/************************************************************************/
/*! \class RtAudio
- \brief Realtime audio i/o C++ class.
+ \brief Realtime audio i/o C++ classes.
RtAudio provides a common API (Application Programming Interface)
- for realtime audio input/output across Linux (native ALSA and
- OSS), SGI, Macintosh OS X (CoreAudio), and Windows (DirectSound
- and ASIO) operating systems.
+ for realtime audio input/output across Linux (native ALSA, Jack,
+ and OSS), SGI, Macintosh OS X (CoreAudio and Jack), and Windows
+ (DirectSound and ASIO) operating systems.
- RtAudio WWW site: http://www-ccrma.stanford.edu/~gary/rtaudio/
+ RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
- RtAudio: a realtime audio i/o C++ class
- Copyright (c) 2001-2002 Gary P. Scavone
+ RtAudio: realtime audio i/o C++ classes
+ Copyright (c) 2001-2008 Gary P. Scavone
Permission is hereby granted, free of charge, to any person
obtaining a copy of this software and associated documentation files
included in all copies or substantial portions of the Software.
Any person wishing to distribute modifications to the Software is
- requested to send the modifications to the original developer so that
- they can be incorporated into the canonical version.
+ asked to send the modifications to the original developer so that
+ they can be incorporated into the canonical version. This is,
+ however, not a binding provision of this license.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
*/
/************************************************************************/
+// RtAudio: Version 4.0.4
#include "RtAudio.h"
-#include <vector>
-#include <stdio.h>
-#include <iostream.h>
+#include <iostream>
// Static variable definitions.
-const unsigned int RtAudio :: SAMPLE_RATES[] = {
+const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
+const unsigned int RtApi::SAMPLE_RATES[] = {
4000, 5512, 8000, 9600, 11025, 16000, 22050,
32000, 44100, 48000, 88200, 96000, 176400, 192000
};
-const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT8 = 1;
-const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT16 = 2;
-const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT24 = 4;
-const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_SINT32 = 8;
-const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_FLOAT32 = 16;
-const RtAudio::RTAUDIO_FORMAT RtAudio :: RTAUDIO_FLOAT64 = 32;
#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__)
#define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
+ #define MUTEX_DESTROY(A) DeleteCriticalSection(A)
#define MUTEX_LOCK(A) EnterCriticalSection(A)
#define MUTEX_UNLOCK(A) LeaveCriticalSection(A)
-#else // pthread API
+#elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
+ // pthread API
#define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
+ #define MUTEX_DESTROY(A) pthread_mutex_destroy(A)
#define MUTEX_LOCK(A) pthread_mutex_lock(A)
#define MUTEX_UNLOCK(A) pthread_mutex_unlock(A)
+#else
+ #define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions
+ #define MUTEX_DESTROY(A) abs(*A) // dummy definitions
#endif
// *************************************************** //
//
-// Public common (OS-independent) methods.
+// RtAudio definitions.
//
// *************************************************** //
-RtAudio :: RtAudio()
+void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()
{
- initialize();
+ apis.clear();
+
+ // The order here will control the order of RtAudio's API search in
+ // the constructor.
+#if defined(__UNIX_JACK__)
+ apis.push_back( UNIX_JACK );
+#endif
+#if defined(__LINUX_ALSA__)
+ apis.push_back( LINUX_ALSA );
+#endif
+#if defined(__LINUX_OSS__)
+ apis.push_back( LINUX_OSS );
+#endif
+#if defined(__WINDOWS_ASIO__)
+ apis.push_back( WINDOWS_ASIO );
+#endif
+#if defined(__WINDOWS_DS__)
+ apis.push_back( WINDOWS_DS );
+#endif
+#if defined(__MACOSX_CORE__)
+ apis.push_back( MACOSX_CORE );
+#endif
+#if defined(__RTAUDIO_DUMMY__)
+ apis.push_back( RTAUDIO_DUMMY );
+#endif
+}
- if (nDevices <= 0) {
- sprintf(message, "RtAudio: no audio devices found!");
- error(RtError::NO_DEVICES_FOUND);
- }
+void RtAudio :: openRtApi( RtAudio::Api api )
+{
+#if defined(__UNIX_JACK__)
+ if ( api == UNIX_JACK )
+ rtapi_ = new RtApiJack();
+#endif
+#if defined(__LINUX_ALSA__)
+ if ( api == LINUX_ALSA )
+ rtapi_ = new RtApiAlsa();
+#endif
+#if defined(__LINUX_OSS__)
+ if ( api == LINUX_OSS )
+ rtapi_ = new RtApiOss();
+#endif
+#if defined(__WINDOWS_ASIO__)
+ if ( api == WINDOWS_ASIO )
+ rtapi_ = new RtApiAsio();
+#endif
+#if defined(__WINDOWS_DS__)
+ if ( api == WINDOWS_DS )
+ rtapi_ = new RtApiDs();
+#endif
+#if defined(__MACOSX_CORE__)
+ if ( api == MACOSX_CORE )
+ rtapi_ = new RtApiCore();
+#endif
+#if defined(__RTAUDIO_DUMMY__)
+ if ( api == RTAUDIO_DUMMY )
+ rtapi_ = new RtApiDummy();
+#endif
}
-RtAudio :: RtAudio(int *streamId,
- int outputDevice, int outputChannels,
- int inputDevice, int inputChannels,
- RTAUDIO_FORMAT format, int sampleRate,
- int *bufferSize, int numberOfBuffers)
+RtAudio :: RtAudio( RtAudio::Api api ) throw()
{
- initialize();
+ rtapi_ = 0;
- if (nDevices <= 0) {
- sprintf(message, "RtAudio: no audio devices found!");
- error(RtError::NO_DEVICES_FOUND);
- }
+ if ( api != UNSPECIFIED ) {
+ // Attempt to open the specified API.
+ openRtApi( api );
+ if ( rtapi_ ) return;
- try {
- *streamId = openStream(outputDevice, outputChannels, inputDevice, inputChannels,
- format, sampleRate, bufferSize, numberOfBuffers);
+ // No compiled support for specified API value. Issue a debug
+ // warning and continue as if no API was specified.
+ std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
}
- catch (RtError &exception) {
- // deallocate the RTAUDIO_DEVICE structures
- if (devices) free(devices);
- throw exception;
+
+ // Iterate through the compiled APIs and return as soon as we find
+ // one with at least one device or we reach the end of the list.
+ std::vector< RtAudio::Api > apis;
+ getCompiledApi( apis );
+ for ( unsigned int i=0; i<apis.size(); i++ ) {
+ openRtApi( apis[i] );
+ if ( rtapi_->getDeviceCount() ) break;
}
+
+ if ( rtapi_ ) return;
+
+ // It should not be possible to get here because the preprocessor
+ // definition __RTAUDIO_DUMMY__ is automatically defined if no
+ // API-specific definitions are passed to the compiler. But just in
+ // case something weird happens, we'll print out an error message.
+ std::cerr << "\nRtAudio: no compiled API support found ... critical error!!\n\n";
+}
+
+RtAudio :: ~RtAudio() throw()
+{
+ delete rtapi_;
}
-RtAudio :: ~RtAudio()
+void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
+ RtAudio::StreamParameters *inputParameters,
+ RtAudioFormat format, unsigned int sampleRate,
+ unsigned int *bufferFrames,
+ RtAudioCallback callback, void *userData,
+ RtAudio::StreamOptions *options )
{
- // close any existing streams
- while ( streams.size() )
- closeStream( streams.begin()->first );
+ return rtapi_->openStream( outputParameters, inputParameters, format,
+ sampleRate, bufferFrames, callback,
+ userData, options );
+}
+
+// *************************************************** //
+//
+// Public RtApi definitions (see end of file for
+// private or protected utility functions).
+//
+// *************************************************** //
- // deallocate the RTAUDIO_DEVICE structures
- if (devices) free(devices);
+RtApi :: RtApi()
+{
+ stream_.state = STREAM_CLOSED;
+ stream_.mode = UNINITIALIZED;
+ stream_.apiHandle = 0;
+ stream_.userBuffer[0] = 0;
+ stream_.userBuffer[1] = 0;
+ MUTEX_INITIALIZE( &stream_.mutex );
+ showWarnings_ = true;
}
-int RtAudio :: openStream(int outputDevice, int outputChannels,
- int inputDevice, int inputChannels,
- RTAUDIO_FORMAT format, int sampleRate,
- int *bufferSize, int numberOfBuffers)
+RtApi :: ~RtApi()
{
- static int streamKey = 0; // Unique stream identifier ... OK for multiple instances.
+ MUTEX_DESTROY( &stream_.mutex );
+}
- if (outputChannels < 1 && inputChannels < 1) {
- sprintf(message,"RtAudio: one or both 'channel' parameters must be greater than zero.");
- error(RtError::INVALID_PARAMETER);
+void RtApi :: openStream( RtAudio::StreamParameters *oParams,
+ RtAudio::StreamParameters *iParams,
+ RtAudioFormat format, unsigned int sampleRate,
+ unsigned int *bufferFrames,
+ RtAudioCallback callback, void *userData,
+ RtAudio::StreamOptions *options )
+{
+ if ( stream_.state != STREAM_CLOSED ) {
+ errorText_ = "RtApi::openStream: a stream is already open!";
+ error( RtError::INVALID_USE );
}
- if ( formatBytes(format) == 0 ) {
- sprintf(message,"RtAudio: 'format' parameter value is undefined.");
- error(RtError::INVALID_PARAMETER);
+ if ( oParams && oParams->nChannels < 1 ) {
+ errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
+ error( RtError::INVALID_USE );
}
- if ( outputChannels > 0 ) {
- if (outputDevice > nDevices || outputDevice < 0) {
- sprintf(message,"RtAudio: 'outputDevice' parameter value (%d) is invalid.", outputDevice);
- error(RtError::INVALID_PARAMETER);
- }
+ if ( iParams && iParams->nChannels < 1 ) {
+ errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
+ error( RtError::INVALID_USE );
}
- if ( inputChannels > 0 ) {
- if (inputDevice > nDevices || inputDevice < 0) {
- sprintf(message,"RtAudio: 'inputDevice' parameter value (%d) is invalid.", inputDevice);
- error(RtError::INVALID_PARAMETER);
- }
+ if ( oParams == NULL && iParams == NULL ) {
+ errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
+ error( RtError::INVALID_USE );
}
- // Allocate a new stream structure.
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) calloc(1, sizeof(RTAUDIO_STREAM));
- if (stream == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
+ if ( formatBytes(format) == 0 ) {
+ errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
+ error( RtError::INVALID_USE );
}
- stream->mode = UNINITIALIZED;
- MUTEX_INITIALIZE(&stream->mutex);
-
- bool result = FAILURE;
- int device, defaultDevice = 0;
- STREAM_MODE mode;
- int channels;
- if ( outputChannels > 0 ) {
-
- mode = OUTPUT;
- channels = outputChannels;
- if ( outputDevice == 0 ) { // Try default device first.
- defaultDevice = getDefaultOutputDevice();
- device = defaultDevice;
+ unsigned int nDevices = getDeviceCount();
+ unsigned int oChannels = 0;
+ if ( oParams ) {
+ oChannels = oParams->nChannels;
+ if ( oParams->deviceId >= nDevices ) {
+ errorText_ = "RtApi::openStream: output device parameter value is invalid.";
+ error( RtError::INVALID_USE );
}
- else
- device = outputDevice - 1;
+ }
- for (int i=-1; i<nDevices; i++) {
- if (i >= 0 ) {
- if ( i == defaultDevice ) continue;
- device = i;
- }
- if (devices[device].probed == false) {
- // If the device wasn't successfully probed before, try it
- // again now.
- clearDeviceInfo(&devices[device]);
- probeDeviceInfo(&devices[device]);
- }
- if ( devices[device].probed )
- result = probeDeviceOpen(device, stream, mode, channels, sampleRate,
- format, bufferSize, numberOfBuffers);
- if (result == SUCCESS) break;
- if ( outputDevice > 0 ) break;
+ unsigned int iChannels = 0;
+ if ( iParams ) {
+ iChannels = iParams->nChannels;
+ if ( iParams->deviceId >= nDevices ) {
+ errorText_ = "RtApi::openStream: input device parameter value is invalid.";
+ error( RtError::INVALID_USE );
}
}
- if ( inputChannels > 0 && ( result == SUCCESS || outputChannels <= 0 ) ) {
+ clearStreamInfo();
+ bool result;
- mode = INPUT;
- channels = inputChannels;
+ if ( oChannels > 0 ) {
- if ( inputDevice == 0 ) { // Try default device first.
- defaultDevice = getDefaultInputDevice();
- device = defaultDevice;
- }
- else
- device = inputDevice - 1;
+ result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
+ sampleRate, format, bufferFrames, options );
+ if ( result == false ) error( RtError::SYSTEM_ERROR );
+ }
- for (int i=-1; i<nDevices; i++) {
- if (i >= 0 ) {
- if ( i == defaultDevice ) continue;
- device = i;
- }
- if (devices[device].probed == false) {
- // If the device wasn't successfully probed before, try it
- // again now.
- clearDeviceInfo(&devices[device]);
- probeDeviceInfo(&devices[device]);
- }
- if ( devices[device].probed )
- result = probeDeviceOpen(device, stream, mode, channels, sampleRate,
- format, bufferSize, numberOfBuffers);
- if (result == SUCCESS) break;
- if ( outputDevice > 0 ) break;
+ if ( iChannels > 0 ) {
+
+ result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
+ sampleRate, format, bufferFrames, options );
+ if ( result == false ) {
+ if ( oChannels > 0 ) closeStream();
+ error( RtError::SYSTEM_ERROR );
}
}
- streams[++streamKey] = (void *) stream;
- if ( result == SUCCESS )
- return streamKey;
-
- // If we get here, all attempted probes failed. Close any opened
- // devices and delete the allocated stream.
- closeStream(streamKey);
- if ( ( outputDevice == 0 && outputChannels > 0 )
- || ( inputDevice == 0 && inputChannels > 0 ) )
- sprintf(message,"RtAudio: no devices found for given parameters.");
- else
- sprintf(message,"RtAudio: unable to open specified device(s) with given stream parameters.");
- error(RtError::INVALID_PARAMETER);
+ stream_.callbackInfo.callback = (void *) callback;
+ stream_.callbackInfo.userData = userData;
- return -1;
+ if ( options ) options->numberOfBuffers = stream_.nBuffers;
+ stream_.state = STREAM_STOPPED;
}
-int RtAudio :: getDeviceCount(void)
+unsigned int RtApi :: getDefaultInputDevice( void )
{
- return nDevices;
+ // Should be implemented in subclasses if possible.
+ return 0;
}
-void RtAudio :: getDeviceInfo(int device, RTAUDIO_DEVICE *info)
+unsigned int RtApi :: getDefaultOutputDevice( void )
{
- if (device > nDevices || device < 1) {
- sprintf(message, "RtAudio: invalid device specifier (%d)!", device);
- error(RtError::INVALID_DEVICE);
- }
-
- int deviceIndex = device - 1;
-
- // If the device wasn't successfully probed before, try it now (or again).
