X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=RtAudio.cpp;h=18d1fd0af29b264fe835aa78649b09e9af5324fa;hb=HEAD;hp=1d235161e79dd26970fa9d671af31801d9918034;hpb=44e8c742d4fd088069888efe491ba16672e67d6b;p=rtaudio-cdist.git diff --git a/RtAudio.cpp b/RtAudio.cpp index 1d23516..18d1fd0 100644 --- a/RtAudio.cpp +++ b/RtAudio.cpp @@ -1,4 +1,4 @@ -/************************************************************************/ +/************************************************************************/ /*! \class RtAudio \brief Realtime audio i/o C++ classes. @@ -47,6 +47,7 @@ #include #include #include +#include // Static variable definitions. const unsigned int RtApi::MAX_SAMPLE_RATES = 14; @@ -98,39 +99,95 @@ 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 ); -#endif -#if defined(__LINUX_ALSA__) - apis.push_back( LINUX_ALSA ); + RtAudio::UNIX_JACK, #endif #if defined(__LINUX_PULSE__) - apis.push_back( LINUX_PULSE ); + RtAudio::LINUX_PULSE, +#endif +#if defined(__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 ) @@ -409,14 +466,14 @@ double RtApi :: getStreamTime( void ) struct timeval then; struct timeval now; - if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 ) + if ( stream_.state != STREAM_RUNNING || (stream_.lastTickTimestamp.tv_sec == 0 && stream_.lastTickTimestamp.tv_usec == 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)); + (then.tv_sec + 0.000001 * then.tv_usec)); #else return stream_.streamTime; #endif @@ -440,6 +497,14 @@ unsigned int RtApi :: getStreamSampleRate( void ) return stream_.sampleRate; } +void RtApi :: startStream( void ) +{ +#if defined( HAVE_GETTIMEOFDAY ) + stream_.lastTickTimestamp.tv_sec = 0; + stream_.lastTickTimestamp.tv_usec = 0; +#endif +} + // *************************************************** // // @@ -1479,12 +1544,17 @@ void RtApiCore :: closeStream( void ) void RtApiCore :: startStream( void ) { verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiCore::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); return; } + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + OSStatus result = noErr; CoreHandle *handle = (CoreHandle *) stream_.apiHandle; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { @@ -1833,7 +1903,7 @@ bool RtApiCore :: callbackEvent( AudioDeviceID deviceId, channelsLeft -= streamChannels; } } - + if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, @@ -1973,7 +2043,7 @@ unsigned int RtApiJack :: getDeviceCount( void ) const char **ports; std::string port, previousPort; unsigned int nChannels = 0, nDevices = 0; - ports = jack_get_ports( client, NULL, NULL, 0 ); + ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 ); if ( ports ) { // Parse the port names up to the first colon (:). size_t iColon = 0; @@ -2012,7 +2082,7 @@ RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) const char **ports; std::string port, previousPort; unsigned int nPorts = 0, nDevices = 0; - ports = jack_get_ports( client, NULL, NULL, 0 ); + ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 ); if ( ports ) { // Parse the port names up to the first colon (:). size_t iColon = 0; @@ -2047,7 +2117,7 @@ RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) // 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 ); + ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput ); if ( ports ) { while ( ports[ nChannels ] ) nChannels++; free( ports ); @@ -2056,7 +2126,7 @@ RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device ) // Jack "output ports" equal RtAudio input channels. nChannels = 0; - ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput ); + ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput ); if ( ports ) { while ( ports[ nChannels ] ) nChannels++; free( ports ); @@ -2130,7 +2200,7 @@ static void jackShutdown( void *infoPointer ) static int jackXrun( void *infoPointer ) { - JackHandle *handle = (JackHandle *) infoPointer; + JackHandle *handle = *((JackHandle **) infoPointer); if ( handle->ports[0] ) handle->xrun[0] = true; if ( handle->ports[1] ) handle->xrun[1] = true; @@ -2168,7 +2238,7 @@ bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne const char **ports; std::string port, previousPort, deviceName; unsigned int nPorts = 0, nDevices = 0; - ports = jack_get_ports( client, NULL, NULL, 0 ); + ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 ); if ( ports ) { // Parse the port names up to the first colon (:). size_t iColon = 0; @@ -2192,22 +2262,24 @@ bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne 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; + if ( ! (options && (options->flags & RTAUDIO_JACK_DONT_CONNECT)) ) { + // 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, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, 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. @@ -2221,7 +2293,7 @@ bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne stream_.sampleRate = jackRate; // Get the latency of the JACK port. - ports = jack_get_ports( client, deviceName.c_str(), NULL, flag ); + ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag ); if ( ports[ firstChannel ] ) { // Added by Ge Wang jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency); @@ -2334,7 +2406,7 @@ bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne else { stream_.mode = mode; jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo ); - jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle ); + jack_set_xrun_callback( handle->client, jackXrun, (void *) &stream_.apiHandle ); jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo ); } @@ -2435,12 +2507,17 @@ void RtApiJack :: closeStream( void ) void RtApiJack :: startStream( void ) { verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiJack::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); return; } + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + JackHandle *handle = (JackHandle *) stream_.apiHandle; int result = jack_activate( handle->client ); if ( result ) { @@ -2453,7 +2530,7 @@ void RtApiJack :: startStream( void ) // 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); + ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput); if ( ports == NULL) { errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!"; goto unlock; @@ -2477,7 +2554,7 @@ void RtApiJack :: startStream( void ) if ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) { result = 1; - ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput ); + ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput ); if ( ports == NULL) { errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!"; goto unlock; @@ -2724,7 +2801,7 @@ RtApiAsio :: RtApiAsio() // CoInitialize beforehand, but it must be for appartment threading // (in which case, CoInitilialize will return S_FALSE here). coInitialized_ = false; - HRESULT hr = CoInitialize( NULL ); + HRESULT hr = CoInitialize( NULL ); if ( FAILED(hr) ) { errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)"; error( RtAudioError::WARNING ); @@ -3163,8 +3240,8 @@ 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 ); @@ -3175,7 +3252,7 @@ bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigne errorText_ = errorStream_.str(); goto error; } - buffersAllocated = true; + buffersAllocated = true; stream_.state = STREAM_STOPPED; // Set flags for buffer conversion. @@ -3314,12 +3391,17 @@ bool stopThreadCalled = false; void RtApiAsio :: startStream() { verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiAsio::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); return; } + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + AsioHandle *handle = (AsioHandle *) stream_.apiHandle; ASIOError result = ASIOStart(); if ( result != ASE_OK ) { @@ -3649,13 +3731,13 @@ static long asioMessages( long selector, long value, void* /*message*/, double* static const char* getAsioErrorString( ASIOError result ) { - struct Messages + struct Messages { ASIOError value; const char*message; }; - static const Messages m[] = + static const Messages m[] = { { ASE_NotPresent, "Hardware input or output is not present or available." }, { ASE_HWMalfunction, "Hardware is malfunctioning." }, @@ -3687,11 +3769,33 @@ static const char* getAsioErrorString( ASIOError result ) #ifndef INITGUID #define INITGUID #endif + +#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 + +#ifdef _MSC_VER + #pragma comment( lib, "ksuser" ) + #pragma comment( lib, "mfplat.lib" ) + #pragma comment( lib, "mfuuid.lib" ) + #pragma comment( lib, "wmcodecdspuuid" ) +#endif + //============================================================================= #define SAFE_RELEASE( objectPtr )\ @@ -3749,8 +3853,9 @@ public: relOutIndex += bufferSize_; } - // "in" index can end on the "out" index but cannot begin at it - if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) { + // the "IN" index CAN BEGIN at the "OUT" index + // the "IN" index CANNOT END at the "OUT" index + if ( inIndex_ < relOutIndex && inIndexEnd >= relOutIndex ) { return false; // not enough space between "in" index and "out" index } @@ -3810,8 +3915,9 @@ public: relInIndex += bufferSize_; } - // "out" index can begin at and end on the "in" index - if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) { + // the "OUT" index CANNOT BEGIN at the "IN" index + // the "OUT" index CAN END at the "IN" index + if ( outIndex_ <= relInIndex && outIndexEnd > relInIndex ) { return false; // not enough space between "out" index and "in" index } @@ -3865,148 +3971,194 @@ private: //----------------------------------------------------------------------------- // 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; - - // for cmath functions - using namespace std; - - outSampleCount = ( unsigned int ) 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 ) +// between HW and the user. The WasapiResampler class is used to perform this conversion between +// HwIn->UserIn and UserO ut->HwOut during the stream callback loop. +class WasapiResampler +{ +public: + WasapiResampler( bool isFloat, unsigned int bitsPerSample, unsigned int channelCount, + unsigned int inSampleRate, unsigned int outSampleRate ) + : _bytesPerSample( bitsPerSample / 8 ) + , _channelCount( channelCount ) + , _sampleRatio( ( float ) outSampleRate / inSampleRate ) + , _transformUnk( NULL ) + , _transform( NULL ) + , _mediaType( NULL ) + , _inputMediaType( NULL ) + , _outputMediaType( NULL ) + + #ifdef __IWMResamplerProps_FWD_DEFINED__ + , _resamplerProps( NULL ) + #endif { - // 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; + // 1. Initialization - 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; - } + MFStartup( MF_VERSION, MFSTARTUP_NOSOCKET ); - // jump to next in sample - inSampleFraction += sampleStep; - } + // 2. Create Resampler Transform Object + + CoCreateInstance( CLSID_CResamplerMediaObject, NULL, CLSCTX_INPROC_SERVER, + IID_IUnknown, ( void** ) &_transformUnk ); + + _transformUnk->QueryInterface( IID_PPV_ARGS( &_transform ) ); + + #ifdef __IWMResamplerProps_FWD_DEFINED__ + _transformUnk->QueryInterface( IID_PPV_ARGS( &_resamplerProps ) ); + _resamplerProps->SetHalfFilterLength( 60 ); // best conversion