned int j=0; j<streamChannels; j++ ) {\r
- out[j*outOffset] = *in++;\r
- }\r
- out += outJump;\r
- in += inJump;\r
- }\r
- channelsLeft -= streamChannels;\r
- }\r
- }\r
- \r
- if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer\r
- convertBuffer( stream_.userBuffer[1],\r
- stream_.deviceBuffer,\r
- stream_.convertInfo[1] );\r
- }\r
- }\r
- }\r
-\r
- unlock:\r
- //MUTEX_UNLOCK( &stream_.mutex );\r
-\r
- RtApi::tickStreamTime();\r
- return SUCCESS;\r
-}\r
-\r
-const char* RtApiCore :: getErrorCode( OSStatus code )\r
-{\r
- switch( code ) {\r
-\r
- case kAudioHardwareNotRunningError:\r
- return "kAudioHardwareNotRunningError";\r
-\r
- case kAudioHardwareUnspecifiedError:\r
- return "kAudioHardwareUnspecifiedError";\r
-\r
- case kAudioHardwareUnknownPropertyError:\r
- return "kAudioHardwareUnknownPropertyError";\r
-\r
- case kAudioHardwareBadPropertySizeError:\r
- return "kAudioHardwareBadPropertySizeError";\r
-\r
- case kAudioHardwareIllegalOperationError:\r
- return "kAudioHardwareIllegalOperationError";\r
-\r
- case kAudioHardwareBadObjectError:\r
- return "kAudioHardwareBadObjectError";\r
-\r
- case kAudioHardwareBadDeviceError:\r
- return "kAudioHardwareBadDeviceError";\r
-\r
- case kAudioHardwareBadStreamError:\r
- return "kAudioHardwareBadStreamError";\r
-\r
- case kAudioHardwareUnsupportedOperationError:\r
- return "kAudioHardwareUnsupportedOperationError";\r
-\r
- case kAudioDeviceUnsupportedFormatError:\r
- return "kAudioDeviceUnsupportedFormatError";\r
-\r
- case kAudioDevicePermissionsError:\r
- return "kAudioDevicePermissionsError";\r
-\r
- default:\r
- return "CoreAudio unknown error";\r
- }\r
-}\r
-\r
- //******************** End of __MACOSX_CORE__ *********************//\r
-#endif\r
-\r
-#if defined(__UNIX_JACK__)\r
-\r
-// JACK is a low-latency audio server, originally written for the\r
-// GNU/Linux operating system and now also ported to OS-X. It can\r
-// connect a number of different applications to an audio device, as\r
-// well as allowing them to share audio between themselves.\r
-//\r
-// When using JACK with RtAudio, "devices" refer to JACK clients that\r
-// have ports connected to the server. The JACK server is typically\r
-// started in a terminal as follows:\r
-//\r
-// .jackd -d alsa -d hw:0\r
-//\r
-// or through an interface program such as qjackctl. Many of the\r
-// parameters normally set for a stream are fixed by the JACK server\r
-// and can be specified when the JACK server is started. In\r
-// particular,\r
-//\r
-// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4\r
-//\r
-// specifies a sample rate of 44100 Hz, a buffer size of 512 sample\r
-// frames, and number of buffers = 4. Once the server is running, it\r
-// is not possible to override these values. If the values are not\r
-// specified in the command-line, the JACK server uses default values.\r
-//\r
-// The JACK server does not have to be running when an instance of\r
-// RtApiJack is created, though the function getDeviceCount() will\r
-// report 0 devices found until JACK has been started. When no\r
-// devices are available (i.e., the JACK server is not running), a\r
-// stream cannot be opened.\r
-\r
-#include <jack/jack.h>\r
-#include <unistd.h>\r
-#include <cstdio>\r
-\r
-// A structure to hold various information related to the Jack API\r
-// implementation.\r
-struct JackHandle {\r
- jack_client_t *client;\r
- jack_port_t **ports[2];\r
- std::string deviceName[2];\r
- bool xrun[2];\r
- pthread_cond_t condition;\r
- int drainCounter; // Tracks callback counts when draining\r
- bool internalDrain; // Indicates if stop is initiated from callback or not.\r
-\r
- JackHandle()\r
- :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }\r
-};\r
-\r
-static void jackSilentError( const char * ) {};\r
-\r
-RtApiJack :: RtApiJack()\r
-{\r
- // Nothing to do here.\r
-#if !defined(__RTAUDIO_DEBUG__)\r
- // Turn off Jack's internal error reporting.\r
- jack_set_error_function( &jackSilentError );\r
-#endif\r
-}\r
-\r
-RtApiJack :: ~RtApiJack()\r
-{\r
- if ( stream_.state != STREAM_CLOSED ) closeStream();\r
-}\r
-\r
-unsigned int RtApiJack :: getDeviceCount( void )\r
-{\r
- // See if we can become a jack client.\r
- jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption;\r
- jack_status_t *status = NULL;\r
- jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );\r
- if ( client == 0 ) return 0;\r
-\r
- const char **ports;\r
- std::string port, previousPort;\r
- unsigned int nChannels = 0, nDevices = 0;\r
- ports = jack_get_ports( client, NULL, NULL, 0 );\r
- if ( ports ) {\r
- // Parse the port names up to the first colon (:).\r
- size_t iColon = 0;\r
- do {\r
- port = (char *) ports[ nChannels ];\r
- iColon = port.find(":");\r
- if ( iColon != std::string::npos ) {\r
- port = port.substr( 0, iColon + 1 );\r
- if ( port != previousPort ) {\r
- nDevices++;\r
- previousPort = port;\r
- }\r
- }\r
- } while ( ports[++nChannels] );\r
- free( ports );\r
- }\r
-\r
- jack_client_close( client );\r
- return nDevices;\r
-}\r
-\r
-RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )\r
-{\r
- RtAudio::DeviceInfo info;\r
- info.probed = false;\r
-\r
- jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption\r
- jack_status_t *status = NULL;\r
- jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );\r
- if ( client == 0 ) {\r
- errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- const char **ports;\r
- std::string port, previousPort;\r
- unsigned int nPorts = 0, nDevices = 0;\r
- ports = jack_get_ports( client, NULL, NULL, 0 );\r
- if ( ports ) {\r
- // Parse the port names up to the first colon (:).\r
- size_t iColon = 0;\r
- do {\r
- port = (char *) ports[ nPorts ];\r
- iColon = port.find(":");\r
- if ( iColon != std::string::npos ) {\r
- port = port.substr( 0, iColon );\r
- if ( port != previousPort ) {\r
- if ( nDevices == device ) info.name = port;\r
- nDevices++;\r
- previousPort = port;\r
- }\r
- }\r
- } while ( ports[++nPorts] );\r
- free( ports );\r
- }\r
-\r
- if ( device >= nDevices ) {\r
- jack_client_close( client );\r
- errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";\r
- error( RtAudioError::INVALID_USE );\r
- return info;\r
- }\r
-\r
- // Get the current jack server sample rate.\r
- info.sampleRates.clear();\r
- info.sampleRates.push_back( jack_get_sample_rate( client ) );\r
-\r
- // Count the available ports containing the client name as device\r
- // channels. Jack "input ports" equal RtAudio output channels.\r
- unsigned int nChannels = 0;\r
- ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );\r
- if ( ports ) {\r
- while ( ports[ nChannels ] ) nChannels++;\r
- free( ports );\r
- info.outputChannels = nChannels;\r
- }\r
-\r
- // Jack "output ports" equal RtAudio input channels.\r
- nChannels = 0;\r
- ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );\r
- if ( ports ) {\r
- while ( ports[ nChannels ] ) nChannels++;\r
- free( ports );\r
- info.inputChannels = nChannels;\r
- }\r
-\r
- if ( info.outputChannels == 0 && info.inputChannels == 0 ) {\r
- jack_client_close(client);\r
- errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- // If device opens for both playback and capture, we determine the channels.\r
- if ( info.outputChannels > 0 && info.inputChannels > 0 )\r
- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;\r
-\r
- // Jack always uses 32-bit floats.\r
- info.nativeFormats = RTAUDIO_FLOAT32;\r
-\r
- // Jack doesn't provide default devices so we'll use the first available one.\r
- if ( device == 0 && info.outputChannels > 0 )\r
- info.isDefaultOutput = true;\r
- if ( device == 0 && info.inputChannels > 0 )\r
- info.isDefaultInput = true;\r
-\r
- jack_client_close(client);\r
- info.probed = true;\r
- return info;\r
-}\r
-\r
-static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )\r
-{\r
- CallbackInfo *info = (CallbackInfo *) infoPointer;\r
-\r
- RtApiJack *object = (RtApiJack *) info->object;\r
- if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;\r
-\r
- return 0;\r
-}\r
-\r
-// This function will be called by a spawned thread when the Jack\r
-// server signals that it is shutting down. It is necessary to handle\r
-// it this way because the jackShutdown() function must return before\r
-// the jack_deactivate() function (in closeStream()) will return.\r
-static void *jackCloseStream( void *ptr )\r
-{\r
- CallbackInfo *info = (CallbackInfo *) ptr;\r
- RtApiJack *object = (RtApiJack *) info->object;\r
-\r
- object->closeStream();\r
-\r
- pthread_exit( NULL );\r
-}\r
-static void jackShutdown( void *infoPointer )\r
-{\r
- CallbackInfo *info = (CallbackInfo *) infoPointer;\r
- RtApiJack *object = (RtApiJack *) info->object;\r
-\r
- // Check current stream state. If stopped, then we'll assume this\r
- // was called as a result of a call to RtApiJack::stopStream (the\r
- // deactivation of a client handle causes this function to be called).\r
- // If not, we'll assume the Jack server is shutting down or some\r
- // other problem occurred and we should close the stream.\r
- if ( object->isStreamRunning() == false ) return;\r
-\r
- ThreadHandle threadId;\r
- pthread_create( &threadId, NULL, jackCloseStream, info );\r
- std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;\r
-}\r
-\r
-static int jackXrun( void *infoPointer )\r
-{\r
- JackHandle *handle = (JackHandle *) infoPointer;\r
-\r
- if ( handle->ports[0] ) handle->xrun[0] = true;\r
- if ( handle->ports[1] ) handle->xrun[1] = true;\r
-\r
- return 0;\r
-}\r
-\r
-bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,\r
- unsigned int firstChannel, unsigned int sampleRate,\r
- RtAudioFormat format, unsigned int *bufferSize,\r
- RtAudio::StreamOptions *options )\r
-{\r
- JackHandle *handle = (JackHandle *) stream_.apiHandle;\r
-\r
- // Look for jack server and try to become a client (only do once per stream).\r
- jack_client_t *client = 0;\r
- if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {\r
- jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;\r
- jack_status_t *status = NULL;\r
- if ( options && !options->streamName.empty() )\r
- client = jack_client_open( options->streamName.c_str(), jackoptions, status );\r
- else\r
- client = jack_client_open( "RtApiJack", jackoptions, status );\r
- if ( client == 0 ) {\r
- errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";\r
- error( RtAudioError::WARNING );\r
- return FAILURE;\r
- }\r
- }\r
- else {\r
- // The handle must have been created on an earlier pass.\r
- client = handle->client;\r
- }\r
-\r
- const char **ports;\r
- std::string port, previousPort, deviceName;\r
- unsigned int nPorts = 0, nDevices = 0;\r
- ports = jack_get_ports( client, NULL, NULL, 0 );\r
- if ( ports ) {\r
- // Parse the port names up to the first colon (:).\r
- size_t iColon = 0;\r
- do {\r
- port = (char *) ports[ nPorts ];\r
- iColon = port.find(":");\r
- if ( iColon != std::string::npos ) {\r
- port = port.substr( 0, iColon );\r
- if ( port != previousPort ) {\r
- if ( nDevices == device ) deviceName = port;\r
- nDevices++;\r
- previousPort = port;\r
- }\r
- }\r
- } while ( ports[++nPorts] );\r
- free( ports );\r
- }\r
-\r
- if ( device >= nDevices ) {\r
- errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";\r
- return FAILURE;\r
- }\r
-\r
- // Count the available ports containing the client name as device\r
- // channels. Jack "input ports" equal RtAudio output channels.\r
- unsigned int nChannels = 0;\r
- unsigned long flag = JackPortIsInput;\r
- if ( mode == INPUT ) flag = JackPortIsOutput;\r
- ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );\r
- if ( ports ) {\r
- while ( ports[ nChannels ] ) nChannels++;\r
- free( ports );\r
- }\r
-\r
- // Compare the jack ports for specified client to the requested number of channels.\r
- if ( nChannels < (channels + firstChannel) ) {\r
- errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Check the jack server sample rate.\r
- unsigned int jackRate = jack_get_sample_rate( client );\r
- if ( sampleRate != jackRate ) {\r
- jack_client_close( client );\r
- errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- stream_.sampleRate = jackRate;\r
-\r
- // Get the latency of the JACK port.\r
- ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );\r
- if ( ports[ firstChannel ] ) {\r
- // Added by Ge Wang\r
- jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);\r
- // the range (usually the min and max are equal)\r
- jack_latency_range_t latrange; latrange.min = latrange.max = 0;\r
- // get the latency range\r
- jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );\r
- // be optimistic, use the min!\r
- stream_.latency[mode] = latrange.min;\r
- //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );\r
- }\r
- free( ports );\r
-\r
- // The jack server always uses 32-bit floating-point data.\r
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;\r
- stream_.userFormat = format;\r
-\r
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;\r
- else stream_.userInterleaved = true;\r
-\r
- // Jack always uses non-interleaved buffers.\r
- stream_.deviceInterleaved[mode] = false;\r
-\r
- // Jack always provides host byte-ordered data.\r
- stream_.doByteSwap[mode] = false;\r
-\r
- // Get the buffer size. The buffer size and number of buffers\r
- // (periods) is set when the jack server is started.\r
- stream_.bufferSize = (int) jack_get_buffer_size( client );\r
- *bufferSize = stream_.bufferSize;\r
-\r
- stream_.nDeviceChannels[mode] = channels;\r
- stream_.nUserChannels[mode] = channels;\r
-\r
- // Set flags for buffer conversion.\r
- stream_.doConvertBuffer[mode] = false;\r
- if ( stream_.userFormat != stream_.deviceFormat[mode] )\r
- stream_.doConvertBuffer[mode] = true;\r
- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&\r
- stream_.nUserChannels[mode] > 1 )\r
- stream_.doConvertBuffer[mode] = true;\r
-\r
- // Allocate our JackHandle structure for the stream.\r
- if ( handle == 0 ) {\r
- try {\r
- handle = new JackHandle;\r
- }\r
- catch ( std::bad_alloc& ) {\r
- errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";\r
- goto error;\r
- }\r
-\r
- if ( pthread_cond_init(&handle->condition, NULL) ) {\r
- errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";\r
- goto error;\r
- }\r
- stream_.apiHandle = (void *) handle;\r
- handle->client = client;\r
- }\r
- handle->deviceName[mode] = deviceName;\r
-\r
- // Allocate necessary internal buffers.\r
- unsigned long bufferBytes;\r
- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );\r
- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );\r
- if ( stream_.userBuffer[mode] == NULL ) {\r
- errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";\r
- goto error;\r
- }\r
-\r
- if ( stream_.doConvertBuffer[mode] ) {\r
-\r
- bool makeBuffer = true;\r
- if ( mode == OUTPUT )\r
- bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );\r
- else { // mode == INPUT\r
- bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );\r
- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {\r
- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);\r
- if ( bufferBytes < bytesOut ) makeBuffer = false;\r
- }\r
- }\r
-\r
- if ( makeBuffer ) {\r
- bufferBytes *= *bufferSize;\r
- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );\r
- if ( stream_.deviceBuffer == NULL ) {\r
- errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";\r
- goto error;\r
- }\r
- }\r
- }\r
-\r
- // Allocate memory for the Jack ports (channels) identifiers.\r
- handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );\r
- if ( handle->ports[mode] == NULL ) {\r
- errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";\r
- goto error;\r
- }\r
-\r
- stream_.device[mode] = device;\r
- stream_.channelOffset[mode] = firstChannel;\r
- stream_.state = STREAM_STOPPED;\r
- stream_.callbackInfo.object = (void *) this;\r
-\r
- if ( stream_.mode == OUTPUT && mode == INPUT )\r
- // We had already set up the stream for output.\r
- stream_.mode = DUPLEX;\r
- else {\r
- stream_.mode = mode;\r
- jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );\r
- jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );\r
- jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );\r
- }\r
-\r
- // Register our ports.\r
- char label[64];\r
- if ( mode == OUTPUT ) {\r
- for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {\r
- snprintf( label, 64, "outport %d", i );\r
- handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,\r
- JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );\r
- }\r
- }\r
- else {\r
- for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {\r
- snprintf( label, 64, "inport %d", i );\r
- handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,\r
- JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );\r
- }\r
- }\r
-\r
- // Setup the buffer conversion information structure. We don't use\r
- // buffers to do channel offsets, so we override that parameter\r
- // here.\r
- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );\r
-\r
- return SUCCESS;\r
-\r
- error:\r
- if ( handle ) {\r
- pthread_cond_destroy( &handle->condition );\r
- jack_client_close( handle->client );\r
-\r
- if ( handle->ports[0] ) free( handle->ports[0] );\r
- if ( handle->ports[1] ) free( handle->ports[1] );\r
-\r
- delete handle;\r
- stream_.apiHandle = 0;\r
- }\r
-\r
- for ( int i=0; i<2; i++ ) {\r
- if ( stream_.userBuffer[i] ) {\r
- free( stream_.userBuffer[i] );\r
- stream_.userBuffer[i] = 0;\r
- }\r
- }\r
-\r
- if ( stream_.deviceBuffer ) {\r
- free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = 0;\r
- }\r
-\r
- return FAILURE;\r
-}\r
-\r
-void RtApiJack :: closeStream( void )\r
-{\r
- if ( stream_.state == STREAM_CLOSED ) {\r
- errorText_ = "RtApiJack::closeStream(): no open stream to close!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- JackHandle *handle = (JackHandle *) stream_.apiHandle;\r
- if ( handle ) {\r
-\r
- if ( stream_.state == STREAM_RUNNING )\r
- jack_deactivate( handle->client );\r
-\r
- jack_client_close( handle->client );\r
- }\r
-\r
- if ( handle ) {\r
- if ( handle->ports[0] ) free( handle->ports[0] );\r
- if ( handle->ports[1] ) free( handle->ports[1] );\r
- pthread_cond_destroy( &handle->condition );\r
- delete handle;\r
- stream_.apiHandle = 0;\r
- }\r
-\r
- for ( int i=0; i<2; i++ ) {\r
- if ( stream_.userBuffer[i] ) {\r
- free( stream_.userBuffer[i] );\r
- stream_.userBuffer[i] = 0;\r
- }\r
- }\r
-\r
- if ( stream_.deviceBuffer ) {\r
- free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = 0;\r
- }\r
-\r
- stream_.mode = UNINITIALIZED;\r
- stream_.state = STREAM_CLOSED;\r
-}\r
-\r
-void RtApiJack :: startStream( void )\r
-{\r
- verifyStream();\r
- if ( stream_.state == STREAM_RUNNING ) {\r
- errorText_ = "RtApiJack::startStream(): the stream is already running!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- JackHandle *handle = (JackHandle *) stream_.apiHandle;\r
- int result = jack_activate( handle->client );\r
- if ( result ) {\r
- errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";\r
- goto unlock;\r
- }\r
-\r
- const char **ports;\r
-\r
- // Get the list of available ports.\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
- result = 1;\r
- ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);\r
- if ( ports == NULL) {\r
- errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";\r
- goto unlock;\r
- }\r
-\r
- // Now make the port connections. Since RtAudio wasn't designed to\r
- // allow the user to select particular channels of a device, we'll\r
- // just open the first "nChannels" ports with offset.\r
- for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {\r
- result = 1;\r
- if ( ports[ stream_.channelOffset[0] + i ] )\r
- result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );\r
- if ( result ) {\r
- free( ports );\r
- errorText_ = "RtApiJack::startStream(): error connecting output ports!";\r
- goto unlock;\r
- }\r
- }\r
- free(ports);\r
- }\r
-\r
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {\r
- result = 1;\r
- ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );\r
- if ( ports == NULL) {\r
- errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";\r
- goto unlock;\r
- }\r
-\r
- // Now make the port connections. See note above.\r
- for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {\r
- result = 1;\r
- if ( ports[ stream_.channelOffset[1] + i ] )\r
- result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );\r
- if ( result ) {\r
- free( ports );\r
- errorText_ = "RtApiJack::startStream(): error connecting input ports!";\r
- goto unlock;\r
- }\r
- }\r
- free(ports);\r
- }\r
-\r
- handle->drainCounter = 0;\r
- handle->internalDrain = false;\r
- stream_.state = STREAM_RUNNING;\r
-\r
- unlock:\r
- if ( result == 0 ) return;\r
- error( RtAudioError::SYSTEM_ERROR );\r
-}\r
-\r
-void RtApiJack :: stopStream( void )\r
-{\r
- verifyStream();\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- JackHandle *handle = (JackHandle *) stream_.apiHandle;\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
-\r
- if ( handle->drainCounter == 0 ) {\r
- handle->drainCounter = 2;\r
- pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled\r
- }\r
- }\r
-\r
- jack_deactivate( handle->client );\r
- stream_.state = STREAM_STOPPED;\r
-}\r
-\r
-void RtApiJack :: abortStream( void )\r
-{\r
- verifyStream();\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- JackHandle *handle = (JackHandle *) stream_.apiHandle;\r
- handle->drainCounter = 2;\r
-\r
- stopStream();\r
-}\r
-\r
-// This function will be called by a spawned thread when the user\r
-// callback function signals that the stream should be stopped or\r
-// aborted. It is necessary to handle it this way because the\r
-// callbackEvent() function must return before the jack_deactivate()\r
-// function will return.\r
-static void *jackStopStream( void *ptr )\r
-{\r
- CallbackInfo *info = (CallbackInfo *) ptr;\r
- RtApiJack *object = (RtApiJack *) info->object;\r
-\r
- object->stopStream();\r
- pthread_exit( NULL );\r
-}\r
-\r
-bool RtApiJack :: callbackEvent( unsigned long nframes )\r
-{\r
- if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;\r
- if ( stream_.state == STREAM_CLOSED ) {\r
- errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";\r
- error( RtAudioError::WARNING );\r
- return FAILURE;\r
- }\r
- if ( stream_.bufferSize != nframes ) {\r
- errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";\r
- error( RtAudioError::WARNING );\r
- return FAILURE;\r
- }\r
-\r
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;\r
- JackHandle *handle = (JackHandle *) stream_.apiHandle;\r
-\r
- // Check if we were draining the stream and signal is finished.\r
- if ( handle->drainCounter > 3 ) {\r
- ThreadHandle threadId;\r
-\r
- stream_.state = STREAM_STOPPING;\r
- if ( handle->internalDrain == true )\r
- pthread_create( &threadId, NULL, jackStopStream, info );\r
- else\r
- pthread_cond_signal( &handle->condition );\r
- return SUCCESS;\r
- }\r
-\r
- // Invoke user callback first, to get fresh output data.\r
- if ( handle->drainCounter == 0 ) {\r
- RtAudioCallback callback = (RtAudioCallback) info->callback;\r
- double streamTime = getStreamTime();\r
- RtAudioStreamStatus status = 0;\r
- if ( stream_.mode != INPUT && handle->xrun[0] == true ) {\r
- status |= RTAUDIO_OUTPUT_UNDERFLOW;\r
- handle->xrun[0] = false;\r
- }\r
- if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {\r
- status |= RTAUDIO_INPUT_OVERFLOW;\r
- handle->xrun[1] = false;\r
- }\r
- int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
- stream_.bufferSize, streamTime, status, info->userData );\r
- if ( cbReturnValue == 2 ) {\r
- stream_.state = STREAM_STOPPING;\r
- handle->drainCounter = 2;\r
- ThreadHandle id;\r
- pthread_create( &id, NULL, jackStopStream, info );\r
- return SUCCESS;\r
- }\r
- else if ( cbReturnValue == 1 ) {\r
- handle->drainCounter = 1;\r
- handle->internalDrain = true;\r
- }\r
- }\r
-\r
- jack_default_audio_sample_t *jackbuffer;\r
- unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
-\r
- if ( handle->drainCounter > 1 ) { // write zeros to the output stream\r
-\r
- for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {\r
- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );\r
- memset( jackbuffer, 0, bufferBytes );\r
- }\r
-\r
- }\r
- else if ( stream_.doConvertBuffer[0] ) {\r
-\r
- convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );\r
-\r
- for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {\r
- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );\r
- memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );\r
- }\r
- }\r
- else { // no buffer conversion\r
- for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {\r
- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );\r
- memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );\r
- }\r
- }\r
-\r
- if ( handle->drainCounter ) {\r
- handle->drainCounter++;\r
- goto unlock;\r
- }\r
- }\r
-\r
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {\r
-\r
- if ( stream_.doConvertBuffer[1] ) {\r
- for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {\r
- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );\r
- memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );\r
- }\r
- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );\r
- }\r
- else { // no buffer conversion\r
- for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {\r
- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );\r
- memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );\r
- }\r
- }\r
- }\r
-\r
- unlock:\r
- RtApi::tickStreamTime();\r
- return SUCCESS;\r
-}\r
- //******************** End of __UNIX_JACK__ *********************//\r
-#endif\r
-\r
-#if defined(__WINDOWS_ASIO__) // ASIO API on Windows\r
-\r
-// The ASIO API is designed around a callback scheme, so this\r
-// implementation is similar to that used for OS-X CoreAudio and Linux\r
-// Jack. The primary constraint with ASIO is that it only allows\r
-// access to a single driver at a time. Thus, it is not possible to\r
-// have more than one simultaneous RtAudio stream.\r
-//\r
-// This implementation also requires a number of external ASIO files\r
-// and a few global variables. The ASIO callback scheme does not\r
-// allow for the passing of user data, so we must create a global\r
-// pointer to our callbackInfo structure.\r
-//\r
-// On unix systems, we make use of a pthread condition variable.\r
-// Since there is no equivalent in Windows, I hacked something based\r
-// on information found in\r
-// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.\r
-\r
-#include "asiosys.h"\r
-#include "asio.h"\r
-#include "iasiothiscallresolver.h"\r
-#include "asiodrivers.h"\r
-#include <cmath>\r
-\r
-static AsioDrivers drivers;\r
-static ASIOCallbacks asioCallbacks;\r
-static ASIODriverInfo driverInfo;\r
-static CallbackInfo *asioCallbackInfo;\r
-static bool asioXRun;\r
-\r
-struct AsioHandle {\r
- int drainCounter; // Tracks callback counts when draining\r
- bool internalDrain; // Indicates if stop is initiated from callback or not.\r
- ASIOBufferInfo *bufferInfos;\r
- HANDLE condition;\r
-\r
- AsioHandle()\r
- :drainCounter(0), internalDrain(false), bufferInfos(0) {}\r
-};\r
-\r
-// Function declarations (definitions at end of section)\r
-static const char* getAsioErrorString( ASIOError result );\r
-static void sampleRateChanged( ASIOSampleRate sRate );\r
-static long asioMessages( long selector, long value, void* message, double* opt );\r
-\r
-RtApiAsio :: RtApiAsio()\r
-{\r
- // ASIO cannot run on a multi-threaded appartment. You can call\r
- // CoInitialize beforehand, but it must be for appartment threading\r
- // (in which case, CoInitilialize will return S_FALSE here).\r
- coInitialized_ = false;\r
- HRESULT hr = CoInitialize( NULL ); \r
- if ( FAILED(hr) ) {\r
- errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";\r
- error( RtAudioError::WARNING );\r
- }\r
- coInitialized_ = true;\r
-\r
- drivers.removeCurrentDriver();\r
- driverInfo.asioVersion = 2;\r
-\r
- // See note in DirectSound implementation about GetDesktopWindow().\r
- driverInfo.sysRef = GetForegroundWindow();\r
-}\r
-\r
-RtApiAsio :: ~RtApiAsio()\r
-{\r
- if ( stream_.state != STREAM_CLOSED ) closeStream();\r
- if ( coInitialized_ ) CoUninitialize();\r
-}\r
-\r
-unsigned int RtApiAsio :: getDeviceCount( void )\r
-{\r
- return (unsigned int) drivers.asioGetNumDev();\r
-}\r
-\r
-RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )\r
-{\r
- RtAudio::DeviceInfo info;\r
- info.probed = false;\r
-\r
- // Get device ID\r
- unsigned int nDevices = getDeviceCount();\r
- if ( nDevices == 0 ) {\r
- errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";\r
- error( RtAudioError::INVALID_USE );\r
- return info;\r
- }\r
-\r
- if ( device >= nDevices ) {\r
- errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";\r
- error( RtAudioError::INVALID_USE );\r
- return info;\r
- }\r
-\r
- // If a stream is already open, we cannot probe other devices. Thus, use the saved results.\r
- if ( stream_.state != STREAM_CLOSED ) {\r
- if ( device >= devices_.size() ) {\r
- errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
- return devices_[ device ];\r
- }\r
-\r
- char driverName[32];\r
- ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );\r
- if ( result != ASE_OK ) {\r
- errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- info.name = driverName;\r
-\r
- if ( !drivers.loadDriver( driverName ) ) {\r
- errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- result = ASIOInit( &driverInfo );\r
- if ( result != ASE_OK ) {\r
- errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- // Determine the device channel information.\r
- long inputChannels, outputChannels;\r
- result = ASIOGetChannels( &inputChannels, &outputChannels );\r
- if ( result != ASE_OK ) {\r
- drivers.removeCurrentDriver();\r
- errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- info.outputChannels = outputChannels;\r
- info.inputChannels = inputChannels;\r
- if ( info.outputChannels > 0 && info.inputChannels > 0 )\r
- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;\r
-\r
- // Determine the supported sample rates.\r
- info.sampleRates.clear();\r
- for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {\r
- result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );\r
- if ( result == ASE_OK )\r
- info.sampleRates.push_back( SAMPLE_RATES[i] );\r
- }\r
-\r
- // Determine supported data types ... just check first channel and assume rest are the same.\r
- ASIOChannelInfo channelInfo;\r
- channelInfo.channel = 0;\r
- channelInfo.isInput = true;\r