- if (devices[deviceIndex].probed == false) {
- clearDeviceInfo(&devices[deviceIndex]);
- probeDeviceInfo(&devices[deviceIndex]);
- }
-
- // Clear the info structure.
- memset(info, 0, sizeof(RTAUDIO_DEVICE));
-
- strncpy(info->name, devices[deviceIndex].name, 128);
- info->probed = devices[deviceIndex].probed;
- if ( info->probed == true ) {
- info->maxOutputChannels = devices[deviceIndex].maxOutputChannels;
- info->maxInputChannels = devices[deviceIndex].maxInputChannels;
- info->maxDuplexChannels = devices[deviceIndex].maxDuplexChannels;
- info->minOutputChannels = devices[deviceIndex].minOutputChannels;
- info->minInputChannels = devices[deviceIndex].minInputChannels;
- info->minDuplexChannels = devices[deviceIndex].minDuplexChannels;
- info->hasDuplexSupport = devices[deviceIndex].hasDuplexSupport;
- info->nSampleRates = devices[deviceIndex].nSampleRates;
- if (info->nSampleRates == -1) {
- info->sampleRates[0] = devices[deviceIndex].sampleRates[0];
- info->sampleRates[1] = devices[deviceIndex].sampleRates[1];
- }
- else {
- for (int i=0; i<info->nSampleRates; i++)
- info->sampleRates[i] = devices[deviceIndex].sampleRates[i];
- }
- info->nativeFormats = devices[deviceIndex].nativeFormats;
- if ( deviceIndex == getDefaultOutputDevice() ||
- deviceIndex == getDefaultInputDevice() )
- info->isDefault = true;
- }
+ // Should be implemented in subclasses if possible.
+ return 0;
+}
+void RtApi :: closeStream( void )
+{
+ // MUST be implemented in subclasses!
return;
}
-char * const RtAudio :: getStreamBuffer(int streamId)
+bool RtApi :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- return stream->userBuffer;
+ // MUST be implemented in subclasses!
+ return FAILURE;
}
-#if defined(__LINUX_ALSA__) || defined(__LINUX_OSS__) || defined(__IRIX_AL__)
-
-extern "C" void *callbackHandler(void * ptr);
-
-void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData)
+void RtApi :: tickStreamTime( void )
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- CALLBACK_INFO *info = (CALLBACK_INFO *) &stream->callbackInfo;
- if ( info->usingCallback ) {
- sprintf(message, "RtAudio: A callback is already set for this stream!");
- error(RtError::WARNING);
- return;
- }
+ // Subclasses that do not provide their own implementation of
+ // getStreamTime should call this function once per buffer I/O to
+ // provide basic stream time support.
- info->callback = (void *) callback;
- info->userData = userData;
- info->usingCallback = true;
- info->object = (void *) this;
- info->streamId = streamId;
+ stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
- int err = pthread_create(&info->thread, NULL, callbackHandler, &stream->callbackInfo);
-
- if (err) {
- info->usingCallback = false;
- sprintf(message, "RtAudio: error starting callback thread!");
- error(RtError::THREAD_ERROR);
- }
+#if defined( HAVE_GETTIMEOFDAY )
+ gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
}
-void RtAudio :: cancelStreamCallback(int streamId)
+long RtApi :: getStreamLatency( void )
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- if (stream->callbackInfo.usingCallback) {
-
- if (stream->state == STREAM_RUNNING)
- stopStream( streamId );
-
- MUTEX_LOCK(&stream->mutex);
+ verifyStream();
- stream->callbackInfo.usingCallback = false;
- pthread_cancel(stream->callbackInfo.thread);
- pthread_join(stream->callbackInfo.thread, NULL);
- stream->callbackInfo.thread = 0;
- stream->callbackInfo.callback = NULL;
- stream->callbackInfo.userData = NULL;
+ long totalLatency = 0;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
+ totalLatency = stream_.latency[0];
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
+ totalLatency += stream_.latency[1];
- MUTEX_UNLOCK(&stream->mutex);
- }
+ return totalLatency;
}
+double RtApi :: getStreamTime( void )
+{
+ verifyStream();
+
+#if defined( HAVE_GETTIMEOFDAY )
+ // Return a very accurate estimate of the stream time by
+ // adding in the elapsed time since the last tick.
+ struct timeval then;
+ struct timeval now;
+
+ if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
+ return stream_.streamTime;
+
+ gettimeofday( &now, NULL );
+ then = stream_.lastTickTimestamp;
+ return stream_.streamTime +
+ ((now.tv_sec + 0.000001 * now.tv_usec) -
+ (then.tv_sec + 0.000001 * then.tv_usec));
+#else
+ return stream_.streamTime;
#endif
+}
+
// *************************************************** //
//
// procedure for each of its audio devices. A single RtAudio duplex
// stream using two different devices is supported here, though it
// cannot be guaranteed to always behave correctly because we cannot
-// synchronize these two callbacks. This same functionality can be
-// achieved with better synchrony by opening two separate streams for
-// the devices and using RtAudio blocking calls (i.e. tickStream()).
-//
-// The possibility of having multiple RtAudio streams accessing the
-// same CoreAudio device is not currently supported. The problem
-// involves the inability to install our callbackHandler function for
-// the same device more than once. I experimented with a workaround
-// for this, but it requires an additional buffer for mixing output
-// data before filling the CoreAudio device buffer. In the end, I
-// decided it wasn't worth supporting.
+// synchronize these two callbacks.
//
-// Property listeners are currently not used. The issue is what could
+// A property listener is installed for over/underrun information.
+// However, no functionality is currently provided to allow property
+// listeners to trigger user handlers because it is unclear what could
// be done if a critical stream parameter (buffer size, sample rate,
// device disconnect) notification arrived. The listeners entail
// quite a bit of extra code and most likely, a user program wouldn't
-// be prepared for the result anyway. Some initial listener code is
-// commented out.
+// be prepared for the result anyway. However, we do provide a flag
+// to the client callback function to inform of an over/underrun.
+//
+// The mechanism for querying and setting system parameters was
+// updated (and perhaps simplified) in OS-X version 10.4. However,
+// since 10.4 support is not necessarily available to all users, I've
+// decided not to update the respective code at this time. Perhaps
+// this will happen when Apple makes 10.4 free for everyone. :-)
+
+// A structure to hold various information related to the CoreAudio API
+// implementation.
+struct CoreHandle {
+ AudioDeviceID id[2]; // device ids
+ UInt32 iStream[2]; // device stream index (first for mono mode)
+ bool xrun[2];
+ char *deviceBuffer;
+ pthread_cond_t condition;
+ int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
+
+ CoreHandle()
+ :deviceBuffer(0), drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
+};
-void RtAudio :: initialize(void)
+RtApiCore :: RtApiCore()
{
- OSStatus err = noErr;
- UInt32 dataSize;
- AudioDeviceID *deviceList = NULL;
- nDevices = 0;
-
- // Find out how many audio devices there are, if any.
- err = AudioHardwareGetPropertyInfo(kAudioHardwarePropertyDevices, &dataSize, NULL);
- if (err != noErr) {
- sprintf(message, "RtAudio: OSX error getting device info!");
- error(RtError::SYSTEM_ERROR);
- }
-
- nDevices = dataSize / sizeof(AudioDeviceID);
- if (nDevices == 0) return;
-
- // Allocate the RTAUDIO_DEVICE structures.
- devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE));
- if (devices == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
- }
-
- // Make space for the devices we are about to get.
- deviceList = (AudioDeviceID *) malloc( dataSize );
- if (deviceList == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
- }
+ // Nothing to do here.
+}
- // Get the array of AudioDeviceIDs.
- err = AudioHardwareGetProperty(kAudioHardwarePropertyDevices, &dataSize, (void *) deviceList);
- if (err != noErr) {
- free(deviceList);
- sprintf(message, "RtAudio: OSX error getting device properties!");
- error(RtError::SYSTEM_ERROR);
- }
+RtApiCore :: ~RtApiCore()
+{
+ // The subclass destructor gets called before the base class
+ // destructor, so close an existing stream before deallocating
+ // apiDeviceId memory.
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
- // Write device identifiers to device structures and then
- // probe the device capabilities.
- for (int i=0; i<nDevices; i++) {
- devices[i].id[0] = deviceList[i];
- //probeDeviceInfo(&devices[i]);
+unsigned int RtApiCore :: getDeviceCount( void )
+{
+ // Find out how many audio devices there are, if any.
+ UInt32 dataSize;
+ OSStatus result = AudioHardwareGetPropertyInfo( kAudioHardwarePropertyDevices, &dataSize, NULL );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
+ error( RtError::WARNING );
+ return 0;
}
- free(deviceList);
+ return dataSize / sizeof( AudioDeviceID );
}
-int RtAudio :: getDefaultInputDevice(void)
+unsigned int RtApiCore :: getDefaultInputDevice( void )
{
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices <= 1 ) return 0;
+
AudioDeviceID id;
UInt32 dataSize = sizeof( AudioDeviceID );
-
OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultInputDevice,
&dataSize, &id );
- if (result != noErr) {
- sprintf( message, "RtAudio: OSX error getting default input device." );
- error(RtError::WARNING);
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
+ error( RtError::WARNING );
return 0;
}
- for ( int i=0; i<nDevices; i++ ) {
- if ( id == devices[i].id[0] ) return i;
+ dataSize *= nDevices;
+ AudioDeviceID deviceList[ nDevices ];
+ result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
+ error( RtError::WARNING );
+ return 0;
}
+ for ( unsigned int i=0; i<nDevices; i++ )
+ if ( id == deviceList[i] ) return i;
+
+ errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
+ error( RtError::WARNING );
return 0;
}
-int RtAudio :: getDefaultOutputDevice(void)
+unsigned int RtApiCore :: getDefaultOutputDevice( void )
{
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices <= 1 ) return 0;
+
AudioDeviceID id;
UInt32 dataSize = sizeof( AudioDeviceID );
-
OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultOutputDevice,
&dataSize, &id );
- if (result != noErr) {
- sprintf( message, "RtAudio: OSX error getting default output device." );
- error(RtError::WARNING);
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
+ error( RtError::WARNING );
return 0;
}
- for ( int i=0; i<nDevices; i++ ) {
- if ( id == devices[i].id[0] ) return i;
+ dataSize *= nDevices;
+ AudioDeviceID deviceList[ nDevices ];
+ result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
+ error( RtError::WARNING );
+ return 0;
}
+ for ( unsigned int i=0; i<nDevices; i++ )
+ if ( id == deviceList[i] ) return i;
+
+ errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
+ error( RtError::WARNING );
return 0;
}
-static bool deviceSupportsFormat( AudioDeviceID id, bool isInput,
- AudioStreamBasicDescription *desc, bool isDuplex )
+RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
{
- OSStatus result = noErr;
- UInt32 dataSize = sizeof( AudioStreamBasicDescription );
-
- result = AudioDeviceGetProperty( id, 0, isInput,
- kAudioDevicePropertyStreamFormatSupported,
- &dataSize, desc );
-
- if (result == kAudioHardwareNoError) {
- if ( isDuplex ) {
- result = AudioDeviceGetProperty( id, 0, true,
- kAudioDevicePropertyStreamFormatSupported,
- &dataSize, desc );
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+ // Get device ID
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices == 0 ) {
+ errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
+ error( RtError::INVALID_USE );
+ }
- if (result != kAudioHardwareNoError)
- return false;
- }
- return true;
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
+ error( RtError::INVALID_USE );
}
- return false;
-}
+ AudioDeviceID deviceList[ nDevices ];
+ UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
+ OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
+ error( RtError::WARNING );
+ return info;
+ }
-void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info)
-{
- OSStatus err = noErr;
+ AudioDeviceID id = deviceList[ device ];
- // Get the device manufacturer and name.
+ // Get the device name.
+ info.name.erase();
char name[256];
- char fullname[512];
- UInt32 dataSize = 256;
- err = AudioDeviceGetProperty( info->id[0], 0, false,
- kAudioDevicePropertyDeviceManufacturer,
- &dataSize, name );
- if (err != noErr) {
- sprintf( message, "RtAudio: OSX error getting device manufacturer." );
- error(RtError::DEBUG_WARNING);
- return;
- }
- strncpy(fullname, name, 256);
- strcat(fullname, ": " );
-
dataSize = 256;
- err = AudioDeviceGetProperty( info->id[0], 0, false,
- kAudioDevicePropertyDeviceName,
- &dataSize, name );
- if (err != noErr) {
- sprintf( message, "RtAudio: OSX error getting device name." );
- error(RtError::DEBUG_WARNING);
- return;
+ result = AudioDeviceGetProperty( id, 0, false,
+ kAudioDevicePropertyDeviceManufacturer,
+ &dataSize, name );
+
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
}
- strncat(fullname, name, 254);
- strncat(info->name, fullname, 128);
+ info.name.append( (const char *)name, strlen(name) );
+ info.name.append( ": " );
- // Get output channel information.
- unsigned int i, minChannels, maxChannels, nStreams = 0;
+ dataSize = 256;
+ result = AudioDeviceGetProperty( id, 0, false,
+ kAudioDevicePropertyDeviceName,
+ &dataSize, name );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
+ }
+ info.name.append( (const char *)name, strlen(name) );
+
+ // Get the output stream "configuration".
AudioBufferList *bufferList = nil;
- err = AudioDeviceGetPropertyInfo( info->id[0], 0, false,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, NULL );
- if (err == noErr && dataSize > 0) {
- bufferList = (AudioBufferList *) malloc( dataSize );
- if (bufferList == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::DEBUG_WARNING);
- return;
- }
-
- err = AudioDeviceGetProperty( info->id[0], 0, false,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, bufferList );
- if (err == noErr) {
- maxChannels = 0;
- minChannels = 1000;
- nStreams = bufferList->mNumberBuffers;
- for ( i=0; i<nStreams; i++ ) {
- maxChannels += bufferList->mBuffers[i].mNumberChannels;
- if ( bufferList->mBuffers[i].mNumberChannels < minChannels )
- minChannels = bufferList->mBuffers[i].mNumberChannels;
- }
- }
- }
- if (err != noErr || dataSize <= 0) {
- sprintf( message, "RtAudio: OSX error getting output channels for device (%s).", info->name );
- error(RtError::DEBUG_WARNING);
- return;
+ result = AudioDeviceGetPropertyInfo( id, 0, false,
+ kAudioDevicePropertyStreamConfiguration,
+ &dataSize, NULL );
+ if (result != noErr || dataSize == 0) {
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
+ }
+
+ // Allocate the AudioBufferList.
+ bufferList = (AudioBufferList *) malloc( dataSize );
+ if ( bufferList == NULL ) {
+ errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
+ error( RtError::WARNING );
+ return info;
+ }
+
+ result = AudioDeviceGetProperty( id, 0, false,
+ kAudioDevicePropertyStreamConfiguration,
+ &dataSize, bufferList );
+ if ( result != noErr ) {
+ free( bufferList );
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
}
- free (bufferList);
- if ( nStreams ) {
- if ( maxChannels > 0 )
- info->maxOutputChannels = maxChannels;
- if ( minChannels > 0 )
- info->minOutputChannels = minChannels;
+ // Get output channel information.