quality + #endif + + // 3. Specify input / output format + + MFCreateMediaType( &_mediaType ); + _mediaType->SetGUID( MF_MT_MAJOR_TYPE, MFMediaType_Audio ); + _mediaType->SetGUID( MF_MT_SUBTYPE, isFloat ? MFAudioFormat_Float : MFAudioFormat_PCM ); + _mediaType->SetUINT32( MF_MT_AUDIO_NUM_CHANNELS, channelCount ); + _mediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, inSampleRate ); + _mediaType->SetUINT32( MF_MT_AUDIO_BLOCK_ALIGNMENT, _bytesPerSample * channelCount ); + _mediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * inSampleRate ); + _mediaType->SetUINT32( MF_MT_AUDIO_BITS_PER_SAMPLE, bitsPerSample ); + _mediaType->SetUINT32( MF_MT_ALL_SAMPLES_INDEPENDENT, TRUE ); + + MFCreateMediaType( &_inputMediaType ); + _mediaType->CopyAllItems( _inputMediaType ); + + _transform->SetInputType( 0, _inputMediaType, 0 ); + + MFCreateMediaType( &_outputMediaType ); + _mediaType->CopyAllItems( _outputMediaType ); + + _outputMediaType->SetUINT32( MF_MT_AUDIO_SAMPLES_PER_SECOND, outSampleRate ); + _outputMediaType->SetUINT32( MF_MT_AUDIO_AVG_BYTES_PER_SECOND, _bytesPerSample * channelCount * outSampleRate ); + + _transform->SetOutputType( 0, _outputMediaType, 0 ); + + // 4. Send stream start messages to Resampler + + _transform->ProcessMessage( MFT_MESSAGE_COMMAND_FLUSH, 0 ); + _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_BEGIN_STREAMING, 0 ); + _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_START_OF_STREAM, 0 ); } - else // else interpolate + + ~WasapiResampler() { - // frame-by-frame, copy each relative input sample into it's corresponding output sample - for ( unsigned int outSample = 0; outSample < outSampleCount; outSample++ ) + // 8. Send stream stop messages to Resampler + + _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_OF_STREAM, 0 ); + _transform->ProcessMessage( MFT_MESSAGE_NOTIFY_END_STREAMING, 0 ); + + // 9. Cleanup + + MFShutdown(); + + SAFE_RELEASE( _transformUnk ); + SAFE_RELEASE( _transform ); + 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 ) + { + unsigned int inputBufferSize = _bytesPerSample * _channelCount * inSampleCount; + if ( _sampleRatio == 1 ) { - unsigned int inSample = ( unsigned int ) inSampleFraction; - float inSampleDec = inSampleFraction - inSample; - unsigned int frameInSample = inSample * channelCount; - unsigned int frameOutSample = outSample * channelCount; + // no sample rate conversion required + memcpy( outBuffer, inBuffer, inputBufferSize ); + outSampleCount = inSampleCount; + return; + } - 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; - } - } + unsigned int outputBufferSize = ( unsigned int ) ceilf( inputBufferSize * _sampleRatio ) + ( _bytesPerSample * _channelCount ); + + IMFMediaBuffer* rInBuffer; + IMFSample* rInSample; + BYTE* rInByteBuffer = NULL; + + // 5. Create Sample object from input data + + MFCreateMemoryBuffer( inputBufferSize, &rInBuffer ); + + rInBuffer->Lock( &rInByteBuffer, NULL, NULL ); + memcpy( rInByteBuffer, inBuffer, inputBufferSize ); + rInBuffer->Unlock(); + rInByteBuffer = NULL; + + rInBuffer->SetCurrentLength( inputBufferSize ); - // jump to next in sample - inSampleFraction += sampleStep; + MFCreateSample( &rInSample ); + rInSample->AddBuffer( rInBuffer ); + + // 6. Pass input data to Resampler + + _transform->ProcessInput( 0, rInSample, 0 ); + + SAFE_RELEASE( rInBuffer ); + SAFE_RELEASE( rInSample ); + + // 7. Perform sample rate conversion + + IMFMediaBuffer* rOutBuffer = NULL; + BYTE* rOutByteBuffer = NULL; + + MFT_OUTPUT_DATA_BUFFER rOutDataBuffer; + DWORD rStatus; + DWORD rBytes = outputBufferSize; // maximum bytes accepted per ProcessOutput + + // 7.1 Create Sample object for output data + + memset( &rOutDataBuffer, 0, sizeof rOutDataBuffer ); + MFCreateSample( &( rOutDataBuffer.pSample ) ); + MFCreateMemoryBuffer( rBytes, &rOutBuffer ); + rOutDataBuffer.pSample->AddBuffer( rOutBuffer ); + rOutDataBuffer.dwStreamID = 0; + rOutDataBuffer.dwStatus = 0; + rOutDataBuffer.pEvents = NULL; + + // 7.2 Get output data from Resampler + + if ( _transform->ProcessOutput( 0, 1, &rOutDataBuffer, &rStatus ) == MF_E_TRANSFORM_NEED_MORE_INPUT ) + { + outSampleCount = 0; + SAFE_RELEASE( rOutBuffer ); + SAFE_RELEASE( rOutDataBuffer.pSample ); + return; } + + // 7.3 Write output data to outBuffer + + SAFE_RELEASE( rOutBuffer ); + rOutDataBuffer.pSample->ConvertToContiguousBuffer( &rOutBuffer ); + rOutBuffer->GetCurrentLength( &rBytes ); + + rOutBuffer->Lock( &rOutByteBuffer, NULL, NULL ); + memcpy( outBuffer, rOutByteBuffer, rBytes ); + rOutBuffer->Unlock(); + rOutByteBuffer = NULL; + + outSampleCount = rBytes / _bytesPerSample / _channelCount; + SAFE_RELEASE( rOutBuffer ); + SAFE_RELEASE( rOutDataBuffer.pSample ); } -} + +private: + unsigned int _bytesPerSample; + unsigned int _channelCount; + float _sampleRatio; + + IUnknown* _transformUnk; + IMFTransform* _transform; + IMFMediaType* _mediaType; + IMFMediaType* _inputMediaType; + IMFMediaType* _outputMediaType; + + #ifdef __IWMResamplerProps_FWD_DEFINED__ + IWMResamplerProps* _resamplerProps; + #endif +}; //----------------------------------------------------------------------------- @@ -4044,10 +4196,9 @@ 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; } //----------------------------------------------------------------------------- @@ -4074,6 +4225,9 @@ 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 ); @@ -4260,7 +4414,9 @@ RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device ) hr = audioClient->GetMixFormat( &deviceFormat ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format."; + char