- if ( info.inputChannels <= 0 ) channelInfo.isInput = false;\r
- result = ASIOGetChannelInfo( &channelInfo );\r
- if ( result != ASE_OK ) {\r
- drivers.removeCurrentDriver();\r
- errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- info.nativeFormats = 0;\r
- if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )\r
- info.nativeFormats |= RTAUDIO_SINT16;\r
- else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )\r
- info.nativeFormats |= RTAUDIO_SINT32;\r
- else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )\r
- info.nativeFormats |= RTAUDIO_FLOAT32;\r
- else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )\r
- info.nativeFormats |= RTAUDIO_FLOAT64;\r
- else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )\r
- info.nativeFormats |= RTAUDIO_SINT24;\r
-\r
- if ( info.outputChannels > 0 )\r
- if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;\r
- if ( info.inputChannels > 0 )\r
- if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;\r
-\r
- info.probed = true;\r
- drivers.removeCurrentDriver();\r
- return info;\r
-}\r
-\r
-static void bufferSwitch( long index, ASIOBool /*processNow*/ )\r
-{\r
- RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;\r
- object->callbackEvent( index );\r
-}\r
-\r
-void RtApiAsio :: saveDeviceInfo( void )\r
-{\r
- devices_.clear();\r
-\r
- unsigned int nDevices = getDeviceCount();\r
- devices_.resize( nDevices );\r
- for ( unsigned int i=0; i<nDevices; i++ )\r
- devices_[i] = getDeviceInfo( i );\r
-}\r
-\r
-bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,\r
- unsigned int firstChannel, unsigned int sampleRate,\r
- RtAudioFormat format, unsigned int *bufferSize,\r
- RtAudio::StreamOptions *options )\r
-{\r
- // For ASIO, a duplex stream MUST use the same driver.\r
- if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) {\r
- errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";\r
- return FAILURE;\r
- }\r
-\r
- char driverName[32];\r
- ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );\r
- if ( result != ASE_OK ) {\r
- errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Only load the driver once for duplex stream.\r
- if ( mode != INPUT || stream_.mode != OUTPUT ) {\r
- // The getDeviceInfo() function will not work when a stream is open\r
- // because ASIO does not allow multiple devices to run at the same\r
- // time. Thus, we'll probe the system before opening a stream and\r
- // save the results for use by getDeviceInfo().\r
- this->saveDeviceInfo();\r
-\r
- if ( !drivers.loadDriver( driverName ) ) {\r
- errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- result = ASIOInit( &driverInfo );\r
- if ( result != ASE_OK ) {\r
- errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- }\r
-\r
- // Check the device channel count.\r
- long inputChannels, outputChannels;\r
- result = ASIOGetChannels( &inputChannels, &outputChannels );\r
- if ( result != ASE_OK ) {\r
- drivers.removeCurrentDriver();\r
- errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||\r
- ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {\r
- drivers.removeCurrentDriver();\r
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- stream_.nDeviceChannels[mode] = channels;\r
- stream_.nUserChannels[mode] = channels;\r
- stream_.channelOffset[mode] = firstChannel;\r
-\r
- // Verify the sample rate is supported.\r
- result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );\r
- if ( result != ASE_OK ) {\r
- drivers.removeCurrentDriver();\r
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Get the current sample rate\r
- ASIOSampleRate currentRate;\r
- result = ASIOGetSampleRate( ¤tRate );\r
- if ( result != ASE_OK ) {\r
- drivers.removeCurrentDriver();\r
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Set the sample rate only if necessary\r
- if ( currentRate != sampleRate ) {\r
- result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );\r
- if ( result != ASE_OK ) {\r
- drivers.removeCurrentDriver();\r
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- }\r
-\r
- // Determine the driver data type.\r
- ASIOChannelInfo channelInfo;\r
- channelInfo.channel = 0;\r
- if ( mode == OUTPUT ) channelInfo.isInput = false;\r
- else channelInfo.isInput = true;\r
- result = ASIOGetChannelInfo( &channelInfo );\r
- if ( result != ASE_OK ) {\r
- drivers.removeCurrentDriver();\r
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Assuming WINDOWS host is always little-endian.\r
- stream_.doByteSwap[mode] = false;\r
- stream_.userFormat = format;\r
- stream_.deviceFormat[mode] = 0;\r
- if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;\r
- if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;\r
- }\r
- else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;\r
- if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;\r
- }\r
- else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {\r
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;\r
- if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;\r
- }\r
- else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {\r
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;\r
- if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;\r
- }\r
- else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT24;\r
- if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;\r
- }\r
-\r
- if ( stream_.deviceFormat[mode] == 0 ) {\r
- drivers.removeCurrentDriver();\r
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Set the buffer size. For a duplex stream, this will end up\r
- // setting the buffer size based on the input constraints, which\r
- // should be ok.\r
- long minSize, maxSize, preferSize, granularity;\r
- result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );\r
- if ( result != ASE_OK ) {\r
- drivers.removeCurrentDriver();\r
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;\r
- else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;\r
- else if ( granularity == -1 ) {\r
- // Make sure bufferSize is a power of two.\r
- int log2_of_min_size = 0;\r
- int log2_of_max_size = 0;\r
-\r
- for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {\r
- if ( minSize & ((long)1 << i) ) log2_of_min_size = i;\r
- if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;\r
- }\r
-\r
- long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );\r
- int min_delta_num = log2_of_min_size;\r
-\r
- for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {\r
- long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );\r
- if (current_delta < min_delta) {\r
- min_delta = current_delta;\r
- min_delta_num = i;\r
- }\r
- }\r
-\r
- *bufferSize = ( (unsigned int)1 << min_delta_num );\r
- if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;\r
- else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;\r
- }\r
- else if ( granularity != 0 ) {\r
- // Set to an even multiple of granularity, rounding up.\r
- *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;\r
- }\r
-\r
- if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) {\r
- drivers.removeCurrentDriver();\r
- errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";\r
- return FAILURE;\r
- }\r
-\r
- stream_.bufferSize = *bufferSize;\r
- stream_.nBuffers = 2;\r
-\r
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;\r
- else stream_.userInterleaved = true;\r
-\r
- // ASIO always uses non-interleaved buffers.\r
- stream_.deviceInterleaved[mode] = false;\r
-\r
- // Allocate, if necessary, our AsioHandle structure for the stream.\r
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;\r
- if ( handle == 0 ) {\r
- try {\r
- handle = new AsioHandle;\r
- }\r
- catch ( std::bad_alloc& ) {\r
- //if ( handle == NULL ) { \r
- drivers.removeCurrentDriver();\r
- errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";\r
- return FAILURE;\r
- }\r
- handle->bufferInfos = 0;\r
-\r
- // Create a manual-reset event.\r
- handle->condition = CreateEvent( NULL, // no security\r
- TRUE, // manual-reset\r
- FALSE, // non-signaled initially\r
- NULL ); // unnamed\r
- stream_.apiHandle = (void *) handle;\r
- }\r
-\r
- // Create the ASIO internal buffers. Since RtAudio sets up input\r
- // and output separately, we'll have to dispose of previously\r
- // created output buffers for a duplex stream.\r
- long inputLatency, outputLatency;\r
- if ( mode == INPUT && stream_.mode == OUTPUT ) {\r
- ASIODisposeBuffers();\r
- if ( handle->bufferInfos ) free( handle->bufferInfos );\r
- }\r
-\r
- // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.\r
- bool buffersAllocated = false;\r
- unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];\r
- handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );\r
- if ( handle->bufferInfos == NULL ) {\r
- errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";\r
- errorText_ = errorStream_.str();\r
- goto error;\r
- }\r
-\r
- ASIOBufferInfo *infos;\r
- infos = handle->bufferInfos;\r
- for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {\r
- infos->isInput = ASIOFalse;\r
- infos->channelNum = i + stream_.channelOffset[0];\r
- infos->buffers[0] = infos->buffers[1] = 0;\r
- }\r
- for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {\r
- infos->isInput = ASIOTrue;\r
- infos->channelNum = i + stream_.channelOffset[1];\r
- infos->buffers[0] = infos->buffers[1] = 0;\r
- }\r
-\r
- // Set up the ASIO callback structure and create the ASIO data buffers.\r
- asioCallbacks.bufferSwitch = &bufferSwitch;\r
- asioCallbacks.sampleRateDidChange = &sampleRateChanged;\r
- asioCallbacks.asioMessage = &asioMessages;\r
- asioCallbacks.bufferSwitchTimeInfo = NULL;\r
- result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );\r
- if ( result != ASE_OK ) {\r
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";\r
- errorText_ = errorStream_.str();\r
- goto error;\r
- }\r
- buffersAllocated = true;\r
-\r
- // Set flags for buffer conversion.\r
- stream_.doConvertBuffer[mode] = false;\r
- if ( stream_.userFormat != stream_.deviceFormat[mode] )\r
- stream_.doConvertBuffer[mode] = true;\r
- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&\r
- stream_.nUserChannels[mode] > 1 )\r
- stream_.doConvertBuffer[mode] = true;\r
-\r
- // Allocate necessary internal buffers\r
- unsigned long bufferBytes;\r
- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );\r
- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );\r
- if ( stream_.userBuffer[mode] == NULL ) {\r
- errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";\r
- goto error;\r
- }\r
-\r
- if ( stream_.doConvertBuffer[mode] ) {\r
-\r
- bool makeBuffer = true;\r
- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );\r
- if ( mode == INPUT ) {\r
- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {\r
- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );\r
- if ( bufferBytes <= bytesOut ) makeBuffer = false;\r
- }\r
- }\r
-\r
- if ( makeBuffer ) {\r
- bufferBytes *= *bufferSize;\r
- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );\r
- if ( stream_.deviceBuffer == NULL ) {\r
- errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";\r
- goto error;\r
- }\r
- }\r
- }\r
-\r
- stream_.sampleRate = sampleRate;\r
- stream_.device[mode] = device;\r
- stream_.state = STREAM_STOPPED;\r
- asioCallbackInfo = &stream_.callbackInfo;\r
- stream_.callbackInfo.object = (void *) this;\r
- if ( stream_.mode == OUTPUT && mode == INPUT )\r
- // We had already set up an output stream.\r
- stream_.mode = DUPLEX;\r
- else\r
- stream_.mode = mode;\r
-\r
- // Determine device latencies\r
- result = ASIOGetLatencies( &inputLatency, &outputLatency );\r
- if ( result != ASE_OK ) {\r
- errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING); // warn but don't fail\r
- }\r
- else {\r
- stream_.latency[0] = outputLatency;\r
- stream_.latency[1] = inputLatency;\r
- }\r
-\r
- // Setup the buffer conversion information structure. We don't use\r
- // buffers to do channel offsets, so we override that parameter\r
- // here.\r
- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );\r
-\r
- return SUCCESS;\r
-\r
- error:\r
- if ( buffersAllocated )\r
- ASIODisposeBuffers();\r
- drivers.removeCurrentDriver();\r
-\r
- if ( handle ) {\r
- CloseHandle( handle->condition );\r
- if ( handle->bufferInfos )\r
- free( handle->bufferInfos );\r
- delete handle;\r
- stream_.apiHandle = 0;\r
- }\r
-\r
- for ( int i=0; i<2; i++ ) {\r
- if ( stream_.userBuffer[i] ) {\r
- free( stream_.userBuffer[i] );\r
- stream_.userBuffer[i] = 0;\r
- }\r
- }\r
-\r
- if ( stream_.deviceBuffer ) {\r
- free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = 0;\r
- }\r
-\r
- return FAILURE;\r
-}\r
-\r
-void RtApiAsio :: closeStream()\r
-{\r
- if ( stream_.state == STREAM_CLOSED ) {\r
- errorText_ = "RtApiAsio::closeStream(): no open stream to close!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- if ( stream_.state == STREAM_RUNNING ) {\r
- stream_.state = STREAM_STOPPED;\r
- ASIOStop();\r
- }\r
- ASIODisposeBuffers();\r
- drivers.removeCurrentDriver();\r
-\r
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;\r
- if ( handle ) {\r
- CloseHandle( handle->condition );\r
- if ( handle->bufferInfos )\r
- free( handle->bufferInfos );\r
- delete handle;\r
- stream_.apiHandle = 0;\r
- }\r
-\r
- for ( int i=0; i<2; i++ ) {\r
- if ( stream_.userBuffer[i] ) {\r
- free( stream_.userBuffer[i] );\r
- stream_.userBuffer[i] = 0;\r
- }\r
- }\r
-\r
- if ( stream_.deviceBuffer ) {\r
- free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = 0;\r
- }\r
-\r
- stream_.mode = UNINITIALIZED;\r
- stream_.state = STREAM_CLOSED;\r
-}\r
-\r
-bool stopThreadCalled = false;\r
-\r
-void RtApiAsio :: startStream()\r
-{\r
- verifyStream();\r
- if ( stream_.state == STREAM_RUNNING ) {\r
- errorText_ = "RtApiAsio::startStream(): the stream is already running!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;\r
- ASIOError result = ASIOStart();\r
- if ( result != ASE_OK ) {\r
- errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
-\r
- handle->drainCounter = 0;\r
- handle->internalDrain = false;\r
- ResetEvent( handle->condition );\r
- stream_.state = STREAM_RUNNING;\r
- asioXRun = false;\r
-\r
- unlock:\r
- stopThreadCalled = false;\r
-\r
- if ( result == ASE_OK ) return;\r
- error( RtAudioError::SYSTEM_ERROR );\r
-}\r
-\r
-void RtApiAsio :: stopStream()\r
-{\r
- verifyStream();\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
- if ( handle->drainCounter == 0 ) {\r
- handle->drainCounter = 2;\r
- WaitForSingleObject( handle->condition, INFINITE ); // block until signaled\r
- }\r
- }\r
-\r
- stream_.state = STREAM_STOPPED;\r
-\r
- ASIOError result = ASIOStop();\r
- if ( result != ASE_OK ) {\r
- errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";\r
- errorText_ = errorStream_.str();\r
- }\r
-\r
- if ( result == ASE_OK ) return;\r
- error( RtAudioError::SYSTEM_ERROR );\r
-}\r
-\r
-void RtApiAsio :: abortStream()\r
-{\r
- verifyStream();\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- // The following lines were commented-out because some behavior was\r
- // noted where the device buffers need to be zeroed to avoid\r
- // continuing sound, even when the device buffers are completely\r
- // disposed. So now, calling abort is the same as calling stop.\r
- // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;\r
- // handle->drainCounter = 2;\r
- stopStream();\r
-}\r
-\r
-// This function will be called by a spawned thread when the user\r
-// callback function signals that the stream should be stopped or\r
-// aborted. It is necessary to handle it this way because the\r
-// callbackEvent() function must return before the ASIOStop()\r
-// function will return.\r
-static unsigned __stdcall asioStopStream( void *ptr )\r
-{\r
- CallbackInfo *info = (CallbackInfo *) ptr;\r
- RtApiAsio *object = (RtApiAsio *) info->object;\r
-\r
- object->stopStream();\r
- _endthreadex( 0 );\r
- return 0;\r
-}\r
-\r
-bool RtApiAsio :: callbackEvent( long bufferIndex )\r
-{\r
- if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;\r
- if ( stream_.state == STREAM_CLOSED ) {\r
- errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";\r
- error( RtAudioError::WARNING );\r
- return FAILURE;\r
- }\r
-\r
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;\r
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;\r
-\r
- // Check if we were draining the stream and signal if finished.\r
- if ( handle->drainCounter > 3 ) {\r
-\r
- stream_.state = STREAM_STOPPING;\r
- if ( handle->internalDrain == false )\r
- SetEvent( handle->condition );\r
- else { // spawn a thread to stop the stream\r
- unsigned threadId;\r
- stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,\r
- &stream_.callbackInfo, 0, &threadId );\r
- }\r
- return SUCCESS;\r
- }\r
-\r
- // Invoke user callback to get fresh output data UNLESS we are\r
- // draining stream.\r
- if ( handle->drainCounter == 0 ) {\r
- RtAudioCallback callback = (RtAudioCallback) info->callback;\r
- double streamTime = getStreamTime();\r
- RtAudioStreamStatus status = 0;\r
- if ( stream_.mode != INPUT && asioXRun == true ) {\r
- status |= RTAUDIO_OUTPUT_UNDERFLOW;\r
- asioXRun = false;\r
- }\r
- if ( stream_.mode != OUTPUT && asioXRun == true ) {\r
- status |= RTAUDIO_INPUT_OVERFLOW;\r
- asioXRun = false;\r
- }\r
- int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
- stream_.bufferSize, streamTime, status, info->userData );\r
- if ( cbReturnValue == 2 ) {\r
- stream_.state = STREAM_STOPPING;\r
- handle->drainCounter = 2;\r
- unsigned threadId;\r
- stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,\r
- &stream_.callbackInfo, 0, &threadId );\r
- return SUCCESS;\r
- }\r
- else if ( cbReturnValue == 1 ) {\r
- handle->drainCounter = 1;\r
- handle->internalDrain = true;\r
- }\r
- }\r
-\r
- unsigned int nChannels, bufferBytes, i, j;\r
- nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
-\r
- bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );\r
-\r
- if ( handle->drainCounter > 1 ) { // write zeros to the output stream\r
-\r
- for ( i=0, j=0; i<nChannels; i++ ) {\r
- if ( handle->bufferInfos[i].isInput != ASIOTrue )\r
- memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );\r
- }\r
-\r
- }\r
- else if ( stream_.doConvertBuffer[0] ) {\r
-\r
- convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );\r
- if ( stream_.doByteSwap[0] )\r
- byteSwapBuffer( stream_.deviceBuffer,\r
- stream_.bufferSize * stream_.nDeviceChannels[0],\r
- stream_.deviceFormat[0] );\r
-\r
- for ( i=0, j=0; i<nChannels; i++ ) {\r
- if ( handle->bufferInfos[i].isInput != ASIOTrue )\r
- memcpy( handle->bufferInfos[i].buffers[bufferIndex],\r
- &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );\r
- }\r
-\r
- }\r
- else {\r
-\r
- if ( stream_.doByteSwap[0] )\r
- byteSwapBuffer( stream_.userBuffer[0],\r
- stream_.bufferSize * stream_.nUserChannels[0],\r
- stream_.userFormat );\r
-\r
- for ( i=0, j=0; i<nChannels; i++ ) {\r
- if ( handle->bufferInfos[i].isInput != ASIOTrue )\r
- memcpy( handle->bufferInfos[i].buffers[bufferIndex],\r
- &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );\r
- }\r
-\r
- }\r
-\r
- if ( handle->drainCounter ) {\r
- handle->drainCounter++;\r
- goto unlock;\r
- }\r
- }\r
-\r
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {\r
-\r
- bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);\r
-\r
- if (stream_.doConvertBuffer[1]) {\r
-\r
- // Always interleave ASIO input data.\r
- for ( i=0, j=0; i<nChannels; i++ ) {\r
- if ( handle->bufferInfos[i].isInput == ASIOTrue )\r
- memcpy( &stream_.deviceBuffer[j++*bufferBytes],\r
- handle->bufferInfos[i].buffers[bufferIndex],\r
- bufferBytes );\r
- }\r
-\r
- if ( stream_.doByteSwap[1] )\r
- byteSwapBuffer( stream_.deviceBuffer,\r
- stream_.bufferSize * stream_.nDeviceChannels[1],\r
- stream_.deviceFormat[1] );\r
- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );\r
-\r
- }\r
- else {\r
- for ( i=0, j=0; i<nChannels; i++ ) {\r
- if ( handle->bufferInfos[i].isInput == ASIOTrue ) {\r
- memcpy( &stream_.userBuffer[1][bufferBytes*j++],\r
- handle->bufferInfos[i].buffers[bufferIndex],\r
- bufferBytes );\r
- }\r
- }\r
-\r
- if ( stream_.doByteSwap[1] )\r
- byteSwapBuffer( stream_.userBuffer[1],\r
- stream_.bufferSize * stream_.nUserChannels[1],\r
- stream_.userFormat );\r
- }\r
- }\r
-\r
- unlock:\r
- // The following call was suggested by Malte Clasen. While the API\r
- // documentation indicates it should not be required, some device\r
- // drivers apparently do not function correctly without it.\r
- ASIOOutputReady();\r
-\r
- RtApi::tickStreamTime();\r
- return SUCCESS;\r
-}\r
-\r
-static void sampleRateChanged( ASIOSampleRate sRate )\r
-{\r
- // The ASIO documentation says that this usually only happens during\r
- // external sync. Audio processing is not stopped by the driver,\r
- // actual sample rate might not have even changed, maybe only the\r
- // sample rate status of an AES/EBU or S/PDIF digital input at the\r
- // audio device.\r
-\r
- RtApi *object = (RtApi *) asioCallbackInfo->object;\r
- try {\r
- object->stopStream();\r
- }\r
- catch ( RtAudioError &exception ) {\r
- std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;\r
- return;\r
- }\r
-\r
- std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;\r
-}\r
-\r
-static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ )\r
-{\r
- long ret = 0;\r
-\r
- switch( selector ) {\r
- case kAsioSelectorSupported:\r
- if ( value == kAsioResetRequest\r
- || value == kAsioEngineVersion\r
- || value == kAsioResyncRequest\r
- || value == kAsioLatenciesChanged\r
- // The following three were added for ASIO 2.0, you don't\r
- // necessarily have to support them.\r
- || value == kAsioSupportsTimeInfo\r
- || value == kAsioSupportsTimeCode\r
- || value == kAsioSupportsInputMonitor)\r
- ret = 1L;\r
- break;\r
- case kAsioResetRequest:\r
- // Defer the task and perform the reset of the driver during the\r
- // next "safe" situation. You cannot reset the driver right now,\r
- // as this code is called from the driver. Reset the driver is\r
- // done by completely destruct is. I.e. ASIOStop(),\r
- // ASIODisposeBuffers(), Destruction Afterwards you initialize the\r
- // driver again.\r
- std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;\r
- ret = 1L;\r
- break;\r
- case kAsioResyncRequest:\r
- // This informs the application that the driver encountered some\r
- // non-fatal data loss. It is used for synchronization purposes\r
- // of different media. Added mainly to work around the Win16Mutex\r
- // problems in Windows 95/98 with the Windows Multimedia system,\r
- // which could lose data because the Mutex was held too long by\r
- // another thread. However a driver can issue it in other\r
- // situations, too.\r
- // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;\r
- asioXRun = true;\r
- ret = 1L;\r
- break;\r
- case kAsioLatenciesChanged:\r
- // This will inform the host application that the drivers were\r
- // latencies changed. Beware, it this does not mean that the\r
- // buffer sizes have changed! You might need to update internal\r
- // delay data.\r
- std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;\r
- ret = 1L;\r
- break;\r
- case kAsioEngineVersion:\r
- // Return the supported ASIO version of the host application. If\r
- // a host application does not implement this selector, ASIO 1.0\r
- // is assumed by the driver.\r
- ret = 2L;\r
- break;\r
- case kAsioSupportsTimeInfo:\r
- // Informs the driver whether the\r
- // asioCallbacks.bufferSwitchTimeInfo() callback is supported.\r
- // For compatibility with ASIO 1.0 drivers the host application\r
- // should always support the "old" bufferSwitch method, too.\r
- ret = 0;\r
- break;\r
- case kAsioSupportsTimeCode:\r
- // Informs the driver whether application is interested in time\r
- // code info. If an application does not need to know about time\r
- // code, the driver has less work to do.\r
- ret = 0;\r
- break;\r
- }\r
- return ret;\r
-}\r
-\r
-static const char* getAsioErrorString( ASIOError result )\r
-{\r
- struct Messages \r
- {\r
- ASIOError value;\r
- const char*message;\r
- };\r
-\r
- static const Messages m[] = \r
- {\r
- { ASE_NotPresent, "Hardware input or output is not present or available." },\r
- { ASE_HWMalfunction, "Hardware is malfunctioning." },\r
- { ASE_InvalidParameter, "Invalid input parameter." },\r
- { ASE_InvalidMode, "Invalid mode." },\r
- { ASE_SPNotAdvancing, "Sample position not advancing." },\r
- { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },\r
- { ASE_NoMemory, "Not enough memory to complete the request." }\r
- };\r
-\r
- for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )\r
- if ( m[i].value == result ) return m[i].message;\r
-\r
- return "Unknown error.";\r
-}\r
-\r
-//******************** End of __WINDOWS_ASIO__ *********************//\r
-#endif\r
-\r
-\r
-#if defined(__WINDOWS_WASAPI__) // Windows WASAPI API\r
-\r
-#include <audioclient.h>\r
-#include <avrt.h>\r
-#include <functiondiscoverykeys.h>\r
-#include <mmdeviceapi.h>\r
-\r
-//=============================================================================\r
-\r
-#define EXIT_ON_ERROR( hr, errorType, errorText )\\r
-if ( FAILED( hr ) )\\r
-{\\r
- errorText_ = __FUNCTION__ ": " errorText;\\r
- error( errorType );\\r
- goto Exit;\\r
-}\r
-\r
-#define SAFE_RELEASE( objectPtr )\\r
-if ( objectPtr )\\r
-{\\r
- objectPtr->Release();\\r
- objectPtr = NULL;\\r
-}\r
-\r
-typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );\r
-\r
-//-----------------------------------------------------------------------------\r
-\r
-// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.\r
-// Therefore we must perform all necessary conversions to user buffers in order to satisfy these\r
-// requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to\r
-// provide intermediate storage for read / write synchronization.\r
-class WasapiBuffer\r
-{\r
-public:\r
- WasapiBuffer()\r
- : buffer_( NULL ),\r
- bufferSize_( 0 ),\r
- inIndex_( 0 ),\r
- outIndex_( 0 ) {}\r
-\r
- ~WasapiBuffer() {\r
- delete buffer_;\r
- }\r
-\r
- // sets the length of the internal ring buffer\r
- void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {\r
- delete buffer_;\r
-\r
- buffer_ = ( char* ) calloc( bufferSize, formatBytes );\r
-\r
- bufferSize_ = bufferSize;\r
- inIndex_ = 0;\r
- outIndex_ = 0;\r
- }\r
-\r
- // attempt to push a buffer into the ring buffer at the current "in" index\r
- bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )\r
- {\r
- if ( !buffer || // incoming buffer is NULL\r
- bufferSize == 0 || // incoming buffer has no data\r
- bufferSize > bufferSize_ ) // incoming buffer too large\r
- {\r
- return false;\r
- }\r
-\r
- unsigned int relOutIndex = outIndex_;\r
- unsigned int inIndexEnd = inIndex_ + bufferSize;\r
- if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {\r
- relOutIndex += bufferSize_;\r
- }\r
-\r
- // "in" index can end on the "out" index but cannot begin at it\r
- if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {\r
- return false; // not enough space between "in" index and "out" index\r
- }\r
-\r
- // copy buffer from external to internal\r
- int fromZeroSize = inIndex_ + bufferSize - bufferSize_;\r
- fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;\r
- int fromInSize = bufferSize - fromZeroSize;\r
-\r
- switch( format )\r
- {\r
- case RTAUDIO_SINT8:\r
- memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );\r
- memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );\r
- break;\r
- case RTAUDIO_SINT16:\r
- memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );\r
- memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );\r
- break;\r
- case RTAUDIO_SINT24:\r
- memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );\r
- memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );\r
- break;\r
- case RTAUDIO_SINT32:\r
- memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );\r
- memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );\r
- break;\r
- case RTAUDIO_FLOAT32:\r
- memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );\r
- memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );\r
- break;\r
- case RTAUDIO_FLOAT64:\r
- memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );\r
- memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );\r
- break;\r
- }\r
-\r
- // update "in" index\r
- inIndex_ += bufferSize;\r
- inIndex_ %= bufferSize_;\r
-\r
- return true;\r
- }\r
-\r
- // attempt to pull a buffer from the ring buffer from the current "out" index\r
- bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )\r
- {\r
- if ( !buffer || // incoming buffer is NULL\r
- bufferSize == 0 || // incoming buffer has no data\r
- bufferSize > bufferSize_ ) // incoming buffer too large\r
- {\r
- return false;\r
- }\r
-\r
- unsigned int relInIndex = inIndex_;\r
- unsigned int outIndexEnd = outIndex_ + bufferSize;\r
- if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {\r
- relInIndex += bufferSize_;\r
- }\r
-\r
- // "out" index can begin at and end on the "in" index\r
- if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {\r
- return false; // not enough space between "out" index and "in" index\r
- }\r
-\r
- // copy buffer from internal to external\r
- int fromZeroSize = outIndex_ + bufferSize - bufferSize_;\r
- fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;\r
- int fromOutSize = bufferSize - fromZeroSize;\r
-\r
- switch( format )\r
- {\r
- case RTAUDIO_SINT8:\r
- memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );\r
- memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );\r
- break;\r
- case RTAUDIO_SINT16:\r
- memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );\r
- memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );\r
- break;\r
- case RTAUDIO_SINT24:\r
- memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );\r
- memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );\r
- break;\r
- case RTAUDIO_SINT32:\r
- memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );\r
- memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );\r
- break;\r
- case RTAUDIO_FLOAT32:\r
- memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );\r
- memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );\r
- break;\r
- case RTAUDIO_FLOAT64:\r
- memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );\r
- memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );\r
- break;\r
- }\r
-\r
- // update "out" index\r
- outIndex_ += bufferSize;\r
- outIndex_ %= bufferSize_;\r
-\r
- return true;\r
- }\r
-\r
-private:\r
- char* buffer_;\r
- unsigned int bufferSize_;\r
- unsigned int inIndex_;\r
- unsigned int outIndex_;\r
-};\r
-\r