+ unsigned int i, nStreams = bufferList->mNumberBuffers;
+ for ( i=0; i<nStreams; i++ )
+ info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
+ free( bufferList );
+
+ // Get the input stream "configuration".
+ result = AudioDeviceGetPropertyInfo( id, 0, true,
+ kAudioDevicePropertyStreamConfiguration,
+ &dataSize, NULL );
+ if (result != noErr || dataSize == 0) {
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
+ }
+
+ // Allocate the AudioBufferList.
+ bufferList = (AudioBufferList *) malloc( dataSize );
+ if ( bufferList == NULL ) {
+ errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
+ error( RtError::WARNING );
+ return info;
+ }
+
+ result = AudioDeviceGetProperty( id, 0, true,
+ kAudioDevicePropertyStreamConfiguration,
+ &dataSize, bufferList );
+ if ( result != noErr ) {
+ free( bufferList );
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
}
// Get input channel information.
- bufferList = nil;
- err = AudioDeviceGetPropertyInfo( info->id[0], 0, true,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, NULL );
- if (err == noErr && dataSize > 0) {
- bufferList = (AudioBufferList *) malloc( dataSize );
- if (bufferList == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::DEBUG_WARNING);
- return;
- }
- err = AudioDeviceGetProperty( info->id[0], 0, true,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, bufferList );
- if (err == noErr) {
- maxChannels = 0;
- minChannels = 1000;
- nStreams = bufferList->mNumberBuffers;
- for ( i=0; i<nStreams; i++ ) {
- if ( bufferList->mBuffers[i].mNumberChannels < minChannels )
- minChannels = bufferList->mBuffers[i].mNumberChannels;
- maxChannels += bufferList->mBuffers[i].mNumberChannels;
- }
- }
- }
- if (err != noErr || dataSize <= 0) {
- sprintf( message, "RtAudio: OSX error getting input channels for device (%s).", info->name );
- error(RtError::DEBUG_WARNING);
- return;
- }
-
- free (bufferList);
- if ( nStreams ) {
- if ( maxChannels > 0 )
- info->maxInputChannels = maxChannels;
- if ( minChannels > 0 )
- info->minInputChannels = minChannels;
- }
+ nStreams = bufferList->mNumberBuffers;
+ for ( i=0; i<nStreams; i++ )
+ info.inputChannels += bufferList->mBuffers[i].mNumberChannels;
+ free( bufferList );
// If device opens for both playback and capture, we determine the channels.
- if (info->maxOutputChannels > 0 && info->maxInputChannels > 0) {
- info->hasDuplexSupport = true;
- info->maxDuplexChannels = (info->maxOutputChannels > info->maxInputChannels) ?
- info->maxInputChannels : info->maxOutputChannels;
- info->minDuplexChannels = (info->minOutputChannels > info->minInputChannels) ?
- info->minInputChannels : info->minOutputChannels;
- }
-
- // Probe the device sample rate and data format parameters. The
- // core audio query mechanism is performed on a "stream"
- // description, which can have a variable number of channels and
- // apply to input or output only.
+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
- // Create a stream description structure.
- AudioStreamBasicDescription description;
- dataSize = sizeof( AudioStreamBasicDescription );
- memset(&description, 0, sizeof(AudioStreamBasicDescription));
+ // Probe the device sample rates.
bool isInput = false;
- if ( info->maxOutputChannels == 0 ) isInput = true;
- bool isDuplex = false;
- if ( info->maxDuplexChannels > 0 ) isDuplex = true;
+ if ( info.outputChannels == 0 ) isInput = true;
// Determine the supported sample rates.
- info->nSampleRates = 0;
- for (i=0; i<MAX_SAMPLE_RATES; i++) {
- description.mSampleRate = (double) SAMPLE_RATES[i];
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
- info->sampleRates[info->nSampleRates++] = SAMPLE_RATES[i];
- }
-
- if (info->nSampleRates == 0) {
- sprintf( message, "RtAudio: No supported sample rates found for OSX device (%s).", info->name );
- error(RtError::DEBUG_WARNING);
- return;
- }
+ result = AudioDeviceGetPropertyInfo( id, 0, isInput,
+ kAudioDevicePropertyAvailableNominalSampleRates,
+ &dataSize, NULL );
- // Check for continuous sample rate support.
- description.mSampleRate = kAudioStreamAnyRate;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) ) {
- info->sampleRates[1] = info->sampleRates[info->nSampleRates-1];
- info->nSampleRates = -1;
+ if ( result != kAudioHardwareNoError || dataSize == 0 ) {
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
}
- // Determine the supported data formats.
- info->nativeFormats = 0;
- description.mFormatID = kAudioFormatLinearPCM;
- description.mBitsPerChannel = 8;
- description.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_SINT8;
- else {
- description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_SINT8;
- }
+ UInt32 nRanges = dataSize / sizeof( AudioValueRange );
+ AudioValueRange rangeList[ nRanges ];
+ result = AudioDeviceGetProperty( id, 0, isInput,
+ kAudioDevicePropertyAvailableNominalSampleRates,
+ &dataSize, &rangeList );
- description.mBitsPerChannel = 16;
- description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_SINT16;
- else {
- description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_SINT16;
+ if ( result != kAudioHardwareNoError ) {
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
}
- description.mBitsPerChannel = 32;
- description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_SINT32;
- else {
- description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_SINT32;
+ Float64 minimumRate = 100000000.0, maximumRate = 0.0;
+ for ( UInt32 i=0; i<nRanges; i++ ) {
+ if ( rangeList[i].mMinimum < minimumRate ) minimumRate = rangeList[i].mMinimum;
+ if ( rangeList[i].mMaximum > maximumRate ) maximumRate = rangeList[i].mMaximum;
}
- description.mBitsPerChannel = 24;
- description.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsAlignedHigh | kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_SINT24;
- else {
- description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_SINT24;
+ info.sampleRates.clear();
+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+ if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate )
+ info.sampleRates.push_back( SAMPLE_RATES[k] );
}
- description.mBitsPerChannel = 32;
- description.mFormatFlags = kLinearPCMFormatFlagIsFloat | kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_FLOAT32;
- else {
- description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_FLOAT32;
+ if ( info.sampleRates.size() == 0 ) {
+ errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
}
- description.mBitsPerChannel = 64;
- description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_FLOAT64;
- else {
- description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( info->id[0], isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_FLOAT64;
- }
+ // CoreAudio always uses 32-bit floating point data for PCM streams.
+ // Thus, any other "physical" formats supported by the device are of
+ // no interest to the client.
+ info.nativeFormats = RTAUDIO_FLOAT32;
- // Check that we have at least one supported format.
- if (info->nativeFormats == 0) {
- sprintf(message, "RtAudio: OSX PCM device (%s) data format not supported by RtAudio.",
- info->name);
- error(RtError::DEBUG_WARNING);
- return;
- }
+ if ( getDefaultOutputDevice() == device )
+ info.isDefaultOutput = true;
+ if ( getDefaultInputDevice() == device )
+ info.isDefaultInput = true;
- info->probed = true;
+ info.probed = true;
+ return info;
}
-OSStatus callbackHandler(AudioDeviceID inDevice,
- const AudioTimeStamp* inNow,
- const AudioBufferList* inInputData,
- const AudioTimeStamp* inInputTime,
- AudioBufferList* outOutputData,
- const AudioTimeStamp* inOutputTime,
- void* infoPointer)
+OSStatus callbackHandler( AudioDeviceID inDevice,
+ const AudioTimeStamp* inNow,
+ const AudioBufferList* inInputData,
+ const AudioTimeStamp* inInputTime,
+ AudioBufferList* outOutputData,
+ const AudioTimeStamp* inOutputTime,
+ void* infoPointer )
{
- CALLBACK_INFO *info = (CALLBACK_INFO *) infoPointer;
+ CallbackInfo *info = (CallbackInfo *) infoPointer;
- RtAudio *object = (RtAudio *) info->object;
- try {
- object->callbackEvent( info->streamId, inDevice, (void *)inInputData, (void *)outOutputData );
- }
- catch (RtError &exception) {
- fprintf(stderr, "\nCallback handler error (%s)!\n\n", exception.getMessage());
+ RtApiCore *object = (RtApiCore *) info->object;
+ if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
return kAudioHardwareUnspecifiedError;
+ else
+ return kAudioHardwareNoError;
+}
+
+OSStatus deviceListener( AudioDeviceID inDevice,
+ UInt32 channel,
+ Boolean isInput,
+ AudioDevicePropertyID propertyID,
+ void* handlePointer )
+{
+ CoreHandle *handle = (CoreHandle *) handlePointer;
+ if ( propertyID == kAudioDeviceProcessorOverload ) {
+ if ( isInput )
+ handle->xrun[1] = true;
+ else
+ handle->xrun[0] = true;
}
return kAudioHardwareNoError;
}
-/*
-OSStatus deviceListener(AudioDeviceID inDevice,
- UInt32 channel,
- Boolean isInput,
- AudioDevicePropertyID propertyID,
- void* infoPointer)
+static bool hasProperty( AudioDeviceID id, UInt32 channel, bool isInput, AudioDevicePropertyID property )
{
- CALLBACK_INFO *info = (CALLBACK_INFO *) infoPointer;
+ OSStatus result = AudioDeviceGetPropertyInfo( id, channel, isInput, property, NULL, NULL );
+ return result == 0;
+}
- RtAudio *object = (RtAudio *) info->object;
- try {
- object->settingChange( info->streamId );
- }
- catch (RtError &exception) {
- fprintf(stderr, "\nDevice listener error (%s)!\n\n", exception.getMessage());
- return kAudioHardwareUnspecifiedError;
+bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ // Get device ID
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices == 0 ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";
+ return FAILURE;
}
- return kAudioHardwareNoError;
-}
-*/
+ if ( device >= nDevices ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiCore::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
-bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream,
- STREAM_MODE mode, int channels,
- int sampleRate, RTAUDIO_FORMAT format,
- int *bufferSize, int numberOfBuffers)
-{
- // Check to make sure we don't already have a stream accessing this device.
- RTAUDIO_STREAM *streamPtr;
- std::map<int, void *>::const_iterator i;
- for ( i=streams.begin(); i!=streams.end(); ++i ) {
- streamPtr = (RTAUDIO_STREAM *) i->second;
- if ( streamPtr->device[0] == device || streamPtr->device[1] == device ) {
- sprintf(message, "RtAudio: no current OS X support for multiple streams accessing the same device!");
- error(RtError::WARNING);
- return FAILURE;
- }
+ AudioDeviceID deviceList[ nDevices ];
+ UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
+ OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";
+ return FAILURE;
}
+ AudioDeviceID id = deviceList[ device ];
+
// Setup for stream mode.
bool isInput = false;
- AudioDeviceID id = devices[device].id[0];
if ( mode == INPUT ) isInput = true;
- // Search for a stream which contains the desired number of channels.
- OSStatus err = noErr;
- UInt32 dataSize;
- unsigned int deviceChannels, nStreams;
+ // Set or disable "hog" mode.
+ dataSize = sizeof( UInt32 );
+ UInt32 doHog = 0;
+ if ( options && options->flags & RTAUDIO_HOG_DEVICE ) doHog = 1;
+ result = AudioHardwareSetProperty( kAudioHardwarePropertyHogModeIsAllowed, dataSize, &doHog );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Get the stream "configuration".
+ AudioBufferList *bufferList;
+ result = AudioDeviceGetPropertyInfo( id, 0, isInput,
+ kAudioDevicePropertyStreamConfiguration,
+ &dataSize, NULL );
+ if (result != noErr || dataSize == 0) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Allocate the AudioBufferList.
+ bufferList = (AudioBufferList *) malloc( dataSize );
+ if ( bufferList == NULL ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";
+ return FAILURE;
+ }
+
+ result = AudioDeviceGetProperty( id, 0, isInput,
+ kAudioDevicePropertyStreamConfiguration,
+ &dataSize, bufferList );
+ if ( result != noErr ) {
+ free( bufferList );
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Search for a stream that contains the desired number of
+ // channels. CoreAudio devices can have an arbitrary number of
+ // streams and each stream can have an arbitrary number of channels.
+ // For each stream, a single buffer of interleaved samples is
+ // provided. RtAudio currently only supports the use of one stream
+ // of interleaved data or multiple consecutive single-channel
+ // streams. Thus, our search below is limited to these two
+ // contexts.
+ unsigned int streamChannels = 0, nStreams = 0;
UInt32 iChannel = 0, iStream = 0;
- AudioBufferList *bufferList = nil;
- err = AudioDeviceGetPropertyInfo( id, 0, isInput,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, NULL );
-
- if (err == noErr && dataSize > 0) {
- bufferList = (AudioBufferList *) malloc( dataSize );
- if (bufferList == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ unsigned int offsetCounter = firstChannel;
+ stream_.deviceInterleaved[mode] = true;
+ nStreams = bufferList->mNumberBuffers;
+ bool foundStream = false;
+
+ for ( iStream=0; iStream<nStreams; iStream++ ) {
+ streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
+ if ( streamChannels >= channels + offsetCounter ) {
+ iChannel += offsetCounter;
+ foundStream = true;
+ break;
}
- err = AudioDeviceGetProperty( id, 0, isInput,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, bufferList );
-
- if (err == noErr) {
- stream->deInterleave[mode] = false;
- nStreams = bufferList->mNumberBuffers;
- for ( iStream=0; iStream<nStreams; iStream++ ) {
- if ( bufferList->mBuffers[iStream].mNumberChannels >= (unsigned int) channels ) break;
- iChannel += bufferList->mBuffers[iStream].mNumberChannels;
+ if ( streamChannels > offsetCounter ) break;
+ offsetCounter -= streamChannels;
+ iChannel += streamChannels;
+ }
+
+ // If we didn't find a single stream above, see if we can meet
+ // the channel specification in mono mode (i.e. using separate
+ // non-interleaved buffers). This can only work if there are N
+ // consecutive one-channel streams, where N is the number of
+ // desired channels (+ channel offset).
+ if ( foundStream == false ) {
+ unsigned int counter = 0;
+ offsetCounter = firstChannel;
+ iChannel = 0;
+ for ( iStream=0; iStream<nStreams; iStream++ ) {
+ streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
+ if ( offsetCounter ) {
+ if ( streamChannels > offsetCounter ) break;
+ offsetCounter -= streamChannels;
}
- // If we didn't find a single stream above, see if we can meet
- // the channel specification in mono mode (i.e. using separate
- // non-interleaved buffers). This can only work if there are N
- // consecutive one-channel streams, where N is the number of
- // desired channels.
