-/************************************************************************/
+/************************************************************************/
/*! \class RtAudio
\brief Realtime audio i/o C++ classes.
return RTAUDIO_VERSION;
}
-void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &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<bool b> class StaticAssert { private: StaticAssert() {} };
+template<> class StaticAssert<true>{ public: StaticAssert() {} };
+class StaticAssertions { StaticAssertions() {
+ StaticAssert<rtaudio_num_api_names == RtAudio::NUM_APIS>();
+}};
+
+void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis )
+{
+ apis = std::vector<RtAudio::Api>(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 )
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 );
#ifndef INITGUID
#define INITGUID
#endif
+
+#include <mfapi.h>
+#include <mferror.h>
+#include <mfplay.h>
+#include <mftransform.h>
+#include <wmcodecdsp.h>
+
#include <audioclient.h>
#include <avrt.h>
#include <mmdeviceapi.h>
#include <functiondiscoverykeys_devpkey.h>
-#include <mfapi.h>
-#include <mferror.h>
-#include <mfplay.h>
-#include <Wmcodecdsp.h>
+#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
//=============================================================================
, _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
_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 );
// 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
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 )
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 ) );
IUnknown* _transformUnk;
IMFTransform* _transform;
- IWMResamplerProps* _resamplerProps;
IMFMediaType* _mediaType;
IMFMediaType* _inputMediaType;
IMFMediaType* _outputMediaType;
+
+ #ifdef __IWMResamplerProps_FWD_DEFINED__
+ IWMResamplerProps* _resamplerProps;
+ #endif
};
//-----------------------------------------------------------------------------
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;
}
//-----------------------------------------------------------------------------
IMMDeviceCollection* captureDevices = NULL;
IMMDeviceCollection* renderDevices = NULL;
+ if ( !deviceEnumerator_ )
+ return 0;
+
// Count capture devices
errorText_.clear();
HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
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;
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 ) ) {
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;
}
// 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 )
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;
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 ) ) {
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;
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 ) ) {
// 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],
// 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,
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
// 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();
}
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__ *********************//
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.
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;
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_ ") <<
RtApiPulse *context = static_cast<RtApiPulse *>( 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_ ") <<
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;
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;
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_ ") <<
projects / rtaudio-cdist.git / blobdiff