-//-----------------------------------------------------------------------------\r
-\r
-// In order to satisfy WASAPI's buffer requirements, we need a means of converting sample rate and\r
-// channel counts between HW and the user. The convertBufferWasapi function is used to perform\r
-// these conversions between HwIn->UserIn and UserOut->HwOut during the stream callback loop.\r
-// This sample rate converter favors speed over quality, and works best with conversions between\r
-// one rate and its multiple. RtApiWasapi will not populate a device's sample rate list with rates\r
-// that may cause artifacts via this conversion.\r
-void convertBufferWasapi( char* outBuffer,\r
- const char* inBuffer,\r
- const unsigned int& inChannelCount,\r
- const unsigned int& outChannelCount,\r
- const unsigned int& inSampleRate,\r
- const unsigned int& outSampleRate,\r
- const unsigned int& inSampleCount,\r
- unsigned int& outSampleCount,\r
- const RtAudioFormat& format )\r
-{\r
- // calculate the new outSampleCount and relative sampleStep\r
- float sampleRatio = ( float ) outSampleRate / inSampleRate;\r
- float sampleStep = 1.0f / sampleRatio;\r
- float inSampleFraction = 0.0f;\r
- unsigned int commonChannelCount = min( inChannelCount, outChannelCount );\r
-\r
- outSampleCount = ( unsigned int ) ( inSampleCount * sampleRatio );\r
-\r
- // frame-by-frame, copy each relative input sample into it's corresponding output sample\r
- for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ )\r
- {\r
- unsigned int inSample = ( unsigned int ) inSampleFraction;\r
-\r
- switch ( format )\r
- {\r
- case RTAUDIO_SINT8:\r
- memcpy( &( ( char* ) outBuffer )[ outSample * outChannelCount ], &( ( char* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( char ) );\r
- break;\r
- case RTAUDIO_SINT16:\r
- memcpy( &( ( short* ) outBuffer )[ outSample * outChannelCount ], &( ( short* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( short ) );\r
- break;\r
- case RTAUDIO_SINT24:\r
- memcpy( &( ( S24* ) outBuffer )[ outSample * outChannelCount ], &( ( S24* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( S24 ) );\r
- break;\r
- case RTAUDIO_SINT32:\r
- memcpy( &( ( int* ) outBuffer )[ outSample * outChannelCount ], &( ( int* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( int ) );\r
- break;\r
- case RTAUDIO_FLOAT32:\r
- memcpy( &( ( float* ) outBuffer )[ outSample * outChannelCount ], &( ( float* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( float ) );\r
- break;\r
- case RTAUDIO_FLOAT64:\r
- memcpy( &( ( double* ) outBuffer )[ outSample * outChannelCount ], &( ( double* ) inBuffer )[ inSample * inChannelCount ], commonChannelCount * sizeof( double ) );\r
- break;\r
- }\r
-\r
- // jump to next in sample\r
- inSampleFraction += sampleStep;\r
- }\r
-}\r
-\r
-//-----------------------------------------------------------------------------\r
-\r
-// A structure to hold various information related to the WASAPI implementation.\r
-struct WasapiHandle\r
-{\r
- IAudioClient* captureAudioClient;\r
- IAudioClient* renderAudioClient;\r
- IAudioCaptureClient* captureClient;\r
- IAudioRenderClient* renderClient;\r
- HANDLE captureEvent;\r
- HANDLE renderEvent;\r
-\r
- WasapiHandle()\r
- : captureAudioClient( NULL ),\r
- renderAudioClient( NULL ),\r
- captureClient( NULL ),\r
- renderClient( NULL ),\r
- captureEvent( NULL ),\r
- renderEvent( NULL ) {}\r
-};\r
-\r
-//=============================================================================\r
-\r
-RtApiWasapi::RtApiWasapi()\r
- : coInitialized_( false ), deviceEnumerator_( NULL )\r
-{\r
- // WASAPI can run either apartment or multi-threaded\r
- HRESULT hr = CoInitialize( NULL );\r
-\r
- if ( !FAILED( hr ) )\r
- coInitialized_ = true;\r
-\r
- // instantiate device enumerator\r
- hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,\r
- CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),\r
- ( void** ) &deviceEnumerator_ );\r
-\r
- if ( FAILED( hr ) ) {\r
- errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator";\r
- error( RtAudioError::DRIVER_ERROR );\r
- }\r
-}\r
-\r
-//-----------------------------------------------------------------------------\r
-\r
-RtApiWasapi::~RtApiWasapi()\r
-{\r
- // if this object previously called CoInitialize()\r
- if ( coInitialized_ ) {\r
- CoUninitialize();\r
- }\r
-\r
- if ( stream_.state != STREAM_CLOSED ) {\r
- closeStream();\r
- }\r
-\r
- SAFE_RELEASE( deviceEnumerator_ );\r
-}\r
-\r
-//=============================================================================\r
-\r
-unsigned int RtApiWasapi::getDeviceCount( void )\r
-{\r
- unsigned int captureDeviceCount = 0;\r
- unsigned int renderDeviceCount = 0;\r
-\r
- IMMDeviceCollection* captureDevices = NULL;\r
- IMMDeviceCollection* renderDevices = NULL;\r
-\r
- // count capture devices\r
- HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture device collection" );\r
-\r
- hr = captureDevices->GetCount( &captureDeviceCount );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture device count" );\r
-\r
- // count render devices\r
- hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render device collection" );\r
-\r
- hr = renderDevices->GetCount( &renderDeviceCount );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render device count" );\r
-\r
-Exit:\r
- // release all references\r
- SAFE_RELEASE( captureDevices );\r
- SAFE_RELEASE( renderDevices );\r
-\r
- return captureDeviceCount + renderDeviceCount;\r
-}\r
-\r
-//-----------------------------------------------------------------------------\r
-\r
-RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )\r
-{\r
- RtAudio::DeviceInfo info;\r
- unsigned int captureDeviceCount = 0;\r
- unsigned int renderDeviceCount = 0;\r
- std::wstring deviceName;\r
- std::string defaultDeviceName;\r
- bool isCaptureDevice = false;\r
-\r
- PROPVARIANT deviceNameProp;\r
- PROPVARIANT defaultDeviceNameProp;\r
-\r
- IMMDeviceCollection* captureDevices = NULL;\r
- IMMDeviceCollection* renderDevices = NULL;\r
- IMMDevice* devicePtr = NULL;\r
- IMMDevice* defaultDevicePtr = NULL;\r
- IAudioClient* audioClient = NULL;\r
- IPropertyStore* devicePropStore = NULL;\r
- IPropertyStore* defaultDevicePropStore = NULL;\r
-\r
- WAVEFORMATEX* deviceFormat = NULL;\r
- WAVEFORMATEX* closestMatchFormat = NULL;\r
-\r
- // probed\r
- info.probed = false;\r
-\r
- // count capture devices\r
- HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture device collection" );\r
-\r
- hr = captureDevices->GetCount( &captureDeviceCount );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture device count" );\r
-\r
- // count render devices\r
- hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render device collection" );\r
-\r
- hr = renderDevices->GetCount( &renderDeviceCount );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render device count" );\r
-\r
- // validate device index\r
- if ( device >= captureDeviceCount + renderDeviceCount )\r
- EXIT_ON_ERROR( -1, RtAudioError::INVALID_USE, "Invalid device index" );\r
-\r
- // determine whether index falls within capture or render devices\r
- //if ( device < captureDeviceCount ) {\r
- if ( device >= renderDeviceCount ) {\r
- //hr = captureDevices->Item( device, &devicePtr );\r
- hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture device handle" );\r
-\r
- isCaptureDevice = true;\r
- }\r
- else {\r
- //hr = renderDevices->Item( device - captureDeviceCount, &devicePtr );\r
- hr = renderDevices->Item( device, &devicePtr );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render device handle" );\r
-\r
- isCaptureDevice = false;\r
- }\r
-\r
- // get default device name\r
- if ( isCaptureDevice ) {\r
- hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve default render device handle" );\r
- }\r
- else {\r
- hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve default capture device handle" );\r
- }\r
-\r
- hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to open default device property store" );\r
-\r
- PropVariantInit( &defaultDeviceNameProp );\r
-\r
- hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve default device property: PKEY_Device_FriendlyName" );\r
-\r
- deviceName = defaultDeviceNameProp.pwszVal;\r
- defaultDeviceName = std::string( deviceName.begin(), deviceName.end() );\r
-\r
- // name\r
- hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to open device property store" );\r
-\r
- PropVariantInit( &deviceNameProp );\r
-\r
- hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve device property: PKEY_Device_FriendlyName" );\r
-\r
- deviceName = deviceNameProp.pwszVal;\r
- info.name = std::string( deviceName.begin(), deviceName.end() );\r
-\r
- // is default\r
- if ( isCaptureDevice ) {\r
- info.isDefaultInput = info.name == defaultDeviceName;\r
- info.isDefaultOutput = false;\r
- }\r
- else {\r
- info.isDefaultInput = false;\r
- info.isDefaultOutput = info.name == defaultDeviceName;\r
- }\r
-\r
- // channel count\r
- hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve device audio client" );\r
-\r
- hr = audioClient->GetMixFormat( &deviceFormat );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve device mix format" );\r
-\r
- if ( isCaptureDevice ) {\r
- info.inputChannels = deviceFormat->nChannels;\r
- info.outputChannels = 0;\r
- info.duplexChannels = 0;\r
- }\r
- else {\r
- info.inputChannels = 0;\r
- info.outputChannels = deviceFormat->nChannels;\r
- info.duplexChannels = 0;\r
- }\r
-\r
- // sample rates\r
- info.sampleRates.clear();\r
-\r
- // allow support for sample rates that are multiples of the base rate\r
- for ( unsigned int i = 0; i < MAX_SAMPLE_RATES; i++ ) {\r
- if ( SAMPLE_RATES[i] < deviceFormat->nSamplesPerSec ) {\r
- if ( deviceFormat->nSamplesPerSec % SAMPLE_RATES[i] == 0 ) {\r
- info.sampleRates.push_back( SAMPLE_RATES[i] );\r
- }\r
- }\r
- else {\r
- if ( SAMPLE_RATES[i] % deviceFormat->nSamplesPerSec == 0 ) {\r
- info.sampleRates.push_back( SAMPLE_RATES[i] );\r
- }\r
- }\r
- }\r
-\r
- // native format\r
- info.nativeFormats = 0;\r
-\r
- if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||\r
- ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&\r
- ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )\r
- {\r
- if ( deviceFormat->wBitsPerSample == 32 ) {\r
- info.nativeFormats |= RTAUDIO_FLOAT32;\r
- }\r
- else if ( deviceFormat->wBitsPerSample == 64 ) {\r
- info.nativeFormats |= RTAUDIO_FLOAT64;\r
- }\r
- }\r
- else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||\r
- ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&\r
- ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )\r
- {\r
- if ( deviceFormat->wBitsPerSample == 8 ) {\r
- info.nativeFormats |= RTAUDIO_SINT8;\r
- }\r
- else if ( deviceFormat->wBitsPerSample == 16 ) {\r
- info.nativeFormats |= RTAUDIO_SINT16;\r
- }\r
- else if ( deviceFormat->wBitsPerSample == 24 ) {\r
- info.nativeFormats |= RTAUDIO_SINT24;\r
- }\r
- else if ( deviceFormat->wBitsPerSample == 32 ) {\r
- info.nativeFormats |= RTAUDIO_SINT32;\r
- }\r
- }\r
-\r
- // probed\r
- info.probed = true;\r
-\r
-Exit:\r
- // release all references\r
- PropVariantClear( &deviceNameProp );\r
- PropVariantClear( &defaultDeviceNameProp );\r
-\r
- SAFE_RELEASE( captureDevices );\r
- SAFE_RELEASE( renderDevices );\r
- SAFE_RELEASE( devicePtr );\r
- SAFE_RELEASE( defaultDevicePtr );\r
- SAFE_RELEASE( audioClient );\r
- SAFE_RELEASE( devicePropStore );\r
- SAFE_RELEASE( defaultDevicePropStore );\r
-\r
- CoTaskMemFree( deviceFormat );\r
- CoTaskMemFree( closestMatchFormat );\r
-\r
- return info;\r
-}\r
-\r
-//-----------------------------------------------------------------------------\r
-\r
-unsigned int RtApiWasapi::getDefaultOutputDevice( void )\r
-{\r
- for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {\r
- if ( getDeviceInfo( i ).isDefaultOutput ) {\r
- return i;\r
- }\r
- }\r
-\r
- return 0;\r
-}\r
-\r
-//-----------------------------------------------------------------------------\r
-\r
-unsigned int RtApiWasapi::getDefaultInputDevice( void )\r
-{\r
- for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {\r
- if ( getDeviceInfo( i ).isDefaultInput ) {\r
- return i;\r
- }\r
- }\r
-\r
- return 0;\r
-}\r
-\r
-//-----------------------------------------------------------------------------\r
-\r
-void RtApiWasapi::closeStream( void )\r
-{\r
- if ( stream_.state == STREAM_CLOSED ) {\r
- errorText_ = "RtApiWasapi::closeStream: No open stream to close";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- if ( stream_.state != STREAM_STOPPED )\r
- stopStream();\r
-\r
- // clean up stream memory\r
- SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )\r
- SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )\r
-\r
- SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )\r
- SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )\r
-\r
- if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )\r
- CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );\r
-\r
- if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )\r
- CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );\r
-\r
- delete stream_.apiHandle;\r
- stream_.apiHandle = NULL;\r
-\r
- for ( int i = 0; i < 2; i++ ) {\r
- if ( stream_.userBuffer[i] ) {\r
- free( stream_.userBuffer[i] );\r
- stream_.userBuffer[i] = 0;\r
- }\r
- }\r
-\r
- if ( stream_.deviceBuffer ) {\r
- free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = 0;\r
- }\r
-\r
- // update stream state\r
- stream_.state = STREAM_CLOSED;\r
-}\r
-\r
-//-----------------------------------------------------------------------------\r
-\r
-void RtApiWasapi::startStream( void )\r
-{\r
- verifyStream();\r
-\r
- if ( stream_.state == STREAM_RUNNING ) {\r
- errorText_ = "RtApiWasapi::startStream: The stream is already running";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- // update stream state\r
- stream_.state = STREAM_RUNNING;\r
-\r
- // create WASAPI stream thread\r
- stream_.callbackInfo.thread = ( unsigned int ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );\r
-\r
- if ( !stream_.callbackInfo.thread ) {\r
- errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread";\r
- error( RtAudioError::THREAD_ERROR );\r
- }\r
- else {\r
- SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );\r
- ResumeThread( ( void* ) stream_.callbackInfo.thread );\r
- }\r
-}\r
-\r
-//-----------------------------------------------------------------------------\r
-\r
-void RtApiWasapi::stopStream( void )\r
-{\r
- verifyStream();\r
-\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- errorText_ = "RtApiWasapi::stopStream: The stream is already stopped";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- // inform stream thread by setting stream state to STREAM_STOPPING\r
- stream_.state = STREAM_STOPPING;\r
-\r
- // wait until stream thread is stopped\r
- while( stream_.state != STREAM_STOPPED ) {\r
- Sleep( 1 );\r
- }\r
-\r
- // Wait for the last buffer to play before stopping.\r
- Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );\r
-\r
- // stop capture client if applicable\r
- if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {\r
- HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();\r
- if ( FAILED( hr ) ) {\r
- errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream";\r
- error( RtAudioError::DRIVER_ERROR );\r
- }\r
- }\r
-\r
- // stop render client if applicable\r
- if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {\r
- HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();\r
- if ( FAILED( hr ) ) {\r
- errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream";\r
- error( RtAudioError::DRIVER_ERROR );\r
- }\r
- }\r
-\r
- // close thread handle\r
- if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {\r
- errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread";\r
- error( RtAudioError::THREAD_ERROR );\r
- }\r
-\r
- stream_.callbackInfo.thread = NULL;\r
-}\r
-\r
-//-----------------------------------------------------------------------------\r
-\r
-void RtApiWasapi::abortStream( void )\r
-{\r
- verifyStream();\r
-\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- errorText_ = "RtApiWasapi::abortStream: The stream is already stopped";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- // inform stream thread by setting stream state to STREAM_STOPPING\r
- stream_.state = STREAM_STOPPING;\r
-\r
- // wait until stream thread is stopped\r
- while ( stream_.state != STREAM_STOPPED ) {\r
- Sleep( 1 );\r
- }\r
-\r
- // stop capture client if applicable\r
- if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {\r
- HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();\r
- if ( FAILED( hr ) ) {\r
- errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream";\r
- error( RtAudioError::DRIVER_ERROR );\r
- }\r
- }\r
-\r
- // stop render client if applicable\r
- if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {\r
- HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();\r
- if ( FAILED( hr ) ) {\r
- errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream";\r
- error( RtAudioError::DRIVER_ERROR );\r
- }\r
- }\r
-\r
- // close thread handle\r
- if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {\r
- errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread";\r
- error( RtAudioError::THREAD_ERROR );\r
- }\r
-\r
- stream_.callbackInfo.thread = NULL;\r
-}\r
-\r
-//-----------------------------------------------------------------------------\r
-\r
-bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,\r
- unsigned int firstChannel, unsigned int sampleRate,\r
- RtAudioFormat format, unsigned int* bufferSize,\r
- RtAudio::StreamOptions* options )\r
-{\r
- bool methodResult = FAILURE;\r
- unsigned int captureDeviceCount = 0;\r
- unsigned int renderDeviceCount = 0;\r
-\r
- IMMDeviceCollection* captureDevices = NULL;\r
- IMMDeviceCollection* renderDevices = NULL;\r
- IMMDevice* devicePtr = NULL;\r
- WAVEFORMATEX* deviceFormat = NULL;\r
-\r
- // create API Handle if not already created\r
- if ( !stream_.apiHandle )\r
- stream_.apiHandle = ( void* ) new WasapiHandle();\r
-\r
- // count capture devices\r
- HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture device collection" );\r
-\r
- hr = captureDevices->GetCount( &captureDeviceCount );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture device count" );\r
-\r
- // count render devices\r
- hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render device collection" );\r
-\r
- hr = renderDevices->GetCount( &renderDeviceCount );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render device count" );\r
-\r
- // validate device index\r
- if ( device >= captureDeviceCount + renderDeviceCount )\r
- EXIT_ON_ERROR( -1, RtAudioError::INVALID_USE, "Invalid device index" );\r
-\r
- // determine whether index falls within capture or render devices\r
- //if ( device < captureDeviceCount ) {\r
- if ( device >= renderDeviceCount ) {\r
- if ( mode != INPUT )\r
- EXIT_ON_ERROR( -1, RtAudioError::INVALID_USE, "Capture device selected as output device" );\r
-\r
- // retrieve captureAudioClient from devicePtr\r
- IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;\r
-\r
- //hr = captureDevices->Item( device, &devicePtr );\r
- hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture device handle" );\r
-\r
- hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,\r
- NULL, ( void** ) &captureAudioClient );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve device audio client" );\r
-\r
- hr = captureAudioClient->GetMixFormat( &deviceFormat );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve device mix format" );\r
-\r
- stream_.nDeviceChannels[mode] = deviceFormat->nChannels;\r
- captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );\r
- }\r
- else {\r
- if ( mode != OUTPUT )\r
- EXIT_ON_ERROR( -1, RtAudioError::INVALID_USE, "Render device selected as input device" );\r
-\r
- // retrieve renderAudioClient from devicePtr\r
- IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;\r
-\r
- //hr = renderDevices->Item( device - captureDeviceCount, &devicePtr );\r
- hr = renderDevices->Item( device, &devicePtr );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render device handle" );\r
-\r
- hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,\r
- NULL, ( void** ) &renderAudioClient );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve device audio client" );\r
-\r
- hr = renderAudioClient->GetMixFormat( &deviceFormat );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve device mix format" );\r
-\r
- stream_.nDeviceChannels[mode] = deviceFormat->nChannels;\r
- renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );\r
- }\r
-\r
- // fill stream data\r
- if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||\r
- ( stream_.mode == INPUT && mode == OUTPUT ) ) {\r
- stream_.mode = DUPLEX;\r
- }\r
- else {\r
- stream_.mode = mode;\r
- }\r
-\r
- stream_.device[mode] = device;\r
- stream_.state = STREAM_STOPPED;\r
- stream_.doByteSwap[mode] = false;\r
- stream_.sampleRate = sampleRate;\r
- stream_.bufferSize = *bufferSize;\r
- stream_.nBuffers = 1;\r
- stream_.nUserChannels[mode] = channels;\r
- stream_.channelOffset[mode] = firstChannel;\r
- stream_.userFormat = format;\r
- stream_.deviceFormat[mode] = getDeviceInfo( device ).nativeFormats;\r
-\r
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED )\r
- stream_.userInterleaved = false;\r
- else\r
- stream_.userInterleaved = true;\r
- stream_.deviceInterleaved[mode] = true;\r
-\r
- // Set flags for buffer conversion.\r
- stream_.doConvertBuffer[mode] = false;\r
- if ( stream_.userFormat != stream_.deviceFormat[mode] )\r
- stream_.doConvertBuffer[mode] = true;\r
- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&\r
- stream_.nUserChannels[mode] > 1 )\r
- stream_.doConvertBuffer[mode] = true;\r
-\r
- if ( stream_.doConvertBuffer[mode] )\r
- setConvertInfo( mode, 0 );\r
-\r
- // Allocate necessary internal buffers\r
- unsigned int bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );\r
-\r
- stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );\r
- if ( !stream_.userBuffer[mode] )\r
- EXIT_ON_ERROR( -1, RtAudioError::MEMORY_ERROR, "Error allocating user buffer memory" );\r
-\r
- if ( stream_.doConvertBuffer[mode] && !stream_.deviceBuffer ) {\r
- unsigned int deviceBufferSize = max( stream_.nUserChannels[INPUT] * stream_.bufferSize * formatBytes( stream_.userFormat ),\r
- stream_.nUserChannels[OUTPUT] * stream_.bufferSize * formatBytes( stream_.userFormat ) );\r
-\r
- stream_.deviceBuffer = ( char* ) calloc( deviceBufferSize, 1 );\r
- if ( !stream_.deviceBuffer )\r
- EXIT_ON_ERROR( -1, RtAudioError::MEMORY_ERROR, "Error allocating device buffer memory" );\r
- }\r
-\r
- if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )\r
- stream_.callbackInfo.priority = 15;\r
- else\r
- stream_.callbackInfo.priority = 0;\r
-\r
- ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback\r
- ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode\r
-\r
- methodResult = SUCCESS;\r
-\r
-Exit:\r
- //clean up\r
-\r
- SAFE_RELEASE( captureDevices );\r
- SAFE_RELEASE( renderDevices );\r
- SAFE_RELEASE( devicePtr );\r
-\r
- CoTaskMemFree( deviceFormat );\r
-\r
- // if method failed, close the stream\r
- if ( methodResult == FAILURE )\r
- closeStream();\r
-\r
- return methodResult;\r
-}\r
-\r
-//=============================================================================\r
-\r
-DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )\r
-{\r
- if ( wasapiPtr )\r
- ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();\r
-\r
- return 0;\r
-}\r
-\r
-DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )\r
-{\r
- if ( wasapiPtr )\r
- ( ( RtApiWasapi* ) wasapiPtr )->stopStream();\r
-\r
- return 0;\r
-}\r
-\r
-DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )\r
-{\r
- if ( wasapiPtr )\r
- ( ( RtApiWasapi* ) wasapiPtr )->abortStream();\r
-\r
- return 0;\r
-}\r
-\r
-//-----------------------------------------------------------------------------\r
-\r
-void RtApiWasapi::wasapiThread()\r
-{\r
- // as this is a new thread, we must CoInitialize it\r
- CoInitialize( NULL );\r
-\r
- HRESULT hr;\r
-\r
- IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;\r
- IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;\r
- IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;\r
- IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;\r
- HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;\r
- HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;\r
-\r
- WAVEFORMATEX* captureFormat = NULL;\r
- WAVEFORMATEX* renderFormat = NULL;\r
- float captureSrRatio = 0.0f;\r
- float renderSrRatio = 0.0f;\r
- WasapiBuffer captureBuffer;\r
- WasapiBuffer renderBuffer;\r
-\r
- // Attempt to assign "Pro Audio" characteristic to thread\r
- HMODULE AvrtDll = LoadLibrary( "AVRT.dll" );\r
- if ( AvrtDll ) {\r
- DWORD taskIndex = 0;\r
- TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );\r
- AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );\r
- FreeLibrary( AvrtDll );\r
- }\r
-\r
- // start capture stream if applicable\r
- if ( captureAudioClient ) {\r
- hr = captureAudioClient->GetMixFormat( &captureFormat );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve device mix format" );\r
-\r
- captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate );\r
-\r
- // initialize capture stream according to desire buffer size\r
- float desiredBufferSize = stream_.bufferSize * captureSrRatio;\r
- REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / captureFormat->nSamplesPerSec );\r
-\r
- if ( !captureClient ) {\r
- hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,\r
- AUDCLNT_STREAMFLAGS_EVENTCALLBACK,\r
- desiredBufferPeriod,\r
- desiredBufferPeriod,\r
- captureFormat,\r
- NULL );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to initialize capture audio client" );\r
-\r
- hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),\r
- ( void** ) &captureClient );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture client handle" );\r
-\r
- // configure captureEvent to trigger on every available capture buffer\r
- captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );\r
- if ( !captureEvent )\r
- EXIT_ON_ERROR( -1, RtAudioError::SYSTEM_ERROR, "Unable to create capture event" );\r
-\r
- hr = captureAudioClient->SetEventHandle( captureEvent );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to set capture event handle" );\r
-\r
- ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;\r
- ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;\r
- }\r
-\r
- unsigned int inBufferSize = 0;\r
- hr = captureAudioClient->GetBufferSize( &inBufferSize );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to get capture buffer size" );\r
-\r
- // scale outBufferSize according to stream->user sample rate ratio\r
- unsigned int outBufferSize = ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT];\r
- inBufferSize *= stream_.nDeviceChannels[INPUT];\r
-\r
- // set captureBuffer size\r
- captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );\r
-\r
- // reset the capture stream\r
- hr = captureAudioClient->Reset();\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to reset capture stream" );\r
-\r
- // start the capture stream\r
- hr = captureAudioClient->Start();\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to start capture stream" );\r
- }\r
-\r
- // start render stream if applicable\r
- if ( renderAudioClient ) {\r
- hr = renderAudioClient->GetMixFormat( &renderFormat );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve device mix format" );\r
-\r
- renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate );\r
-\r
- // initialize render stream according to desire buffer size\r
- float desiredBufferSize = stream_.bufferSize * renderSrRatio;\r
- REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec );\r
-\r
- if ( !renderClient ) {\r
- hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,\r
- AUDCLNT_STREAMFLAGS_EVENTCALLBACK,\r
- desiredBufferPeriod,\r
- desiredBufferPeriod,\r
- renderFormat,\r
- NULL );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to initialize render audio client" );\r
-\r
- hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),\r
- ( void** ) &renderClient );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render client handle" );\r
-\r
- // configure renderEvent to trigger on every available render buffer\r
- renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );\r
- if ( !renderEvent )\r
- EXIT_ON_ERROR( -1, RtAudioError::SYSTEM_ERROR, "Unable to create render event" );\r
-\r
- hr = renderAudioClient->SetEventHandle( renderEvent );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to set render event handle" );\r
-\r
- ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;\r
- ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;\r
- }\r
-\r
- unsigned int outBufferSize = 0;\r
- hr = renderAudioClient->GetBufferSize( &outBufferSize );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to get render buffer size" );\r
-\r
- // scale inBufferSize according to user->stream sample rate ratio\r
- unsigned int inBufferSize = ( unsigned int ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT];\r
- outBufferSize *= stream_.nDeviceChannels[OUTPUT];\r
-\r
- // set renderBuffer size\r
- renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );\r
-\r
- // reset the render stream\r
- hr = renderAudioClient->Reset();\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to reset render stream" );\r
-\r
- // start the render stream\r
- hr = renderAudioClient->Start();\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to start render stream" );\r
- }\r
-\r
- // declare local stream variables\r
- RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;\r
-\r
- BYTE* streamBuffer = NULL;\r
- unsigned long captureFlags = 0;\r
-\r
- unsigned int bufferFrameCount = 0;\r
- unsigned int numFramesPadding = 0;\r
- unsigned int convBufferSize = 0;\r
-\r
- bool callbackPushed = false;\r
- bool callbackPulled = false;\r
- bool callbackStopped = false;\r
-\r
- int callbackResult = 0;\r
-\r
- // convBuffer is used to store converted buffers between WASAPI and the user\r
- char* convBuffer = NULL;\r
-\r
- if ( stream_.mode == INPUT ) {\r
- convBuffer = ( char* ) malloc( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ) );\r
- }\r
- else if ( stream_.mode == OUTPUT ) {\r
- convBuffer = ( char* ) malloc( ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );\r
- }\r
- else if ( stream_.mode == DUPLEX ) {\r
- convBuffer = ( char* ) malloc( max( ( size_t ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),\r