- iChannel = 0;
- if ( iStream >= nStreams && nStreams >= (unsigned int) channels ) {
- int counter = 0;
- for ( iStream=0; iStream<nStreams; iStream++ ) {
- if ( bufferList->mBuffers[iStream].mNumberChannels == 1 )
- counter++;
- else
- counter = 0;
- if ( counter == channels ) {
- iStream -= channels - 1;
- iChannel -= channels - 1;
- stream->deInterleave[mode] = true;
- break;
- }
- iChannel += bufferList->mBuffers[iStream].mNumberChannels;
- }
+ else if ( streamChannels == 1 )
+ counter++;
+ else
+ counter = 0;
+ if ( counter == channels ) {
+ iStream -= channels - 1;
+ iChannel -= channels - 1;
+ stream_.deviceInterleaved[mode] = false;
+ foundStream = true;
+ break;
}
+ iChannel += streamChannels;
}
}
- if (err != noErr || dataSize <= 0) {
- if ( bufferList ) free( bufferList );
- sprintf( message, "RtAudio: OSX error getting channels for device (%s).", devices[device].name );
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
+ free( bufferList );
- if (iStream >= nStreams) {
- free (bufferList);
- sprintf( message, "RtAudio: unable to find OSX audio stream on device (%s) for requested channels (%d).",
- devices[device].name, channels );
- error(RtError::DEBUG_WARNING);
+ if ( foundStream == false ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: unable to find OS-X stream on device (" << device << ") for requested channels.";
+ errorText_ = errorStream_.str();
return FAILURE;
}
- // This is ok even for mono mode ... it gets updated later.
- deviceChannels = bufferList->mBuffers[iStream].mNumberChannels;
- free (bufferList);
-
// Determine the buffer size.
AudioValueRange bufferRange;
- dataSize = sizeof(AudioValueRange);
- err = AudioDeviceGetProperty( id, 0, isInput,
- kAudioDevicePropertyBufferSizeRange,
- &dataSize, &bufferRange);
- if (err != noErr) {
- sprintf( message, "RtAudio: OSX error getting buffer size range for device (%s).",
- devices[device].name );
- error(RtError::DEBUG_WARNING);
+ dataSize = sizeof( AudioValueRange );
+ result = AudioDeviceGetProperty( id, 0, isInput,
+ kAudioDevicePropertyBufferFrameSizeRange,
+ &dataSize, &bufferRange );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";
+ errorText_ = errorStream_.str();
return FAILURE;
}
- long bufferBytes = *bufferSize * deviceChannels * formatBytes(RTAUDIO_FLOAT32);
- if (bufferRange.mMinimum > bufferBytes) bufferBytes = (int) bufferRange.mMinimum;
- else if (bufferRange.mMaximum < bufferBytes) bufferBytes = (int) bufferRange.mMaximum;
+ if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;
+ else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;
// Set the buffer size. For mono mode, I'm assuming we only need to
- // make this setting for the first channel.
- UInt32 theSize = (UInt32) bufferBytes;
- dataSize = sizeof( UInt32);
- err = AudioDeviceSetProperty(id, NULL, 0, isInput,
- kAudioDevicePropertyBufferSize,
- dataSize, &theSize);
- if (err != noErr) {
- sprintf( message, "RtAudio: OSX error setting the buffer size for device (%s).",
- devices[device].name );
- error(RtError::DEBUG_WARNING);
+ // make this setting for the master channel.
+ UInt32 theSize = (UInt32) *bufferSize;
+ dataSize = sizeof( UInt32 );
+ result = AudioDeviceSetProperty( id, NULL, 0, isInput,
+ kAudioDevicePropertyBufferFrameSize,
+ dataSize, &theSize );
+
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";
+ errorText_ = errorStream_.str();
return FAILURE;
}
// If attempting to setup a duplex stream, the bufferSize parameter
// MUST be the same in both directions!
- *bufferSize = bufferBytes / ( deviceChannels * formatBytes(RTAUDIO_FLOAT32) );
- if ( stream->mode == OUTPUT && mode == INPUT && *bufferSize != stream->bufferSize ) {
- sprintf( message, "RtAudio: OSX error setting buffer size for duplex stream on device (%s).",
- devices[device].name );
- error(RtError::DEBUG_WARNING);
+ *bufferSize = theSize;
+ if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";
+ errorText_ = errorStream_.str();
return FAILURE;
}
- stream->bufferSize = *bufferSize;
- stream->nBuffers = 1;
+ stream_.bufferSize = *bufferSize;
+ stream_.nBuffers = 1;
+
+ // Get the stream ID(s) so we can set the stream format. In mono
+ // mode, we'll have to do this for each stream (channel).
+ AudioStreamID streamIDs[ nStreams ];
+ dataSize = nStreams * sizeof( AudioStreamID );
+ result = AudioDeviceGetProperty( id, 0, isInput,
+ kAudioDevicePropertyStreams,
+ &dataSize, &streamIDs );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream ID(s) for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // Set the stream format description. Do for each channel in mono mode.
+ // Now set the stream format. Also, check the physical format of the
+ // device and change that if necessary.
AudioStreamBasicDescription description;
dataSize = sizeof( AudioStreamBasicDescription );
- if ( stream->deInterleave[mode] ) nStreams = channels;
- else nStreams = 1;
- for ( unsigned int i=0; i<nStreams; i++, iChannel++ ) {
-
- err = AudioDeviceGetProperty( id, iChannel, isInput,
- kAudioDevicePropertyStreamFormat,
- &dataSize, &description );
- if (err != noErr) {
- sprintf( message, "RtAudio: OSX error getting stream format for device (%s).", devices[device].name );
- error(RtError::DEBUG_WARNING);
+ if ( stream_.deviceInterleaved[mode] ) nStreams = 1;
+ else nStreams = channels;
+
+ bool updateFormat;
+ for ( unsigned int i=0; i<nStreams; i++ ) {
+
+ result = AudioStreamGetProperty( streamIDs[iStream+i], 0,
+ kAudioStreamPropertyVirtualFormat,
+ &dataSize, &description );
+
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
+ errorText_ = errorStream_.str();
return FAILURE;
}
- // Set the sample rate and data format id.
- description.mSampleRate = (double) sampleRate;
- description.mFormatID = kAudioFormatLinearPCM;
- err = AudioDeviceSetProperty( id, NULL, iChannel, isInput,
- kAudioDevicePropertyStreamFormat,
- dataSize, &description );
- if (err != noErr) {
- sprintf( message, "RtAudio: OSX error setting sample rate or data format for device (%s).", devices[device].name );
- error(RtError::DEBUG_WARNING);
+ // Set the sample rate and data format id. However, only make the
+ // change if the sample rate is not within 1.0 of the desired
+ // rate and the format is not linear pcm.
+ updateFormat = false;
+ if ( fabs( description.mSampleRate - (double)sampleRate ) > 1.0 ) {
+ description.mSampleRate = (double) sampleRate;
+ updateFormat = true;
+ }
+
+ if ( description.mFormatID != kAudioFormatLinearPCM ) {
+ description.mFormatID = kAudioFormatLinearPCM;
+ updateFormat = true;
+ }
+
+ if ( updateFormat ) {
+ result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0,
+ kAudioStreamPropertyVirtualFormat,
+ dataSize, &description );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // Now check the physical format.
+ result = AudioStreamGetProperty( streamIDs[iStream+i], 0,
+ kAudioStreamPropertyPhysicalFormat,
+ &dataSize, &description );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
+ errorText_ = errorStream_.str();
return FAILURE;
}
+
+ if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 24 ) {
+ description.mFormatID = kAudioFormatLinearPCM;
+ AudioStreamBasicDescription testDescription = description;
+ unsigned long formatFlags;
+
+ // We'll try higher bit rates first and then work our way down.
+ testDescription.mBitsPerChannel = 32;
+ formatFlags = description.mFormatFlags | kLinearPCMFormatFlagIsFloat & ~kLinearPCMFormatFlagIsSignedInteger;
+ testDescription.mFormatFlags = formatFlags;
+ result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
+ if ( result == noErr ) continue;
+
+ testDescription = description;
+ testDescription.mBitsPerChannel = 32;
+ formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger) & ~kLinearPCMFormatFlagIsFloat;
+ testDescription.mFormatFlags = formatFlags;
+ result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
+ if ( result == noErr ) continue;
+
+ testDescription = description;
+ testDescription.mBitsPerChannel = 24;
+ testDescription.mFormatFlags = formatFlags;
+ result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
+ if ( result == noErr ) continue;
+
+ testDescription = description;
+ testDescription.mBitsPerChannel = 16;
+ testDescription.mFormatFlags = formatFlags;
+ result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
+ if ( result == noErr ) continue;
+
+ testDescription = description;
+ testDescription.mBitsPerChannel = 8;
+ testDescription.mFormatFlags = formatFlags;
+ result = AudioStreamSetProperty( streamIDs[iStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
}
- // Check whether we need byte-swapping (assuming OS X host is big-endian).
- iChannel -= nStreams;
- err = AudioDeviceGetProperty( id, iChannel, isInput,
- kAudioDevicePropertyStreamFormat,
- &dataSize, &description );
- if (err != noErr) {
- sprintf( message, "RtAudio: OSX error getting stream format for device (%s).", devices[device].name );
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ // Get the stream latency. There can be latency in both the device
+ // and the stream. First, attempt to get the device latency on the
+ // master channel or the first open channel. Errors that might
+ // occur here are not deemed critical.
+ UInt32 latency, channel = 0;
+ dataSize = sizeof( UInt32 );
+ AudioDevicePropertyID property = kAudioDevicePropertyLatency;
+ for ( int i=0; i<2; i++ ) {
+ if ( hasProperty( id, channel, isInput, property ) == true ) break;
+ channel = iChannel + 1 + i;
+ }
+ if ( channel <= iChannel + 1 ) {
+ result = AudioDeviceGetProperty( id, channel, isInput, property, &dataSize, &latency );
+ if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
+ else {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ }
+ }
+
+ // Now try to get the stream latency. For "mono" mode, I assume the
+ // latency is equal for all single-channel streams.
+ result = AudioStreamGetProperty( streamIDs[iStream], 0, property, &dataSize, &latency );
+ if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] += latency;
+ else {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream latency for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
}
- stream->doByteSwap[mode] = false;
- if ( !description.mFormatFlags & kLinearPCMFormatFlagIsBigEndian )
- stream->doByteSwap[mode] = true;
+ // Byte-swapping: According to AudioHardware.h, the stream data will
+ // always be presented in native-endian format, so we should never
+ // need to byte swap.
+ stream_.doByteSwap[mode] = false;
// From the CoreAudio documentation, PCM data must be supplied as
// 32-bit floats.
- stream->userFormat = format;
- stream->deviceFormat[mode] = RTAUDIO_FLOAT32;
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+
+ if ( stream_.deviceInterleaved[mode] )
+ stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
+ else // mono mode
+ stream_.nDeviceChannels[mode] = channels;
+ stream_.nUserChannels[mode] = channels;
+ stream_.channelOffset[mode] = iChannel; // offset within a CoreAudio stream
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate our CoreHandle structure for the stream.
+ CoreHandle *handle = 0;
+ if ( stream_.apiHandle == 0 ) {
+ try {
+ handle = new CoreHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";
+ goto error;
+ }
- if ( stream->deInterleave[mode] )
- stream->nDeviceChannels[mode] = channels;
+ if ( pthread_cond_init( &handle->condition, NULL ) ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";
+ goto error;
+ }
+ stream_.apiHandle = (void *) handle;
+ }
else
- stream->nDeviceChannels[mode] = description.mChannelsPerFrame;
- stream->nUserChannels[mode] = channels;
-
- // Set handle and flags for buffer conversion.
- stream->handle[mode] = iStream;
- stream->doConvertBuffer[mode] = false;
- if (stream->userFormat != stream->deviceFormat[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] > 1 && stream->deInterleave[mode])
- stream->doConvertBuffer[mode] = true;
+ handle = (CoreHandle *) stream_.apiHandle;
+ handle->iStream[mode] = iStream;
+ handle->id[mode] = id;
// Allocate necessary internal buffers.
- if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) {
-
- long buffer_bytes;
- if (stream->nUserChannels[0] >= stream->nUserChannels[1])
- buffer_bytes = stream->nUserChannels[0];
- else
- buffer_bytes = stream->nUserChannels[1];
-
- buffer_bytes *= *bufferSize * formatBytes(stream->userFormat);
- if (stream->userBuffer) free(stream->userBuffer);
- stream->userBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->userBuffer == NULL)
- goto memory_error;
+ unsigned long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
}
- if ( stream->deInterleave[mode] ) {
+ // If possible, we will make use of the CoreAudio stream buffers as
+ // "device buffers". However, we can't do this if the device
+ // buffers are non-interleaved ("mono" mode).
+ if ( !stream_.deviceInterleaved[mode] && stream_.doConvertBuffer[mode] ) {
- long buffer_bytes;
bool makeBuffer = true;
- if ( mode == OUTPUT )
- buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- else { // mode == INPUT
- buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]);
- if ( stream->mode == OUTPUT && stream->deviceBuffer ) {
- long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- if ( buffer_bytes < bytes_out ) makeBuffer = false;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
}
}
if ( makeBuffer ) {
- buffer_bytes *= *bufferSize;
- if (stream->deviceBuffer) free(stream->deviceBuffer);
- stream->deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->deviceBuffer == NULL)
- goto memory_error;
-
- // If not de-interleaving, we point stream->deviceBuffer to the
- // OS X supplied device buffer before doing any necessary data
- // conversions. This presents a problem if we have a duplex
- // stream using one device which needs de-interleaving and
- // another device which doesn't. So, save a pointer to our own
- // device buffer in the CALLBACK_INFO structure.
- stream->callbackInfo.buffers = stream->deviceBuffer;
- }
- }
-
- stream->sampleRate = sampleRate;
- stream->device[mode] = device;
- stream->state = STREAM_STOPPED;
- stream->callbackInfo.object = (void *) this;
- stream->callbackInfo.waitTime = (unsigned long) (200000.0 * stream->bufferSize / stream->sampleRate);
- stream->callbackInfo.device[mode] = id;
- if ( stream->mode == OUTPUT && mode == INPUT && stream->device[0] == device )
- // Only one callback procedure per device.
- stream->mode = DUPLEX;
- else {
- err = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream->callbackInfo );
- if (err != noErr) {
- sprintf( message, "RtAudio: OSX error setting callback for device (%s).", devices[device].name );
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+
+ // Save a pointer to our own device buffer in the CoreHandle
+ // structure because we may need to use the stream_.deviceBuffer
+ // variable to point to the CoreAudio buffer before buffer
+ // conversion (if we have a duplex stream with two different
+ // conversion schemes).
+ handle->deviceBuffer = stream_.deviceBuffer;
}
- if ( stream->mode == OUTPUT && mode == INPUT )
- stream->mode = DUPLEX;
- else
- stream->mode = mode;
}
- // If we wanted to use property listeners, they would be setup here.
+ stream_.sampleRate = sampleRate;
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ stream_.callbackInfo.object = (void *) this;
- return SUCCESS;
+ // Setup the buffer conversion information structure. We override
+ // the channel offset value and perform our own setting for that
+ // here.