error[256]; + snprintf(error, sizeof(error), "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format (%d)", hr); + errorText_ = error; goto Exit; } @@ -4416,6 +4572,7 @@ void RtApiWasapi::closeStream( void ) void RtApiWasapi::startStream( void ) { verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiWasapi::startStream: The stream is already running."; @@ -4423,6 +4580,10 @@ void RtApiWasapi::startStream( void ) return; } + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + // update stream state stream_.state = STREAM_RUNNING; @@ -4462,26 +4623,6 @@ void RtApiWasapi::stopStream( void ) // 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."; @@ -4512,26 +4653,6 @@ void RtApiWasapi::abortStream( void ) 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."; @@ -4599,7 +4720,7 @@ 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; @@ -4619,28 +4740,66 @@ 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; } - // retrieve renderAudioClient from devicePtr + 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; + } + + 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 ) ) { @@ -4651,13 +4810,13 @@ 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; } @@ -4693,7 +4852,8 @@ 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 ) @@ -4786,6 +4946,8 @@ void RtApiWasapi::wasapiThread() float renderSrRatio = 0.0f; WasapiBuffer captureBuffer; WasapiBuffer renderBuffer; + WasapiResampler* captureResampler = NULL; + WasapiResampler* renderResampler = NULL; // declare local stream variables RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback; @@ -4794,7 +4956,8 @@ void RtApiWasapi::wasapiThread() unsigned int bufferFrameCount = 0; unsigned int numFramesPadding = 0; unsigned int convBufferSize = 0; - bool callbackPushed = false; + bool loopbackEnabled = stream_.device[INPUT] == stream_.device[OUTPUT]; + bool callbackPushed = true; bool callbackPulled = false; bool callbackStopped = false; int callbackResult = 0; @@ -4804,14 +4967,15 @@ void RtApiWasapi::wasapiThread() unsigned int convBuffSize = 0; unsigned int deviceBuffSize = 0; - errorText_.clear(); + std::string errorText; 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" ); + TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = + ( TAvSetMmThreadCharacteristicsPtr ) (void(*)()) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" ); AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex ); FreeLibrary( AvrtDll ); } @@ -4820,112 +4984,119 @@ void RtApiWasapi::wasapiThread() if ( captureAudioClient ) { hr = captureAudioClient->GetMixFormat( &captureFormat ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format."; + errorText = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format."; goto Exit; } - captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate ); + // init captureResampler + captureResampler = new WasapiResampler( stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[INPUT] == RTAUDIO_FLOAT64, + formatBytes( stream_.deviceFormat[INPUT] ) * 8, stream_.nDeviceChannels[INPUT], + 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 ); + captureSrRatio = ( ( float ) captureFormat->nSamplesPerSec / stream_.sampleRate ); 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 ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client."; + 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."; + 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; + // 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; + } + + ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent; } - hr = captureAudioClient->SetEventHandle( captureEvent ); + ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient; + + // reset the capture stream + hr = captureAudioClient->Reset(); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle."; + errorText = "RtApiWasapi::wasapiThread: Unable to reset capture stream."; goto Exit; } - ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient; - ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent; + // start the capture stream + hr = captureAudioClient->Start(); + if ( FAILED( hr ) ) { + errorText = "RtApiWasapi::wasapiThread: Unable to start capture stream."; + goto Exit; + } } unsigned int inBufferSize = 0; hr = captureAudioClient->GetBufferSize( &inBufferSize ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size."; + 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]; + unsigned int outBufferSize = ( unsigned int ) ceilf( 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."; + errorText = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format."; goto Exit; } - renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate ); + // init renderResampler + renderResampler = new WasapiResampler( stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT32 || stream_.deviceFormat[OUTPUT] == RTAUDIO_FLOAT64, + formatBytes( stream_.deviceFormat[OUTPUT] ) * 8, stream_.nDeviceChannels[OUTPUT], + stream_.sampleRate, renderFormat->nSamplesPerSec ); - // initialize render stream according to desire buffer size - float desiredBufferSize = stream_.bufferSize * renderSrRatio; - REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) desiredBufferSize * 10000000 / renderFormat->nSamplesPerSec ); + renderSrRatio = ( ( float ) renderFormat->nSamplesPerSec / stream_.sampleRate ); if ( !renderClient ) { hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_EVENTCALLBACK, - desiredBufferPeriod, - desiredBufferPeriod, + 0, + 0, renderFormat, NULL ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client."; + 