- ( size_t ) ( stream_.bufferSize * renderSrRatio ) * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) ) );\r
- }\r
-\r
- // stream process loop\r
- while ( stream_.state != STREAM_STOPPING ) {\r
- if ( !callbackPulled ) {\r
- // Callback Input\r
- // ==============\r
- // 1. Pull callback buffer from inputBuffer\r
- // 2. If 1. was successful: Convert callback buffer to user sample rate and channel count\r
- // Convert callback buffer to user format\r
-\r
- if ( captureAudioClient ) {\r
- // Pull callback buffer from inputBuffer\r
- callbackPulled = captureBuffer.pullBuffer( convBuffer,\r
- ( unsigned int ) ( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT],\r
- stream_.deviceFormat[INPUT] );\r
-\r
- if ( callbackPulled ) {\r
- // Convert callback buffer to user sample rate and channel count\r
- convertBufferWasapi( stream_.deviceBuffer,\r
- convBuffer,\r
- stream_.nDeviceChannels[INPUT],\r
- stream_.nUserChannels[INPUT],\r
- captureFormat->nSamplesPerSec,\r
- stream_.sampleRate,\r
- ( unsigned int ) ( stream_.bufferSize * captureSrRatio ),\r
- convBufferSize,\r
- stream_.deviceFormat[INPUT] );\r
-\r
- if ( stream_.doConvertBuffer[INPUT] ) {\r
- // Convert callback buffer to user format\r
- convertBuffer( stream_.userBuffer[INPUT],\r
- stream_.deviceBuffer,\r
- stream_.convertInfo[INPUT] );\r
- }\r
- else {\r
- // no conversion, simple copy deviceBuffer to userBuffer\r
- memcpy( stream_.userBuffer[INPUT],\r
- stream_.deviceBuffer,\r
- stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );\r
- }\r
- }\r
- }\r
- else {\r
- // if there is no capture stream, set callbackPulled flag\r
- callbackPulled = true;\r
- }\r
-\r
- // Execute Callback\r
- // ================\r
- // 1. Execute user callback method\r
- // 2. Handle return value from callback\r
-\r
- // if callback has not requested the stream to stop\r
- if ( callbackPulled && !callbackStopped ) {\r
- // Execute user callback method\r
- callbackResult = callback( stream_.userBuffer[OUTPUT],\r
- stream_.userBuffer[INPUT],\r
- stream_.bufferSize,\r
- getStreamTime(),\r
- captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,\r
- stream_.callbackInfo.userData );\r
-\r
- // Handle return value from callback\r
- if ( callbackResult == 1 ) {\r
- // instantiate a thread to stop this thread\r
- HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, NULL, NULL );\r
-\r
- if ( !threadHandle ) {\r
- EXIT_ON_ERROR( -1, RtAudioError::THREAD_ERROR, "Unable to instantiate stream stop thread" );\r
- }\r
- else if ( !CloseHandle( threadHandle ) ) {\r
- EXIT_ON_ERROR( -1, RtAudioError::THREAD_ERROR, "Unable to close stream stop thread handle" );\r
- }\r
-\r
- callbackStopped = true;\r
- }\r
- else if ( callbackResult == 2 ) {\r
- // instantiate a thread to stop this thread\r
- HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, NULL, NULL );\r
-\r
- if ( !threadHandle ) {\r
- EXIT_ON_ERROR( -1, RtAudioError::THREAD_ERROR, "Unable to instantiate stream abort thread" );\r
- }\r
- else if ( !CloseHandle( threadHandle ) ) {\r
- EXIT_ON_ERROR( -1, RtAudioError::THREAD_ERROR, "Unable to close stream abort thread handle" );\r
- }\r
-\r
- callbackStopped = true;\r
- }\r
- }\r
- }\r
-\r
- // Callback Output\r
- // ===============\r
- // 1. Convert callback buffer to stream format\r
- // 2. Convert callback buffer to stream sample rate and channel count\r
- // 3. Push callback buffer into outputBuffer\r
-\r
- if ( renderAudioClient && callbackPulled ) {\r
- if ( stream_.doConvertBuffer[OUTPUT] ) {\r
- // Convert callback buffer to stream format\r
- convertBuffer( stream_.deviceBuffer,\r
- stream_.userBuffer[OUTPUT],\r
- stream_.convertInfo[OUTPUT] );\r
-\r
- // Convert callback buffer to stream sample rate and channel count\r
- convertBufferWasapi( convBuffer,\r
- stream_.deviceBuffer,\r
- stream_.nUserChannels[OUTPUT],\r
- stream_.nDeviceChannels[OUTPUT],\r
- stream_.sampleRate,\r
- renderFormat->nSamplesPerSec,\r
- stream_.bufferSize,\r
- convBufferSize,\r
- stream_.deviceFormat[OUTPUT] );\r
- }\r
- else {\r
- // Convert callback buffer to stream sample rate and channel count\r
- convertBufferWasapi( convBuffer,\r
- stream_.userBuffer[OUTPUT],\r
- stream_.nUserChannels[OUTPUT],\r
- stream_.nDeviceChannels[OUTPUT],\r
- stream_.sampleRate,\r
- renderFormat->nSamplesPerSec,\r
- stream_.bufferSize,\r
- convBufferSize,\r
- stream_.deviceFormat[OUTPUT] );\r
- }\r
-\r
- // Push callback buffer into outputBuffer\r
- callbackPushed = renderBuffer.pushBuffer( convBuffer,\r
- convBufferSize * stream_.nDeviceChannels[OUTPUT],\r
- stream_.deviceFormat[OUTPUT] );\r
- }\r
-\r
- // Stream Capture\r
- // ==============\r
- // 1. Get capture buffer from stream\r
- // 2. Push capture buffer into inputBuffer\r
- // 3. If 2. was successful: Release capture buffer\r
-\r
- if ( captureAudioClient ) {\r
- // if the callback input buffer was not pulled from captureBuffer, wait for next capture event\r
- if ( !callbackPulled ) {\r
- WaitForSingleObject( captureEvent, INFINITE );\r
- }\r
-\r
- // Get capture buffer from stream\r
- hr = captureClient->GetBuffer( &streamBuffer,\r
- &bufferFrameCount,\r
- &captureFlags, NULL, NULL );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve capture buffer" );\r
-\r
- if ( bufferFrameCount != 0 ) {\r
- // Push capture buffer into inputBuffer\r
- if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,\r
- bufferFrameCount * stream_.nDeviceChannels[INPUT],\r
- stream_.deviceFormat[INPUT] ) )\r
- {\r
- // Release capture buffer\r
- hr = captureClient->ReleaseBuffer( bufferFrameCount );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to release capture buffer" );\r
- }\r
- else\r
- {\r
- // Inform WASAPI that capture was unsuccessful\r
- hr = captureClient->ReleaseBuffer( 0 );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to release capture buffer" );\r
- }\r
- }\r
- else\r
- {\r
- // Inform WASAPI that capture was unsuccessful\r
- hr = captureClient->ReleaseBuffer( 0 );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to release capture buffer" );\r
- }\r
- }\r
-\r
- // Stream Render\r
- // =============\r
- // 1. Get render buffer from stream\r
- // 2. Pull next buffer from outputBuffer\r
- // 3. If 2. was successful: Fill render buffer with next buffer\r
- // Release render buffer\r
-\r
- if ( renderAudioClient ) {\r
- // if the callback output buffer was not pushed to renderBuffer, wait for next render event\r
- if ( callbackPulled && !callbackPushed ) {\r
- WaitForSingleObject( renderEvent, INFINITE );\r
- }\r
-\r
- // Get render buffer from stream\r
- hr = renderAudioClient->GetBufferSize( &bufferFrameCount );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render buffer size" );\r
-\r
- hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render buffer padding" );\r
-\r
- bufferFrameCount -= numFramesPadding;\r
-\r
- if ( bufferFrameCount != 0 ) {\r
- hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to retrieve render buffer" );\r
-\r
- // Pull next buffer from outputBuffer\r
- // Fill render buffer with next buffer\r
- if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,\r
- bufferFrameCount * stream_.nDeviceChannels[OUTPUT],\r
- stream_.deviceFormat[OUTPUT] ) )\r
- {\r
- // Release render buffer\r
- hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to release render buffer" );\r
- }\r
- else\r
- {\r
- // Inform WASAPI that render was unsuccessful\r
- hr = renderClient->ReleaseBuffer( 0, 0 );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to release render buffer" );\r
- }\r
- }\r
- else\r
- {\r
- // Inform WASAPI that render was unsuccessful\r
- hr = renderClient->ReleaseBuffer( 0, 0 );\r
- EXIT_ON_ERROR( hr, RtAudioError::DRIVER_ERROR, "Unable to release render buffer" );\r
- }\r
- }\r
-\r
- // if the callback buffer was pushed renderBuffer reset callbackPulled flag\r
- if ( callbackPushed ) {\r
- callbackPulled = false;\r
- }\r
-\r
- // tick stream time\r
- RtApi::tickStreamTime();\r
- }\r
-\r
-Exit:\r
- // clean up\r
- CoTaskMemFree( captureFormat );\r
- CoTaskMemFree( renderFormat );\r
-\r
- //delete convBuffer;\r
- free ( convBuffer );\r
-\r
- CoUninitialize();\r
-\r
- // update stream state\r
- stream_.state = STREAM_STOPPED;\r
-}\r
-\r
-//******************** End of __WINDOWS_WASAPI__ *********************//\r
-#endif\r
-\r
-\r
-#if defined(__WINDOWS_DS__) // Windows DirectSound API\r
-\r
-// Modified by Robin Davies, October 2005\r
-// - Improvements to DirectX pointer chasing. \r
-// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.\r
-// - Auto-call CoInitialize for DSOUND and ASIO platforms.\r
-// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007\r
-// Changed device query structure for RtAudio 4.0.7, January 2010\r
-\r
-#include <dsound.h>\r
-#include <assert.h>\r
-#include <algorithm>\r
-\r
-#if defined(__MINGW32__)\r
- // missing from latest mingw winapi\r
-#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */\r
-#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */\r
-#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */\r
-#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */\r
-#endif\r
-\r
-#define MINIMUM_DEVICE_BUFFER_SIZE 32768\r
-\r
-#ifdef _MSC_VER // if Microsoft Visual C++\r
-#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.\r
-#endif\r
-\r
-static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )\r
-{\r
- if ( pointer > bufferSize ) pointer -= bufferSize;\r
- if ( laterPointer < earlierPointer ) laterPointer += bufferSize;\r
- if ( pointer < earlierPointer ) pointer += bufferSize;\r
- return pointer >= earlierPointer && pointer < laterPointer;\r
-}\r
-\r
-// A structure to hold various information related to the DirectSound\r
-// API implementation.\r
-struct DsHandle {\r
- unsigned int drainCounter; // Tracks callback counts when draining\r
- bool internalDrain; // Indicates if stop is initiated from callback or not.\r
- void *id[2];\r
- void *buffer[2];\r
- bool xrun[2];\r
- UINT bufferPointer[2]; \r
- DWORD dsBufferSize[2];\r
- DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.\r
- HANDLE condition;\r
-\r
- DsHandle()\r
- :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; }\r
-};\r
-\r
-// Declarations for utility functions, callbacks, and structures\r
-// specific to the DirectSound implementation.\r
-static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,\r
- LPCTSTR description,\r
- LPCTSTR module,\r
- LPVOID lpContext );\r
-\r
-static const char* getErrorString( int code );\r
-\r
-static unsigned __stdcall callbackHandler( void *ptr );\r
-\r
-struct DsDevice {\r
- LPGUID id[2];\r
- bool validId[2];\r
- bool found;\r
- std::string name;\r
-\r
- DsDevice()\r
- : found(false) { validId[0] = false; validId[1] = false; }\r
-};\r
-\r
-struct DsProbeData {\r
- bool isInput;\r
- std::vector<struct DsDevice>* dsDevices;\r
-};\r
-\r
-RtApiDs :: RtApiDs()\r
-{\r
- // Dsound will run both-threaded. If CoInitialize fails, then just\r
- // accept whatever the mainline chose for a threading model.\r
- coInitialized_ = false;\r
- HRESULT hr = CoInitialize( NULL );\r
- if ( !FAILED( hr ) ) coInitialized_ = true;\r
-}\r
-\r
-RtApiDs :: ~RtApiDs()\r
-{\r
- if ( coInitialized_ ) CoUninitialize(); // balanced call.\r
- if ( stream_.state != STREAM_CLOSED ) closeStream();\r
-}\r
-\r
-// The DirectSound default output is always the first device.\r
-unsigned int RtApiDs :: getDefaultOutputDevice( void )\r
-{\r
- return 0;\r
-}\r
-\r
-// The DirectSound default input is always the first input device,\r
-// which is the first capture device enumerated.\r
-unsigned int RtApiDs :: getDefaultInputDevice( void )\r
-{\r
- return 0;\r
-}\r
-\r
-unsigned int RtApiDs :: getDeviceCount( void )\r
-{\r
- // Set query flag for previously found devices to false, so that we\r
- // can check for any devices that have disappeared.\r
- for ( unsigned int i=0; i<dsDevices.size(); i++ )\r
- dsDevices[i].found = false;\r
-\r
- // Query DirectSound devices.\r
- struct DsProbeData probeInfo;\r
- probeInfo.isInput = false;\r
- probeInfo.dsDevices = &dsDevices;\r
- HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- }\r
-\r
- // Query DirectSoundCapture devices.\r
- probeInfo.isInput = true;\r
- result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- }\r
-\r
- // Clean out any devices that may have disappeared.\r
- std::vector< int > indices;\r
- for ( unsigned int i=0; i<dsDevices.size(); i++ )\r
- if ( dsDevices[i].found == false ) indices.push_back( i );\r
- unsigned int nErased = 0;\r
- for ( unsigned int i=0; i<indices.size(); i++ )\r
- dsDevices.erase( dsDevices.begin()-nErased++ );\r
-\r
- return static_cast<unsigned int>(dsDevices.size());\r
-}\r
-\r
-RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )\r
-{\r
- RtAudio::DeviceInfo info;\r
- info.probed = false;\r
-\r
- if ( dsDevices.size() == 0 ) {\r
- // Force a query of all devices\r
- getDeviceCount();\r
- if ( dsDevices.size() == 0 ) {\r
- errorText_ = "RtApiDs::getDeviceInfo: no devices found!";\r
- error( RtAudioError::INVALID_USE );\r
- return info;\r
- }\r
- }\r
-\r
- if ( device >= dsDevices.size() ) {\r
- errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";\r
- error( RtAudioError::INVALID_USE );\r
- return info;\r
- }\r
-\r
- HRESULT result;\r
- if ( dsDevices[ device ].validId[0] == false ) goto probeInput;\r
-\r
- LPDIRECTSOUND output;\r
- DSCAPS outCaps;\r
- result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- goto probeInput;\r
- }\r
-\r
- outCaps.dwSize = sizeof( outCaps );\r
- result = output->GetCaps( &outCaps );\r
- if ( FAILED( result ) ) {\r
- output->Release();\r
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- goto probeInput;\r
- }\r
-\r
- // Get output channel information.\r
- info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;\r
-\r
- // Get sample rate information.\r
- info.sampleRates.clear();\r
- for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {\r
- if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&\r
- SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate )\r
- info.sampleRates.push_back( SAMPLE_RATES[k] );\r
- }\r
-\r
- // Get format information.\r
- if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;\r
- if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;\r
-\r
- output->Release();\r
-\r
- if ( getDefaultOutputDevice() == device )\r
- info.isDefaultOutput = true;\r
-\r
- if ( dsDevices[ device ].validId[1] == false ) {\r
- info.name = dsDevices[ device ].name;\r
- info.probed = true;\r
- return info;\r
- }\r
-\r
- probeInput:\r
-\r
- LPDIRECTSOUNDCAPTURE input;\r
- result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- DSCCAPS inCaps;\r
- inCaps.dwSize = sizeof( inCaps );\r
- result = input->GetCaps( &inCaps );\r
- if ( FAILED( result ) ) {\r
- input->Release();\r
- errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- // Get input channel information.\r
- info.inputChannels = inCaps.dwChannels;\r
-\r
- // Get sample rate and format information.\r
- std::vector<unsigned int> rates;\r
- if ( inCaps.dwChannels >= 2 ) {\r
- if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;\r
- if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;\r
- if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;\r
- if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;\r
- if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;\r
- if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;\r
- if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;\r
- if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;\r
-\r
- if ( info.nativeFormats & RTAUDIO_SINT16 ) {\r
- if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );\r
- if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );\r
- if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );\r
- if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );\r
- }\r
- else if ( info.nativeFormats & RTAUDIO_SINT8 ) {\r
- if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );\r
- if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );\r
- if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );\r
- if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );\r
- }\r
- }\r
- else if ( inCaps.dwChannels == 1 ) {\r
- if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;\r
- if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;\r
- if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;\r
- if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;\r
- if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;\r
- if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;\r
- if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;\r
- if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;\r
-\r
- if ( info.nativeFormats & RTAUDIO_SINT16 ) {\r
- if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );\r
- if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );\r
- if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );\r
- if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );\r
- }\r
- else if ( info.nativeFormats & RTAUDIO_SINT8 ) {\r
- if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );\r
- if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );\r
- if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );\r
- if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );\r
- }\r
- }\r
- else info.inputChannels = 0; // technically, this would be an error\r
-\r
- input->Release();\r
-\r
- if ( info.inputChannels == 0 ) return info;\r
-\r
- // Copy the supported rates to the info structure but avoid duplication.\r
- bool found;\r
- for ( unsigned int i=0; i<rates.size(); i++ ) {\r
- found = false;\r
- for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {\r
- if ( rates[i] == info.sampleRates[j] ) {\r
- found = true;\r
- break;\r
- }\r
- }\r
- if ( found == false ) info.sampleRates.push_back( rates[i] );\r
- }\r
- std::sort( info.sampleRates.begin(), info.sampleRates.end() );\r
-\r
- // If device opens for both playback and capture, we determine the channels.\r
- if ( info.outputChannels > 0 && info.inputChannels > 0 )\r
- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;\r
-\r
- if ( device == 0 ) info.isDefaultInput = true;\r
-\r
- // Copy name and return.\r
- info.name = dsDevices[ device ].name;\r
- info.probed = true;\r
- return info;\r
-}\r
-\r
-bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,\r
- unsigned int firstChannel, unsigned int sampleRate,\r
- RtAudioFormat format, unsigned int *bufferSize,\r
- RtAudio::StreamOptions *options )\r
-{\r
- if ( channels + firstChannel > 2 ) {\r
- errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";\r
- return FAILURE;\r
- }\r
-\r
- size_t nDevices = dsDevices.size();\r
- if ( nDevices == 0 ) {\r
- // This should not happen because a check is made before this function is called.\r
- errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";\r
- return FAILURE;\r
- }\r
-\r
- if ( device >= nDevices ) {\r
- // This should not happen because a check is made before this function is called.\r
- errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";\r
- return FAILURE;\r
- }\r
-\r
- if ( mode == OUTPUT ) {\r
- if ( dsDevices[ device ].validId[0] == false ) {\r
- errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- }\r
- else { // mode == INPUT\r
- if ( dsDevices[ device ].validId[1] == false ) {\r
- errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- }\r
-\r
- // According to a note in PortAudio, using GetDesktopWindow()\r
- // instead of GetForegroundWindow() is supposed to avoid problems\r
- // that occur when the application's window is not the foreground\r
- // window. Also, if the application window closes before the\r
- // DirectSound buffer, DirectSound can crash. In the past, I had\r
- // problems when using GetDesktopWindow() but it seems fine now\r
- // (January 2010). I'll leave it commented here.\r
- // HWND hWnd = GetForegroundWindow();\r
- HWND hWnd = GetDesktopWindow();\r
-\r
- // Check the numberOfBuffers parameter and limit the lowest value to\r
- // two. This is a judgement call and a value of two is probably too\r
- // low for capture, but it should work for playback.\r
- int nBuffers = 0;\r
- if ( options ) nBuffers = options->numberOfBuffers;\r
- if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;\r
- if ( nBuffers < 2 ) nBuffers = 3;\r
-\r
- // Check the lower range of the user-specified buffer size and set\r
- // (arbitrarily) to a lower bound of 32.\r
- if ( *bufferSize < 32 ) *bufferSize = 32;\r
-\r
- // Create the wave format structure. The data format setting will\r
- // be determined later.\r
- WAVEFORMATEX waveFormat;\r
- ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );\r
- waveFormat.wFormatTag = WAVE_FORMAT_PCM;\r
- waveFormat.nChannels = channels + firstChannel;\r
- waveFormat.nSamplesPerSec = (unsigned long) sampleRate;\r
-\r
- // Determine the device buffer size. By default, we'll use the value\r
- // defined above (32K), but we will grow it to make allowances for\r
- // very large software buffer sizes.\r
- DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;\r
- DWORD dsPointerLeadTime = 0;\r
-\r
- void *ohandle = 0, *bhandle = 0;\r
- HRESULT result;\r
- if ( mode == OUTPUT ) {\r
-\r
- LPDIRECTSOUND output;\r
- result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- DSCAPS outCaps;\r
- outCaps.dwSize = sizeof( outCaps );\r
- result = output->GetCaps( &outCaps );\r
- if ( FAILED( result ) ) {\r
- output->Release();\r
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Check channel information.\r
- if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {\r
- errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Check format information. Use 16-bit format unless not\r
- // supported or user requests 8-bit.\r
- if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&\r
- !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {\r
- waveFormat.wBitsPerSample = 16;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;\r
- }\r
- else {\r
- waveFormat.wBitsPerSample = 8;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT8;\r
- }\r
- stream_.userFormat = format;\r
-\r
- // Update wave format structure and buffer information.\r
- waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;\r
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;\r
- dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;\r
-\r
- // If the user wants an even bigger buffer, increase the device buffer size accordingly.\r
- while ( dsPointerLeadTime * 2U > dsBufferSize )\r
- dsBufferSize *= 2;\r
-\r
- // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.\r
- // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );\r
- // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.\r
- result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );\r
- if ( FAILED( result ) ) {\r
- output->Release();\r
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Even though we will write to the secondary buffer, we need to\r
- // access the primary buffer to set the correct output format\r
- // (since the default is 8-bit, 22 kHz!). Setup the DS primary\r
- // buffer description.\r
- DSBUFFERDESC bufferDescription;\r
- ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );\r
- bufferDescription.dwSize = sizeof( DSBUFFERDESC );\r
- bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;\r
-\r
- // Obtain the primary buffer\r
- LPDIRECTSOUNDBUFFER buffer;\r
- result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );\r
- if ( FAILED( result ) ) {\r
- output->Release();\r
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Set the primary DS buffer sound format.\r
- result = buffer->SetFormat( &waveFormat );\r
- if ( FAILED( result ) ) {\r
- output->Release();\r
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Setup the secondary DS buffer description.\r
- ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );\r
- bufferDescription.dwSize = sizeof( DSBUFFERDESC );\r
- bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |\r
- DSBCAPS_GLOBALFOCUS |\r
- DSBCAPS_GETCURRENTPOSITION2 |\r
- DSBCAPS_LOCHARDWARE ); // Force hardware mixing\r
- bufferDescription.dwBufferBytes = dsBufferSize;\r
- bufferDescription.lpwfxFormat = &waveFormat;\r
-\r
- // Try to create the secondary DS buffer. If that doesn't work,\r
- // try to use software mixing. Otherwise, there's a problem.\r
- result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );\r
- if ( FAILED( result ) ) {\r
- bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |\r
- DSBCAPS_GLOBALFOCUS |\r
- DSBCAPS_GETCURRENTPOSITION2 |\r
- DSBCAPS_LOCSOFTWARE ); // Force software mixing\r
- result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );\r
- if ( FAILED( result ) ) {\r
- output->Release();\r
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- }\r
-\r
- // Get the buffer size ... might be different from what we specified.\r
- DSBCAPS dsbcaps;\r
- dsbcaps.dwSize = sizeof( DSBCAPS );\r
- result = buffer->GetCaps( &dsbcaps );\r
- if ( FAILED( result ) ) {\r
- output->Release();\r
- buffer->Release();\r
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- dsBufferSize = dsbcaps.dwBufferBytes;\r
-\r
- // Lock the DS buffer\r
- LPVOID audioPtr;\r
- DWORD dataLen;\r
- result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );\r
- if ( FAILED( result ) ) {\r
- output->Release();\r
- buffer->Release();\r
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Zero the DS buffer\r
- ZeroMemory( audioPtr, dataLen );\r
-\r
- // Unlock the DS buffer\r
- result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );\r
- if ( FAILED( result ) ) {\r
- output->Release();\r
- buffer->Release();\r
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- ohandle = (void *) output;\r
- bhandle = (void *) buffer;\r
- }\r
-\r
- if ( mode == INPUT ) {\r
-\r
- LPDIRECTSOUNDCAPTURE input;\r
- result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- DSCCAPS inCaps;\r
- inCaps.dwSize = sizeof( inCaps );\r
- result = input->GetCaps( &inCaps );\r
- if ( FAILED( result ) ) {\r
- input->Release();\r
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Check channel information.\r
- if ( inCaps.dwChannels < channels + firstChannel ) {\r
- errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";\r
- return FAILURE;\r
- }\r
-\r
- // Check format information. Use 16-bit format unless user\r
- // requests 8-bit.\r
- DWORD deviceFormats;\r
- if ( channels + firstChannel == 2 ) {\r
- deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;\r
- if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {\r
- waveFormat.wBitsPerSample = 8;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT8;\r
- }\r
- else { // assume 16-bit is supported\r
- waveFormat.wBitsPerSample = 16;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;\r
- }\r
- }\r
- else { // channel == 1\r
- deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;\r
- if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {\r
- waveFormat.wBitsPerSample = 8;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT8;\r
- }\r
- else { // assume 16-bit is supported\r
- waveFormat.wBitsPerSample = 16;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;\r
- }\r
- }\r
- stream_.userFormat = format;\r
-\r
- // Update wave format structure and buffer information.\r
- waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;\r
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;\r
- dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;\r
-\r
- // If the user wants an even bigger buffer, increase the device buffer size accordingly.\r
- while ( dsPointerLeadTime * 2U > dsBufferSize )\r
- dsBufferSize *= 2;\r
-\r
- // Setup the secondary DS buffer description.\r
- DSCBUFFERDESC bufferDescription;\r
- ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );\r
- bufferDescription.dwSize = sizeof( DSCBUFFERDESC );\r
- bufferDescription.dwFlags = 0;\r
- bufferDescription.dwReserved = 0;\r
- bufferDescription.dwBufferBytes = dsBufferSize;\r
- bufferDescription.lpwfxFormat = &waveFormat;\r
-\r
- // Create the capture buffer.\r
- LPDIRECTSOUNDCAPTUREBUFFER buffer;\r
- result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );\r
- if ( FAILED( result ) ) {\r
- input->Release();\r
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Get the buffer size ... might be different from what we specified.\r
- DSCBCAPS dscbcaps;\r
- dscbcaps.dwSize = sizeof( DSCBCAPS );\r
- result = buffer->GetCaps( &dscbcaps );\r
- if ( FAILED( result ) ) {\r
- input->Release();\r
- buffer->Release();\r
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- dsBufferSize = dscbcaps.dwBufferBytes;\r
-\r
- // NOTE: We could have a problem here if this is a duplex stream\r
- // and the play and capture hardware buffer sizes are different\r
- // (I'm actually not sure if that is a problem or not).\r
- // Currently, we are not verifying that.\r
-\r
- // Lock the capture buffer\r
- LPVOID audioPtr;\r
- DWORD dataLen;\r
- result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );\r
- if ( FAILED( result ) ) {\r
- input->Release();\r
- buffer->Release();\r
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Zero the buffer\r
- ZeroMemory( audioPtr, dataLen );\r
-\r
- // Unlock the buffer\r
- result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );\r
- if ( FAILED( result ) ) {\r
- input->Release();\r
- buffer->Release();\r
- errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- ohandle = (void *) input;\r
- bhandle = (void *) buffer;\r
- }\r
-\r
- // Set various stream parameters\r
- DsHandle *handle = 0;\r
- stream_.nDeviceChannels[mode] = channels + firstChannel;\r
- stream_.nUserChannels[mode] = channels;\r
- stream_.bufferSize = *bufferSize;\r
- stream_.channelOffset[mode] = firstChannel;\r
- stream_.deviceInterleaved[mode] = true;\r
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;\r
- else stream_.userInterleaved = true;\r
-\r
- // Set flag for buffer conversion\r
- stream_.doConvertBuffer[mode] = false;\r
- if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])\r
- stream_.doConvertBuffer[mode] = true;\r
- if (stream_.userFormat != stream_.deviceFormat[mode])\r
- stream_.doConvertBuffer[mode] = true;\r
- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&\r
- stream_.nUserChannels[mode] > 1 )\r
- stream_.doConvertBuffer[mode] = true;\r
-\r
- // Allocate necessary internal buffers\r
- long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );\r
- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );\r
- if ( stream_.userBuffer[mode] == NULL ) {\r
- errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";\r
- goto error;\r
- }\r
-\r
- if ( stream_.doConvertBuffer[mode] ) {\r
-\r
- bool makeBuffer = true;\r
- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );\r
- if ( mode == INPUT ) {\r
- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {\r
- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );\r
- if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;\r
- }\r
- }\r
-\r
- if ( makeBuffer ) {\r
- bufferBytes *= *bufferSize;\r
- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );\r
- if ( stream_.deviceBuffer == NULL ) {\r
- errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";\r
- goto error;\r
- }\r
- }\r
- }\r
-\r
- // Allocate our DsHandle structures for the stream.\r
- if ( stream_.apiHandle == 0 ) {\r
- try {\r
- handle = new DsHandle;\r
- }\r
- catch ( std::bad_alloc& ) {\r
- errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";\r
- goto error;\r
- }\r
-\r
- // Create a manual-reset event.\r
- handle->condition = CreateEvent( NULL, // no security\r
- TRUE, // manual-reset\r
- FALSE, // non-signaled initially\r
- NULL ); // unnamed\r
- stream_.apiHandle = (void *) handle;\r
- }\r
- else\r
- handle = (DsHandle *) stream_.apiHandle;\r
- handle->id[mode] = ohandle;\r
- handle->buffer[mode] = bhandle;\r
- handle->dsBufferSize[mode] = dsBufferSize;\r
- handle->dsPointerLeadTime[mode] = dsPointerLeadTime;\r
-\r
- stream_.device[mode] = device;\r
- stream_.state = STREAM_STOPPED;\r
- if ( stream_.mode == OUTPUT && mode == INPUT )\r
- // We had already set up an output stream.\r
- stream_.mode = DUPLEX;\r
- else\r
- stream_.mode = mode;\r
- stream_.nBuffers = nBuffers;\r
- stream_.sampleRate = sampleRate;\r
-\r
- // Setup the buffer conversion information structure.\r
- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );\r
-\r
- // Setup the callback thread.\r
- if ( stream_.callbackInfo.isRunning == false ) {\r
- unsigned threadId;\r
- stream_.callbackInfo.isRunning = true;\r
- stream_.callbackInfo.object = (void *) this;\r
- stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,\r
- &stream_.callbackInfo, 0, &threadId );\r
- if ( stream_.callbackInfo.thread == 0 ) {\r
- errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";\r
- goto error;\r
- }\r
-\r
- // Boost DS thread priority\r
- SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );\r
- }\r
- return SUCCESS;\r
-\r
- error:\r
- if ( handle ) {\r
- if ( handle->buffer[0] ) { // the object pointer can be NULL and valid\r
- LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];\r
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];\r
- if ( buffer ) buffer->Release();\r
- object->Release();\r
- }\r
- if ( handle->buffer[1] ) {\r
- LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];\r
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];\r
- if ( buffer ) buffer->Release();\r
- object->Release();\r
- }\r
- CloseHandle( handle->condition );\r
- delete handle;\r
- stream_.apiHandle = 0;\r
- }\r
-\r
- for ( int i=0; i<2; i++ ) {\r
- if ( stream_.userBuffer[i] ) {\r
- free( stream_.userBuffer[i] );\r
- stream_.userBuffer[i] = 0;\r
- }\r
- }\r
-\r
- if ( stream_.deviceBuffer ) {\r
- free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = 0;\r
- }\r
-\r
- stream_.state = STREAM_CLOSED;\r
- return FAILURE;\r
-}\r
-\r
-void RtApiDs :: closeStream()\r
-{\r
- if ( stream_.state == STREAM_CLOSED ) {\r
- errorText_ = "RtApiDs::closeStream(): no open stream to close!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- // Stop the callback thread.\r
- stream_.callbackInfo.isRunning = false;\r
- WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );\r
- CloseHandle( (HANDLE) stream_.callbackInfo.thread );\r
-\r
- DsHandle *handle = (DsHandle *) stream_.apiHandle;\r
- if ( handle ) {\r
- if ( handle->buffer[0] ) { // the object pointer can be NULL and valid\r
- LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];\r
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];\r
- if ( buffer ) {\r
- buffer->Stop();\r
- buffer->Release();\r
- }\r
- object->Release();\r
- }\r
- if ( handle->buffer[1] ) {\r
- LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];\r
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];\r
- if ( buffer ) {\r
- buffer->Stop();\r
- buffer->Release();\r
- }\r
- object->Release();\r
- }\r
- CloseHandle( handle->condition );\r
- delete handle;\r
- stream_.apiHandle = 0;\r
- }\r
-\r
- for ( int i=0; i<2; i++ ) {\r
- if ( stream_.userBuffer[i] ) {\r
- free( stream_.userBuffer[i] );\r
- stream_.userBuffer[i] = 0;\r
- }\r
- }\r
-\r
- if ( stream_.deviceBuffer ) {\r
- free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = 0;\r
- }\r
-\r
- stream_.mode = UNINITIALIZED;\r
- stream_.state = STREAM_CLOSED;\r
-}\r
-\r
-void RtApiDs :: startStream()\r
-{\r
- verifyStream();\r
- if ( stream_.state == STREAM_RUNNING ) {\r
- errorText_ = "RtApiDs::startStream(): the stream is already running!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- DsHandle *handle = (DsHandle *) stream_.apiHandle;\r
-\r
- // Increase scheduler frequency on lesser windows (a side-effect of\r
- // increasing timer accuracy). On greater windows (Win2K or later),\r
- // this is already in effect.\r
- timeBeginPeriod( 1 ); \r
-\r
- buffersRolling = false;\r
- duplexPrerollBytes = 0;\r
-\r
- if ( stream_.mode == DUPLEX ) {\r
- // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.\r
- duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );\r
- }\r
-\r
- HRESULT result = 0;\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
-\r
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];\r
- result = buffer->Play( 0, 0, DSBPLAY_LOOPING );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
- }\r
-\r
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {\r
-\r
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];\r
- result = buffer->Start( DSCBSTART_LOOPING );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
- }\r
-\r
- handle->drainCounter = 0;\r
- handle->internalDrain = false;\r
- ResetEvent( handle->condition );\r
- stream_.state = STREAM_RUNNING;\r
-\r
- unlock:\r
- if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );\r
-}\r
-\r
-void RtApiDs :: stopStream()\r
-{\r
- verifyStream();\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- HRESULT result = 0;\r
- LPVOID audioPtr;\r
- DWORD dataLen;\r
- DsHandle *handle = (DsHandle *) stream_.apiHandle;\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
- if ( handle->drainCounter == 0 ) {\r
- handle->drainCounter = 2;\r
- WaitForSingleObject( handle->condition, INFINITE ); // block until signaled\r
- }\r
-\r
- stream_.state = STREAM_STOPPED;\r
-\r
- // Stop the buffer and clear memory\r
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];\r
- result = buffer->Stop();\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
-\r
- // Lock the buffer and clear it so that if we start to play again,\r
- // we won't have old data playing.\r
- result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
-\r
- // Zero the DS buffer\r
- ZeroMemory( audioPtr, dataLen );\r
-\r
- // Unlock the DS buffer\r
- result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
-\r
- // If we start playing again, we must begin at beginning of buffer.\r
- handle->bufferPointer[0] = 0;\r
- }\r
-\r
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {\r
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];\r
- audioPtr = NULL;\r
- dataLen = 0;\r
-\r
- stream_.state = STREAM_STOPPED;\r
-\r
- result = buffer->Stop();\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
-\r
- // Lock the buffer and clear it so that if we start to play again,\r
- // we won't have old data playing.\r
- result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
-\r
- // Zero the DS buffer\r
- ZeroMemory( audioPtr, dataLen );\r
-\r
- // Unlock the DS buffer\r
- result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
-\r
- // If we start recording again, we must begin at beginning of buffer.\r
- handle->bufferPointer[1] = 0;\r
- }\r
-\r
- unlock:\r
- timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.\r
- if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );\r
-}\r
-\r
-void RtApiDs :: abortStream()\r
-{\r
- verifyStream();\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- DsHandle *handle = (DsHandle *) stream_.apiHandle;\r
- handle->drainCounter = 2;\r
-\r
- stopStream();\r
-}\r
-\r
-void RtApiDs :: callbackEvent()\r
-{\r
- if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {\r
- Sleep( 50 ); // sleep 50 milliseconds\r
- return;\r
- }\r
-\r
- if ( stream_.state == STREAM_CLOSED ) {\r
- errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;\r
- DsHandle *handle = (DsHandle *) stream_.apiHandle;\r
-\r
- // Check if we were draining the stream and signal is finished.\r
- if ( handle->drainCounter > stream_.nBuffers + 2 ) {\r
-\r
- stream_.state = STREAM_STOPPING;\r
- if ( handle->internalDrain == false )\r
- SetEvent( handle->condition );\r
- else\r
- stopStream();\r
- return;\r
- }\r
-\r
- // Invoke user callback to get fresh output data UNLESS we are\r
- // draining stream.\r
- if ( handle->drainCounter == 0 ) {\r
- RtAudioCallback callback = (RtAudioCallback) info->callback;\r
- double streamTime = getStreamTime();\r
- RtAudioStreamStatus status = 0;\r
- if ( stream_.mode != INPUT && handle->xrun[0] == true ) {\r
- status |= RTAUDIO_OUTPUT_UNDERFLOW;\r
- handle->xrun[0] = false;\r
- }\r
- if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {\r
- status |= RTAUDIO_INPUT_OVERFLOW;\r
- handle->xrun[1] = false;\r
- }\r
- int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
- stream_.bufferSize, streamTime, status, info->userData );\r
- if ( cbReturnValue == 2 ) {\r
- stream_.state = STREAM_STOPPING;\r
- handle->drainCounter = 2;\r
- abortStream();\r
- return;\r
- }\r
- else if ( cbReturnValue == 1 ) {\r
- handle->drainCounter = 1;\r
- handle->internalDrain = true;\r
- }\r
- }\r
-\r
- HRESULT result;\r
- DWORD currentWritePointer, safeWritePointer;\r
- DWORD currentReadPointer, safeReadPointer;\r
- UINT nextWritePointer;\r
-\r
- LPVOID buffer1 = NULL;\r
- LPVOID buffer2 = NULL;\r
- DWORD bufferSize1 = 0;\r
- DWORD bufferSize2 = 0;\r
-\r
- char *buffer;\r
- long bufferBytes;\r
-\r
- if ( buffersRolling == false ) {\r
- if ( stream_.mode == DUPLEX ) {\r
- //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );\r
-\r
- // It takes a while for the devices to get rolling. As a result,\r
- // there's no guarantee that the capture and write device pointers\r
- // will move in lockstep. Wait here for both devices to start\r
- // rolling, and then set our buffer pointers accordingly.\r
- // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600\r
- // bytes later than the write buffer.\r
-\r
- // Stub: a serious risk of having a pre-emptive scheduling round\r
- // take place between the two GetCurrentPosition calls... but I'm\r
- // really not sure how to solve the problem. Temporarily boost to\r
- // Realtime priority, maybe; but I'm not sure what priority the\r
- // DirectSound service threads run at. We *should* be roughly\r
- // within a ms or so of correct.\r
-\r
- LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];\r
- LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];\r
-\r
- DWORD startSafeWritePointer, startSafeReadPointer;\r
-\r
- result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::SYSTEM_ERROR );\r
- return;\r
- }\r
- result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::SYSTEM_ERROR );\r
- return;\r
- }\r
- while ( true ) {\r
- result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::SYSTEM_ERROR );\r
- return;\r
- }\r
- result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::SYSTEM_ERROR );\r
- return;\r
- }\r
- if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;\r
- Sleep( 1 );\r
- }\r
-\r
- //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );\r
-\r
- handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];\r
- if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];\r
- handle->bufferPointer[1] = safeReadPointer;\r
- }\r
- else if ( stream_.mode == OUTPUT ) {\r
-\r
- // Set the proper nextWritePosition after initial startup.\r
- LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];\r
- result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::SYSTEM_ERROR );\r
- return;\r
- }\r
- handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];\r
- if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];\r
- }\r
-\r
- buffersRolling = true;\r
- }\r
-\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
- \r
- LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];\r
-\r
- if ( handle->drainCounter > 1 ) { // write zeros to the output stream\r
- bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];\r
- bufferBytes *= formatBytes( stream_.userFormat );\r
- memset( stream_.userBuffer[0], 0, bufferBytes );\r
- }\r
-\r
- // Setup parameters and do buffer conversion if necessary.\r
- if ( stream_.doConvertBuffer[0] ) {\r
- buffer = stream_.deviceBuffer;\r
- convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );\r
- bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];\r
- bufferBytes *= formatBytes( stream_.deviceFormat[0] );\r
- }\r
- else {\r
- buffer = stream_.userBuffer[0];\r
- bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];\r
- bufferBytes *= formatBytes( stream_.userFormat );\r
- }\r
-\r
- // No byte swapping necessary in DirectSound implementation.\r
-\r
- // Ahhh ... windoze. 16-bit data is signed but 8-bit data is\r
- // unsigned. So, we need to convert our signed 8-bit data here to\r
- // unsigned.\r
- if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )\r
- for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );\r
-\r
- DWORD dsBufferSize = handle->dsBufferSize[0];\r
- nextWritePointer = handle->bufferPointer[0];\r
-\r
- DWORD endWrite, leadPointer;\r
- while ( true ) {\r
- // Find out where the read and "safe write" pointers are.\r
- result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::SYSTEM_ERROR );\r
- return;\r
- }\r
-\r
- // We will copy our output buffer into the region between\r
- // safeWritePointer and leadPointer. If leadPointer is not\r
- // beyond the next endWrite position, wait until it is.\r
- leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];\r
- //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;\r
- if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;\r
- if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset\r
- endWrite = nextWritePointer + bufferBytes;\r
-\r
- // Check whether the entire write region is behind the play pointer.\r
- if ( leadPointer >= endWrite ) break;\r
-\r
- // If we are here, then we must wait until the leadPointer advances\r
- // beyond the end of our next write region. We use the\r
- // Sleep() function to suspend operation until that happens.\r
- double millis = ( endWrite - leadPointer ) * 1000.0;\r
- millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);\r
- if ( millis < 1.0 ) millis = 1.0;\r
- Sleep( (DWORD) millis );\r
- }\r
-\r
- if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )\r
- || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) { \r
- // We've strayed into the forbidden zone ... resync the read pointer.\r
- handle->xrun[0] = true;\r
- nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;\r
- if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;\r
- handle->bufferPointer[0] = nextWritePointer;\r
- endWrite = nextWritePointer + bufferBytes;\r
- }\r
-\r
- // Lock free space in the buffer\r
- result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,\r
- &bufferSize1, &buffer2, &bufferSize2, 0 );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::SYSTEM_ERROR );\r
- return;\r
- }\r
-\r
- // Copy our buffer into the DS buffer\r
- CopyMemory( buffer1, buffer, bufferSize1 );\r
- if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );\r
-\r
- // Update our buffer offset and unlock sound buffer\r
- dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::SYSTEM_ERROR );\r
- return;\r
- }\r
- nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;\r
- handle->bufferPointer[0] = nextWritePointer;\r
-\r
- if ( handle->drainCounter ) {\r
- handle->drainCounter++;\r
- goto unlock;\r
- }\r
- }\r
-\r
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {\r
-\r
- // Setup parameters.\r
- if ( stream_.doConvertBuffer[1] ) {\r
- buffer = stream_.deviceBuffer;\r
- bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];\r
- bufferBytes *= formatBytes( stream_.deviceFormat[1] );\r
- }\r
- else {\r
- buffer = stream_.userBuffer[1];\r
- bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];\r
- bufferBytes *= formatBytes( stream_.userFormat );\r
- }\r
-\r
- LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];\r
- long nextReadPointer = handle->bufferPointer[1];\r
- DWORD dsBufferSize = handle->dsBufferSize[1];\r
-\r
- // Find out where the write and "safe read" pointers are.\r
- result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::SYSTEM_ERROR );\r
- return;\r
- }\r
-\r
- if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset\r
- DWORD endRead = nextReadPointer + bufferBytes;\r
-\r
- // Handling depends on whether we are INPUT or DUPLEX. \r
- // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,\r
- // then a wait here will drag the write pointers into the forbidden zone.\r
- // \r
- // In DUPLEX mode, rather than wait, we will back off the read pointer until \r
- // it's in a safe position. This causes dropouts, but it seems to be the only \r
- // practical way to sync up the read and write pointers reliably, given the \r
- // the very complex relationship between phase and increment of the read and write \r
- // pointers.\r
- //\r
- // In order to minimize audible dropouts in DUPLEX mode, we will\r
- // provide a pre-roll period of 0.5 seconds in which we return\r
- // zeros from the read buffer while the pointers sync up.\r
-\r
- if ( stream_.mode == DUPLEX ) {\r
- if ( safeReadPointer < endRead ) {\r
- if ( duplexPrerollBytes <= 0 ) {\r
- // Pre-roll time over. Be more agressive.\r
- int adjustment = endRead-safeReadPointer;\r
-\r
- handle->xrun[1] = true;\r
- // Two cases:\r
- // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,\r
- // and perform fine adjustments later.\r
- // - small adjustments: back off by twice as much.\r
- if ( adjustment >= 2*bufferBytes )\r
- nextReadPointer = safeReadPointer-2*bufferBytes;\r
- else\r
- nextReadPointer = safeReadPointer-bufferBytes-adjustment;\r
-\r
- if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;\r
-\r
- }\r
- else {\r
- // In pre=roll time. Just do it.\r
- nextReadPointer = safeReadPointer - bufferBytes;\r
- while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;\r
- }\r
- endRead = nextReadPointer + bufferBytes;\r
- }\r
- }\r
- else { // mode == INPUT\r
- while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {\r
- // See comments for playback.\r
- double millis = (endRead - safeReadPointer) * 1000.0;\r
- millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);\r
- if ( millis < 1.0 ) millis = 1.0;\r
- Sleep( (DWORD) millis );\r
-\r
- // Wake up and find out where we are now.\r
- result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::SYSTEM_ERROR );\r
- return;\r
- }\r
- \r
- if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset\r
- }\r
- }\r
-\r
- // Lock free space in the buffer\r
- result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,\r
- &bufferSize1, &buffer2, &bufferSize2, 0 );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::SYSTEM_ERROR );\r
- return;\r
- }\r
-\r
- if ( duplexPrerollBytes <= 0 ) {\r
- // Copy our buffer into the DS buffer\r
- CopyMemory( buffer, buffer1, bufferSize1 );\r
- if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );\r
- }\r
- else {\r
- memset( buffer, 0, bufferSize1 );\r
- if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );\r
- duplexPrerollBytes -= bufferSize1 + bufferSize2;\r
- }\r
-\r
- // Update our buffer offset and unlock sound buffer\r
- nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;\r
- dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );\r
- if ( FAILED( result ) ) {\r
- errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::SYSTEM_ERROR );\r
- return;\r
- }\r
- handle->bufferPointer[1] = nextReadPointer;\r
-\r
- // No byte swapping necessary in DirectSound implementation.\r
-\r
- // If necessary, convert 8-bit data from unsigned to signed.\r
- if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )\r
- for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );\r
-\r
- // Do buffer conversion if necessary.\r
- if ( stream_.doConvertBuffer[1] )\r
- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );\r
- }\r
-\r
- unlock:\r
- RtApi::tickStreamTime();\r
-}\r
-\r
-// Definitions for utility functions and callbacks\r
-// specific to the DirectSound implementation.\r
-\r
-static unsigned __stdcall callbackHandler( void *ptr )\r
-{\r
- CallbackInfo *info = (CallbackInfo *) ptr;\r
- RtApiDs *object = (RtApiDs *) info->object;\r
- bool* isRunning = &info->isRunning;\r
-\r
- while ( *isRunning == true ) {\r
- object->callbackEvent();\r
- }\r
-\r
- _endthreadex( 0 );\r
- return 0;\r
-}\r
-\r
-#include "tchar.h"\r
-\r
-static std::string convertTChar( LPCTSTR name )\r
-{\r
-#if defined( UNICODE ) || defined( _UNICODE )\r
- int length = WideCharToMultiByte(CP_UTF8, 0, name, -1, NULL, 0, NULL, NULL);\r
- std::string s( length-1, '\0' );\r
- WideCharToMultiByte(CP_UTF8, 0, name, -1, &s[0], length, NULL, NULL);\r
-#else\r
- std::string s( name );\r
-#endif\r
-\r
- return s;\r
-}\r
-\r
-static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,\r
- LPCTSTR description,\r
- LPCTSTR /*module*/,\r
- LPVOID lpContext )\r
-{\r
- struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;\r
- std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;\r
-\r
- HRESULT hr;\r
- bool validDevice = false;\r
- if ( probeInfo.isInput == true ) {\r
- DSCCAPS caps;\r
- LPDIRECTSOUNDCAPTURE object;\r
-\r
- hr = DirectSoundCaptureCreate( lpguid, &object, NULL );\r
- if ( hr != DS_OK ) return TRUE;\r
-\r
- caps.dwSize = sizeof(caps);\r
- hr = object->GetCaps( &caps );\r
- if ( hr == DS_OK ) {\r
- if ( caps.dwChannels > 0 && caps.dwFormats > 0 )\r
- validDevice = true;\r
- }\r
- object->Release();\r
- }\r
- else {\r
- DSCAPS caps;\r
- LPDIRECTSOUND object;\r
- hr = DirectSoundCreate( lpguid, &object, NULL );\r
- if ( hr != DS_OK ) return TRUE;\r
-\r
- caps.dwSize = sizeof(caps);\r
- hr = object->GetCaps( &caps );\r
- if ( hr == DS_OK ) {\r
- if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )\r
- validDevice = true;\r
- }\r
- object->Release();\r
- }\r
-\r
- // If good device, then save its name and guid.\r
- std::string name = convertTChar( description );\r
- //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )\r
- if ( lpguid == NULL )\r
- name = "Default Device";\r
- if ( validDevice ) {\r
- for ( unsigned int i=0; i<dsDevices.size(); i++ ) {\r
- if ( dsDevices[i].name == name ) {\r
- dsDevices[i].found = true;\r
- if ( probeInfo.isInput ) {\r
- dsDevices[i].id[1] = lpguid;\r
- dsDevices[i].validId[1] = true;\r
- }\r
- else {\r
- dsDevices[i].id[0] = lpguid;\r
- dsDevices[i].validId[0] = true;\r
- }\r
- return TRUE;\r
- }\r
- }\r
-\r
- DsDevice device;\r
- device.name = name;\r
- device.found = true;\r
- if ( probeInfo.isInput ) {\r
- device.id[1] = lpguid;\r
- device.validId[1] = true;\r
- }\r
- else {\r
- device.id[0] = lpguid;\r
- device.validId[0] = true;\r
- }\r
- dsDevices.push_back( device );\r
- }\r
-\r
- return TRUE;\r
-}\r
-\r
-static const char* getErrorString( int code )\r
-{\r
- switch ( code ) {\r
-\r
- case DSERR_ALLOCATED:\r
- return "Already allocated";\r
-\r
- case DSERR_CONTROLUNAVAIL:\r
- return "Control unavailable";\r
-\r
- case DSERR_INVALIDPARAM:\r
- return "Invalid parameter";\r
-\r
- case DSERR_INVALIDCALL:\r
- return "Invalid call";\r
-\r
- case DSERR_GENERIC:\r
- return "Generic error";\r
-\r
- case DSERR_PRIOLEVELNEEDED:\r
- return "Priority level needed";\r
-\r
- case DSERR_OUTOFMEMORY:\r
- return "Out of memory";\r
-\r
- case DSERR_BADFORMAT:\r
- return "The sample rate or the channel format is not supported";\r
-\r
- case DSERR_UNSUPPORTED:\r
- return "Not supported";\r
-\r
- case DSERR_NODRIVER:\r
- return "No driver";\r
-\r
- case DSERR_ALREADYINITIALIZED:\r
- return "Already initialized";\r
-\r
- case DSERR_NOAGGREGATION:\r
- return "No aggregation";\r
-\r
- case DSERR_BUFFERLOST:\r
- return "Buffer lost";\r
-\r
- case DSERR_OTHERAPPHASPRIO:\r
- return "Another application already has priority";\r
-\r
- case DSERR_UNINITIALIZED:\r
- return "Uninitialized";\r
-\r
- default:\r
- return "DirectSound unknown error";\r
- }\r
-}\r
-//******************** End of __WINDOWS_DS__ *********************//\r
-#endif\r
-\r
-\r
-#if defined(__LINUX_ALSA__)\r
-\r
-#include <alsa/asoundlib.h>\r
-#include <unistd.h>\r
-\r
- // A structure to hold various information related to the ALSA API\r
- // implementation.\r
-struct AlsaHandle {\r
- snd_pcm_t *handles[2];\r
- bool synchronized;\r
- bool xrun[2];\r
- pthread_cond_t runnable_cv;\r
- bool runnable;\r
-\r
- AlsaHandle()\r
- :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }\r
-};\r
-\r
-static void *alsaCallbackHandler( void * ptr );\r
-\r
-RtApiAlsa :: RtApiAlsa()\r
-{\r
- // Nothing to do here.\r
-}\r
-\r
-RtApiAlsa :: ~RtApiAlsa()\r
-{\r
- if ( stream_.state != STREAM_CLOSED ) closeStream();\r
-}\r
-\r
-unsigned int RtApiAlsa :: getDeviceCount( void )\r
-{\r
- unsigned nDevices = 0;\r
- int result, subdevice, card;\r
- char name[64];\r
- snd_ctl_t *handle;\r
-\r
- // Count cards and devices\r
- card = -1;\r
- snd_card_next( &card );\r
- while ( card >= 0 ) {\r
- sprintf( name, "hw:%d", card );\r
- result = snd_ctl_open( &handle, name, 0 );\r
- if ( result < 0 ) {\r
- errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- goto nextcard;\r
- }\r
- subdevice = -1;\r
- while( 1 ) {\r
- result = snd_ctl_pcm_next_device( handle, &subdevice );\r
- if ( result < 0 ) {\r
- errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- break;\r
- }\r
- if ( subdevice < 0 )\r
- break;\r
- nDevices++;\r
- }\r
- nextcard:\r
- snd_ctl_close( handle );\r
- snd_card_next( &card );\r
- }\r
-\r
- result = snd_ctl_open( &handle, "default", 0 );\r
- if (result == 0) {\r
- nDevices++;\r
- snd_ctl_close( handle );\r
- }\r
-\r
- return nDevices;\r
-}\r
-\r
-RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )\r
-{\r
- RtAudio::DeviceInfo info;\r
- info.probed = false;\r
-\r
- unsigned nDevices = 0;\r
- int result, subdevice, card;\r
- char name[64];\r
- snd_ctl_t *chandle;\r
-\r
- // Count cards and devices\r
- card = -1;\r
- snd_card_next( &card );\r
- while ( card >= 0 ) {\r
- sprintf( name, "hw:%d", card );\r
- result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );\r
- if ( result < 0 ) {\r
- errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- goto nextcard;\r
- }\r
- subdevice = -1;\r
- while( 1 ) {\r
- result = snd_ctl_pcm_next_device( chandle, &subdevice );\r
- if ( result < 0 ) {\r
- errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- break;\r
- }\r
- if ( subdevice < 0 ) break;\r
- if ( nDevices == device ) {\r
- sprintf( name, "hw:%d,%d", card, subdevice );\r
- goto foundDevice;\r
- }\r
- nDevices++;\r
- }\r
- nextcard:\r
- snd_ctl_close( chandle );\r
- snd_card_next( &card );\r
- }\r
-\r
- result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );\r
- if ( result == 0 ) {\r
- if ( nDevices == device ) {\r
- strcpy( name, "default" );\r
- goto foundDevice;\r
- }\r
- nDevices++;\r
- }\r
-\r
- if ( nDevices == 0 ) {\r
- errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";\r
- error( RtAudioError::INVALID_USE );\r
- return info;\r
- }\r
-\r
- if ( device >= nDevices ) {\r
- errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";\r
- error( RtAudioError::INVALID_USE );\r
- return info;\r
- }\r
-\r
- foundDevice:\r
-\r
- // If a stream is already open, we cannot probe the stream devices.\r
- // Thus, use the saved results.\r
- if ( stream_.state != STREAM_CLOSED &&\r
- ( stream_.device[0] == device || stream_.device[1] == device ) ) {\r
- snd_ctl_close( chandle );\r
- if ( device >= devices_.size() ) {\r
- errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
- return devices_[ device ];\r
- }\r
-\r
- int openMode = SND_PCM_ASYNC;\r
- snd_pcm_stream_t stream;\r
- snd_pcm_info_t *pcminfo;\r
- snd_pcm_info_alloca( &pcminfo );\r
- snd_pcm_t *phandle;\r
- snd_pcm_hw_params_t *params;\r
- snd_pcm_hw_params_alloca( ¶ms );\r
-\r
- // First try for playback unless default device (which has subdev -1)\r
- stream = SND_PCM_STREAM_PLAYBACK;\r
- snd_pcm_info_set_stream( pcminfo, stream );\r
- if ( subdevice != -1 ) {\r
- snd_pcm_info_set_device( pcminfo, subdevice );\r
- snd_pcm_info_set_subdevice( pcminfo, 0 );\r
-\r
- result = snd_ctl_pcm_info( chandle, pcminfo );\r
- if ( result < 0 ) {\r
- // Device probably doesn't support playback.\r
- goto captureProbe;\r
- }\r
- }\r
-\r
- result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );\r
- if ( result < 0 ) {\r
- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- goto captureProbe;\r
- }\r
-\r
- // The device is open ... fill the parameter structure.\r
- result = snd_pcm_hw_params_any( phandle, params );\r
- if ( result < 0 ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- goto captureProbe;\r
- }\r
-\r
- // Get output channel information.\r
- unsigned int value;\r
- result = snd_pcm_hw_params_get_channels_max( params, &value );\r
- if ( result < 0 ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- goto captureProbe;\r
- }\r
- info.outputChannels = value;\r
- snd_pcm_close( phandle );\r
-\r
- captureProbe:\r
- stream = SND_PCM_STREAM_CAPTURE;\r
- snd_pcm_info_set_stream( pcminfo, stream );\r
-\r
- // Now try for capture unless default device (with subdev = -1)\r
- if ( subdevice != -1 ) {\r
- result = snd_ctl_pcm_info( chandle, pcminfo );\r
- snd_ctl_close( chandle );\r
- if ( result < 0 ) {\r
- // Device probably doesn't support capture.\r
- if ( info.outputChannels == 0 ) return info;\r
- goto probeParameters;\r
- }\r
- }\r
- else\r
- snd_ctl_close( chandle );\r
-\r
- result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);\r
- if ( result < 0 ) {\r
- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- if ( info.outputChannels == 0 ) return info;\r
- goto probeParameters;\r
- }\r
-\r
- // The device is open ... fill the parameter structure.\r