+ if ( stream_.doConvertBuffer[mode] ) {
+ setConvertInfo( mode, 0 );
- memory_error:
- if (stream->userBuffer) {
- free(stream->userBuffer);
- stream->userBuffer = 0;
+ // Add channel offset for interleaved channels.
+ if ( firstChannel > 0 && stream_.deviceInterleaved[mode] ) {
+ if ( mode == OUTPUT ) {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+ stream_.convertInfo[mode].outOffset[k] += firstChannel;
+ }
+ else {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+ stream_.convertInfo[mode].inOffset[k] += firstChannel;
+ }
+ }
}
- sprintf(message, "RtAudio: OSX error allocating buffer memory (%s).", devices[device].name);
- error(RtError::WARNING);
- return FAILURE;
-}
-void RtAudio :: cancelStreamCallback(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
+ // Only one callback procedure per device.
+ stream_.mode = DUPLEX;
+ else {
+ result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ stream_.mode = DUPLEX;
+ else
+ stream_.mode = mode;
+ }
- if (stream->callbackInfo.usingCallback) {
+ // Setup the device property listener for over/underload.
+ result = AudioDeviceAddPropertyListener( id, 0, isInput,
+ kAudioDeviceProcessorOverload,
+ deviceListener, (void *) handle );
- if (stream->state == STREAM_RUNNING)
- stopStream( streamId );
+ return SUCCESS;
- MUTEX_LOCK(&stream->mutex);
+ error:
+ if ( handle ) {
+ pthread_cond_destroy( &handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
- stream->callbackInfo.usingCallback = false;
- stream->callbackInfo.userData = NULL;
- stream->state = STREAM_STOPPED;
- stream->callbackInfo.callback = NULL;
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
- MUTEX_UNLOCK(&stream->mutex);
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
}
+
+ return FAILURE;
}
-void RtAudio :: closeStream(int streamId)
+void RtApiCore :: closeStream( void )
{
- // We don't want an exception to be thrown here because this
- // function is called by our class destructor. So, do our own
- // streamId check.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
- error(RtError::WARNING);
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiCore::closeStream(): no open stream to close!";
+ error( RtError::WARNING );
return;
}
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId];
-
- AudioDeviceID id;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- id = devices[stream->device[0]].id[0];
- if (stream->state == STREAM_RUNNING)
- AudioDeviceStop( id, callbackHandler );
- AudioDeviceRemoveIOProc( id, callbackHandler );
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( stream_.state == STREAM_RUNNING )
+ AudioDeviceStop( handle->id[0], callbackHandler );
+ AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
}
- if (stream->mode == INPUT || ( stream->mode == DUPLEX && stream->device[0] != stream->device[1]) ) {
- id = devices[stream->device[1]].id[0];
- if (stream->state == STREAM_RUNNING)
- AudioDeviceStop( id, callbackHandler );
- AudioDeviceRemoveIOProc( id, callbackHandler );
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+ if ( stream_.state == STREAM_RUNNING )
+ AudioDeviceStop( handle->id[1], callbackHandler );
+ AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
}
- pthread_mutex_destroy(&stream->mutex);
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
- if (stream->userBuffer)
- free(stream->userBuffer);
+ if ( handle->deviceBuffer ) {
+ free( handle->deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
- if ( stream->deInterleave[0] || stream->deInterleave[1] )
- free(stream->callbackInfo.buffers);
+ // Destroy pthread condition variable.
+ pthread_cond_destroy( &handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
- free(stream);
- streams.erase(streamId);
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
}
-void RtAudio :: startStream(int streamId)
+void RtApiCore :: startStream( void )
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiCore::startStream(): the stream is already running!";
+ error( RtError::WARNING );
+ return;
+ }
- if (stream->state == STREAM_RUNNING)
- goto unlock;
+ MUTEX_LOCK( &stream_.mutex );
- OSStatus err;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
+ OSStatus result = noErr;
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- err = AudioDeviceStart(devices[stream->device[0]].id[0], callbackHandler);
- if (err != noErr) {
- sprintf(message, "RtAudio: OSX error starting callback procedure on device (%s).",
- devices[stream->device[0]].name);
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
+ result = AudioDeviceStart( handle->id[0], callbackHandler );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
}
- if (stream->mode == INPUT || ( stream->mode == DUPLEX && stream->device[0] != stream->device[1]) ) {
+ if ( stream_.mode == INPUT ||
+ ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
- err = AudioDeviceStart(devices[stream->device[1]].id[0], callbackHandler);
- if (err != noErr) {
- sprintf(message, "RtAudio: OSX error starting input callback procedure on device (%s).",
- devices[stream->device[0]].name);
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
+ result = AudioDeviceStart( handle->id[1], callbackHandler );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
}
- stream->callbackInfo.streamId = streamId;
- stream->state = STREAM_RUNNING;
- stream->callbackInfo.blockTick = true;
- stream->callbackInfo.stopStream = false;
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ stream_.state = STREAM_RUNNING;
unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result == noErr ) return;
+ error( RtError::SYSTEM_ERROR );
}
-void RtAudio :: stopStream(int streamId)
+void RtApiCore :: stopStream( void )
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
+ }
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK( &stream_.mutex );
- if (stream->state == STREAM_STOPPED)
- goto unlock;
+ OSStatus result = noErr;
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- OSStatus err;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 1;
+ pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
+ }
- err = AudioDeviceStop(devices[stream->device[0]].id[0], callbackHandler);
- if (err != noErr) {
- sprintf(message, "RtAudio: OSX error stopping callback procedure on device (%s).",
- devices[stream->device[0]].name);
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
+ result = AudioDeviceStop( handle->id[0], callbackHandler );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
}
- if (stream->mode == INPUT || ( stream->mode == DUPLEX && stream->device[0] != stream->device[1]) ) {
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
- err = AudioDeviceStop(devices[stream->device[1]].id[0], callbackHandler);
- if (err != noErr) {
- sprintf(message, "RtAudio: OSX error stopping input callback procedure on device (%s).",
- devices[stream->device[0]].name);
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
+ result = AudioDeviceStop( handle->id[1], callbackHandler );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
}
- stream->state = STREAM_STOPPED;
-
unlock:
- MUTEX_UNLOCK(&stream->mutex);
-}
+ MUTEX_UNLOCK( &stream_.mutex );
-void RtAudio :: abortStream(int streamId)
-{
- stopStream( streamId );
+ stream_.state = STREAM_STOPPED;
+ if ( result == noErr ) return;
+ error( RtError::SYSTEM_ERROR );
}
-// I don't know how this function can be implemented.
-int RtAudio :: streamWillBlock(int streamId)
+void RtApiCore :: abortStream( void )
{
- sprintf(message, "RtAudio: streamWillBlock() cannot be implemented for OS X.");
- error(RtError::WARNING);
- return 0;
-}
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
+ }
-void RtAudio :: tickStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ handle->drainCounter = 1;
- if (stream->state == STREAM_STOPPED)
- return;
+ stopStream();
+}
- if (stream->callbackInfo.usingCallback) {
- sprintf(message, "RtAudio: tickStream() should not be used when a callback function is set!");
- error(RtError::WARNING);
- return;
+bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
+ const AudioBufferList *inBufferList,
+ const AudioBufferList *outBufferList )
+{
+ if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtError::WARNING );
+ return FAILURE;
}
- // Block waiting here until the user data is processed in callbackEvent().
- while ( stream->callbackInfo.blockTick )
- usleep(stream->callbackInfo.waitTime);
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
- MUTEX_LOCK(&stream->mutex);
+ // Check if we were draining the stream and signal is finished.
+ if ( handle->drainCounter > 3 ) {
+ if ( handle->internalDrain == false )
+ pthread_cond_signal( &handle->condition );
+ else
+ stopStream();
+ return SUCCESS;
+ }
- stream->callbackInfo.blockTick = true;
+ MUTEX_LOCK( &stream_.mutex );
- MUTEX_UNLOCK(&stream->mutex);
-}
+ AudioDeviceID outputDevice = handle->id[0];
-void RtAudio :: callbackEvent( int streamId, DEVICE_ID deviceId, void *inData, void *outData )
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- CALLBACK_INFO *info;
- AudioBufferList *inBufferList = (AudioBufferList *) inData;
- AudioBufferList *outBufferList = (AudioBufferList *) outData;
-
- if (stream->state == STREAM_STOPPED) return;
-
- info = (CALLBACK_INFO *) &stream->callbackInfo;
- if ( !info->usingCallback ) {
- // Block waiting here until we get new user data in tickStream().
- while ( !info->blockTick )
- usleep(info->waitTime);
- }
- else if ( info->stopStream ) {
- // Check if the stream should be stopped (via the previous user
- // callback return value). We stop the stream here, rather than
- // after the function call, so that output data can first be
- // processed.
- this->stopStream(info->streamId);
- return;
+ // Invoke user callback to get fresh output data UNLESS we are
+ // draining stream or duplex mode AND the input/output devices are
+ // different AND this function is called for the input device.
+ if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
+ }
+ handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( handle->drainCounter == 2 ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ abortStream();
+ return SUCCESS;
+ }
+ else if ( handle->drainCounter == 1 )
+ handle->internalDrain = true;
}
- MUTEX_LOCK(&stream->mutex);
+ if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
- if ( stream->mode == INPUT || ( stream->mode == DUPLEX && deviceId == info->device[1] ) ) {
+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
- if (stream->doConvertBuffer[1]) {
-
- if ( stream->deInterleave[1] ) {
- stream->deviceBuffer = (char *) stream->callbackInfo.buffers;
- int bufferBytes = inBufferList->mBuffers[stream->handle[1]].mDataByteSize;
- for ( int i=0; i<stream->nDeviceChannels[1]; i++ ) {
- memcpy(&stream->deviceBuffer[i*bufferBytes],
- inBufferList->mBuffers[stream->handle[1]+i].mData, bufferBytes );
+ if ( stream_.deviceInterleaved[0] ) {
+ memset( outBufferList->mBuffers[handle->iStream[0]].mData,
+ 0,
+ outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
+ }
+ else {
+ for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+ memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
+ 0,
+ outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
}
}
+ }
+ else if ( stream_.doConvertBuffer[0] ) {
+
+ if ( stream_.deviceInterleaved[0] )
+ stream_.deviceBuffer = (char *) outBufferList->mBuffers[handle->iStream[0]].mData;
else
- stream->deviceBuffer = (char *) inBufferList->mBuffers[stream->handle[1]].mData;
+ stream_.deviceBuffer = handle->deviceBuffer;
+
+ convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
- if ( stream->doByteSwap[1] )
- byteSwapBuffer(stream->deviceBuffer,
- stream->bufferSize * stream->nDeviceChannels[1],
- stream->deviceFormat[1]);
- convertStreamBuffer(stream, INPUT);
+ if ( !stream_.deviceInterleaved[0] ) {
+ UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
+ for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+ memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
+ &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
+ }
+ }
}
else {
- memcpy(stream->userBuffer,
- inBufferList->mBuffers[stream->handle[1]].mData,
- inBufferList->mBuffers[stream->handle[1]].mDataByteSize );
-
- if (stream->doByteSwap[1])
- byteSwapBuffer(stream->userBuffer,
- stream->bufferSize * stream->nUserChannels[1],
- stream->userFormat);
+ if ( stream_.deviceInterleaved[0] ) {
+ memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
+ stream_.userBuffer[0],
+ outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
+ }
+ else {
+ UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
+ for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+ memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
+ &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
+ }
+ }
}
- }
- // Don't invoke the user callback if duplex mode, the input/output
- // devices are different, and this function is called for the output
- // device.
- if ( info->usingCallback && (stream->mode != DUPLEX || deviceId == info->device[1] ) ) {
- RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) info->callback;
- info->stopStream = callback(stream->userBuffer, stream->bufferSize, info->userData);
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
+ }
}
- if ( stream->mode == OUTPUT || ( stream->mode == DUPLEX && deviceId == info->device[0] ) ) {
+ AudioDeviceID inputDevice = handle->id[1];
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
- if (stream->doConvertBuffer[0]) {
+ if ( stream_.doConvertBuffer[1] ) {
- if ( !stream->deInterleave[0] )
- stream->deviceBuffer = (char *) outBufferList->mBuffers[stream->handle[0]].mData;
- else
- stream->deviceBuffer = (char *) stream->callbackInfo.buffers;
-
- convertStreamBuffer(stream, OUTPUT);
- if ( stream->doByteSwap[0] )
- byteSwapBuffer(stream->deviceBuffer,
- stream->bufferSize * stream->nDeviceChannels[0],
- stream->deviceFormat[0]);
-
- if ( stream->deInterleave[0] ) {
- int bufferBytes = outBufferList->mBuffers[stream->handle[0]].mDataByteSize;
- for ( int i=0; i<stream->nDeviceChannels[0]; i++ ) {
- memcpy(outBufferList->mBuffers[stream->handle[0]+i].mData,
- &stream->deviceBuffer[i*bufferBytes], bufferBytes );
+ if ( stream_.deviceInterleaved[1] )
+ stream_.deviceBuffer = (char *) inBufferList->mBuffers[handle->iStream[1]].mData;
+ else {
+ stream_.deviceBuffer = (char *) handle->deviceBuffer;
+ UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
+ for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
+ memcpy( &stream_.deviceBuffer[i*bufferBytes],
+ inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
}
}
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+
}
else {
- if (stream->doByteSwap[0])
- byteSwapBuffer(stream->userBuffer,
- stream->bufferSize * stream->nUserChannels[0],
- stream->userFormat);
-
- memcpy(outBufferList->mBuffers[stream->handle[0]].mData,
- stream->userBuffer,
- outBufferList->mBuffers[stream->handle[0]].mDataByteSize );
+ memcpy( stream_.userBuffer[1],
+ inBufferList->mBuffers[handle->iStream[1]].mData,
+ inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
}
}
- if ( !info->usingCallback && (stream->mode != DUPLEX || deviceId == info->device[1] ) )
- info->blockTick = false;
-
- MUTEX_UNLOCK(&stream->mutex);
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
+ RtApi::tickStreamTime();
+ return SUCCESS;
}
-void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData)
+const char* RtApiCore :: getErrorCode( OSStatus code )
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ switch( code ) {
- stream->callbackInfo.callback = (void *) callback;
- stream->callbackInfo.userData = userData;
- stream->callbackInfo.usingCallback = true;
-}
+ case kAudioHardwareNotRunningError:
+ return "kAudioHardwareNotRunningError";
-//******************** End of __MACOSX_CORE__ *********************//
+ case kAudioHardwareUnspecifiedError:
+ return "kAudioHardwareUnspecifiedError";
-#elif defined(__LINUX_ALSA__)
+ case kAudioHardwareUnknownPropertyError:
+ return "kAudioHardwareUnknownPropertyError";
-#define MAX_DEVICES 16
+ case kAudioHardwareBadPropertySizeError:
+ return "kAudioHardwareBadPropertySizeError";
-void RtAudio :: initialize(void)
-{
- int card, result, device;
- char name[32];
- const char *cardId;
- char deviceNames[MAX_DEVICES][32];
- snd_ctl_t *handle;
- snd_ctl_card_info_t *info;
- snd_ctl_card_info_alloca(&info);
+ case kAudioHardwareIllegalOperationError:
+ return "kAudioHardwareIllegalOperationError";
- // Count cards and devices
- nDevices = 0;
- card = -1;
- snd_card_next(&card);
- while ( card >= 0 ) {
- sprintf(name, "hw:%d", card);
- result = snd_ctl_open(&handle, name, 0);
- if (result < 0) {
- sprintf(message, "RtAudio: ALSA control open (%i): %s.", card, snd_strerror(result));
- error(RtError::DEBUG_WARNING);
- goto next_card;
- }
- result = snd_ctl_card_info(handle, info);
- if (result < 0) {
- sprintf(message, "RtAudio: ALSA control hardware info (%i): %s.", card, snd_strerror(result));
- error(RtError::DEBUG_WARNING);
- goto next_card;
- }
- cardId = snd_ctl_card_info_get_id(info);
- device = -1;
- while (1) {
- result = snd_ctl_pcm_next_device(handle, &device);
- if (result < 0) {
- sprintf(message, "RtAudio: ALSA control next device (%i): %s.", card, snd_strerror(result));
- error(RtError::DEBUG_WARNING);
- break;
- }
- if (device < 0)
- break;
- if ( strlen(cardId) )
- sprintf( deviceNames[nDevices++], "hw:%s,%d", cardId, device );
- else
- sprintf( deviceNames[nDevices++], "hw:%d,%d", card, device );
- if ( nDevices > MAX_DEVICES ) break;
- }
- if ( nDevices > MAX_DEVICES ) break;
- next_card:
- snd_ctl_close(handle);
- snd_card_next(&card);
- }
+ case kAudioHardwareBadObjectError:
+ return "kAudioHardwareBadObjectError";
- if (nDevices == 0) return;
+ case kAudioHardwareBadDeviceError:
+ return "kAudioHardwareBadDeviceError";
- // Allocate the RTAUDIO_DEVICE structures.
- devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE));
- if (devices == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
- }
+ case kAudioHardwareBadStreamError:
+ return "kAudioHardwareBadStreamError";
- // Write device ascii identifiers to device structures and then
- // probe the device capabilities.
- for (int i=0; i<nDevices; i++) {
- strncpy(devices[i].name, deviceNames[i], 32);
- //probeDeviceInfo(&devices[i]);
- }
+ case kAudioHardwareUnsupportedOperationError:
+ return "kAudioHardwareUnsupportedOperationError";
+
+ case kAudioDeviceUnsupportedFormatError:
+ return "kAudioDeviceUnsupportedFormatError";
+
+ case kAudioDevicePermissionsError:
+ return "kAudioDevicePermissionsError";
+
+ default:
+ return "CoreAudio unknown error";
+ }
}
-int RtAudio :: getDefaultInputDevice(void)
+//******************** End of __MACOSX_CORE__ *********************//
+#endif
+
+#if defined(__UNIX_JACK__)
+
+// JACK is a low-latency audio server, originally written for the
+// GNU/Linux operating system and now also ported to OS-X. It can
+// connect a number of different applications to an audio device, as
+// well as allowing them to share audio between themselves.
+//
+// When using JACK with RtAudio, "devices" refer to JACK clients that
+// have ports connected to the server. The JACK server is typically
+// started in a terminal as follows:
+//
+// .jackd -d alsa -d hw:0
+//
+// or through an interface program such as qjackctl. Many of the
+// parameters normally set for a stream are fixed by the JACK server
+// and can be specified when the JACK server is started. In
+// particular,
+//
+// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
+//
+// specifies a sample rate of 44100 Hz, a buffer size of 512 sample
+// frames, and number of buffers = 4. Once the server is running, it
+// is not possible to override these values. If the values are not
+// specified in the command-line, the JACK server uses default values.
+//
+// The JACK server does not have to be running when an instance of
+// RtApiJack is created, though the function getDeviceCount() will
+// report 0 devices found until JACK has been started. When no
+// devices are available (i.e., the JACK server is not running), a
+// stream cannot be opened.
+
+#include <jack/jack.h>
+#include <unistd.h>
+
+// A structure to hold various information related to the Jack API
+// implementation.
+struct JackHandle {
+ jack_client_t *client;
+ jack_port_t **ports[2];
+ std::string deviceName[2];
+ bool xrun[2];
+ pthread_cond_t condition;
+ int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
+
+ JackHandle()
+ :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
+};
+
+RtApiJack :: RtApiJack()
{
- // No ALSA API functions for default devices.
- return 0;
+ // Nothing to do here.
}
-int RtAudio :: getDefaultOutputDevice(void)
+RtApiJack :: ~RtApiJack()
{
- // No ALSA API functions for default devices.
- return 0;
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
}
-void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info)
+unsigned int RtApiJack :: getDeviceCount( void )
{
- int err;
- int open_mode = SND_PCM_ASYNC;
- snd_pcm_t *handle;
- snd_ctl_t *chandle;
- snd_pcm_stream_t stream;
- snd_pcm_info_t *pcminfo;
- snd_pcm_info_alloca(&pcminfo);
- snd_pcm_hw_params_t *params;
- snd_pcm_hw_params_alloca(¶ms);
- char name[32];
- char *card;
-
- // Open the control interface for this card.
- strncpy( name, info->name, 32 );
- card = strtok(name, ",");
- err = snd_ctl_open(&chandle, card, 0);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA control open (%s): %s.", card, snd_strerror(err));
- error(RtError::DEBUG_WARNING);
- return;
+ // See if we can become a jack client.
+ jack_client_t *client = jack_client_new( "RtApiJackCount" );
+ if ( client == 0 ) return 0;
+
+ const char **ports;
+ std::string port, previousPort;
+ unsigned int nChannels = 0, nDevices = 0;
+ ports = jack_get_ports( client, NULL, NULL, 0 );
+ if ( ports ) {
+ // Parse the port names up to the first colon (:).
+ unsigned int iColon = 0;
+ do {
+ port = (char *) ports[ nChannels ];
+ iColon = port.find(":");
+ if ( iColon != std::string::npos ) {
+ port = port.substr( 0, iColon + 1 );
+ if ( port != previousPort ) {
+ nDevices++;
+ previousPort = port;
+ }
+ }
+ } while ( ports[++nChannels] );
+ free( ports );
}
- unsigned int dev = (unsigned int) atoi( strtok(NULL, ",") );
-
- // First try for playback
- stream = SND_PCM_STREAM_PLAYBACK;
- snd_pcm_info_set_device(pcminfo, dev);
- snd_pcm_info_set_subdevice(pcminfo, 0);
- snd_pcm_info_set_stream(pcminfo, stream);
- if ((err = snd_ctl_pcm_info(chandle, pcminfo)) < 0) {
- if (err == -ENOENT) {
- sprintf(message, "RtAudio: ALSA pcm device (%s) doesn't handle output!", info->name);
- error(RtError::DEBUG_WARNING);
- }
- else {
- sprintf(message, "RtAudio: ALSA snd_ctl_pcm_info error for device (%s) output: %s",
- info->name, snd_strerror(err));
- error(RtError::DEBUG_WARNING);
- }
- goto capture_probe;
- }
+ jack_client_close( client );
+ return nDevices;
+}
- err = snd_pcm_open(&handle, info->name, stream, open_mode | SND_PCM_NONBLOCK );
- if (err < 0) {
- if ( err == EBUSY )
- sprintf(message, "RtAudio: ALSA pcm playback device (%s) is busy: %s.",
- info->name, snd_strerror(err));
- else
- sprintf(message, "RtAudio: ALSA pcm playback open (%s) error: %s.",
- info->name, snd_strerror(err));
- error(RtError::DEBUG_WARNING);
- goto capture_probe;
+RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ jack_client_t *client = jack_client_new( "RtApiJackInfo" );
+ if ( client == 0 ) {
+ errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
+ error( RtError::WARNING );
+ return info;
+ }
+
+ const char **ports;
+ std::string port, previousPort;
+ unsigned int nPorts = 0, nDevices = 0;
+ ports = jack_get_ports( client, NULL, NULL, 0 );
+ if ( ports ) {
+ // Parse the port names up to the first colon (:).
+ unsigned int iColon = 0;
+ do {
+ port = (char *) ports[ nPorts ];
+ iColon = port.find(":");
+ if ( iColon != std::string::npos ) {
+ port = port.substr( 0, iColon );
+ if ( port != previousPort ) {
+ if ( nDevices == device ) info.name = port;
+ nDevices++;
+ previousPort = port;
+ }
+ }
+ } while ( ports[++nPorts] );
+ free( ports );
}
- // We have an open device ... allocate the parameter structure.
- err = snd_pcm_hw_params_any(handle, params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA hardware probe error (%s): %s.",
- info->name, snd_strerror(err));
- error(RtError::WARNING);
- goto capture_probe;
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
+ error( RtError::INVALID_USE );
}
- // Get output channel information.
- info->minOutputChannels = snd_pcm_hw_params_get_channels_min(params);
- info->maxOutputChannels = snd_pcm_hw_params_get_channels_max(params);
-
- snd_pcm_close(handle);
-
- capture_probe:
- // Now try for capture
- stream = SND_PCM_STREAM_CAPTURE;
- snd_pcm_info_set_stream(pcminfo, stream);
+ // Get the current jack server sample rate.
+ info.sampleRates.clear();
+ info.sampleRates.push_back( jack_get_sample_rate( client ) );
- err = snd_ctl_pcm_info(chandle, pcminfo);
- snd_ctl_close(chandle);
- if ( err < 0 ) {
- if (err == -ENOENT) {
- sprintf(message, "RtAudio: ALSA pcm device (%s) doesn't handle input!", info->name);
- error(RtError::DEBUG_WARNING);
- }
- else {
- sprintf(message, "RtAudio: ALSA snd_ctl_pcm_info error for device (%s) input: %s",
- info->name, snd_strerror(err));
- error(RtError::DEBUG_WARNING);
- }
- if (info->maxOutputChannels == 0)
- // didn't open for playback either ... device invalid
- return;
- goto probe_parameters;
+ // Count the available ports containing the client name as device
+ // channels. Jack "input ports" equal RtAudio output channels.
+ unsigned int nChannels = 0;
+ ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
+ if ( ports ) {
+ while ( ports[ nChannels ] ) nChannels++;
+ free( ports );
+ info.outputChannels = nChannels;
}
- err = snd_pcm_open(&handle, info->name, stream, open_mode | SND_PCM_NONBLOCK);
- if (err < 0) {
- if ( err == EBUSY )
- sprintf(message, "RtAudio: ALSA pcm capture device (%s) is busy: %s.",
- info->name, snd_strerror(err));
- else
- sprintf(message, "RtAudio: ALSA pcm capture open (%s) error: %s.",
- info->name, snd_strerror(err));
- error(RtError::DEBUG_WARNING);
- if (info->maxOutputChannels == 0)
- // didn't open for playback either ... device invalid
- return;
- goto probe_parameters;
- }
-
- // We have an open capture device ... allocate the parameter structure.
- err = snd_pcm_hw_params_any(handle, params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA hardware probe error (%s): %s.",
- info->name, snd_strerror(err));
- error(RtError::WARNING);
- if (info->maxOutputChannels > 0)
- goto probe_parameters;
- else
- return;
+ // Jack "output ports" equal RtAudio input channels.
+ nChannels = 0;
+ ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
+ if ( ports ) {
+ while ( ports[ nChannels ] ) nChannels++;
+ free( ports );
+ info.inputChannels = nChannels;
}
- // Get input channel information.
- info->minInputChannels = snd_pcm_hw_params_get_channels_min(params);
- info->maxInputChannels = snd_pcm_hw_params_get_channels_max(params);
-
- snd_pcm_close(handle);
+ if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
+ jack_client_close(client);
+ errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
+ error( RtError::WARNING );
+ return info;
+ }
// If device opens for both playback and capture, we determine the channels.
- if (info->maxOutputChannels == 0 || info->maxInputChannels == 0)
- goto probe_parameters;
-
- info->hasDuplexSupport = true;
- info->maxDuplexChannels = (info->maxOutputChannels > info->maxInputChannels) ?
- info->maxInputChannels : info->maxOutputChannels;
- info->minDuplexChannels = (info->minOutputChannels > info->minInputChannels) ?
- info->minInputChannels : info->minOutputChannels;
-
- probe_parameters:
- // At this point, we just need to figure out the supported data
- // formats and sample rates. We'll proceed by opening the device in
- // the direction with the maximum number of channels, or playback if
- // they are equal. This might limit our sample rate options, but so
- // be it.
-
- if (info->maxOutputChannels >= info->maxInputChannels)
- stream = SND_PCM_STREAM_PLAYBACK;
- else
- stream = SND_PCM_STREAM_CAPTURE;
+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
- err = snd_pcm_open(&handle, info->name, stream, open_mode);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA pcm (%s) won't reopen during probe: %s.",
- info->name, snd_strerror(err));
- error(RtError::WARNING);
- return;
- }
+ // Jack always uses 32-bit floats.
+ info.nativeFormats = RTAUDIO_FLOAT32;
- // We have an open device ... allocate the parameter structure.
- err = snd_pcm_hw_params_any(handle, params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA hardware reopen probe error (%s): %s.",
- info->name, snd_strerror(err));
- error(RtError::WARNING);
- return;
- }
+ // Jack doesn't provide default devices so we'll use the first available one.
+ if ( device == 0 && info.outputChannels > 0 )
+ info.isDefaultOutput = true;
+ if ( device == 0 && info.inputChannels > 0 )
+ info.isDefaultInput = true;
- // Test a non-standard sample rate to see if continuous rate is supported.
- int dir = 0;
- if (snd_pcm_hw_params_test_rate(handle, params, 35500, dir) == 0) {
- // It appears that continuous sample rate support is available.
- info->nSampleRates = -1;
- info->sampleRates[0] = snd_pcm_hw_params_get_rate_min(params, &dir);
- info->sampleRates[1] = snd_pcm_hw_params_get_rate_max(params, &dir);
- }
- else {
- // No continuous rate support ... test our discrete set of sample rate values.