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."; + errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle."; goto Exit; } @@ -4933,70 +5104,75 @@ void RtApiWasapi::wasapiThread() renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL ); if ( !renderEvent ) { errorType = RtAudioError::SYSTEM_ERROR; - errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event."; + 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."; + errorText = "RtApiWasapi::wasapiThread: Unable to set render event handle."; goto Exit; } ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient; ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent; + + // 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; + } } unsigned int outBufferSize = 0; hr = renderAudioClient->GetBufferSize( &outBufferSize ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size."; + 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]; + unsigned int inBufferSize = ( unsigned int ) ceilf( 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 ) { - using namespace std; // for roundf - convBuffSize = ( size_t ) roundf( stream_.bufferSize * captureSrRatio ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ); + // malloc buffer memory + if ( stream_.mode == INPUT ) + { + using namespace std; // for ceilf + convBuffSize = ( size_t ) ( ceilf( 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] ); + else if ( stream_.mode == OUTPUT ) + { + convBuffSize = ( size_t ) ( ceilf( 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] ) ); + else if ( stream_.mode == DUPLEX ) + { + convBuffSize = std::max( ( size_t ) ( ceilf( stream_.bufferSize * captureSrRatio ) ) * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ), + ( size_t ) ( ceilf( 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 ); + convBuffSize *= 2; // allow overflow for *SrRatio remainders + convBuffer = ( char* ) calloc( convBuffSize, 1 ); + stream_.deviceBuffer = ( char* ) calloc( deviceBuffSize, 1 ); if ( !convBuffer || !stream_.deviceBuffer ) { errorType = RtAudioError::MEMORY_ERROR; - errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory."; + errorText = "RtApiWasapi::wasapiThread: Error allocating device buffer memory."; goto Exit; } @@ -5009,23 +5185,43 @@ 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], @@ -5060,18 +5256,21 @@ void RtApiWasapi::wasapiThread() captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0, stream_.callbackInfo.userData ); + // tick stream time + RtApi::tickStreamTime(); + // 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."; + 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."; + errorText = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle."; goto Exit; } @@ -5082,12 +5281,12 @@ void RtApiWasapi::wasapiThread() HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL ); if ( !threadHandle ) { errorType = RtAudioError::THREAD_ERROR; - errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread."; + 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."; + errorText = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle."; goto Exit; } @@ -5102,24 +5301,32 @@ 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] ); - } + } + else { + // no further conversion, simple copy userBuffer to deviceBuffer + memcpy( stream_.deviceBuffer, + stream_.userBuffer[OUTPUT], + stream_.bufferSize * stream_.nUserChannels[OUTPUT] * formatBytes( stream_.userFormat ) ); + } - // 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, @@ -5140,7 +5347,7 @@ 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 @@ -5148,7 +5355,7 @@ void RtApiWasapi::wasapiThread() &bufferFrameCount, &captureFlags, NULL, NULL ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer."; + errorText = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer."; goto Exit; } @@ -5161,7 +5368,7 @@ void RtApiWasapi::wasapiThread() // Release capture buffer hr = captureClient->ReleaseBuffer( bufferFrameCount ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; + errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; goto Exit; } } @@ -5170,7 +5377,7 @@ void RtApiWasapi::wasapiThread() // Inform WASAPI that capture was unsuccessful hr = captureClient->ReleaseBuffer( 0 ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; + errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; goto Exit; } } @@ -5180,7 +5387,7 @@ void RtApiWasapi::wasapiThread() // Inform WASAPI that capture was unsuccessful hr = captureClient->ReleaseBuffer( 0 ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; + errorText = "RtApiWasapi::wasapiThread: Unable to release capture buffer."; goto Exit; } } @@ -5202,13 +5409,13 @@ void RtApiWasapi::wasapiThread() // Get render buffer from stream hr = renderAudioClient->GetBufferSize( &bufferFrameCount ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size."; + 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."; + errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding."; goto Exit; } @@ -5217,7 +5424,7 @@ void RtApiWasapi::wasapiThread() if ( bufferFrameCount != 0 ) { hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer."; + errorText = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer."; goto Exit; } @@ -5230,7 +5437,7 @@ void RtApiWasapi::wasapiThread() // Release render buffer hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer."; + errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer."; goto Exit; } } @@ -5239,7 +5446,7 @@ void RtApiWasapi::wasapiThread() // Inform WASAPI that render was unsuccessful hr = renderClient->ReleaseBuffer( 