- result = snd_pcm_hw_params_any( phandle, params );\r
- if ( result < 0 ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- if ( info.outputChannels == 0 ) return info;\r
- goto probeParameters;\r
- }\r
-\r
- result = snd_pcm_hw_params_get_channels_max( params, &value );\r
- if ( result < 0 ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- if ( info.outputChannels == 0 ) return info;\r
- goto probeParameters;\r
- }\r
- info.inputChannels = value;\r
- snd_pcm_close( phandle );\r
-\r
- // If device opens for both playback and capture, we determine the channels.\r
- if ( info.outputChannels > 0 && info.inputChannels > 0 )\r
- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;\r
-\r
- // ALSA doesn't provide default devices so we'll use the first available one.\r
- if ( device == 0 && info.outputChannels > 0 )\r
- info.isDefaultOutput = true;\r
- if ( device == 0 && info.inputChannels > 0 )\r
- info.isDefaultInput = true;\r
-\r
- probeParameters:\r
- // At this point, we just need to figure out the supported data\r
- // formats and sample rates. We'll proceed by opening the device in\r
- // the direction with the maximum number of channels, or playback if\r
- // they are equal. This might limit our sample rate options, but so\r
- // be it.\r
-\r
- if ( info.outputChannels >= info.inputChannels )\r
- stream = SND_PCM_STREAM_PLAYBACK;\r
- else\r
- stream = SND_PCM_STREAM_CAPTURE;\r
- snd_pcm_info_set_stream( pcminfo, stream );\r
-\r
- result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);\r
- if ( result < 0 ) {\r
- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- // The device is open ... fill the parameter structure.\r
- result = snd_pcm_hw_params_any( phandle, params );\r
- if ( result < 0 ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- // Test our discrete set of sample rate values.\r
- info.sampleRates.clear();\r
- for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {\r
- if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 )\r
- info.sampleRates.push_back( SAMPLE_RATES[i] );\r
- }\r
- if ( info.sampleRates.size() == 0 ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- // Probe the supported data formats ... we don't care about endian-ness just yet\r
- snd_pcm_format_t format;\r
- info.nativeFormats = 0;\r
- format = SND_PCM_FORMAT_S8;\r
- if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )\r
- info.nativeFormats |= RTAUDIO_SINT8;\r
- format = SND_PCM_FORMAT_S16;\r
- if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )\r
- info.nativeFormats |= RTAUDIO_SINT16;\r
- format = SND_PCM_FORMAT_S24;\r
- if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )\r
- info.nativeFormats |= RTAUDIO_SINT24;\r
- format = SND_PCM_FORMAT_S32;\r
- if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )\r
- info.nativeFormats |= RTAUDIO_SINT32;\r
- format = SND_PCM_FORMAT_FLOAT;\r
- if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )\r
- info.nativeFormats |= RTAUDIO_FLOAT32;\r
- format = SND_PCM_FORMAT_FLOAT64;\r
- if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )\r
- info.nativeFormats |= RTAUDIO_FLOAT64;\r
-\r
- // Check that we have at least one supported format\r
- if ( info.nativeFormats == 0 ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- // Get the device name\r
- char *cardname;\r
- result = snd_card_get_name( card, &cardname );\r
- if ( result >= 0 ) {\r
- sprintf( name, "hw:%s,%d", cardname, subdevice );\r
- free( cardname );\r
- }\r
- info.name = name;\r
-\r
- // That's all ... close the device and return\r
- snd_pcm_close( phandle );\r
- info.probed = true;\r
- return info;\r
-}\r
-\r
-void RtApiAlsa :: saveDeviceInfo( void )\r
-{\r
- devices_.clear();\r
-\r
- unsigned int nDevices = getDeviceCount();\r
- devices_.resize( nDevices );\r
- for ( unsigned int i=0; i<nDevices; i++ )\r
- devices_[i] = getDeviceInfo( i );\r
-}\r
-\r
-bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,\r
- unsigned int firstChannel, unsigned int sampleRate,\r
- RtAudioFormat format, unsigned int *bufferSize,\r
- RtAudio::StreamOptions *options )\r
-\r
-{\r
-#if defined(__RTAUDIO_DEBUG__)\r
- snd_output_t *out;\r
- snd_output_stdio_attach(&out, stderr, 0);\r
-#endif\r
-\r
- // I'm not using the "plug" interface ... too much inconsistent behavior.\r
-\r
- unsigned nDevices = 0;\r
- int result, subdevice, card;\r
- char name[64];\r
- snd_ctl_t *chandle;\r
-\r
- if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )\r
- snprintf(name, sizeof(name), "%s", "default");\r
- else {\r
- // Count cards and devices\r
- card = -1;\r
- snd_card_next( &card );\r
- while ( card >= 0 ) {\r
- sprintf( name, "hw:%d", card );\r
- result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );\r
- if ( result < 0 ) {\r
- errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- subdevice = -1;\r
- while( 1 ) {\r
- result = snd_ctl_pcm_next_device( chandle, &subdevice );\r
- if ( result < 0 ) break;\r
- if ( subdevice < 0 ) break;\r
- if ( nDevices == device ) {\r
- sprintf( name, "hw:%d,%d", card, subdevice );\r
- snd_ctl_close( chandle );\r
- goto foundDevice;\r
- }\r
- nDevices++;\r
- }\r
- snd_ctl_close( chandle );\r
- snd_card_next( &card );\r
- }\r
-\r
- result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );\r
- if ( result == 0 ) {\r
- if ( nDevices == device ) {\r
- strcpy( name, "default" );\r
- goto foundDevice;\r
- }\r
- nDevices++;\r
- }\r
-\r
- if ( nDevices == 0 ) {\r
- // This should not happen because a check is made before this function is called.\r
- errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";\r
- return FAILURE;\r
- }\r
-\r
- if ( device >= nDevices ) {\r
- // This should not happen because a check is made before this function is called.\r
- errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";\r
- return FAILURE;\r
- }\r
- }\r
-\r
- foundDevice:\r
-\r
- // The getDeviceInfo() function will not work for a device that is\r
- // already open. Thus, we'll probe the system before opening a\r
- // stream and save the results for use by getDeviceInfo().\r
- if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once\r
- this->saveDeviceInfo();\r
-\r
- snd_pcm_stream_t stream;\r
- if ( mode == OUTPUT )\r
- stream = SND_PCM_STREAM_PLAYBACK;\r
- else\r
- stream = SND_PCM_STREAM_CAPTURE;\r
-\r
- snd_pcm_t *phandle;\r
- int openMode = SND_PCM_ASYNC;\r
- result = snd_pcm_open( &phandle, name, stream, openMode );\r
- if ( result < 0 ) {\r
- if ( mode == OUTPUT )\r
- errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";\r
- else\r
- errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Fill the parameter structure.\r
- snd_pcm_hw_params_t *hw_params;\r
- snd_pcm_hw_params_alloca( &hw_params );\r
- result = snd_pcm_hw_params_any( phandle, hw_params );\r
- if ( result < 0 ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
-#if defined(__RTAUDIO_DEBUG__)\r
- fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );\r
- snd_pcm_hw_params_dump( hw_params, out );\r
-#endif\r
-\r
- // Set access ... check user preference.\r
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {\r
- stream_.userInterleaved = false;\r
- result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );\r
- if ( result < 0 ) {\r
- result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );\r
- stream_.deviceInterleaved[mode] = true;\r
- }\r
- else\r
- stream_.deviceInterleaved[mode] = false;\r
- }\r
- else {\r
- stream_.userInterleaved = true;\r
- result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );\r
- if ( result < 0 ) {\r
- result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );\r
- stream_.deviceInterleaved[mode] = false;\r
- }\r
- else\r
- stream_.deviceInterleaved[mode] = true;\r
- }\r
-\r
- if ( result < 0 ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Determine how to set the device format.\r
- stream_.userFormat = format;\r
- snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;\r
-\r
- if ( format == RTAUDIO_SINT8 )\r
- deviceFormat = SND_PCM_FORMAT_S8;\r
- else if ( format == RTAUDIO_SINT16 )\r
- deviceFormat = SND_PCM_FORMAT_S16;\r
- else if ( format == RTAUDIO_SINT24 )\r
- deviceFormat = SND_PCM_FORMAT_S24;\r
- else if ( format == RTAUDIO_SINT32 )\r
- deviceFormat = SND_PCM_FORMAT_S32;\r
- else if ( format == RTAUDIO_FLOAT32 )\r
- deviceFormat = SND_PCM_FORMAT_FLOAT;\r
- else if ( format == RTAUDIO_FLOAT64 )\r
- deviceFormat = SND_PCM_FORMAT_FLOAT64;\r
-\r
- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {\r
- stream_.deviceFormat[mode] = format;\r
- goto setFormat;\r
- }\r
-\r
- // The user requested format is not natively supported by the device.\r
- deviceFormat = SND_PCM_FORMAT_FLOAT64;\r
- if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {\r
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;\r
- goto setFormat;\r
- }\r
-\r
- deviceFormat = SND_PCM_FORMAT_FLOAT;\r
- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {\r
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;\r
- goto setFormat;\r
- }\r
-\r
- deviceFormat = SND_PCM_FORMAT_S32;\r
- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;\r
- goto setFormat;\r
- }\r
-\r
- deviceFormat = SND_PCM_FORMAT_S24;\r
- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT24;\r
- goto setFormat;\r
- }\r
-\r
- deviceFormat = SND_PCM_FORMAT_S16;\r
- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;\r
- goto setFormat;\r
- }\r
-\r
- deviceFormat = SND_PCM_FORMAT_S8;\r
- if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT8;\r
- goto setFormat;\r
- }\r
-\r
- // If we get here, no supported format was found.\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
-\r
- setFormat:\r
- result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );\r
- if ( result < 0 ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Determine whether byte-swaping is necessary.\r
- stream_.doByteSwap[mode] = false;\r
- if ( deviceFormat != SND_PCM_FORMAT_S8 ) {\r
- result = snd_pcm_format_cpu_endian( deviceFormat );\r
- if ( result == 0 )\r
- stream_.doByteSwap[mode] = true;\r
- else if (result < 0) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- }\r
-\r
- // Set the sample rate.\r
- result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );\r
- if ( result < 0 ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Determine the number of channels for this device. We support a possible\r
- // minimum device channel number > than the value requested by the user.\r
- stream_.nUserChannels[mode] = channels;\r
- unsigned int value;\r
- result = snd_pcm_hw_params_get_channels_max( hw_params, &value );\r
- unsigned int deviceChannels = value;\r
- if ( result < 0 || deviceChannels < channels + firstChannel ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- result = snd_pcm_hw_params_get_channels_min( hw_params, &value );\r
- if ( result < 0 ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- deviceChannels = value;\r
- if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;\r
- stream_.nDeviceChannels[mode] = deviceChannels;\r
-\r
- // Set the device channels.\r
- result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );\r
- if ( result < 0 ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Set the buffer (or period) size.\r
- int dir = 0;\r
- snd_pcm_uframes_t periodSize = *bufferSize;\r
- result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );\r
- if ( result < 0 ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- *bufferSize = periodSize;\r
-\r
- // Set the buffer number, which in ALSA is referred to as the "period".\r
- unsigned int periods = 0;\r
- if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;\r
- if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;\r
- if ( periods < 2 ) periods = 4; // a fairly safe default value\r
- result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );\r
- if ( result < 0 ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // If attempting to setup a duplex stream, the bufferSize parameter\r
- // MUST be the same in both directions!\r
- if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- stream_.bufferSize = *bufferSize;\r
-\r
- // Install the hardware configuration\r
- result = snd_pcm_hw_params( phandle, hw_params );\r
- if ( result < 0 ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
-#if defined(__RTAUDIO_DEBUG__)\r
- fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");\r
- snd_pcm_hw_params_dump( hw_params, out );\r
-#endif\r
-\r
- // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.\r
- snd_pcm_sw_params_t *sw_params = NULL;\r
- snd_pcm_sw_params_alloca( &sw_params );\r
- snd_pcm_sw_params_current( phandle, sw_params );\r
- snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );\r
- snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );\r
- snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );\r
-\r
- // The following two settings were suggested by Theo Veenker\r
- //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );\r
- //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );\r
-\r
- // here are two options for a fix\r
- //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );\r
- snd_pcm_uframes_t val;\r
- snd_pcm_sw_params_get_boundary( sw_params, &val );\r
- snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );\r
-\r
- result = snd_pcm_sw_params( phandle, sw_params );\r
- if ( result < 0 ) {\r
- snd_pcm_close( phandle );\r
- errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
-#if defined(__RTAUDIO_DEBUG__)\r
- fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");\r
- snd_pcm_sw_params_dump( sw_params, out );\r
-#endif\r
-\r
- // Set flags for buffer conversion\r
- stream_.doConvertBuffer[mode] = false;\r
- if ( stream_.userFormat != stream_.deviceFormat[mode] )\r
- stream_.doConvertBuffer[mode] = true;\r
- if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )\r
- stream_.doConvertBuffer[mode] = true;\r
- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&\r
- stream_.nUserChannels[mode] > 1 )\r
- stream_.doConvertBuffer[mode] = true;\r
-\r
- // Allocate the ApiHandle if necessary and then save.\r
- AlsaHandle *apiInfo = 0;\r
- if ( stream_.apiHandle == 0 ) {\r
- try {\r
- apiInfo = (AlsaHandle *) new AlsaHandle;\r
- }\r
- catch ( std::bad_alloc& ) {\r
- errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";\r
- goto error;\r
- }\r
-\r
- if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {\r
- errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";\r
- goto error;\r
- }\r
-\r
- stream_.apiHandle = (void *) apiInfo;\r
- apiInfo->handles[0] = 0;\r
- apiInfo->handles[1] = 0;\r
- }\r
- else {\r
- apiInfo = (AlsaHandle *) stream_.apiHandle;\r
- }\r
- apiInfo->handles[mode] = phandle;\r
- phandle = 0;\r
-\r
- // Allocate necessary internal buffers.\r
- unsigned long bufferBytes;\r
- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );\r
- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );\r
- if ( stream_.userBuffer[mode] == NULL ) {\r
- errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";\r
- goto error;\r
- }\r
-\r
- if ( stream_.doConvertBuffer[mode] ) {\r
-\r
- bool makeBuffer = true;\r
- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );\r
- if ( mode == INPUT ) {\r
- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {\r
- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );\r
- if ( bufferBytes <= bytesOut ) makeBuffer = false;\r
- }\r
- }\r
-\r
- if ( makeBuffer ) {\r
- bufferBytes *= *bufferSize;\r
- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );\r
- if ( stream_.deviceBuffer == NULL ) {\r
- errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";\r
- goto error;\r
- }\r
- }\r
- }\r
-\r
- stream_.sampleRate = sampleRate;\r
- stream_.nBuffers = periods;\r
- stream_.device[mode] = device;\r
- stream_.state = STREAM_STOPPED;\r
-\r
- // Setup the buffer conversion information structure.\r
- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );\r
-\r
- // Setup thread if necessary.\r
- if ( stream_.mode == OUTPUT && mode == INPUT ) {\r
- // We had already set up an output stream.\r
- stream_.mode = DUPLEX;\r
- // Link the streams if possible.\r
- apiInfo->synchronized = false;\r
- if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )\r
- apiInfo->synchronized = true;\r
- else {\r
- errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";\r
- error( RtAudioError::WARNING );\r
- }\r
- }\r
- else {\r
- stream_.mode = mode;\r
-\r
- // Setup callback thread.\r
- stream_.callbackInfo.object = (void *) this;\r
-\r
- // Set the thread attributes for joinable and realtime scheduling\r
- // priority (optional). The higher priority will only take affect\r
- // if the program is run as root or suid. Note, under Linux\r
- // processes with CAP_SYS_NICE privilege, a user can change\r
- // scheduling policy and priority (thus need not be root). See\r
- // POSIX "capabilities".\r
- pthread_attr_t attr;\r
- pthread_attr_init( &attr );\r
- pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );\r
-\r
-#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)\r
- if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {\r
- // We previously attempted to increase the audio callback priority\r
- // to SCHED_RR here via the attributes. However, while no errors\r
- // were reported in doing so, it did not work. So, now this is\r
- // done in the alsaCallbackHandler function.\r
- stream_.callbackInfo.doRealtime = true;\r
- int priority = options->priority;\r
- int min = sched_get_priority_min( SCHED_RR );\r
- int max = sched_get_priority_max( SCHED_RR );\r
- if ( priority < min ) priority = min;\r
- else if ( priority > max ) priority = max;\r
- stream_.callbackInfo.priority = priority;\r
- }\r
-#endif\r
-\r
- stream_.callbackInfo.isRunning = true;\r
- result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );\r
- pthread_attr_destroy( &attr );\r
- if ( result ) {\r
- stream_.callbackInfo.isRunning = false;\r
- errorText_ = "RtApiAlsa::error creating callback thread!";\r
- goto error;\r
- }\r
- }\r
-\r
- return SUCCESS;\r
-\r
- error:\r
- if ( apiInfo ) {\r
- pthread_cond_destroy( &apiInfo->runnable_cv );\r
- if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );\r
- if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );\r
- delete apiInfo;\r
- stream_.apiHandle = 0;\r
- }\r
-\r
- if ( phandle) snd_pcm_close( phandle );\r
-\r
- for ( int i=0; i<2; i++ ) {\r
- if ( stream_.userBuffer[i] ) {\r
- free( stream_.userBuffer[i] );\r
- stream_.userBuffer[i] = 0;\r
- }\r
- }\r
-\r
- if ( stream_.deviceBuffer ) {\r
- free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = 0;\r
- }\r
-\r
- stream_.state = STREAM_CLOSED;\r
- return FAILURE;\r
-}\r
-\r
-void RtApiAlsa :: closeStream()\r
-{\r
- if ( stream_.state == STREAM_CLOSED ) {\r
- errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;\r
- stream_.callbackInfo.isRunning = false;\r
- MUTEX_LOCK( &stream_.mutex );\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- apiInfo->runnable = true;\r
- pthread_cond_signal( &apiInfo->runnable_cv );\r
- }\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- pthread_join( stream_.callbackInfo.thread, NULL );\r
-\r
- if ( stream_.state == STREAM_RUNNING ) {\r
- stream_.state = STREAM_STOPPED;\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )\r
- snd_pcm_drop( apiInfo->handles[0] );\r
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX )\r
- snd_pcm_drop( apiInfo->handles[1] );\r
- }\r
-\r
- if ( apiInfo ) {\r
- pthread_cond_destroy( &apiInfo->runnable_cv );\r
- if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );\r
- if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );\r
- delete apiInfo;\r
- stream_.apiHandle = 0;\r
- }\r
-\r
- for ( int i=0; i<2; i++ ) {\r
- if ( stream_.userBuffer[i] ) {\r
- free( stream_.userBuffer[i] );\r
- stream_.userBuffer[i] = 0;\r
- }\r
- }\r
-\r
- if ( stream_.deviceBuffer ) {\r
- free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = 0;\r
- }\r
-\r
- stream_.mode = UNINITIALIZED;\r
- stream_.state = STREAM_CLOSED;\r
-}\r
-\r
-void RtApiAlsa :: startStream()\r
-{\r
- // This method calls snd_pcm_prepare if the device isn't already in that state.\r
-\r
- verifyStream();\r
- if ( stream_.state == STREAM_RUNNING ) {\r
- errorText_ = "RtApiAlsa::startStream(): the stream is already running!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- int result = 0;\r
- snd_pcm_state_t state;\r
- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;\r
- snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
- state = snd_pcm_state( handle[0] );\r
- if ( state != SND_PCM_STATE_PREPARED ) {\r
- result = snd_pcm_prepare( handle[0] );\r
- if ( result < 0 ) {\r
- errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
- }\r
- }\r
-\r
- if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {\r
- result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open\r
- state = snd_pcm_state( handle[1] );\r
- if ( state != SND_PCM_STATE_PREPARED ) {\r
- result = snd_pcm_prepare( handle[1] );\r
- if ( result < 0 ) {\r
- errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
- }\r
- }\r
-\r
- stream_.state = STREAM_RUNNING;\r
-\r
- unlock:\r
- apiInfo->runnable = true;\r
- pthread_cond_signal( &apiInfo->runnable_cv );\r
- MUTEX_UNLOCK( &stream_.mutex );\r
-\r
- if ( result >= 0 ) return;\r
- error( RtAudioError::SYSTEM_ERROR );\r
-}\r
-\r
-void RtApiAlsa :: stopStream()\r
-{\r
- verifyStream();\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- stream_.state = STREAM_STOPPED;\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- int result = 0;\r
- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;\r
- snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
- if ( apiInfo->synchronized ) \r
- result = snd_pcm_drop( handle[0] );\r
- else\r
- result = snd_pcm_drain( handle[0] );\r
- if ( result < 0 ) {\r
- errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
- }\r
-\r
- if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {\r
- result = snd_pcm_drop( handle[1] );\r
- if ( result < 0 ) {\r
- errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
- }\r
-\r
- unlock:\r
- apiInfo->runnable = false; // fixes high CPU usage when stopped\r
- MUTEX_UNLOCK( &stream_.mutex );\r
-\r
- if ( result >= 0 ) return;\r
- error( RtAudioError::SYSTEM_ERROR );\r
-}\r
-\r
-void RtApiAlsa :: abortStream()\r
-{\r
- verifyStream();\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- stream_.state = STREAM_STOPPED;\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- int result = 0;\r
- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;\r
- snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
- result = snd_pcm_drop( handle[0] );\r
- if ( result < 0 ) {\r
- errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
- }\r
-\r
- if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {\r
- result = snd_pcm_drop( handle[1] );\r
- if ( result < 0 ) {\r
- errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
- }\r
-\r
- unlock:\r
- apiInfo->runnable = false; // fixes high CPU usage when stopped\r
- MUTEX_UNLOCK( &stream_.mutex );\r
-\r
- if ( result >= 0 ) return;\r
- error( RtAudioError::SYSTEM_ERROR );\r
-}\r
-\r
-void RtApiAlsa :: callbackEvent()\r
-{\r
- AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- MUTEX_LOCK( &stream_.mutex );\r
- while ( !apiInfo->runnable )\r
- pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );\r
-\r
- if ( stream_.state != STREAM_RUNNING ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- return;\r
- }\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- }\r
-\r
- if ( stream_.state == STREAM_CLOSED ) {\r
- errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- int doStopStream = 0;\r
- RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;\r
- double streamTime = getStreamTime();\r
- RtAudioStreamStatus status = 0;\r
- if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {\r
- status |= RTAUDIO_OUTPUT_UNDERFLOW;\r
- apiInfo->xrun[0] = false;\r
- }\r
- if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {\r
- status |= RTAUDIO_INPUT_OVERFLOW;\r
- apiInfo->xrun[1] = false;\r
- }\r
- doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
- stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );\r
-\r
- if ( doStopStream == 2 ) {\r
- abortStream();\r
- return;\r
- }\r
-\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- // The state might change while waiting on a mutex.\r
- if ( stream_.state == STREAM_STOPPED ) goto unlock;\r
-\r
- int result;\r
- char *buffer;\r
- int channels;\r
- snd_pcm_t **handle;\r
- snd_pcm_sframes_t frames;\r
- RtAudioFormat format;\r
- handle = (snd_pcm_t **) apiInfo->handles;\r
-\r
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {\r
-\r
- // Setup parameters.\r
- if ( stream_.doConvertBuffer[1] ) {\r
- buffer = stream_.deviceBuffer;\r
- channels = stream_.nDeviceChannels[1];\r
- format = stream_.deviceFormat[1];\r
- }\r
- else {\r
- buffer = stream_.userBuffer[1];\r
- channels = stream_.nUserChannels[1];\r
- format = stream_.userFormat;\r
- }\r
-\r
- // Read samples from device in interleaved/non-interleaved format.\r
- if ( stream_.deviceInterleaved[1] )\r
- result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );\r
- else {\r
- void *bufs[channels];\r
- size_t offset = stream_.bufferSize * formatBytes( format );\r
- for ( int i=0; i<channels; i++ )\r
- bufs[i] = (void *) (buffer + (i * offset));\r
- result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );\r
- }\r
-\r
- if ( result < (int) stream_.bufferSize ) {\r
- // Either an error or overrun occured.\r
- if ( result == -EPIPE ) {\r
- snd_pcm_state_t state = snd_pcm_state( handle[1] );\r
- if ( state == SND_PCM_STATE_XRUN ) {\r
- apiInfo->xrun[1] = true;\r
- result = snd_pcm_prepare( handle[1] );\r
- if ( result < 0 ) {\r
- errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- }\r
- }\r
- else {\r
- errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- }\r
- }\r
- else {\r
- errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- }\r
- error( RtAudioError::WARNING );\r
- goto tryOutput;\r
- }\r
-\r
- // Do byte swapping if necessary.\r
- if ( stream_.doByteSwap[1] )\r
- byteSwapBuffer( buffer, stream_.bufferSize * channels, format );\r
-\r
- // Do buffer conversion if necessary.\r
- if ( stream_.doConvertBuffer[1] )\r
- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );\r
-\r
- // Check stream latency\r
- result = snd_pcm_delay( handle[1], &frames );\r
- if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;\r
- }\r
-\r
- tryOutput:\r
-\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
-\r
- // Setup parameters and do buffer conversion if necessary.\r
- if ( stream_.doConvertBuffer[0] ) {\r
- buffer = stream_.deviceBuffer;\r
- convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );\r
- channels = stream_.nDeviceChannels[0];\r
- format = stream_.deviceFormat[0];\r
- }\r
- else {\r
- buffer = stream_.userBuffer[0];\r
- channels = stream_.nUserChannels[0];\r
- format = stream_.userFormat;\r
- }\r
-\r
- // Do byte swapping if necessary.\r
- if ( stream_.doByteSwap[0] )\r
- byteSwapBuffer(buffer, stream_.bufferSize * channels, format);\r
-\r
- // Write samples to device in interleaved/non-interleaved format.\r
- if ( stream_.deviceInterleaved[0] )\r
- result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );\r
- else {\r
- void *bufs[channels];\r
- size_t offset = stream_.bufferSize * formatBytes( format );\r
- for ( int i=0; i<channels; i++ )\r
- bufs[i] = (void *) (buffer + (i * offset));\r
- result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );\r
- }\r
-\r
- if ( result < (int) stream_.bufferSize ) {\r
- // Either an error or underrun occured.\r
- if ( result == -EPIPE ) {\r
- snd_pcm_state_t state = snd_pcm_state( handle[0] );\r
- if ( state == SND_PCM_STATE_XRUN ) {\r
- apiInfo->xrun[0] = true;\r
- result = snd_pcm_prepare( handle[0] );\r
- if ( result < 0 ) {\r
- errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- }\r
- }\r
- else {\r
- errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- }\r
- }\r
- else {\r
- errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";\r
- errorText_ = errorStream_.str();\r
- }\r
- error( RtAudioError::WARNING );\r
- goto unlock;\r
- }\r
-\r
- // Check stream latency\r
- result = snd_pcm_delay( handle[0], &frames );\r
- if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;\r
- }\r
-\r
- unlock:\r
- MUTEX_UNLOCK( &stream_.mutex );\r
-\r
- RtApi::tickStreamTime();\r
- if ( doStopStream == 1 ) this->stopStream();\r
-}\r
-\r
-static void *alsaCallbackHandler( void *ptr )\r
-{\r
- CallbackInfo *info = (CallbackInfo *) ptr;\r
- RtApiAlsa *object = (RtApiAlsa *) info->object;\r
- bool *isRunning = &info->isRunning;\r
-\r
-#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)\r
- if ( &info->doRealtime ) {\r
- pthread_t tID = pthread_self(); // ID of this thread\r
- sched_param prio = { info->priority }; // scheduling priority of thread\r
- pthread_setschedparam( tID, SCHED_RR, &prio );\r
- }\r
-#endif\r
-\r
- while ( *isRunning == true ) {\r
- pthread_testcancel();\r
- object->callbackEvent();\r
- }\r
-\r
- pthread_exit( NULL );\r
-}\r
-\r
-//******************** End of __LINUX_ALSA__ *********************//\r
-#endif\r
-\r
-#if defined(__LINUX_PULSE__)\r
-\r
-// Code written by Peter Meerwald, pmeerw@pmeerw.net\r
-// and Tristan Matthews.\r
-\r
-#include <pulse/error.h>\r
-#include <pulse/simple.h>\r
-#include <cstdio>\r
-\r
-static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,\r
- 44100, 48000, 96000, 0};\r
-\r
-struct rtaudio_pa_format_mapping_t {\r
- RtAudioFormat rtaudio_format;\r
- pa_sample_format_t pa_format;\r
-};\r
-\r
-static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {\r
- {RTAUDIO_SINT16, PA_SAMPLE_S16LE},\r
- {RTAUDIO_SINT32, PA_SAMPLE_S32LE},\r
- {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},\r
- {0, PA_SAMPLE_INVALID}};\r
-\r
-struct PulseAudioHandle {\r
- pa_simple *s_play;\r
- pa_simple *s_rec;\r
- pthread_t thread;\r
- pthread_cond_t runnable_cv;\r
- bool runnable;\r
- PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }\r
-};\r
-\r
-RtApiPulse::~RtApiPulse()\r
-{\r
- if ( stream_.state != STREAM_CLOSED )\r
- closeStream();\r
-}\r
-\r
-unsigned int RtApiPulse::getDeviceCount( void )\r
-{\r
- return 1;\r
-}\r
-\r
-RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )\r
-{\r
- RtAudio::DeviceInfo info;\r
- info.probed = true;\r
- info.name = "PulseAudio";\r
- info.outputChannels = 2;\r
- info.inputChannels = 2;\r
- info.duplexChannels = 2;\r
- info.isDefaultOutput = true;\r
- info.isDefaultInput = true;\r
-\r