- info->nSampleRates = 0;
- for (int i=0; i<MAX_SAMPLE_RATES; i++) {
- if (snd_pcm_hw_params_test_rate(handle, params, SAMPLE_RATES[i], dir) == 0) {
- info->sampleRates[info->nSampleRates] = SAMPLE_RATES[i];
- info->nSampleRates++;
- }
- }
- if (info->nSampleRates == 0) {
- snd_pcm_close(handle);
- return;
- }
- }
+ jack_client_close(client);
+ info.probed = true;
+ return info;
+}
- // Probe the supported data formats ... we don't care about endian-ness just yet
- snd_pcm_format_t format;
- info->nativeFormats = 0;
- format = SND_PCM_FORMAT_S8;
- if (snd_pcm_hw_params_test_format(handle, params, format) == 0)
- info->nativeFormats |= RTAUDIO_SINT8;
- format = SND_PCM_FORMAT_S16;
- if (snd_pcm_hw_params_test_format(handle, params, format) == 0)
- info->nativeFormats |= RTAUDIO_SINT16;
- format = SND_PCM_FORMAT_S24;
- if (snd_pcm_hw_params_test_format(handle, params, format) == 0)
- info->nativeFormats |= RTAUDIO_SINT24;
- format = SND_PCM_FORMAT_S32;
- if (snd_pcm_hw_params_test_format(handle, params, format) == 0)
- info->nativeFormats |= RTAUDIO_SINT32;
- format = SND_PCM_FORMAT_FLOAT;
- if (snd_pcm_hw_params_test_format(handle, params, format) == 0)
- info->nativeFormats |= RTAUDIO_FLOAT32;
- format = SND_PCM_FORMAT_FLOAT64;
- if (snd_pcm_hw_params_test_format(handle, params, format) == 0)
- info->nativeFormats |= RTAUDIO_FLOAT64;
+int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
+{
+ CallbackInfo *info = (CallbackInfo *) infoPointer;
- // Check that we have at least one supported format
- if (info->nativeFormats == 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA PCM device (%s) data format not supported by RtAudio.",
- info->name);
- error(RtError::WARNING);
- return;
- }
+ RtApiJack *object = (RtApiJack *) info->object;
+ if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
- // That's all ... close the device and return
- snd_pcm_close(handle);
- info->probed = true;
- return;
+ return 0;
}
-bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream,
- STREAM_MODE mode, int channels,
- int sampleRate, RTAUDIO_FORMAT format,
- int *bufferSize, int numberOfBuffers)
+void jackShutdown( void *infoPointer )
{
-#if defined(__RTAUDIO_DEBUG__)
- snd_output_t *out;
- snd_output_stdio_attach(&out, stderr, 0);
-#endif
-
- // I'm not using the "plug" interface ... too much inconsistent behavior.
- const char *name = devices[device].name;
+ CallbackInfo *info = (CallbackInfo *) infoPointer;
+ RtApiJack *object = (RtApiJack *) info->object;
+
+ // Check current stream state. If stopped, then we'll assume this
+ // was called as a result of a call to RtApiJack::stopStream (the
+ // deactivation of a client handle causes this function to be called).
+ // If not, we'll assume the Jack server is shutting down or some
+ // other problem occurred and we should close the stream.
+ if ( object->isStreamRunning() == false ) return;
+
+ object->closeStream();
+ std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
+}
- snd_pcm_stream_t alsa_stream;
- if (mode == OUTPUT)
- alsa_stream = SND_PCM_STREAM_PLAYBACK;
- else
- alsa_stream = SND_PCM_STREAM_CAPTURE;
-
- int err;
- snd_pcm_t *handle;
- int alsa_open_mode = SND_PCM_ASYNC;
- err = snd_pcm_open(&handle, name, alsa_stream, alsa_open_mode);
- if (err < 0) {
- sprintf(message,"RtAudio: ALSA pcm device (%s) won't open: %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
+int jackXrun( void *infoPointer )
+{
+ JackHandle *handle = (JackHandle *) infoPointer;
- // Fill the parameter structure.
- snd_pcm_hw_params_t *hw_params;
- snd_pcm_hw_params_alloca(&hw_params);
- err = snd_pcm_hw_params_any(handle, hw_params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error getting parameter handle (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
+ if ( handle->ports[0] ) handle->xrun[0] = true;
+ if ( handle->ports[1] ) handle->xrun[1] = true;
-#if defined(__RTAUDIO_DEBUG__)
- fprintf(stderr, "\nRtAudio: ALSA dump hardware params just after device open:\n\n");
- snd_pcm_hw_params_dump(hw_params, out);
-#endif
+ return 0;
+}
+bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
- // Set access ... try interleaved access first, then non-interleaved
- if ( !snd_pcm_hw_params_test_access( handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED) ) {
- err = snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
- }
- else if ( !snd_pcm_hw_params_test_access( handle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED) ) {
- err = snd_pcm_hw_params_set_access(handle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED);
- stream->deInterleave[mode] = true;
+ // Look for jack server and try to become a client (only do once per stream).
+ jack_client_t *client = 0;
+ if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
+ if ( options && !options->streamName.empty() )
+ client = jack_client_new( options->streamName.c_str() );
+ else
+ client = jack_client_new( "RtApiJack" );
+ if ( client == 0 ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
+ error( RtError::WARNING );
+ return FAILURE;
+ }
}
else {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA device (%s) access not supported by RtAudio.", name);
- error(RtError::WARNING);
- return FAILURE;
+ // The handle must have been created on an earlier pass.
+ client = handle->client;
+ }
+
+ const char **ports;
+ std::string port, previousPort, deviceName;
+ unsigned int nPorts = 0, nDevices = 0;
+ ports = jack_get_ports( client, NULL, NULL, 0 );
+ if ( ports ) {
+ // Parse the port names up to the first colon (:).
+ unsigned int iColon = 0;
+ do {
+ port = (char *) ports[ nPorts ];
+ iColon = port.find(":");
+ if ( iColon != std::string::npos ) {
+ port = port.substr( 0, iColon );
+ if ( port != previousPort ) {
+ if ( nDevices == device ) deviceName = port;
+ nDevices++;
+ previousPort = port;
+ }
+ }
+ } while ( ports[++nPorts] );
+ free( ports );
}
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error setting access ( (%s): %s.", name, snd_strerror(err));
- error(RtError::WARNING);
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
return FAILURE;
}
- // Determine how to set the device format.
- stream->userFormat = format;
- snd_pcm_format_t device_format;
-
- if (format == RTAUDIO_SINT8)
- device_format = SND_PCM_FORMAT_S8;
- else if (format == RTAUDIO_SINT16)
- device_format = SND_PCM_FORMAT_S16;
- else if (format == RTAUDIO_SINT24)
- device_format = SND_PCM_FORMAT_S24;
- else if (format == RTAUDIO_SINT32)
- device_format = SND_PCM_FORMAT_S32;
- else if (format == RTAUDIO_FLOAT32)
- device_format = SND_PCM_FORMAT_FLOAT;
- else if (format == RTAUDIO_FLOAT64)
- device_format = SND_PCM_FORMAT_FLOAT64;
-
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = format;
- goto set_format;
+ // 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 );
}
- // The user requested format is not natively supported by the device.
- device_format = SND_PCM_FORMAT_FLOAT64;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = RTAUDIO_FLOAT64;
- goto set_format;
+ // 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;
}
- device_format = SND_PCM_FORMAT_FLOAT;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = RTAUDIO_FLOAT32;
- goto set_format;
+ // Check the jack server sample rate.
+ unsigned int jackRate = jack_get_sample_rate( client );
+ if ( sampleRate != jackRate ) {
+ jack_client_close( client );
+ errorStream_ << "RtApiJack::probeDeviceOpen: the requested sample rate (" << sampleRate << ") is different than the JACK server rate (" << jackRate << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
+ stream_.sampleRate = jackRate;
- device_format = SND_PCM_FORMAT_S32;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- goto set_format;
- }
+ // Get the latency of the JACK port.
+ ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
+ if ( ports[ firstChannel ] )
+ stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
+ free( ports );
- device_format = SND_PCM_FORMAT_S24;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = RTAUDIO_SINT24;
- goto set_format;
- }
+ // The jack server always uses 32-bit floating-point data.
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ stream_.userFormat = format;
- device_format = SND_PCM_FORMAT_S16;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- goto set_format;
- }
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
- device_format = SND_PCM_FORMAT_S8;
- if (snd_pcm_hw_params_test_format(handle, hw_params, device_format) == 0) {
- stream->deviceFormat[mode] = RTAUDIO_SINT8;
- goto set_format;
- }
+ // Jack always uses non-interleaved buffers.
+ stream_.deviceInterleaved[mode] = false;
- // If we get here, no supported format was found.
- sprintf(message,"RtAudio: ALSA pcm device (%s) data format not supported by RtAudio.", name);
- snd_pcm_close(handle);
- error(RtError::WARNING);
- return FAILURE;
+ // Jack always provides host byte-ordered data.
+ stream_.doByteSwap[mode] = false;
- set_format:
- err = snd_pcm_hw_params_set_format(handle, hw_params, device_format);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error setting format (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
+ // Get the buffer size. The buffer size and number of buffers
+ // (periods) is set when the jack server is started.
+ stream_.bufferSize = (int) jack_get_buffer_size( client );
+ *bufferSize = stream_.bufferSize;
- // Determine whether byte-swaping is necessary.
- stream->doByteSwap[mode] = false;
- if (device_format != SND_PCM_FORMAT_S8) {
- err = snd_pcm_format_cpu_endian(device_format);
- if (err == 0)
- stream->doByteSwap[mode] = true;
- else if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error getting format endian-ness (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+ stream_.nDeviceChannels[mode] = channels;
+ stream_.nUserChannels[mode] = channels;
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate our JackHandle structure for the stream.
+ if ( handle == 0 ) {
+ try {
+ handle = new JackHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
+ goto error;
}
- }
- // Set the sample rate.
- err = snd_pcm_hw_params_set_rate(handle, hw_params, (unsigned int)sampleRate, 0);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error setting sample rate (%d) on device (%s): %s.",
- sampleRate, name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+ if ( pthread_cond_init(&handle->condition, NULL) ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
+ goto error;
+ }
+ stream_.apiHandle = (void *) handle;
+ handle->client = client;
}
+ handle->deviceName[mode] = deviceName;
- // Determine the number of channels for this device. We support a possible
- // minimum device channel number > than the value requested by the user.
- stream->nUserChannels[mode] = channels;
- int device_channels = snd_pcm_hw_params_get_channels_max(hw_params);
- if (device_channels < channels) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: channels (%d) not supported by device (%s).",
- channels, name);
- error(RtError::WARNING);
- return FAILURE;
+ // Allocate necessary internal buffers.
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
}
- device_channels = snd_pcm_hw_params_get_channels_min(hw_params);
- if (device_channels < channels) device_channels = channels;
- stream->nDeviceChannels[mode] = device_channels;
-
- // Set the device channels.
- err = snd_pcm_hw_params_set_channels(handle, hw_params, device_channels);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error setting channels (%d) on device (%s): %s.",
- device_channels, name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
+ if ( stream_.doConvertBuffer[mode] ) {
- // Set the buffer number, which in ALSA is referred to as the "period".
- int dir;
- int periods = numberOfBuffers;
- // Even though the hardware might allow 1 buffer, it won't work reliably.
- if (periods < 2) periods = 2;
- err = snd_pcm_hw_params_get_periods_min(hw_params, &dir);
- if (err > periods) periods = err;
- err = snd_pcm_hw_params_get_periods_max(hw_params, &dir);
- if (err < periods) periods = err;
-
- err = snd_pcm_hw_params_set_periods(handle, hw_params, periods, 0);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error setting periods (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
+ bool makeBuffer = true;
+ if ( mode == OUTPUT )
+ bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ else { // mode == INPUT
+ bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
+ if ( bufferBytes < bytesOut ) makeBuffer = false;
+ }
+ }
- // Set the buffer (or period) size.
- err = snd_pcm_hw_params_get_period_size_min(hw_params, &dir);
- if (err > *bufferSize) *bufferSize = err;
-
- err = snd_pcm_hw_params_set_period_size(handle, hw_params, *bufferSize, 0);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error setting period size (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
}
- // If attempting to setup a duplex stream, the bufferSize parameter
- // MUST be the same in both directions!
- if ( stream->mode == OUTPUT && mode == INPUT && *bufferSize != stream->bufferSize ) {
- sprintf( message, "RtAudio: ALSA error setting buffer size for duplex stream on device (%s).",
- name );
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ // Allocate memory for the Jack ports (channels) identifiers.
+ handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
+ if ( handle->ports[mode] == NULL ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
+ goto error;
}
- stream->bufferSize = *bufferSize;
+ stream_.device[mode] = device;
+ stream_.channelOffset[mode] = firstChannel;
+ stream_.state = STREAM_STOPPED;
+ stream_.callbackInfo.object = (void *) this;
- // Install the hardware configuration
- err = snd_pcm_hw_params(handle, hw_params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error installing hardware configuration (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ // We had already set up the stream for output.
+ stream_.mode = DUPLEX;
+ else {
+ stream_.mode = mode;
+ jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
+ jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );
+ jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
}
-#if defined(__RTAUDIO_DEBUG__)
- fprintf(stderr, "\nRtAudio: ALSA dump hardware params after installation:\n\n");
- snd_pcm_hw_params_dump(hw_params, out);
-#endif
-
- /*
- // Install the software configuration
- snd_pcm_sw_params_t *sw_params = NULL;
- snd_pcm_sw_params_alloca(&sw_params);
- snd_pcm_sw_params_current(handle, sw_params);
- err = snd_pcm_sw_params(handle, sw_params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message, "RtAudio: ALSA error installing software configuration (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+ // Register our ports.
+ char label[64];
+ if ( mode == OUTPUT ) {
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ snprintf( label, 64, "outport %d", i );
+ handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
+ JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
+ }
+ }
+ else {
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ snprintf( label, 64, "inport %d", i );
+ handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
+ JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
+ }
}
- */
-
- // Set handle and flags for buffer conversion
- stream->handle[mode] = handle;
- stream->doConvertBuffer[mode] = false;
- if (stream->userFormat != stream->deviceFormat[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] > 1 && stream->deInterleave[mode])
- stream->doConvertBuffer[mode] = true;
- // Allocate necessary internal buffers
- if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) {
+ // Setup the buffer conversion information structure. We don't use
+ // buffers to do channel offsets, so we override that parameter
+ // here.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
- long buffer_bytes;
- if (stream->nUserChannels[0] >= stream->nUserChannels[1])
- buffer_bytes = stream->nUserChannels[0];
- else
- buffer_bytes = stream->nUserChannels[1];
+ return SUCCESS;
- buffer_bytes *= *bufferSize * formatBytes(stream->userFormat);
- if (stream->userBuffer) free(stream->userBuffer);
- stream->userBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->userBuffer == NULL)
- goto memory_error;
- }
+ error:
+ if ( handle ) {
+ pthread_cond_destroy( &handle->condition );
+ jack_client_close( handle->client );
- if ( stream->doConvertBuffer[mode] ) {
+ if ( handle->ports[0] ) free( handle->ports[0] );
+ if ( handle->ports[1] ) free( handle->ports[1] );
- long buffer_bytes;
- bool makeBuffer = true;
- if ( mode == OUTPUT )
- buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- else { // mode == INPUT
- buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]);
- if ( stream->mode == OUTPUT && stream->deviceBuffer ) {
- long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- if ( buffer_bytes < bytes_out ) makeBuffer = false;
- }
- }
+ delete handle;
+ stream_.apiHandle = 0;
+ }
- if ( makeBuffer ) {
- buffer_bytes *= *bufferSize;
- if (stream->deviceBuffer) free(stream->deviceBuffer);
- stream->deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->deviceBuffer == NULL)
- goto memory_error;
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
}
}
- stream->device[mode] = device;
- stream->state = STREAM_STOPPED;
- if ( stream->mode == OUTPUT && mode == INPUT )
- // We had already set up an output stream.