0, 0 ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer."; + errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer."; goto Exit; } } @@ -5249,7 +5456,7 @@ void RtApiWasapi::wasapiThread() // Inform WASAPI that render was unsuccessful hr = renderClient->ReleaseBuffer( 0, 0 ); if ( FAILED( hr ) ) { - errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer."; + errorText = "RtApiWasapi::wasapiThread: Unable to release render buffer."; goto Exit; } } @@ -5257,9 +5464,9 @@ 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(); } } @@ -5270,16 +5477,19 @@ Exit: CoTaskMemFree( renderFormat ); free ( convBuffer ); + delete renderResampler; + delete captureResampler; CoUninitialize(); // update stream state stream_.state = STREAM_STOPPED; - if ( errorText_.empty() ) - return; - else + if ( !errorText.empty() ) + { + errorText_ = errorText; error( errorType ); + } } //******************** End of __WINDOWS_WASAPI__ *********************// @@ -5289,12 +5499,14 @@ Exit: #if defined(__WINDOWS_DS__) // Windows DirectSound API // Modified by Robin Davies, October 2005 -// - Improvements to DirectX pointer chasing. +// - 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 +#include #include #include #include @@ -5331,7 +5543,7 @@ struct DsHandle { void *id[2]; void *buffer[2]; bool xrun[2]; - UINT bufferPointer[2]; + UINT bufferPointer[2]; DWORD dsBufferSize[2]; DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by. HANDLE condition; @@ -6168,18 +6380,23 @@ void RtApiDs :: closeStream() void RtApiDs :: startStream() { verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiDs::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); return; } + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + 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 ); + timeBeginPeriod( 1 ); buffersRolling = false; duplexPrerollBytes = 0; @@ -6224,6 +6441,7 @@ void RtApiDs :: startStream() void RtApiDs :: stopStream() { verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_STOPPED ) { errorText_ = "RtApiDs::stopStream(): the stream is already stopped!"; error( RtAudioError::WARNING ); @@ -6500,7 +6718,7 @@ void RtApiDs :: callbackEvent() } if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - + LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0]; if ( handle->drainCounter > 1 ) { // write zeros to the output stream @@ -6567,7 +6785,7 @@ void RtApiDs :: callbackEvent() } if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize ) - || dsPointerBetween( endWrite, 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; @@ -6641,14 +6859,14 @@ void RtApiDs :: callbackEvent() if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset DWORD endRead = nextReadPointer + bufferBytes; - // Handling depends on whether we are INPUT or DUPLEX. + // 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 + // + // 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 @@ -6699,7 +6917,7 @@ void RtApiDs :: callbackEvent() error( RtAudioError::SYSTEM_ERROR ); return; } - + if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset } } @@ -6941,7 +7159,7 @@ unsigned int RtApiAlsa :: getDeviceCount( void ) unsigned nDevices = 0; int result, subdevice, card; char name[64]; - snd_ctl_t *handle; + snd_ctl_t *handle = 0; // Count cards and devices card = -1; @@ -6950,6 +7168,7 @@ unsigned int RtApiAlsa :: getDeviceCount( void ) 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 ); @@ -6969,7 +7188,8 @@ unsigned int RtApiAlsa :: getDeviceCount( void ) nDevices++; } nextcard: - snd_ctl_close( handle ); + if ( handle ) + snd_ctl_close( handle ); snd_card_next( &card ); } @@ -6990,7 +7210,7 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) unsigned nDevices = 0; int result, subdevice, card; char name[64]; - snd_ctl_t *chandle; + snd_ctl_t *chandle = 0; // Count cards and devices card = -1; @@ -7000,6 +7220,7 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) 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 ); @@ -7022,7 +7243,8 @@ RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device ) nDevices++; } nextcard: - snd_ctl_close( chandle ); + if ( chandle ) + snd_ctl_close( chandle ); snd_card_next( &card ); } @@ -7331,10 +7553,12 @@ 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. @@ -7734,30 +7958,41 @@ 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 ) { - // 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; + 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; - stream_.callbackInfo.priority = priority; + param.sched_priority = priority; + + // Set the policy BEFORE the priority. Otherwise it fails. + pthread_attr_setschedpolicy(&attr, SCHED_RR); + pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM); + // This is definitely required. Otherwise it fails. + pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED); + pthread_attr_setschedparam(&attr, ¶m); } + else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); #endif stream_.callbackInfo.isRunning = true; result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo ); pthread_attr_destroy( &attr ); if ( result ) { - stream_.callbackInfo.isRunning = false; - errorText_ = "RtApiAlsa::error creating callback thread!"; - goto error; + // Failed. Try instead with default attributes. + result = pthread_create( &stream_.callbackInfo.thread, NULL, alsaCallbackHandler, &stream_.callbackInfo ); + if ( result ) { + stream_.callbackInfo.isRunning = false; + errorText_ = "RtApiAlsa::error creating callback thread!"; + goto error; + } } } @@ -7845,6 +8080,7 @@ void RtApiAlsa :: startStream() // This