- for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )\r
- info.sampleRates.push_back( *sr );\r
-\r
- info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;\r
-\r
- return info;\r
-}\r
-\r
-static void *pulseaudio_callback( void * user )\r
-{\r
- CallbackInfo *cbi = static_cast<CallbackInfo *>( user );\r
- RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );\r
- volatile bool *isRunning = &cbi->isRunning;\r
-\r
- while ( *isRunning ) {\r
- pthread_testcancel();\r
- context->callbackEvent();\r
- }\r
-\r
- pthread_exit( NULL );\r
-}\r
-\r
-void RtApiPulse::closeStream( void )\r
-{\r
- PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );\r
-\r
- stream_.callbackInfo.isRunning = false;\r
- if ( pah ) {\r
- MUTEX_LOCK( &stream_.mutex );\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- pah->runnable = true;\r
- pthread_cond_signal( &pah->runnable_cv );\r
- }\r
- MUTEX_UNLOCK( &stream_.mutex );\r
-\r
- pthread_join( pah->thread, 0 );\r
- if ( pah->s_play ) {\r
- pa_simple_flush( pah->s_play, NULL );\r
- pa_simple_free( pah->s_play );\r
- }\r
- if ( pah->s_rec )\r
- pa_simple_free( pah->s_rec );\r
-\r
- pthread_cond_destroy( &pah->runnable_cv );\r
- delete pah;\r
- stream_.apiHandle = 0;\r
- }\r
-\r
- if ( stream_.userBuffer[0] ) {\r
- free( stream_.userBuffer[0] );\r
- stream_.userBuffer[0] = 0;\r
- }\r
- if ( stream_.userBuffer[1] ) {\r
- free( stream_.userBuffer[1] );\r
- stream_.userBuffer[1] = 0;\r
- }\r
-\r
- stream_.state = STREAM_CLOSED;\r
- stream_.mode = UNINITIALIZED;\r
-}\r
-\r
-void RtApiPulse::callbackEvent( void )\r
-{\r
- PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );\r
-\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- MUTEX_LOCK( &stream_.mutex );\r
- while ( !pah->runnable )\r
- pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );\r
-\r
- if ( stream_.state != STREAM_RUNNING ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- return;\r
- }\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- }\r
-\r
- if ( stream_.state == STREAM_CLOSED ) {\r
- errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "\r
- "this shouldn't happen!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;\r
- double streamTime = getStreamTime();\r
- RtAudioStreamStatus status = 0;\r
- int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],\r
- stream_.bufferSize, streamTime, status,\r
- stream_.callbackInfo.userData );\r
-\r
- if ( doStopStream == 2 ) {\r
- abortStream();\r
- return;\r
- }\r
-\r
- MUTEX_LOCK( &stream_.mutex );\r
- void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];\r
- void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];\r
-\r
- if ( stream_.state != STREAM_RUNNING )\r
- goto unlock;\r
-\r
- int pa_error;\r
- size_t bytes;\r
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
- if ( stream_.doConvertBuffer[OUTPUT] ) {\r
- convertBuffer( stream_.deviceBuffer,\r
- stream_.userBuffer[OUTPUT],\r
- stream_.convertInfo[OUTPUT] );\r
- bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *\r
- formatBytes( stream_.deviceFormat[OUTPUT] );\r
- } else\r
- bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *\r
- formatBytes( stream_.userFormat );\r
-\r
- if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {\r
- errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<\r
- pa_strerror( pa_error ) << ".";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- }\r
- }\r
-\r
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {\r
- if ( stream_.doConvertBuffer[INPUT] )\r
- bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *\r
- formatBytes( stream_.deviceFormat[INPUT] );\r
- else\r
- bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *\r
- formatBytes( stream_.userFormat );\r
- \r
- if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {\r
- errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<\r
- pa_strerror( pa_error ) << ".";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- }\r
- if ( stream_.doConvertBuffer[INPUT] ) {\r
- convertBuffer( stream_.userBuffer[INPUT],\r
- stream_.deviceBuffer,\r
- stream_.convertInfo[INPUT] );\r
- }\r
- }\r
-\r
- unlock:\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- RtApi::tickStreamTime();\r
-\r
- if ( doStopStream == 1 )\r
- stopStream();\r
-}\r
-\r
-void RtApiPulse::startStream( void )\r
-{\r
- PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );\r
-\r
- if ( stream_.state == STREAM_CLOSED ) {\r
- errorText_ = "RtApiPulse::startStream(): the stream is not open!";\r
- error( RtAudioError::INVALID_USE );\r
- return;\r
- }\r
- if ( stream_.state == STREAM_RUNNING ) {\r
- errorText_ = "RtApiPulse::startStream(): the stream is already running!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- stream_.state = STREAM_RUNNING;\r
-\r
- pah->runnable = true;\r
- pthread_cond_signal( &pah->runnable_cv );\r
- MUTEX_UNLOCK( &stream_.mutex );\r
-}\r
-\r
-void RtApiPulse::stopStream( void )\r
-{\r
- PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );\r
-\r
- if ( stream_.state == STREAM_CLOSED ) {\r
- errorText_ = "RtApiPulse::stopStream(): the stream is not open!";\r
- error( RtAudioError::INVALID_USE );\r
- return;\r
- }\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- stream_.state = STREAM_STOPPED;\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- if ( pah && pah->s_play ) {\r
- int pa_error;\r
- if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {\r
- errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<\r
- pa_strerror( pa_error ) << ".";\r
- errorText_ = errorStream_.str();\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- error( RtAudioError::SYSTEM_ERROR );\r
- return;\r
- }\r
- }\r
-\r
- stream_.state = STREAM_STOPPED;\r
- MUTEX_UNLOCK( &stream_.mutex );\r
-}\r
-\r
-void RtApiPulse::abortStream( void )\r
-{\r
- PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );\r
-\r
- if ( stream_.state == STREAM_CLOSED ) {\r
- errorText_ = "RtApiPulse::abortStream(): the stream is not open!";\r
- error( RtAudioError::INVALID_USE );\r
- return;\r
- }\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- stream_.state = STREAM_STOPPED;\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- if ( pah && pah->s_play ) {\r
- int pa_error;\r
- if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {\r
- errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<\r
- pa_strerror( pa_error ) << ".";\r
- errorText_ = errorStream_.str();\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- error( RtAudioError::SYSTEM_ERROR );\r
- return;\r
- }\r
- }\r
-\r
- stream_.state = STREAM_STOPPED;\r
- MUTEX_UNLOCK( &stream_.mutex );\r
-}\r
-\r
-bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,\r
- unsigned int channels, unsigned int firstChannel,\r
- unsigned int sampleRate, RtAudioFormat format,\r
- unsigned int *bufferSize, RtAudio::StreamOptions *options )\r
-{\r
- PulseAudioHandle *pah = 0;\r
- unsigned long bufferBytes = 0;\r
- pa_sample_spec ss;\r
-\r
- if ( device != 0 ) return false;\r
- if ( mode != INPUT && mode != OUTPUT ) return false;\r
- if ( channels != 1 && channels != 2 ) {\r
- errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";\r
- return false;\r
- }\r
- ss.channels = channels;\r
-\r
- if ( firstChannel != 0 ) return false;\r
-\r
- bool sr_found = false;\r
- for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {\r
- if ( sampleRate == *sr ) {\r
- sr_found = true;\r
- stream_.sampleRate = sampleRate;\r
- ss.rate = sampleRate;\r
- break;\r
- }\r
- }\r
- if ( !sr_found ) {\r
- errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";\r
- return false;\r
- }\r
-\r
- bool sf_found = 0;\r
- for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;\r
- sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {\r
- if ( format == sf->rtaudio_format ) {\r
- sf_found = true;\r
- stream_.userFormat = sf->rtaudio_format;\r
- stream_.deviceFormat[mode] = stream_.userFormat;\r
- ss.format = sf->pa_format;\r
- break;\r
- }\r
- }\r
- if ( !sf_found ) { // Use internal data format conversion.\r
- stream_.userFormat = format;\r
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;\r
- ss.format = PA_SAMPLE_FLOAT32LE;\r
- }\r
-\r
- // Set other stream parameters.\r
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;\r
- else stream_.userInterleaved = true;\r
- stream_.deviceInterleaved[mode] = true;\r
- stream_.nBuffers = 1;\r
- stream_.doByteSwap[mode] = false;\r
- stream_.nUserChannels[mode] = channels;\r
- stream_.nDeviceChannels[mode] = channels + firstChannel;\r
- stream_.channelOffset[mode] = 0;\r
- std::string streamName = "RtAudio";\r
-\r
- // Set flags for buffer conversion.\r
- stream_.doConvertBuffer[mode] = false;\r
- if ( stream_.userFormat != stream_.deviceFormat[mode] )\r
- stream_.doConvertBuffer[mode] = true;\r
- if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )\r
- stream_.doConvertBuffer[mode] = true;\r
-\r
- // Allocate necessary internal buffers.\r
- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );\r
- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );\r
- if ( stream_.userBuffer[mode] == NULL ) {\r
- errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";\r
- goto error;\r
- }\r
- stream_.bufferSize = *bufferSize;\r
-\r
- if ( stream_.doConvertBuffer[mode] ) {\r
-\r
- bool makeBuffer = true;\r
- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );\r
- if ( mode == INPUT ) {\r
- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {\r
- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );\r
- if ( bufferBytes <= bytesOut ) makeBuffer = false;\r
- }\r
- }\r
-\r
- if ( makeBuffer ) {\r
- bufferBytes *= *bufferSize;\r
- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );\r
- if ( stream_.deviceBuffer == NULL ) {\r
- errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";\r
- goto error;\r
- }\r
- }\r
- }\r
-\r
- stream_.device[mode] = device;\r
-\r
- // Setup the buffer conversion information structure.\r
- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );\r
-\r
- if ( !stream_.apiHandle ) {\r
- PulseAudioHandle *pah = new PulseAudioHandle;\r
- if ( !pah ) {\r
- errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";\r
- goto error;\r
- }\r
-\r
- stream_.apiHandle = pah;\r
- if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {\r
- errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";\r
- goto error;\r
- }\r
- }\r
- pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );\r
-\r
- int error;\r
- if ( !options->streamName.empty() ) streamName = options->streamName;\r
- switch ( mode ) {\r
- case INPUT:\r
- pa_buffer_attr buffer_attr;\r
- buffer_attr.fragsize = bufferBytes;\r
- buffer_attr.maxlength = -1;\r
-\r
- pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );\r
- if ( !pah->s_rec ) {\r
- errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";\r
- goto error;\r
- }\r
- break;\r
- case OUTPUT:\r
- pah->s_play = pa_simple_new( NULL, "RtAudio", PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );\r
- if ( !pah->s_play ) {\r
- errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";\r
- goto error;\r
- }\r
- break;\r
- default:\r
- goto error;\r
- }\r
-\r
- if ( stream_.mode == UNINITIALIZED )\r
- stream_.mode = mode;\r
- else if ( stream_.mode == mode )\r
- goto error;\r
- else\r
- stream_.mode = DUPLEX;\r
-\r
- if ( !stream_.callbackInfo.isRunning ) {\r
- stream_.callbackInfo.object = this;\r
- stream_.callbackInfo.isRunning = true;\r
- if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) {\r
- errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";\r
- goto error;\r
- }\r
- }\r
-\r
- stream_.state = STREAM_STOPPED;\r
- return true;\r
- \r
- error:\r
- if ( pah && stream_.callbackInfo.isRunning ) {\r
- pthread_cond_destroy( &pah->runnable_cv );\r
- delete pah;\r
- stream_.apiHandle = 0;\r
- }\r
-\r
- for ( int i=0; i<2; i++ ) {\r
- if ( stream_.userBuffer[i] ) {\r
- free( stream_.userBuffer[i] );\r
- stream_.userBuffer[i] = 0;\r
- }\r
- }\r
-\r
- if ( stream_.deviceBuffer ) {\r
- free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = 0;\r
- }\r
-\r
- return FAILURE;\r
-}\r
-\r
-//******************** End of __LINUX_PULSE__ *********************//\r
-#endif\r
-\r
-#if defined(__LINUX_OSS__)\r
-\r
-#include <unistd.h>\r
-#include <sys/ioctl.h>\r
-#include <unistd.h>\r
-#include <fcntl.h>\r
-#include <sys/soundcard.h>\r
-#include <errno.h>\r
-#include <math.h>\r
-\r
-static void *ossCallbackHandler(void * ptr);\r
-\r
-// A structure to hold various information related to the OSS API\r
-// implementation.\r
-struct OssHandle {\r
- int id[2]; // device ids\r
- bool xrun[2];\r
- bool triggered;\r
- pthread_cond_t runnable;\r
-\r
- OssHandle()\r
- :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }\r
-};\r
-\r
-RtApiOss :: RtApiOss()\r
-{\r
- // Nothing to do here.\r
-}\r
-\r
-RtApiOss :: ~RtApiOss()\r
-{\r
- if ( stream_.state != STREAM_CLOSED ) closeStream();\r
-}\r
-\r
-unsigned int RtApiOss :: getDeviceCount( void )\r
-{\r
- int mixerfd = open( "/dev/mixer", O_RDWR, 0 );\r
- if ( mixerfd == -1 ) {\r
- errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";\r
- error( RtAudioError::WARNING );\r
- return 0;\r
- }\r
-\r
- oss_sysinfo sysinfo;\r
- if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {\r
- close( mixerfd );\r
- errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";\r
- error( RtAudioError::WARNING );\r
- return 0;\r
- }\r
-\r
- close( mixerfd );\r
- return sysinfo.numaudios;\r
-}\r
-\r
-RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )\r
-{\r
- RtAudio::DeviceInfo info;\r
- info.probed = false;\r
-\r
- int mixerfd = open( "/dev/mixer", O_RDWR, 0 );\r
- if ( mixerfd == -1 ) {\r
- errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- oss_sysinfo sysinfo;\r
- int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );\r
- if ( result == -1 ) {\r
- close( mixerfd );\r
- errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- unsigned nDevices = sysinfo.numaudios;\r
- if ( nDevices == 0 ) {\r
- close( mixerfd );\r
- errorText_ = "RtApiOss::getDeviceInfo: no devices found!";\r
- error( RtAudioError::INVALID_USE );\r
- return info;\r
- }\r
-\r
- if ( device >= nDevices ) {\r
- close( mixerfd );\r
- errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";\r
- error( RtAudioError::INVALID_USE );\r
- return info;\r
- }\r
-\r
- oss_audioinfo ainfo;\r
- ainfo.dev = device;\r
- result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );\r
- close( mixerfd );\r
- if ( result == -1 ) {\r
- errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- // Probe channels\r
- if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;\r
- if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;\r
- if ( ainfo.caps & PCM_CAP_DUPLEX ) {\r
- if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )\r
- info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;\r
- }\r
-\r
- // Probe data formats ... do for input\r
- unsigned long mask = ainfo.iformats;\r
- if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )\r
- info.nativeFormats |= RTAUDIO_SINT16;\r
- if ( mask & AFMT_S8 )\r
- info.nativeFormats |= RTAUDIO_SINT8;\r
- if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )\r
- info.nativeFormats |= RTAUDIO_SINT32;\r
- if ( mask & AFMT_FLOAT )\r
- info.nativeFormats |= RTAUDIO_FLOAT32;\r
- if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )\r
- info.nativeFormats |= RTAUDIO_SINT24;\r
-\r
- // Check that we have at least one supported format\r
- if ( info.nativeFormats == 0 ) {\r
- errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- return info;\r
- }\r
-\r
- // Probe the supported sample rates.\r
- info.sampleRates.clear();\r
- if ( ainfo.nrates ) {\r
- for ( unsigned int i=0; i<ainfo.nrates; i++ ) {\r
- for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {\r
- if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {\r
- info.sampleRates.push_back( SAMPLE_RATES[k] );\r
- break;\r
- }\r
- }\r
- }\r
- }\r
- else {\r
- // Check min and max rate values;\r
- for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {\r
- if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] )\r
- info.sampleRates.push_back( SAMPLE_RATES[k] );\r
- }\r
- }\r
-\r
- if ( info.sampleRates.size() == 0 ) {\r
- errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";\r
- errorText_ = errorStream_.str();\r
- error( RtAudioError::WARNING );\r
- }\r
- else {\r
- info.probed = true;\r
- info.name = ainfo.name;\r
- }\r
-\r
- return info;\r
-}\r
-\r
-\r
-bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,\r
- unsigned int firstChannel, unsigned int sampleRate,\r
- RtAudioFormat format, unsigned int *bufferSize,\r
- RtAudio::StreamOptions *options )\r
-{\r
- int mixerfd = open( "/dev/mixer", O_RDWR, 0 );\r
- if ( mixerfd == -1 ) {\r
- errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";\r
- return FAILURE;\r
- }\r
-\r
- oss_sysinfo sysinfo;\r
- int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );\r
- if ( result == -1 ) {\r
- close( mixerfd );\r
- errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";\r
- return FAILURE;\r
- }\r
-\r
- unsigned nDevices = sysinfo.numaudios;\r
- if ( nDevices == 0 ) {\r
- // This should not happen because a check is made before this function is called.\r
- close( mixerfd );\r
- errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";\r
- return FAILURE;\r
- }\r
-\r
- if ( device >= nDevices ) {\r
- // This should not happen because a check is made before this function is called.\r
- close( mixerfd );\r
- errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";\r
- return FAILURE;\r
- }\r
-\r
- oss_audioinfo ainfo;\r
- ainfo.dev = device;\r
- result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );\r
- close( mixerfd );\r
- if ( result == -1 ) {\r
- errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Check if device supports input or output\r
- if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||\r
- ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {\r
- if ( mode == OUTPUT )\r
- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";\r
- else\r
- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- int flags = 0;\r
- OssHandle *handle = (OssHandle *) stream_.apiHandle;\r
- if ( mode == OUTPUT )\r
- flags |= O_WRONLY;\r
- else { // mode == INPUT\r
- if (stream_.mode == OUTPUT && stream_.device[0] == device) {\r
- // We just set the same device for playback ... close and reopen for duplex (OSS only).\r
- close( handle->id[0] );\r
- handle->id[0] = 0;\r
- if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {\r
- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- // Check that the number previously set channels is the same.\r
- if ( stream_.nUserChannels[0] != channels ) {\r
- errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- flags |= O_RDWR;\r
- }\r
- else\r
- flags |= O_RDONLY;\r
- }\r
-\r
- // Set exclusive access if specified.\r
- if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;\r
-\r
- // Try to open the device.\r
- int fd;\r
- fd = open( ainfo.devnode, flags, 0 );\r
- if ( fd == -1 ) {\r
- if ( errno == EBUSY )\r
- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";\r
- else\r
- errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // For duplex operation, specifically set this mode (this doesn't seem to work).\r
- /*\r
- if ( flags | O_RDWR ) {\r
- result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );\r
- if ( result == -1) {\r
- errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- }\r
- */\r
-\r
- // Check the device channel support.\r
- stream_.nUserChannels[mode] = channels;\r
- if ( ainfo.max_channels < (int)(channels + firstChannel) ) {\r
- close( fd );\r
- errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Set the number of channels.\r
- int deviceChannels = channels + firstChannel;\r
- result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );\r
- if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {\r
- close( fd );\r
- errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- stream_.nDeviceChannels[mode] = deviceChannels;\r
-\r
- // Get the data format mask\r
- int mask;\r
- result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );\r
- if ( result == -1 ) {\r
- close( fd );\r
- errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Determine how to set the device format.\r
- stream_.userFormat = format;\r
- int deviceFormat = -1;\r
- stream_.doByteSwap[mode] = false;\r
- if ( format == RTAUDIO_SINT8 ) {\r
- if ( mask & AFMT_S8 ) {\r
- deviceFormat = AFMT_S8;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT8;\r
- }\r
- }\r
- else if ( format == RTAUDIO_SINT16 ) {\r
- if ( mask & AFMT_S16_NE ) {\r
- deviceFormat = AFMT_S16_NE;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;\r
- }\r
- else if ( mask & AFMT_S16_OE ) {\r
- deviceFormat = AFMT_S16_OE;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;\r
- stream_.doByteSwap[mode] = true;\r
- }\r
- }\r
- else if ( format == RTAUDIO_SINT24 ) {\r
- if ( mask & AFMT_S24_NE ) {\r
- deviceFormat = AFMT_S24_NE;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT24;\r
- }\r
- else if ( mask & AFMT_S24_OE ) {\r
- deviceFormat = AFMT_S24_OE;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT24;\r
- stream_.doByteSwap[mode] = true;\r
- }\r
- }\r
- else if ( format == RTAUDIO_SINT32 ) {\r
- if ( mask & AFMT_S32_NE ) {\r
- deviceFormat = AFMT_S32_NE;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;\r
- }\r
- else if ( mask & AFMT_S32_OE ) {\r
- deviceFormat = AFMT_S32_OE;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;\r
- stream_.doByteSwap[mode] = true;\r
- }\r
- }\r
-\r
- if ( deviceFormat == -1 ) {\r
- // The user requested format is not natively supported by the device.\r
- if ( mask & AFMT_S16_NE ) {\r
- deviceFormat = AFMT_S16_NE;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;\r
- }\r
- else if ( mask & AFMT_S32_NE ) {\r
- deviceFormat = AFMT_S32_NE;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;\r
- }\r
- else if ( mask & AFMT_S24_NE ) {\r
- deviceFormat = AFMT_S24_NE;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT24;\r
- }\r
- else if ( mask & AFMT_S16_OE ) {\r
- deviceFormat = AFMT_S16_OE;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT16;\r
- stream_.doByteSwap[mode] = true;\r
- }\r
- else if ( mask & AFMT_S32_OE ) {\r
- deviceFormat = AFMT_S32_OE;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;\r
- stream_.doByteSwap[mode] = true;\r
- }\r
- else if ( mask & AFMT_S24_OE ) {\r
- deviceFormat = AFMT_S24_OE;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT24;\r
- stream_.doByteSwap[mode] = true;\r
- }\r
- else if ( mask & AFMT_S8) {\r
- deviceFormat = AFMT_S8;\r
- stream_.deviceFormat[mode] = RTAUDIO_SINT8;\r
- }\r
- }\r
-\r
- if ( stream_.deviceFormat[mode] == 0 ) {\r
- // This really shouldn't happen ...\r
- close( fd );\r
- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Set the data format.\r
- int temp = deviceFormat;\r
- result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );\r
- if ( result == -1 || deviceFormat != temp ) {\r
- close( fd );\r
- errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Attempt to set the buffer size. According to OSS, the minimum\r
- // number of buffers is two. The supposed minimum buffer size is 16\r
- // bytes, so that will be our lower bound. The argument to this\r
- // call is in the form 0xMMMMSSSS (hex), where the buffer size (in\r
- // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.\r
- // We'll check the actual value used near the end of the setup\r
- // procedure.\r
- int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;\r
- if ( ossBufferBytes < 16 ) ossBufferBytes = 16;\r
- int buffers = 0;\r
- if ( options ) buffers = options->numberOfBuffers;\r
- if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;\r
- if ( buffers < 2 ) buffers = 3;\r
- temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );\r
- result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );\r
- if ( result == -1 ) {\r
- close( fd );\r
- errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- stream_.nBuffers = buffers;\r
-\r
- // Save buffer size (in sample frames).\r
- *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );\r
- stream_.bufferSize = *bufferSize;\r
-\r
- // Set the sample rate.\r
- int srate = sampleRate;\r
- result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );\r
- if ( result == -1 ) {\r
- close( fd );\r
- errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
-\r
- // Verify the sample rate setup worked.\r
- if ( abs( srate - sampleRate ) > 100 ) {\r
- close( fd );\r
- errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";\r
- errorText_ = errorStream_.str();\r
- return FAILURE;\r
- }\r
- stream_.sampleRate = sampleRate;\r
-\r
- if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {\r
- // We're doing duplex setup here.\r
- stream_.deviceFormat[0] = stream_.deviceFormat[1];\r
- stream_.nDeviceChannels[0] = deviceChannels;\r
- }\r
-\r
- // Set interleaving parameters.\r
- stream_.userInterleaved = true;\r
- stream_.deviceInterleaved[mode] = true;\r
- if ( options && options->flags & RTAUDIO_NONINTERLEAVED )\r
- stream_.userInterleaved = false;\r
-\r
- // Set flags for buffer conversion\r
- stream_.doConvertBuffer[mode] = false;\r
- if ( stream_.userFormat != stream_.deviceFormat[mode] )\r
- stream_.doConvertBuffer[mode] = true;\r
- if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )\r
- stream_.doConvertBuffer[mode] = true;\r
- if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&\r
- stream_.nUserChannels[mode] > 1 )\r
- stream_.doConvertBuffer[mode] = true;\r
-\r
- // Allocate the stream handles if necessary and then save.\r
- if ( stream_.apiHandle == 0 ) {\r
- try {\r
- handle = new OssHandle;\r
- }\r
- catch ( std::bad_alloc& ) {\r
- errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";\r
- goto error;\r
- }\r
-\r
- if ( pthread_cond_init( &handle->runnable, NULL ) ) {\r
- errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";\r
- goto error;\r
- }\r
-\r
- stream_.apiHandle = (void *) handle;\r
- }\r
- else {\r
- handle = (OssHandle *) stream_.apiHandle;\r
- }\r
- handle->id[mode] = fd;\r
-\r
- // Allocate necessary internal buffers.\r
- unsigned long bufferBytes;\r
- bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );\r
- stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );\r
- if ( stream_.userBuffer[mode] == NULL ) {\r
- errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";\r
- goto error;\r
- }\r
-\r
- if ( stream_.doConvertBuffer[mode] ) {\r
-\r
- bool makeBuffer = true;\r
- bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );\r
- if ( mode == INPUT ) {\r
- if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {\r
- unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );\r
- if ( bufferBytes <= bytesOut ) makeBuffer = false;\r
- }\r
- }\r
-\r
- if ( makeBuffer ) {\r
- bufferBytes *= *bufferSize;\r
- if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );\r
- if ( stream_.deviceBuffer == NULL ) {\r
- errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";\r
- goto error;\r
- }\r
- }\r
- }\r
-\r
- stream_.device[mode] = device;\r
- stream_.state = STREAM_STOPPED;\r
-\r
- // Setup the buffer conversion information structure.\r
- if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );\r
-\r
- // Setup thread if necessary.\r
- if ( stream_.mode == OUTPUT && mode == INPUT ) {\r
- // We had already set up an output stream.\r
- stream_.mode = DUPLEX;\r
- if ( stream_.device[0] == device ) handle->id[0] = fd;\r
- }\r
- else {\r
- stream_.mode = mode;\r
-\r
- // Setup callback thread.\r
- stream_.callbackInfo.object = (void *) this;\r
-\r
- // Set the thread attributes for joinable and realtime scheduling\r
- // priority. The higher priority will only take affect if the\r
- // program is run as root or suid.\r
- pthread_attr_t attr;\r
- pthread_attr_init( &attr );\r
- pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );\r
-#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)\r
- if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {\r
- struct sched_param param;\r
- int priority = options->priority;\r
- int min = sched_get_priority_min( SCHED_RR );\r
- int max = sched_get_priority_max( SCHED_RR );\r
- if ( priority < min ) priority = min;\r
- else if ( priority > max ) priority = max;\r
- param.sched_priority = priority;\r
- pthread_attr_setschedparam( &attr, ¶m );\r
- pthread_attr_setschedpolicy( &attr, SCHED_RR );\r
- }\r
- else\r
- pthread_attr_setschedpolicy( &attr, SCHED_OTHER );\r
-#else\r
- pthread_attr_setschedpolicy( &attr, SCHED_OTHER );\r
-#endif\r
-\r
- stream_.callbackInfo.isRunning = true;\r
- result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );\r
- pthread_attr_destroy( &attr );\r
- if ( result ) {\r
- stream_.callbackInfo.isRunning = false;\r
- errorText_ = "RtApiOss::error creating callback thread!";\r
- goto error;\r
- }\r
- }\r
-\r
- return SUCCESS;\r
-\r
- error:\r
- if ( handle ) {\r
- pthread_cond_destroy( &handle->runnable );\r
- if ( handle->id[0] ) close( handle->id[0] );\r
- if ( handle->id[1] ) close( handle->id[1] );\r
- delete handle;\r
- stream_.apiHandle = 0;\r
- }\r
-\r
- for ( int i=0; i<2; i++ ) {\r
- if ( stream_.userBuffer[i] ) {\r
- free( stream_.userBuffer[i] );\r
- stream_.userBuffer[i] = 0;\r
- }\r
- }\r
-\r
- if ( stream_.deviceBuffer ) {\r
- free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = 0;\r
- }\r
-\r
- return FAILURE;\r
-}\r
-\r
-void RtApiOss :: closeStream()\r
-{\r
- if ( stream_.state == STREAM_CLOSED ) {\r
- errorText_ = "RtApiOss::closeStream(): no open stream to close!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- OssHandle *handle = (OssHandle *) stream_.apiHandle;\r
- stream_.callbackInfo.isRunning = false;\r
- MUTEX_LOCK( &stream_.mutex );\r
- if ( stream_.state == STREAM_STOPPED )\r
- pthread_cond_signal( &handle->runnable );\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- pthread_join( stream_.callbackInfo.thread, NULL );\r
-\r
- if ( stream_.state == STREAM_RUNNING ) {\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )\r
- ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );\r
- else\r
- ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );\r
- stream_.state = STREAM_STOPPED;\r
- }\r
-\r
- if ( handle ) {\r
- pthread_cond_destroy( &handle->runnable );\r
- if ( handle->id[0] ) close( handle->id[0] );\r
- if ( handle->id[1] ) close( handle->id[1] );\r
- delete handle;\r
- stream_.apiHandle = 0;\r
- }\r
-\r
- for ( int i=0; i<2; i++ ) {\r
- if ( stream_.userBuffer[i] ) {\r
- free( stream_.userBuffer[i] );\r
- stream_.userBuffer[i] = 0;\r
