X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=RtAudio.cpp;h=6b1ca063a9dc5d17ff292b0751a46b5a0cd238d9;hb=7b58989d83e63ecce877c7935eecc7b2b81dc34d;hp=7fb2798f90092f7333e147f036da8141900a9105;hpb=470f62b4afada9b89b19d1bd57955c417d2e7005;p=rtaudio-cdist.git diff --git a/RtAudio.cpp b/RtAudio.cpp index 7fb2798..6b1ca06 100644 --- a/RtAudio.cpp +++ b/RtAudio.cpp @@ -1,4 +1,4 @@ -/************************************************************************/ +/************************************************************************/ /*! \class RtAudio \brief Realtime audio i/o C++ classes. @@ -98,39 +98,95 @@ std::string RtAudio :: getVersion( void ) return RTAUDIO_VERSION; } -void RtAudio :: getCompiledApi( std::vector &apis ) -{ - apis.clear(); +// Define API names and display names. +// Must be in same order as API enum. +extern "C" { +const char* rtaudio_api_names[][2] = { + { "unspecified" , "Unknown" }, + { "alsa" , "ALSA" }, + { "pulse" , "Pulse" }, + { "oss" , "OpenSoundSystem" }, + { "jack" , "Jack" }, + { "core" , "CoreAudio" }, + { "wasapi" , "WASAPI" }, + { "asio" , "ASIO" }, + { "ds" , "DirectSound" }, + { "dummy" , "Dummy" }, +}; +const unsigned int rtaudio_num_api_names = + sizeof(rtaudio_api_names)/sizeof(rtaudio_api_names[0]); - // The order here will control the order of RtAudio's API search in - // the constructor. +// The order here will control the order of RtAudio's API search in +// the constructor. +extern "C" const RtAudio::Api rtaudio_compiled_apis[] = { #if defined(__UNIX_JACK__) - apis.push_back( UNIX_JACK ); + RtAudio::UNIX_JACK, #endif #if defined(__LINUX_PULSE__) - apis.push_back( LINUX_PULSE ); + RtAudio::LINUX_PULSE, #endif #if defined(__LINUX_ALSA__) - apis.push_back( LINUX_ALSA ); + RtAudio::LINUX_ALSA, #endif #if defined(__LINUX_OSS__) - apis.push_back( LINUX_OSS ); + RtAudio::LINUX_OSS, #endif #if defined(__WINDOWS_ASIO__) - apis.push_back( WINDOWS_ASIO ); + RtAudio::WINDOWS_ASIO, #endif #if defined(__WINDOWS_WASAPI__) - apis.push_back( WINDOWS_WASAPI ); + RtAudio::WINDOWS_WASAPI, #endif #if defined(__WINDOWS_DS__) - apis.push_back( WINDOWS_DS ); + RtAudio::WINDOWS_DS, #endif #if defined(__MACOSX_CORE__) - apis.push_back( MACOSX_CORE ); + RtAudio::MACOSX_CORE, #endif #if defined(__RTAUDIO_DUMMY__) - apis.push_back( RTAUDIO_DUMMY ); + RtAudio::RTAUDIO_DUMMY, #endif + RtAudio::UNSPECIFIED, +}; +extern "C" const unsigned int rtaudio_num_compiled_apis = + sizeof(rtaudio_compiled_apis)/sizeof(rtaudio_compiled_apis[0])-1; +} + +// This is a compile-time check that rtaudio_num_api_names == RtAudio::NUM_APIS. +// If the build breaks here, check that they match. +template class StaticAssert { private: StaticAssert() {} }; +template<> class StaticAssert{ public: StaticAssert() {} }; +class StaticAssertions { StaticAssertions() { + StaticAssert(); +}}; + +void RtAudio :: getCompiledApi( std::vector &apis ) +{ + apis = std::vector(rtaudio_compiled_apis, + rtaudio_compiled_apis + rtaudio_num_compiled_apis); +} + +std::string RtAudio :: getApiName( RtAudio::Api api ) +{ + if (api < 0 || api >= RtAudio::NUM_APIS) + return ""; + return rtaudio_api_names[api][0]; +} + +std::string RtAudio :: getApiDisplayName( RtAudio::Api api ) +{ + if (api < 0 || api >= RtAudio::NUM_APIS) + return "Unknown"; + return rtaudio_api_names[api][1]; +} + +RtAudio::Api RtAudio :: getCompiledApiByName( const std::string &name ) +{ + unsigned int i=0; + for (i = 0; i < rtaudio_num_compiled_apis; ++i) + if (name == rtaudio_api_names[rtaudio_compiled_apis[i]][0]) + return rtaudio_compiled_apis[i]; + return RtAudio::UNSPECIFIED; } void RtAudio :: openRtApi( RtAudio::Api api ) @@ -3165,8 +3221,8 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne 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 + // but only accept the preferred buffer size as parameter for ASIOCreateBuffers (e.g. Creative's 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 ); @@ -3689,18 +3745,32 @@ static const char* getAsioErrorString( ASIOError result ) #ifndef INITGUID #define INITGUID #endif + +#include +#include +#include +#include +#include + #include #include #include #include -#include -#include -#include -#include +#ifndef MF_E_TRANSFORM_NEED_MORE_INPUT + #define MF_E_TRANSFORM_NEED_MORE_INPUT _HRESULT_TYPEDEF_(0xc00d6d72) +#endif + +#ifndef MFSTARTUP_NOSOCKET + #define MFSTARTUP_NOSOCKET 0x1 +#endif -#pragma comment( lib, "mfplat.lib" ) -#pragma comment( lib, "wmcodecdspuuid" ) +#ifdef _MSC_VER + #pragma comment( lib, "ksuser" ) + #pragma comment( lib, "mfplat.lib" ) + #pragma