method calls snd_pcm_prepare if the device isn't already in that state. verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiAlsa::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); @@ -7853,6 +8089,10 @@ void RtApiAlsa :: startStream() MUTEX_LOCK( &stream_.mutex ); + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + int result = 0; snd_pcm_state_t state; AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; @@ -7909,7 +8149,7 @@ void RtApiAlsa :: stopStream() AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle; snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles; if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) { - if ( apiInfo->synchronized ) + if ( apiInfo->synchronized ) result = snd_pcm_drop( handle[0] ); else result = snd_pcm_drain( handle[0] ); @@ -8172,11 +8412,11 @@ 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 ) { - pthread_t tID = pthread_self(); // ID of this thread - sched_param prio = { info->priority }; // scheduling priority of thread - pthread_setschedparam( tID, SCHED_RR, &prio ); + std::cerr << "RtAudio alsa: " << + (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") << + "running realtime scheduling" << std::endl; } #endif @@ -8198,6 +8438,7 @@ static void *alsaCallbackHandler( void *ptr ) #include #include +#include #include static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000, @@ -8234,8 +8475,33 @@ unsigned int RtApiPulse::getDeviceCount( void ) return 1; } +void RtApiPulse::sinkInfoCallback(pa_context*, const pa_sink_info* info, int, void* arg) +{ + RtApiPulse* api = (RtApiPulse *) arg; + if (info) { + api->channels_ = info->sample_spec.channels; + } + pa_threaded_mainloop_signal(api->mainloop_, 0); +} + +void RtApiPulse::contextStateCallback(pa_context* c, void* arg) +{ + pa_threaded_mainloop* mainloop = (pa_threaded_mainloop*) arg; + + switch (pa_context_get_state(c)) { + case PA_CONTEXT_READY: + case PA_CONTEXT_TERMINATED: + case PA_CONTEXT_FAILED: + pa_threaded_mainloop_signal(mainloop, 0); + break; + default: + break; + } +} + RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ ) { + /* Set up some defaults in case we crash and burn */ RtAudio::DeviceInfo info; info.probed = true; info.name = "PulseAudio"; @@ -8251,6 +8517,72 @@ RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ ) info.preferredSampleRate = 48000; info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32; + /* Get the number of output channels from pulseaudio. A simple task, you say? + "What is your mainloop?" */ + mainloop_ = pa_threaded_mainloop_new(); + if (!mainloop_) { + return info; + } + + pa_threaded_mainloop_start(mainloop_); + pa_threaded_mainloop_lock(mainloop_); + + /* "And what is your context?" */ + pa_context* context = pa_context_new(pa_threaded_mainloop_get_api(mainloop_), "RtAudio"); + if (!context) { + pa_threaded_mainloop_unlock(mainloop_); + pa_threaded_mainloop_stop(mainloop_); + pa_threaded_mainloop_free(mainloop_); + mainloop_ = 0; + return info; + } + + pa_context_set_state_callback(context, contextStateCallback, mainloop_); + + pa_context_connect(context, 0, (pa_context_flags_t) 0, 0); + + /* "And what is your favourite colour?" */ + int connected = 0; + pa_context_state_t state = pa_context_get_state(context); + for (; !connected; state = pa_context_get_state(context)) { + switch (state) { + case PA_CONTEXT_READY: + connected = 1; + continue; + case PA_CONTEXT_FAILED: + case PA_CONTEXT_TERMINATED: + /* Blue! No, I mean red! */ + pa_threaded_mainloop_unlock(mainloop_); + pa_context_disconnect(context); + pa_context_unref(context); + pa_threaded_mainloop_stop(mainloop_); + pa_threaded_mainloop_free(mainloop_); + mainloop_ = 0; + return info; + default: + pa_threaded_mainloop_wait(mainloop_); + break; + } + } + + pa_operation* op = pa_context_get_sink_info_by_index(context, 0, sinkInfoCallback, this); + + if (op) { + pa_operation_unref(op); + } + + pa_threaded_mainloop_wait(mainloop_); + pa_threaded_mainloop_unlock(mainloop_); + + pa_context_disconnect(context); + pa_context_unref(context); + + pa_threaded_mainloop_stop(mainloop_); + pa_threaded_mainloop_free(mainloop_); + mainloop_ = 0; + + info.outputChannels = channels_; + return info; } @@ -8260,6 +8592,14 @@ static void *pulseaudio_callback( void * user ) RtApiPulse *context = static_cast( cbi->object ); volatile bool *isRunning = &cbi->isRunning; +#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_ ") << + "running realtime scheduling" << std::endl; + } +#endif + while ( *isRunning ) { pthread_testcancel(); context->callbackEvent(); @@ -8377,7 +8717,7 @@ void RtApiPulse::callbackEvent( void ) 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 ) << "."; @@ -8395,12 +8735,21 @@ void RtApiPulse::callbackEvent( void ) MUTEX_UNLOCK( &stream_.mutex ); RtApi::tickStreamTime(); + if (pah->s_play) { + int e = 0; + pa_usec_t const lat = pa_simple_get_latency(pah->s_play, &e); + if (e == 0) { + stream_.latency[0] = lat * stream_.sampleRate / 1000000; + } + } + if ( doStopStream == 1 ) stopStream(); } void RtApiPulse::startStream( void ) { + RtApi::startStream(); PulseAudioHandle *pah = static_cast( stream_.apiHandle ); if ( stream_.state == STREAM_CLOSED ) { @@ -8416,6 +8765,10 @@ void RtApiPulse::startStream( void ) MUTEX_LOCK( &stream_.mutex ); + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + stream_.state = STREAM_RUNNING; pah->runnable = true; @@ -8439,6 +8792,7 @@ void RtApiPulse::stopStream( void ) } stream_.state = STREAM_STOPPED; + pah->runnable = false; MUTEX_LOCK( &stream_.mutex ); if ( pah && pah->s_play ) { @@ -8473,6 +8827,7 @@ void RtApiPulse::abortStream( void ) } stream_.state = STREAM_STOPPED; + pah->runnable = false; MUTEX_LOCK( &stream_.mutex ); if ( pah && pah->s_play ) { @@ -8502,10 +8857,6 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, 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; @@ -8625,7 +8976,38 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, } break; case OUTPUT: - pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error ); + /* XXX: hard-coded for DCP-o-matic */ + pa_channel_map map; + pa_channel_map_init(&map); + /* XXX: need to check 7.1 */ + map.channels = channels; + + if (channels > 0) { + map.map[0] = PA_CHANNEL_POSITION_FRONT_LEFT; + } + if (channels > 1) { + map.map[1] = PA_CHANNEL_POSITION_FRONT_RIGHT; + } + if (channels > 2) { + map.map[2] = PA_CHANNEL_POSITION_FRONT_CENTER; + } + if (channels > 3) { + map.map[3] = PA_CHANNEL_POSITION_LFE; + } + if (channels > 4) { + map.map[4] = PA_CHANNEL_POSITION_REAR_LEFT; + } + if (channels > 5) { + map.map[5] = PA_CHANNEL_POSITION_REAR_RIGHT; + } + if (channels > 6) { + map.map[6] = PA_CHANNEL_POSITION_SIDE_LEFT; + } + if (channels > 7) { + map.map[7] = PA_CHANNEL_POSITION_SIDE_RIGHT; + } + + pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, &map, NULL, &error ); if ( !pah->s_play ) { errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server."; goto error; @@ -8644,16 +9026,57 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, if ( !stream_.callbackInfo.isRunning ) { stream_.callbackInfo.object = this; + + stream_.state = STREAM_STOPPED; + // 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 (e.g. NetBSD 1.6.x with GNU Pthread) + if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) { + stream_.callbackInfo.doRealtime = true; + 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; + + // Set the policy BEFORE the priority. Otherwise it fails. + pthread_attr_setschedpolicy(&attr, SCHED_RR); + pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM); + // This is definitely required. Otherwise it fails. + pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED); + pthread_attr_setschedparam(&attr, ¶m); + } + else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#else + pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); +#endif + stream_.callbackInfo.isRunning = true; - if ( pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo) != 0 ) { - errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread."; - goto error; + int result = pthread_create( &pah->thread, &attr, pulseaudio_callback, (void *)&stream_.callbackInfo); + pthread_attr_destroy(&attr); + if(result != 0) { + // Failed. Try instead with default attributes. + result = pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo); + if(result != 0) { + stream_.callbackInfo.isRunning = false; + errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread."; + goto error; + } } } - stream_.state = STREAM_STOPPED; - return true; - + return SUCCESS; + error: if ( pah && stream_.callbackInfo.isRunning ) { pthread_cond_destroy( &pah->runnable_cv ); @@ -8673,6 +9096,7 @@ bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode, stream_.deviceBuffer = 0; } + stream_.state = STREAM_CLOSED; return FAILURE; } @@ -9234,8 +9658,9 @@ 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; int priority = options->priority; int min = sched_get_priority_min( SCHED_RR ); @@ -9243,8 +9668,13 @@ bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned 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 ); + + // Set the policy BEFORE the priority. Otherwise it fails. + pthread_attr_setschedpolicy(&attr, SCHED_RR); + pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM); + // This is definitely required. Otherwise it fails. + pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED); + pthread_attr_setschedparam(&attr, ¶m); } else pthread_attr_setschedpolicy( &attr, SCHED_OTHER ); @@ -9256,9 +9686,13 @@ bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned 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; + // Failed. Try instead with default attributes. + result = pthread_create( &stream_.callbackInfo.thread, NULL, ossCallbackHandler, &stream_.callbackInfo ); + if ( result ) { + stream_.callbackInfo.isRunning = false; + errorText_ = "RtApiOss::error creating callback thread!"; + goto error; + } } } @@ -9285,6 +9719,7 @@ bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned stream_.deviceBuffer = 0; } + stream_.state = STREAM_CLOSED; return FAILURE; } @@ -9339,6 +9774,7 @@ void RtApiOss :: closeStream() void RtApiOss :: startStream() { verifyStream(); + RtApi::startStream(); if ( stream_.state == STREAM_RUNNING ) { errorText_ = "RtApiOss::startStream(): the stream is already running!"; error( RtAudioError::WARNING ); @@ -9347,6 +9783,10 @@ void RtApiOss :: startStream() MUTEX_LOCK( &stream_.mutex ); + #if defined( HAVE_GETTIMEOFDAY ) + gettimeofday( &stream_.lastTickTimestamp, NULL ); + #endif + stream_.state = STREAM_RUNNING; // No need to do anything else here ... OSS automatically starts @@ -9614,6 +10054,14 @@ static void *ossCallbackHandler( void *ptr ) RtApiOss *object = (RtApiOss *) info->object; bool *isRunning = &info->isRunning; +#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_ ") << + "running realtime scheduling" << std::endl; + } +#endif + while ( *isRunning == true ) { pthread_testcancel(); object->callbackEvent(); @@ -10330,4 +10778,3 @@ void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat // End: // // vim: et sts=2 sw=2 -