- }\r
- }\r
-\r
- if ( stream_.deviceBuffer ) {\r
- free( stream_.deviceBuffer );\r
- stream_.deviceBuffer = 0;\r
- }\r
-\r
- stream_.mode = UNINITIALIZED;\r
- stream_.state = STREAM_CLOSED;\r
-}\r
-\r
-void RtApiOss :: startStream()\r
-{\r
- verifyStream();\r
- if ( stream_.state == STREAM_RUNNING ) {\r
- errorText_ = "RtApiOss::startStream(): the stream is already running!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- stream_.state = STREAM_RUNNING;\r
-\r
- // No need to do anything else here ... OSS automatically starts\r
- // when fed samples.\r
-\r
- MUTEX_UNLOCK( &stream_.mutex );\r
-\r
- OssHandle *handle = (OssHandle *) stream_.apiHandle;\r
- pthread_cond_signal( &handle->runnable );\r
-}\r
-\r
-void RtApiOss :: stopStream()\r
-{\r
- verifyStream();\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- // The state might change while waiting on a mutex.\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- return;\r
- }\r
-\r
- int result = 0;\r
- OssHandle *handle = (OssHandle *) stream_.apiHandle;\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
-\r
- // Flush the output with zeros a few times.\r
- char *buffer;\r
- int samples;\r
- RtAudioFormat format;\r
-\r
- if ( stream_.doConvertBuffer[0] ) {\r
- buffer = stream_.deviceBuffer;\r
- samples = stream_.bufferSize * stream_.nDeviceChannels[0];\r
- format = stream_.deviceFormat[0];\r
- }\r
- else {\r
- buffer = stream_.userBuffer[0];\r
- samples = stream_.bufferSize * stream_.nUserChannels[0];\r
- format = stream_.userFormat;\r
- }\r
-\r
- memset( buffer, 0, samples * formatBytes(format) );\r
- for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {\r
- result = write( handle->id[0], buffer, samples * formatBytes(format) );\r
- if ( result == -1 ) {\r
- errorText_ = "RtApiOss::stopStream: audio write error.";\r
- error( RtAudioError::WARNING );\r
- }\r
- }\r
-\r
- result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );\r
- if ( result == -1 ) {\r
- errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
- handle->triggered = false;\r
- }\r
-\r
- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {\r
- result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );\r
- if ( result == -1 ) {\r
- errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
- }\r
-\r
- unlock:\r
- stream_.state = STREAM_STOPPED;\r
- MUTEX_UNLOCK( &stream_.mutex );\r
-\r
- if ( result != -1 ) return;\r
- error( RtAudioError::SYSTEM_ERROR );\r
-}\r
-\r
-void RtApiOss :: abortStream()\r
-{\r
- verifyStream();\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- // The state might change while waiting on a mutex.\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- return;\r
- }\r
-\r
- int result = 0;\r
- OssHandle *handle = (OssHandle *) stream_.apiHandle;\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
- result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );\r
- if ( result == -1 ) {\r
- errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
- handle->triggered = false;\r
- }\r
-\r
- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {\r
- result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );\r
- if ( result == -1 ) {\r
- errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";\r
- errorText_ = errorStream_.str();\r
- goto unlock;\r
- }\r
- }\r
-\r
- unlock:\r
- stream_.state = STREAM_STOPPED;\r
- MUTEX_UNLOCK( &stream_.mutex );\r
-\r
- if ( result != -1 ) return;\r
- error( RtAudioError::SYSTEM_ERROR );\r
-}\r
-\r
-void RtApiOss :: callbackEvent()\r
-{\r
- OssHandle *handle = (OssHandle *) stream_.apiHandle;\r
- if ( stream_.state == STREAM_STOPPED ) {\r
- MUTEX_LOCK( &stream_.mutex );\r
- pthread_cond_wait( &handle->runnable, &stream_.mutex );\r
- if ( stream_.state != STREAM_RUNNING ) {\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- return;\r
- }\r
- MUTEX_UNLOCK( &stream_.mutex );\r
- }\r
-\r
- if ( stream_.state == STREAM_CLOSED ) {\r
- errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";\r
- error( RtAudioError::WARNING );\r
- return;\r
- }\r
-\r
- // Invoke user callback to get fresh output data.\r
- int doStopStream = 0;\r
- RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;\r
- double streamTime = getStreamTime();\r
- RtAudioStreamStatus status = 0;\r
- if ( stream_.mode != INPUT && handle->xrun[0] == true ) {\r
- status |= RTAUDIO_OUTPUT_UNDERFLOW;\r
- handle->xrun[0] = false;\r
- }\r
- if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {\r
- status |= RTAUDIO_INPUT_OVERFLOW;\r
- handle->xrun[1] = false;\r
- }\r
- doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],\r
- stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );\r
- if ( doStopStream == 2 ) {\r
- this->abortStream();\r
- return;\r
- }\r
-\r
- MUTEX_LOCK( &stream_.mutex );\r
-\r
- // The state might change while waiting on a mutex.\r
- if ( stream_.state == STREAM_STOPPED ) goto unlock;\r
-\r
- int result;\r
- char *buffer;\r
- int samples;\r
- RtAudioFormat format;\r
-\r
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {\r
-\r
- // Setup parameters and do buffer conversion if necessary.\r
- if ( stream_.doConvertBuffer[0] ) {\r
- buffer = stream_.deviceBuffer;\r
- convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );\r
- samples = stream_.bufferSize * stream_.nDeviceChannels[0];\r
- format = stream_.deviceFormat[0];\r
- }\r
- else {\r
- buffer = stream_.userBuffer[0];\r
- samples = stream_.bufferSize * stream_.nUserChannels[0];\r
- format = stream_.userFormat;\r
- }\r
-\r
- // Do byte swapping if necessary.\r
- if ( stream_.doByteSwap[0] )\r
- byteSwapBuffer( buffer, samples, format );\r
-\r
- if ( stream_.mode == DUPLEX && handle->triggered == false ) {\r
- int trig = 0;\r
- ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );\r
- result = write( handle->id[0], buffer, samples * formatBytes(format) );\r
- trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;\r
- ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );\r
- handle->triggered = true;\r
- }\r
- else\r
- // Write samples to device.\r
- result = write( handle->id[0], buffer, samples * formatBytes(format) );\r
-\r
- if ( result == -1 ) {\r
- // We'll assume this is an underrun, though there isn't a\r
- // specific means for determining that.\r
- handle->xrun[0] = true;\r
- errorText_ = "RtApiOss::callbackEvent: audio write error.";\r
- error( RtAudioError::WARNING );\r
- // Continue on to input section.\r
- }\r
- }\r
-\r
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {\r
-\r
- // Setup parameters.\r
- if ( stream_.doConvertBuffer[1] ) {\r
- buffer = stream_.deviceBuffer;\r
- samples = stream_.bufferSize * stream_.nDeviceChannels[1];\r
- format = stream_.deviceFormat[1];\r
- }\r
- else {\r
- buffer = stream_.userBuffer[1];\r
- samples = stream_.bufferSize * stream_.nUserChannels[1];\r
- format = stream_.userFormat;\r
- }\r
-\r
- // Read samples from device.\r
- result = read( handle->id[1], buffer, samples * formatBytes(format) );\r
-\r
- if ( result == -1 ) {\r
- // We'll assume this is an overrun, though there isn't a\r
- // specific means for determining that.\r
- handle->xrun[1] = true;\r
- errorText_ = "RtApiOss::callbackEvent: audio read error.";\r
- error( RtAudioError::WARNING );\r
- goto unlock;\r
- }\r
-\r
- // Do byte swapping if necessary.\r
- if ( stream_.doByteSwap[1] )\r
- byteSwapBuffer( buffer, samples, format );\r
-\r
- // Do buffer conversion if necessary.\r
- if ( stream_.doConvertBuffer[1] )\r
- convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );\r
- }\r
-\r
- unlock:\r
- MUTEX_UNLOCK( &stream_.mutex );\r
-\r
- RtApi::tickStreamTime();\r
- if ( doStopStream == 1 ) this->stopStream();\r
-}\r
-\r
-static void *ossCallbackHandler( void *ptr )\r
-{\r
- CallbackInfo *info = (CallbackInfo *) ptr;\r
- RtApiOss *object = (RtApiOss *) info->object;\r
- bool *isRunning = &info->isRunning;\r
-\r
- while ( *isRunning == true ) {\r
- pthread_testcancel();\r
- object->callbackEvent();\r
- }\r
-\r
- pthread_exit( NULL );\r
-}\r
-\r
-//******************** End of __LINUX_OSS__ *********************//\r
-#endif\r
-\r
-\r
-// *************************************************** //\r
-//\r
-// Protected common (OS-independent) RtAudio methods.\r
-//\r
-// *************************************************** //\r
-\r
-// This method can be modified to control the behavior of error\r
-// message printing.\r
-void RtApi :: error( RtAudioError::Type type )\r
-{\r
- errorStream_.str(""); // clear the ostringstream\r
-\r
- RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;\r
- if ( errorCallback ) {\r
- // abortStream() can generate new error messages. Ignore them. Just keep original one.\r
-\r
- if ( firstErrorOccurred_ )\r
- return;\r
-\r
- firstErrorOccurred_ = true;\r
- const std::string errorMessage = errorText_;\r
-\r
- if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {\r
- stream_.callbackInfo.isRunning = false; // exit from the thread\r
- abortStream();\r
- }\r
-\r
- errorCallback( type, errorMessage );\r
- firstErrorOccurred_ = false;\r
- return;\r
- }\r
-\r
- if ( type == RtAudioError::WARNING && showWarnings_ == true )\r
- std::cerr << '\n' << errorText_ << "\n\n";\r
- else if ( type != RtAudioError::WARNING )\r
- throw( RtAudioError( errorText_, type ) );\r
-}\r
-\r
-void RtApi :: verifyStream()\r
-{\r
- if ( stream_.state == STREAM_CLOSED ) {\r
- errorText_ = "RtApi:: a stream is not open!";\r
- error( RtAudioError::INVALID_USE );\r
- }\r
-}\r
-\r
-void RtApi :: clearStreamInfo()\r
-{\r
- stream_.mode = UNINITIALIZED;\r
- stream_.state = STREAM_CLOSED;\r
- stream_.sampleRate = 0;\r
- stream_.bufferSize = 0;\r
- stream_.nBuffers = 0;\r
- stream_.userFormat = 0;\r
- stream_.userInterleaved = true;\r
- stream_.streamTime = 0.0;\r
- stream_.apiHandle = 0;\r
- stream_.deviceBuffer = 0;\r
- stream_.callbackInfo.callback = 0;\r
- stream_.callbackInfo.userData = 0;\r
- stream_.callbackInfo.isRunning = false;\r
- stream_.callbackInfo.errorCallback = 0;\r
- for ( int i=0; i<2; i++ ) {\r
- stream_.device[i] = 11111;\r
- stream_.doConvertBuffer[i] = false;\r
- stream_.deviceInterleaved[i] = true;\r
- stream_.doByteSwap[i] = false;\r
- stream_.nUserChannels[i] = 0;\r
- stream_.nDeviceChannels[i] = 0;\r
- stream_.channelOffset[i] = 0;\r
- stream_.deviceFormat[i] = 0;\r
- stream_.latency[i] = 0;\r
- stream_.userBuffer[i] = 0;\r
- stream_.convertInfo[i].channels = 0;\r
- stream_.convertInfo[i].inJump = 0;\r
- stream_.convertInfo[i].outJump = 0;\r
- stream_.convertInfo[i].inFormat = 0;\r
- stream_.convertInfo[i].outFormat = 0;\r
- stream_.convertInfo[i].inOffset.clear();\r
- stream_.convertInfo[i].outOffset.clear();\r
- }\r
-}\r
-\r
-unsigned int RtApi :: formatBytes( RtAudioFormat format )\r
-{\r
- if ( format == RTAUDIO_SINT16 )\r
- return 2;\r
- else if ( format == RTAUDIO_SINT32 || format == RTAUDIO_FLOAT32 )\r
- return 4;\r
- else if ( format == RTAUDIO_FLOAT64 )\r
- return 8;\r
- else if ( format == RTAUDIO_SINT24 )\r
- return 3;\r
- else if ( format == RTAUDIO_SINT8 )\r
- return 1;\r
-\r
- errorText_ = "RtApi::formatBytes: undefined format.";\r
- error( RtAudioError::WARNING );\r
-\r
- return 0;\r
-}\r
-\r
-void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )\r
-{\r
- if ( mode == INPUT ) { // convert device to user buffer\r
- stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];\r
- stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];\r
- stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];\r
- stream_.convertInfo[mode].outFormat = stream_.userFormat;\r
- }\r
- else { // convert user to device buffer\r
- stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];\r
- stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];\r
- stream_.convertInfo[mode].inFormat = stream_.userFormat;\r
- stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];\r
- }\r
-\r
- if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )\r
- stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;\r
- else\r
- stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;\r
-\r
- // Set up the interleave/deinterleave offsets.\r
- if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {\r
- if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||\r
- ( mode == INPUT && stream_.userInterleaved ) ) {\r
- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {\r
- stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );\r
- stream_.convertInfo[mode].outOffset.push_back( k );\r
- stream_.convertInfo[mode].inJump = 1;\r
- }\r
- }\r
- else {\r
- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {\r
- stream_.convertInfo[mode].inOffset.push_back( k );\r
- stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );\r
- stream_.convertInfo[mode].outJump = 1;\r
- }\r
- }\r
- }\r
- else { // no (de)interleaving\r
- if ( stream_.userInterleaved ) {\r
- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {\r
- stream_.convertInfo[mode].inOffset.push_back( k );\r
- stream_.convertInfo[mode].outOffset.push_back( k );\r
- }\r
- }\r
- else {\r
- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {\r
- stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );\r
- stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );\r
- stream_.convertInfo[mode].inJump = 1;\r
- stream_.convertInfo[mode].outJump = 1;\r
- }\r
- }\r
- }\r
-\r
- // Add channel offset.\r
- if ( firstChannel > 0 ) {\r
- if ( stream_.deviceInterleaved[mode] ) {\r
- if ( mode == OUTPUT ) {\r
- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )\r
- stream_.convertInfo[mode].outOffset[k] += firstChannel;\r
- }\r
- else {\r
- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )\r
- stream_.convertInfo[mode].inOffset[k] += firstChannel;\r
- }\r
- }\r
- else {\r
- if ( mode == OUTPUT ) {\r
- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )\r
- stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );\r
- }\r
- else {\r
- for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )\r
- stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );\r
- }\r
- }\r
- }\r
-}\r
-\r
-void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )\r
-{\r
- // This function does format conversion, input/output channel compensation, and\r
- // data interleaving/deinterleaving. 24-bit integers are assumed to occupy\r
- // the lower three bytes of a 32-bit integer.\r
-\r
- // Clear our device buffer when in/out duplex device channels are different\r
- if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&\r
- ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )\r
- memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );\r
-\r
- int j;\r
- if (info.outFormat == RTAUDIO_FLOAT64) {\r
- Float64 scale;\r
- Float64 *out = (Float64 *)outBuffer;\r
-\r
- if (info.inFormat == RTAUDIO_SINT8) {\r
- signed char *in = (signed char *)inBuffer;\r
- scale = 1.0 / 127.5;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];\r
- out[info.outOffset[j]] += 0.5;\r
- out[info.outOffset[j]] *= scale;\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_SINT16) {\r
- Int16 *in = (Int16 *)inBuffer;\r
- scale = 1.0 / 32767.5;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];\r
- out[info.outOffset[j]] += 0.5;\r
- out[info.outOffset[j]] *= scale;\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_SINT24) {\r
- Int24 *in = (Int24 *)inBuffer;\r
- scale = 1.0 / 8388607.5;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());\r
- out[info.outOffset[j]] += 0.5;\r
- out[info.outOffset[j]] *= scale;\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_SINT32) {\r
- Int32 *in = (Int32 *)inBuffer;\r
- scale = 1.0 / 2147483647.5;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];\r
- out[info.outOffset[j]] += 0.5;\r
- out[info.outOffset[j]] *= scale;\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_FLOAT32) {\r
- Float32 *in = (Float32 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_FLOAT64) {\r
- // Channel compensation and/or (de)interleaving only.\r
- Float64 *in = (Float64 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = in[info.inOffset[j]];\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- }\r
- else if (info.outFormat == RTAUDIO_FLOAT32) {\r
- Float32 scale;\r
- Float32 *out = (Float32 *)outBuffer;\r
-\r
- if (info.inFormat == RTAUDIO_SINT8) {\r
- signed char *in = (signed char *)inBuffer;\r
- scale = (Float32) ( 1.0 / 127.5 );\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];\r
- out[info.outOffset[j]] += 0.5;\r
- out[info.outOffset[j]] *= scale;\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_SINT16) {\r
- Int16 *in = (Int16 *)inBuffer;\r
- scale = (Float32) ( 1.0 / 32767.5 );\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];\r
- out[info.outOffset[j]] += 0.5;\r
- out[info.outOffset[j]] *= scale;\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_SINT24) {\r
- Int24 *in = (Int24 *)inBuffer;\r
- scale = (Float32) ( 1.0 / 8388607.5 );\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());\r
- out[info.outOffset[j]] += 0.5;\r
- out[info.outOffset[j]] *= scale;\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_SINT32) {\r
- Int32 *in = (Int32 *)inBuffer;\r
- scale = (Float32) ( 1.0 / 2147483647.5 );\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];\r
- out[info.outOffset[j]] += 0.5;\r
- out[info.outOffset[j]] *= scale;\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_FLOAT32) {\r
- // Channel compensation and/or (de)interleaving only.\r
- Float32 *in = (Float32 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = in[info.inOffset[j]];\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_FLOAT64) {\r
- Float64 *in = (Float64 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- }\r
- else if (info.outFormat == RTAUDIO_SINT32) {\r
- Int32 *out = (Int32 *)outBuffer;\r
- if (info.inFormat == RTAUDIO_SINT8) {\r
- signed char *in = (signed char *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];\r
- out[info.outOffset[j]] <<= 24;\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_SINT16) {\r
- Int16 *in = (Int16 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];\r
- out[info.outOffset[j]] <<= 16;\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_SINT24) {\r
- Int24 *in = (Int24 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();\r
- out[info.outOffset[j]] <<= 8;\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_SINT32) {\r
- // Channel compensation and/or (de)interleaving only.\r
- Int32 *in = (Int32 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = in[info.inOffset[j]];\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_FLOAT32) {\r
- Float32 *in = (Float32 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_FLOAT64) {\r
- Float64 *in = (Float64 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.5 - 0.5);\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- }\r
- else if (info.outFormat == RTAUDIO_SINT24) {\r
- Int24 *out = (Int24 *)outBuffer;\r
- if (info.inFormat == RTAUDIO_SINT8) {\r
- signed char *in = (signed char *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);\r
- //out[info.outOffset[j]] <<= 16;\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_SINT16) {\r
- Int16 *in = (Int16 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 8);\r
- //out[info.outOffset[j]] <<= 8;\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_SINT24) {\r
- // Channel compensation and/or (de)interleaving only.\r
- Int24 *in = (Int24 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = in[info.inOffset[j]];\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_SINT32) {\r
- Int32 *in = (Int32 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] >> 8);\r
- //out[info.outOffset[j]] >>= 8;\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_FLOAT32) {\r
- Float32 *in = (Float32 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_FLOAT64) {\r
- Float64 *in = (Float64 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388607.5 - 0.5);\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- }\r
- else if (info.outFormat == RTAUDIO_SINT16) {\r
- Int16 *out = (Int16 *)outBuffer;\r
- if (info.inFormat == RTAUDIO_SINT8) {\r
- signed char *in = (signed char *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];\r
- out[info.outOffset[j]] <<= 8;\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_SINT16) {\r
- // Channel compensation and/or (de)interleaving only.\r
- Int16 *in = (Int16 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = in[info.inOffset[j]];\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_SINT24) {\r
- Int24 *in = (Int24 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_SINT32) {\r
- Int32 *in = (Int32 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_FLOAT32) {\r
- Float32 *in = (Float32 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_FLOAT64) {\r
- Float64 *in = (Float64 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.5 - 0.5);\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- }\r
- else if (info.outFormat == RTAUDIO_SINT8) {\r
- signed char *out = (signed char *)outBuffer;\r
- if (info.inFormat == RTAUDIO_SINT8) {\r
- // Channel compensation and/or (de)interleaving only.\r
- signed char *in = (signed char *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = in[info.inOffset[j]];\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- if (info.inFormat == RTAUDIO_SINT16) {\r
- Int16 *in = (Int16 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_SINT24) {\r
- Int24 *in = (Int24 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_SINT32) {\r
- Int32 *in = (Int32 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_FLOAT32) {\r
- Float32 *in = (Float32 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- else if (info.inFormat == RTAUDIO_FLOAT64) {\r
- Float64 *in = (Float64 *)inBuffer;\r
- for (unsigned int i=0; i<stream_.bufferSize; i++) {\r
- for (j=0; j<info.channels; j++) {\r
- out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.5 - 0.5);\r
- }\r
- in += info.inJump;\r
- out += info.outJump;\r
- }\r
- }\r
- }\r
-}\r
-\r
-//static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }\r
-//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }\r
-//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }\r
-\r
-void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )\r
-{\r
- register char val;\r
- register char *ptr;\r
-\r
- ptr = buffer;\r
- if ( format == RTAUDIO_SINT16 ) {\r
- for ( unsigned int i=0; i<samples; i++ ) {\r
- // Swap 1st and 2nd bytes.\r
- val = *(ptr);\r
- *(ptr) = *(ptr+1);\r
- *(ptr+1) = val;\r
-\r
- // Increment 2 bytes.\r
- ptr += 2;\r
- }\r
- }\r
- else if ( format == RTAUDIO_SINT32 ||\r
- format == RTAUDIO_FLOAT32 ) {\r
- for ( unsigned int i=0; i<samples; i++ ) {\r
- // Swap 1st and 4th bytes.\r
- val = *(ptr);\r
- *(ptr) = *(ptr+3);\r
- *(ptr+3) = val;\r
-\r
- // Swap 2nd and 3rd bytes.\r
- ptr += 1;\r
- val = *(ptr);\r
- *(ptr) = *(ptr+1);\r
- *(ptr+1) = val;\r
-\r
- // Increment 3 more bytes.\r
- ptr += 3;\r
- }\r
- }\r
- else if ( format == RTAUDIO_SINT24 ) {\r
- for ( unsigned int i=0; i<samples; i++ ) {\r
- // Swap 1st and 3rd bytes.\r
- val = *(ptr);\r
- *(ptr) = *(ptr+2);\r
- *(ptr+2) = val;\r
-\r
- // Increment 2 more bytes.\r
- ptr += 2;\r
- }\r
- }\r
- else if ( format == RTAUDIO_FLOAT64 ) {\r
- for ( unsigned int i=0; i<samples; i++ ) {\r
- // Swap 1st and 8th bytes\r
- val = *(ptr);\r
- *(ptr) = *(ptr+7);\r
- *(ptr+7) = val;\r
-\r
- // Swap 2nd and 7th bytes\r
- ptr += 1;\r
- val = *(ptr);\r
- *(ptr) = *(ptr+5);\r
- *(ptr+5) = val;\r
-\r
- // Swap 3rd and 6th bytes\r
- ptr += 1;\r
- val = *(ptr);\r
- *(ptr) = *(ptr+3);\r
- *(ptr+3) = val;\r
-\r
- // Swap 4th and 5th bytes\r
- ptr += 1;\r
- val = *(ptr);\r
- *(ptr) = *(ptr+1);\r
- *(ptr+1) = val;\r
-\r
- // Increment 5 more bytes.\r
- ptr += 5;\r
- }\r
- }\r
-}\r
-\r
- // Indentation settings for Vim and Emacs\r
- //\r
- // Local Variables:\r
- // c-basic-offset: 2\r
- // indent-tabs-mode: nil\r
- // End:\r
- //\r
- // vim: et sts=2 sw=2\r
-\r
+/************************************************************************/
+/*! \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) 2001-2017 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 5.0.0
+
+#include "RtAudio.h"
+#include <iostream>
+#include <cstdlib>
+#include <cstring>
+#include <climits>
+#include <cmath>
+#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 )
+{
+ return RTAUDIO_VERSION;
+}
+
+void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis )
+{
+ 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()
+{
+ 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;
+#if defined( HAVE_GETTIMEOFDAY )
+ gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
+}
+
+unsigned int RtApi :: getStreamSampleRate( void )
+{
+ verifyStream();
+
+ return stream_.sampleRate;
+}
+
+
+// *************************************************** //
+//
+// OS/API-specific methods.
+//
+// *************************************************** //
+
+#if defined(__MACOSX_CORE__)
+
+// The OS X CoreAudio API is designed to use a separate callback
+// procedure for each of its audio devices. A single RtAudio duplex
+// stream using two different devices is supported here, though it
+// cannot be guaranteed to always behave correctly because we cannot
+// synchronize these two callbacks.
+//
+// A property listener is installed for over/underrun information.
+// However, no functionality is currently provided to allow property
+// listeners to trigger user handlers because it is unclear what could
+// be done if a critical stream parameter (buffer size, sample rate,
+// device disconnect) notification arrived. The listeners entail
+// quite a bit of extra code and most likely, a user program wouldn't
+// be prepared for the result anyway. However, we do provide a flag
+// to the client callback function to inform of an over/underrun.
+
+// 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; }
+};
+
+#if !defined(__RTAUDIO_DEBUG__)
+static void jackSilentError( const char * ) {};
+#endif
+
+RtApiJack :: RtApiJack()
+ :shouldAutoconnect_(true) {
+ // 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 );
+
+ if ( options && options->flags & RTAUDIO_JACK_DONT_CONNECT ) shouldAutoconnect_ = false;
+
+ 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 ( shouldAutoconnect_ && (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 ( shouldAutoconnect_ && (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 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 sampleRatioInv = ( float ) 1 / sampleRatio;
+ float sampleStep = 1.0f / sampleRatio;
+ float inSampleFraction = 0.0f;
+
+ outSampleCount = ( unsigned int ) std::roundf( inSampleCount * sampleRatio );
+
+ // if inSampleRate is a multiple of outSampleRate (or vice versa) there's no need to interpolate
+ if ( floor( sampleRatio ) == sampleRatio || floor( sampleRatioInv ) == sampleRatioInv )
+ {
+ // 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;
+ }
+ }
+ else // else interpolate
+ {
+ // 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;
+ float inSampleDec = inSampleFraction - inSample;
+ unsigned int frameInSample = inSample * channelCount;
+ unsigned int frameOutSample = outSample * channelCount;
+
+ switch ( format )
+ {
+ case RTAUDIO_SINT8:
+ {
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )
+ {
+ char fromSample = ( ( char* ) inBuffer )[ frameInSample + channel ];
+ char toSample = ( ( char* ) inBuffer )[ frameInSample + channelCount + channel ];
+ char sampleDiff = ( char ) ( ( toSample - fromSample ) * inSampleDec );
+ ( ( char* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
+ }
+ break;
+ }
+ case RTAUDIO_SINT16:
+ {
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )
+ {
+ short fromSample = ( ( short* ) inBuffer )[ frameInSample + channel ];
+ short toSample = ( ( short* ) inBuffer )[ frameInSample + channelCount + channel ];
+ short sampleDiff = ( short ) ( ( toSample - fromSample ) * inSampleDec );
+ ( ( short* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
+ }
+ break;
+ }
+ case RTAUDIO_SINT24:
+ {
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )
+ {
+ int fromSample = ( ( S24* ) inBuffer )[ frameInSample + channel ].asInt();
+ int toSample = ( ( S24* ) inBuffer )[ frameInSample + channelCount + channel ].asInt();
+ int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec );
+ ( ( S24* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
+ }
+ break;
+ }
+ case RTAUDIO_SINT32:
+ {
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )
+ {
+ int fromSample = ( ( int* ) inBuffer )[ frameInSample + channel ];
+ int toSample = ( ( int* ) inBuffer )[ frameInSample + channelCount + channel ];
+ int sampleDiff = ( int ) ( ( toSample - fromSample ) * inSampleDec );
+ ( ( int* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
+ }
+ break;
+ }
+ case RTAUDIO_FLOAT32:
+ {
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )
+ {
+ float fromSample = ( ( float* ) inBuffer )[ frameInSample + channel ];
+ float toSample = ( ( float* ) inBuffer )[ frameInSample + channelCount + channel ];
+ float sampleDiff = ( toSample - fromSample ) * inSampleDec;
+ ( ( float* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
+ }
+ break;
+ }
+ case RTAUDIO_FLOAT64:
+ {
+ for ( unsigned int channel = 0; channel < channelCount; channel++ )
+ {
+ double fromSample = ( ( double* ) inBuffer )[ frameInSample + channel ];
+ double toSample = ( ( double* ) inBuffer )[ frameInSample + channelCount + channel ];
+ double sampleDiff = ( toSample - fromSample ) * inSampleDec;
+ ( ( double* ) outBuffer )[ frameOutSample + channel ] = fromSample + sampleDiff;
+ }
+ 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 <mmsystem.h>
+#include <mmreg.h>
+#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 ( stream_.state != STREAM_CLOSED ) closeStream();
+ if ( coInitialized_ ) CoUninitialize(); // balanced call.
+}
+
+// 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 = 0;
+
+ // 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 ) {
+ handle = 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:
+ if ( handle )
+ 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 = 0;
+
+ // 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 ) {
+ chandle = 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:
+ if ( chandle )
+ 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;
+#ifdef AFMT_FLOAT
+ if ( mask & AFMT_FLOAT )
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+#endif
+ 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 - (int)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
+
projects / rtaudio-cdist.git / blobdiff