comment( lib, "mfuuid.lib" ) + #pragma comment( lib, "wmcodecdspuuid" ) +#endif //============================================================================= @@ -3887,10 +3957,13 @@ public: , _sampleRatio( ( float ) outSampleRate / inSampleRate ) , _transformUnk( NULL ) , _transform( NULL ) - , _resamplerProps( NULL ) , _mediaType( NULL ) , _inputMediaType( NULL ) , _outputMediaType( NULL ) + + #ifdef __IWMResamplerProps_FWD_DEFINED__ + , _resamplerProps( NULL ) + #endif { // 1. Initialization @@ -3903,10 +3976,12 @@ public: _transformUnk->QueryInterface( IID_PPV_ARGS( &_transform ) ); - _transformUnk->QueryInterface( IID_PPV_ARGS( &_resamplerProps ) ); - _resamplerProps->SetHalfFilterLength( 60 ); // best conversion quality + #ifdef __IWMResamplerProps_FWD_DEFINED__ + _transformUnk->QueryInterface( IID_PPV_ARGS( &_resamplerProps ) ); + _resamplerProps->SetHalfFilterLength( 60 ); // best conversion quality + #endif - // 3. Specify input / output format + // 3. Specify input / output format MFCreateMediaType( &_mediaType ); _mediaType->SetGUID( MF_MT_MAJOR_TYPE, MFMediaType_Audio ); @@ -3933,17 +4008,17 @@ public: // 4. Send stream start messages to Resampler - _transform->ProcessMessage( MFT_MESSAGE_COMMAND_FLUSH, NULL ); - _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_BEGIN_STREAMING, NULL ); - _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_START_OF_STREAM, NULL ); + _transform->ProcessMessage( MFT_MESSAGE_COMMAND_FLUSH, 0 ); + _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_BEGIN_STREAMING, 0 ); + _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_START_OF_STREAM, 0 ); } ~WasapiResampler() { // 8. Send stream stop messages to Resampler - _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_OF_STREAM, NULL ); - _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_STREAMING, NULL ); + _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_OF_STREAM, 0 ); + _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_STREAMING, 0 ); // 9. Cleanup @@ -3951,10 +4026,13 @@ public: SAFE_RELEASE( _transformUnk ); SAFE_RELEASE( _transform ); - SAFE_RELEASE( _resamplerProps ); SAFE_RELEASE( _mediaType ); SAFE_RELEASE( _inputMediaType ); SAFE_RELEASE( _outputMediaType ); + + #ifdef __IWMResamplerProps_FWD_DEFINED__ + SAFE_RELEASE( _resamplerProps ); + #endif } void Convert( char* outBuffer, const char* inBuffer, unsigned int inSampleCount, unsigned int& outSampleCount ) @@ -4004,7 +4082,7 @@ public: DWORD rStatus; DWORD rBytes = outputBufferSize; // maximum bytes accepted per ProcessOutput - // 7.1 Create Sample object for output data + // 7.1 Create Sample object for output data memset( &rOutDataBuffer, 0, sizeof rOutDataBuffer ); MFCreateSample( &( rOutDataBuffer.pSample ) ); @@ -4047,10 +4125,13 @@ private: IUnknown* _transformUnk; IMFTransform* _transform; - IWMResamplerProps* _resamplerProps; IMFMediaType* _mediaType; IMFMediaType* _inputMediaType; IMFMediaType* _outputMediaType; + + #ifdef __IWMResamplerProps_FWD_DEFINED__ + IWMResamplerProps* _resamplerProps; + #endif }; //----------------------------------------------------------------------------- @@ -4089,10 +4170,9 @@ RtApiWasapi::RtApiWasapi() CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ), ( void** ) &deviceEnumerator_ ); - if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator"; - error( RtAudioError::DRIVER_ERROR ); - } + // If this runs on an old Windows, it will fail. Ignore and proceed. + if ( FAILED( hr ) ) + deviceEnumerator_ = NULL; } //----------------------------------------------------------------------------- @@ -4119,6 +4199,9 @@ unsigned int RtApiWasapi::getDeviceCount( void ) IMMDeviceCollection* captureDevices = NULL; IMMDeviceCollection* renderDevices = NULL; + if ( !deviceEnumerator_ ) + return 0; + // Count capture devices errorText_.clear(); HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices ); @@ -4644,7 +4727,7 @@ bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigne goto Exit; } - // determine whether index falls within capture or render devices + // if device index falls within capture devices if ( device >= renderDeviceCount ) { if ( mode != INPUT ) { errorType = RtAudioError::INVALID_USE; @@ -4664,28 +4747,66 @@ bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigne hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &captureAudioClient ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client."; + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device audio client."; goto Exit; } hr = captureAudioClient->GetMixFormat( &deviceFormat ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format."; + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture 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."; + + // if device index falls within render devices and is configured for loopback + if ( device < renderDeviceCount && mode == INPUT ) + { + // if renderAudioClient is not initialised, initialise it now + IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; + if ( !renderAudioClient ) + { + probeDeviceOpen( device, OUTPUT, channels, firstChannel, sampleRate, format, bufferSize, options ); + } + + // retrieve captureAudioClient from devicePtr + IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient; + + 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** ) &captureAudioClient ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client."; + goto Exit; + } + + hr = captureAudioClient->GetMixFormat( &deviceFormat ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format."; goto Exit; } - // retrieve renderAudioClient from devicePtr + stream_.nDeviceChannels[mode] = deviceFormat->nChannels; + captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] ); + } + + // if device index falls within render devices and is configured for output + if ( device < renderDeviceCount && mode == OUTPUT ) + { + // if renderAudioClient is already initialised, don't initialise it again IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient; + if ( renderAudioClient ) + { + methodResult = SUCCESS; + goto Exit; + } hr = renderDevices->Item( device, &devicePtr ); if ( FAILED( hr ) ) { @@ -4696,13 +4817,13 @@ bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigne hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &renderAudioClient ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client."; + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device audio client."; goto Exit; } hr = renderAudioClient->GetMixFormat( &deviceFormat ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format."; + errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device mix format."; goto Exit; } @@ -4738,7 +4859,8 @@ bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigne // Set flags for buffer conversion. stream_.doConvertBuffer[mode] = false; if ( stream_.userFormat != stream_.deviceFormat[mode] || - stream_.nUserChannels != stream_.nDeviceChannels ) + stream_.nUserChannels[0] != stream_.nDeviceChannels[0] || + stream_.nUserChannels[1] != stream_.nDeviceChannels[1] ) stream_.doConvertBuffer[mode] = true; else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] && stream_.nUserChannels[mode] > 1 ) @@ -4841,6 +4963,7 @@ void RtApiWasapi::wasapiThread() unsigned int bufferFrameCount = 0; unsigned int numFramesPadding = 0; unsigned int convBufferSize = 0; + bool loopbackEnabled = stream_.device[INPUT] == stream_.device[OUTPUT]; bool callbackPushed = true; bool callbackPulled = false; bool callbackStopped = false; @@ -4878,15 +5001,11 @@ void RtApiWasapi::wasapiThread() 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, + loopbackEnabled ? AUDCLNT_STREAMFLAGS_LOOPBACK : AUDCLNT_STREAMFLAGS_EVENTCALLBACK, + 0, + 0, captureFormat, NULL ); if ( FAILED( hr ) ) { @@ -4901,22 +5020,27 @@ void RtApiWasapi::wasapiThread() 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; - } + // don't configure captureEvent if in loopback mode + if ( !loopbackEnabled ) + { + // 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; + hr = captureAudioClient->SetEventHandle( captureEvent ); + if ( FAILED( hr ) ) { + errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle."; + goto Exit; + } + + ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent; } ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient; - ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent; } unsigned int inBufferSize = 0; @@ -4963,15 +5087,11 @@ void RtApiWasapi::wasapiThread() 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, + 0, + 0, renderFormat, NULL ); if ( FAILED( hr ) ) { @@ -5071,23 +5191,43 @@ void RtApiWasapi::wasapiThread() // 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 ( captureAudioClient ) + { + int samplesToPull = ( unsigned int ) floorf( stream_.bufferSize * captureSrRatio ); + if ( captureSrRatio != 1 ) + { + // account for remainders + samplesToPull--; + } + + convBufferSize = 0; + while ( convBufferSize < stream_.bufferSize ) + { + // Pull callback buffer from inputBuffer + callbackPulled = captureBuffer.pullBuffer( convBuffer, + samplesToPull * stream_.nDeviceChannels[INPUT], + stream_.deviceFormat[INPUT] ); + + if ( !callbackPulled ) + { + break; + } - 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] ); + unsigned int deviceBufferOffset = convBufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); + unsigned int convSamples = 0; + + captureResampler->Convert( stream_.deviceBuffer + deviceBufferOffset, + convBuffer, + samplesToPull, + convSamples ); + + convBufferSize += convSamples; + samplesToPull = 1; // now pull one sample at a time until we have stream_.bufferSize samples + } + if ( callbackPulled ) + { if ( stream_.doConvertBuffer[INPUT] ) { // Convert callback buffer to user format convertBuffer( stream_.userBuffer[INPUT], @@ -5164,24 +5304,26 @@ void RtApiWasapi::wasapiThread() // 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] ); + if ( renderAudioClient && callbackPulled ) + { + // if the last call to renderBuffer.PushBuffer() was successful + if ( callbackPushed || convBufferSize == 0 ) + { + 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] ); + // Convert callback buffer to stream sample rate + renderResampler->Convert( convBuffer, + stream_.deviceBuffer, + stream_.bufferSize, + convBufferSize ); + } // Push callback buffer into outputBuffer callbackPushed = renderBuffer.pushBuffer( convBuffer, @@ -5202,7 +5344,7 @@ void RtApiWasapi::wasapiThread() if ( captureAudioClient ) { // if the callback input buffer was not pulled from captureBuffer, wait for next capture event if ( !callbackPulled ) { - WaitForSingleObject( captureEvent, INFINITE ); + WaitForSingleObject( loopbackEnabled ? renderEvent : captureEvent, INFINITE ); } // Get capture buffer from stream @@ -5319,7 +5461,10 @@ void RtApiWasapi::wasapiThread() // if the callback buffer was pushed renderBuffer reset callbackPulled flag if ( callbackPushed ) { + // unsetting the callbackPulled flag lets the stream know that + // the audio device is ready for another callback output buffer. callbackPulled = false; + // tick stream time RtApi::tickStreamTime(); } @@ -5332,16 +5477,16 @@ Exit: CoTaskMemFree( renderFormat ); free ( convBuffer ); + delete renderResampler; + delete captureResampler; CoUninitialize(); + if ( !errorText_.empty() ) + error( errorType ); + // update stream state stream_.state = STREAM_STOPPED; - - if ( errorText_.empty() ) - return; - else - error( errorType ); } //******************** End of __WINDOWS_WASAPI__ *********************// @@ -7395,10 +7540,12 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne if ( result == 0 ) { if ( nDevices == device ) { strcpy( name, "default" ); + snd_ctl_close( chandle ); goto foundDevice; } nDevices++; } + snd_ctl_close( chandle ); if ( nDevices == 0 ) { // This should not happen because a check is made before this function is called. @@ -7798,7 +7945,7 @@ bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne 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) +#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { stream_.callbackInfo.doRealtime = true; struct sched_param param; @@ -8247,7 +8394,7 @@ static void *alsaCallbackHandler( void *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) +#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) if ( info->doRealtime ) { std::cerr << "RtAudio alsa: " << (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") << @@ -8335,7 +8482,7 @@ static void *pulseaudio_callback( void * user ) RtApiPulse *context = static_cast( cbi->object ); volatile bool *isRunning = &cbi->isRunning; -#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) +#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) if (cbi->doRealtime) { std::cerr << "RtAudio pulse: " << (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") << @@ -8738,7 +8885,7 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, 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) +#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { stream_.callbackInfo.doRealtime = true; struct sched_param param; @@ -9359,7 +9506,7 @@ bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned 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) +#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { stream_.callbackInfo.doRealtime = true; struct sched_param param; @@ -9750,7 +9897,7 @@ static void *ossCallbackHandler( void *ptr ) RtApiOss *object = (RtApiOss *) info->object; bool *isRunning = &info->isRunning; -#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread) +#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread) if (info->doRealtime) { std::cerr << "RtAudio oss: " << (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<