- stream->mode = DUPLEX;
- else
- stream->mode = mode;
- stream->nBuffers = periods;
- stream->sampleRate = sampleRate;
-
- return SUCCESS;
-
- memory_error:
- if (stream->handle[0]) {
- snd_pcm_close(stream->handle[0]);
- stream->handle[0] = 0;
- }
- if (stream->handle[1]) {
- snd_pcm_close(stream->handle[1]);
- stream->handle[1] = 0;
- }
- if (stream->userBuffer) {
- free(stream->userBuffer);
- stream->userBuffer = 0;
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
}
- sprintf(message, "RtAudio: ALSA error allocating buffer memory (%s).", name);
- error(RtError::WARNING);
+
return FAILURE;
}
-void RtAudio :: closeStream(int streamId)
+void RtApiJack :: closeStream( void )
{
- // We don't want an exception to be thrown here because this
- // function is called by our class destructor. So, do our own
- // streamId check.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
- error(RtError::WARNING);
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiJack::closeStream(): no open stream to close!";
+ error( RtError::WARNING );
return;
}
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId];
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ if ( handle ) {
- if (stream->callbackInfo.usingCallback) {
- pthread_cancel(stream->callbackInfo.thread);
- pthread_join(stream->callbackInfo.thread, NULL);
- }
+ if ( stream_.state == STREAM_RUNNING )
+ jack_deactivate( handle->client );
- if (stream->state == STREAM_RUNNING) {
- if (stream->mode == OUTPUT || stream->mode == DUPLEX)
- snd_pcm_drop(stream->handle[0]);
- if (stream->mode == INPUT || stream->mode == DUPLEX)
- snd_pcm_drop(stream->handle[1]);
+ jack_client_close( handle->client );
}
- pthread_mutex_destroy(&stream->mutex);
-
- if (stream->handle[0])
- snd_pcm_close(stream->handle[0]);
-
- if (stream->handle[1])
- snd_pcm_close(stream->handle[1]);
+ if ( handle ) {
+ if ( handle->ports[0] ) free( handle->ports[0] );
+ if ( handle->ports[1] ) free( handle->ports[1] );
+ pthread_cond_destroy( &handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
- if (stream->userBuffer)
- free(stream->userBuffer);
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
- if (stream->deviceBuffer)
- free(stream->deviceBuffer);
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
- free(stream);
- streams.erase(streamId);
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
}
-void RtAudio :: startStream(int streamId)
+void RtApiJack :: startStream( void )
{
- // This method calls snd_pcm_prepare if the device isn't already in that state.
-
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiJack::startStream(): the stream is already running!";
+ error( RtError::WARNING );
+ return;
+ }
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK(&stream_.mutex);
- if (stream->state == STREAM_RUNNING)
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ int result = jack_activate( handle->client );
+ if ( result ) {
+ errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
goto unlock;
+ }
- int err;
- snd_pcm_state_t state;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- state = snd_pcm_state(stream->handle[0]);
- if (state != SND_PCM_STATE_PREPARED) {
- err = snd_pcm_prepare(stream->handle[0]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error preparing pcm device (%s): %s.",
- devices[stream->device[0]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
+ const char **ports;
+
+ // Get the list of available ports.
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ result = 1;
+ ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);
+ if ( ports == NULL) {
+ errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
+ goto unlock;
}
- }
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- state = snd_pcm_state(stream->handle[1]);
- if (state != SND_PCM_STATE_PREPARED) {
- err = snd_pcm_prepare(stream->handle[1]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error preparing pcm device (%s): %s.",
- devices[stream->device[1]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
+ // Now make the port connections. Since RtAudio wasn't designed to
+ // allow the user to select particular channels of a device, we'll
+ // just open the first "nChannels" ports with offset.
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ result = 1;
+ if ( ports[ stream_.channelOffset[0] + i ] )
+ result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
+ if ( result ) {
+ free( ports );
+ errorText_ = "RtApiJack::startStream(): error connecting output ports!";
+ goto unlock;
}
}
+ free(ports);
}
- stream->state = STREAM_RUNNING;
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
-}
-
-void RtAudio :: stopStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- int err;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- err = snd_pcm_drain(stream->handle[0]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.",
- devices[stream->device[0]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ result = 1;
+ ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );
+ if ( ports == NULL) {
+ errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
+ goto unlock;
}
- }
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- err = snd_pcm_drain(stream->handle[1]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.",
- devices[stream->device[1]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
+ // Now make the port connections. See note above.
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ result = 1;
+ if ( ports[ stream_.channelOffset[1] + i ] )
+ result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
+ if ( result ) {
+ free( ports );
+ errorText_ = "RtApiJack::startStream(): error connecting input ports!";
+ goto unlock;
+ }
}
+ free(ports);
}
- stream->state = STREAM_STOPPED;
+
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ stream_.state = STREAM_RUNNING;
unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
+
+ if ( result == 0 ) return;
+ error( RtError::SYSTEM_ERROR );
}
-void RtAudio :: abortStream(int streamId)
+void RtApiJack :: stopStream( void )
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
+ }
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK( &stream_.mutex );
- if (stream->state == STREAM_STOPPED)
- goto unlock;
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- int err;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- err = snd_pcm_drop(stream->handle[0]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.",
- devices[stream->device[0]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 1;
+ pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
}
}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- err = snd_pcm_drop(stream->handle[1]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error draining pcm device (%s): %s.",
- devices[stream->device[1]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
- stream->state = STREAM_STOPPED;
+ jack_deactivate( handle->client );
+ stream_.state = STREAM_STOPPED;
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK( &stream_.mutex );
}
-int RtAudio :: streamWillBlock(int streamId)
+void RtApiJack :: abortStream( void )
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- int err = 0, frames = 0;
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- err = snd_pcm_avail_update(stream->handle[0]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error getting available frames for device (%s): %s.",
- devices[stream->device[0]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
-
- frames = err;
-
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- err = snd_pcm_avail_update(stream->handle[1]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error getting available frames for device (%s): %s.",
- devices[stream->device[1]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- if (frames > err) frames = err;
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
}
- frames = stream->bufferSize - frames;
- if (frames < 0) frames = 0;
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ handle->drainCounter = 1;
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
- return frames;
+ stopStream();
}
-void RtAudio :: tickStream(int streamId)
+bool RtApiJack :: callbackEvent( unsigned long nframes )
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- int stopStream = 0;
- if (stream->state == STREAM_STOPPED) {
- if (stream->callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds
- return;
+ if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtError::WARNING );
+ return FAILURE;
}
- else if (stream->callbackInfo.usingCallback) {
- RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback;
- stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData);
+ if ( stream_.bufferSize != nframes ) {
+ errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
+ error( RtError::WARNING );
+ return FAILURE;
}
- MUTEX_LOCK(&stream->mutex);
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
- // The state might change while waiting on a mutex.
- if (stream->state == STREAM_STOPPED)
- goto unlock;
+ // Check if we were draining the stream and signal is finished.
+ if ( handle->drainCounter > 3 ) {
+ if ( handle->internalDrain == false )
+ pthread_cond_signal( &handle->condition );
+ else
+ stopStream();
+ return SUCCESS;
+ }
- int err;
- char *buffer;
- int channels;
- RTAUDIO_FORMAT format;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
+ MUTEX_LOCK( &stream_.mutex );
- // Setup parameters and do buffer conversion if necessary.
- if (stream->doConvertBuffer[0]) {
- convertStreamBuffer(stream, OUTPUT);
- buffer = stream->deviceBuffer;
- channels = stream->nDeviceChannels[0];
- format = stream->deviceFormat[0];
+ // Invoke user callback first, to get fresh output data.
+ if ( handle->drainCounter == 0 ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
}
- else {
- buffer = stream->userBuffer;
- channels = stream->nUserChannels[0];
- format = stream->userFormat;
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
}
+ handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( handle->drainCounter == 2 ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ abortStream();
+ return SUCCESS;
+ }
+ else if ( handle->drainCounter == 1 )
+ handle->internalDrain = true;
+ }
- // Do byte swapping if necessary.
- if (stream->doByteSwap[0])
- byteSwapBuffer(buffer, stream->bufferSize * channels, format);
+ jack_default_audio_sample_t *jackbuffer;
+ unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- // Write samples to device in interleaved/non-interleaved format.
- if (stream->deInterleave[0]) {
- void *bufs[channels];
- size_t offset = stream->bufferSize * formatBytes(format);
- for (int i=0; i<channels; i++)
- bufs[i] = (void *) (buffer + (i * offset));
- err = snd_pcm_writen(stream->handle[0], bufs, stream->bufferSize);
- }
- else
- err = snd_pcm_writei(stream->handle[0], buffer, stream->bufferSize);
+ if ( handle->drainCounter > 0 ) { // write zeros to the output stream
- if (err < stream->bufferSize) {
- // Either an error or underrun occured.
- if (err == -EPIPE) {
- snd_pcm_state_t state = snd_pcm_state(stream->handle[0]);
- if (state == SND_PCM_STATE_XRUN) {
- sprintf(message, "RtAudio: ALSA underrun detected.");
- error(RtError::WARNING);
- err = snd_pcm_prepare(stream->handle[0]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error preparing handle after underrun: %s.",
- snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
- else {
- sprintf(message, "RtAudio: ALSA error, current state is %s.",
- snd_pcm_state_name(state));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- goto unlock;
- }
- else {
- sprintf(message, "RtAudio: ALSA audio write error for device (%s): %s.",
- devices[stream->device[0]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
+ for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+ memset( jackbuffer, 0, bufferBytes );
}
+
}
- }
+ else if ( stream_.doConvertBuffer[0] ) {
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
+ convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
- // Setup parameters.
- if (stream->doConvertBuffer[1]) {
- buffer = stream->deviceBuffer;
- channels = stream->nDeviceChannels[1];
- format = stream->deviceFormat[1];
+ for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+ memcpy( jackbuffer, &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
+ }
}
- else {
- buffer = stream->userBuffer;
- channels = stream->nUserChannels[1];
- format = stream->userFormat;
+ else { // no buffer conversion
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
+ memcpy( jackbuffer, &stream_.userBuffer[0][i*bufferBytes], bufferBytes );
+ }
}
- // Read samples from device in interleaved/non-interleaved format.
- if (stream->deInterleave[1]) {
- void *bufs[channels];
- size_t offset = stream->bufferSize * formatBytes(format);
- for (int i=0; i<channels; i++)
- bufs[i] = (void *) (buffer + (i * offset));
- err = snd_pcm_readn(stream->handle[1], bufs, stream->bufferSize);
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
}
- else
- err = snd_pcm_readi(stream->handle[1], buffer, stream->bufferSize);
+ }
- if (err < stream->bufferSize) {
- // Either an error or underrun occured.
- if (err == -EPIPE) {
- snd_pcm_state_t state = snd_pcm_state(stream->handle[1]);
- if (state == SND_PCM_STATE_XRUN) {
- sprintf(message, "RtAudio: ALSA overrun detected.");
- error(RtError::WARNING);
- err = snd_pcm_prepare(stream->handle[1]);
- if (err < 0) {
- sprintf(message, "RtAudio: ALSA error preparing handle after overrun: %s.",
- snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
- else {
- sprintf(message, "RtAudio: ALSA error, current state is %s.",
- snd_pcm_state_name(state));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- goto unlock;
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ if ( stream_.doConvertBuffer[1] ) {
+ for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
+ memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
}
- else {
- sprintf(message, "RtAudio: ALSA audio read error for device (%s): %s.",
- devices[stream->device[1]].name, snd_strerror(err));
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+ }
+ else { // no buffer conversion
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
+ memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
}
}
-
- // Do byte swapping if necessary.
- if (stream->doByteSwap[1])
- byteSwapBuffer(buffer, stream->bufferSize * channels, format);
-
- // Do buffer conversion if necessary.
- if (stream->doConvertBuffer[1])
- convertStreamBuffer(stream, INPUT);
}
unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK(&stream_.mutex);
- if (stream->callbackInfo.usingCallback && stopStream)
- this->stopStream(streamId);
+ RtApi::tickStreamTime();
+ return SUCCESS;
}
+//******************** End of __UNIX_JACK__ *********************//
+#endif
-extern "C" void *callbackHandler(void *ptr)
-{
- CALLBACK_INFO *info = (CALLBACK_INFO *) ptr;
- RtAudio *object = (RtAudio *) info->object;
- int stream = info->streamId;
- bool *usingCallback = &info->usingCallback;
+#if defined(__WINDOWS_ASIO__) // ASIO API on Windows
- while ( *usingCallback ) {
- pthread_testcancel();
- try {
- object->tickStream(stream);
- }
- catch (RtError &exception) {
- fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n",
- exception.getMessage());
- break;
- }
- }
+// The ASIO API is designed around a callback scheme, so this
+// implementation is similar to that used for OS-X CoreAudio and Linux
+// Jack. The primary constraint with ASIO is that it only allows
+// access to a single driver at a time. Thus, it is not possible to
+// have more than one simultaneous RtAudio stream.
+//
+// This implementation also requires a number of external ASIO files
+// and a few global variables. The ASIO callback scheme does not
+// allow for the passing of user data, so we must create a global
+// pointer to our callbackInfo structure.
+//
+// On unix systems, we make use of a pthread condition variable.
+// Since there is no equivalent in Windows, I hacked something based
+// on information found in
+// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
- return 0;
-}
+#include "asiosys.h"
+#include "asio.h"
+#include "iasiothiscallresolver.h"
+#include "asiodrivers.h"
+#include <cmath>
-//******************** End of __LINUX_ALSA__ *********************//
+AsioDrivers drivers;
+ASIOCallbacks asioCallbacks;
+ASIODriverInfo driverInfo;
+CallbackInfo *asioCallbackInfo;
+bool asioXRun;
-#elif defined(__LINUX_OSS__)
+struct AsioHandle {
+ int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
+ ASIOBufferInfo *bufferInfos;
+ HANDLE condition;
-#include <sys/stat.h>
-#include <sys/types.h>
-#include <sys/ioctl.h>
-#include <unistd.h>
-#include <fcntl.h>
-#include <sys/soundcard.h>
-#include <errno.h>
-#include <math.h>
+ AsioHandle()
+ :drainCounter(0), internalDrain(false), bufferInfos(0) {}
+};
-#define DAC_NAME "/dev/dsp"
-#define MAX_DEVICES 16
-#define MAX_CHANNELS 16
+// Function declarations (definitions at end of section)
+static const char* getAsioErrorString( ASIOError result );
+void sampleRateChanged( ASIOSampleRate sRate );
+long asioMessages( long selector, long value, void* message, double* opt );
-void RtAudio :: initialize(void)
+RtApiAsio :: RtApiAsio()
{
- // Count cards and devices
- nDevices = 0;
-
- // We check /dev/dsp before probing devices. /dev/dsp is supposed to
- // be a link to the "default" audio device, of the form /dev/dsp0,
- // /dev/dsp1, etc... However, I've seen many cases where /dev/dsp was a
- // real device, so we need to check for that. Also, sometimes the
- // link is to /dev/dspx and other times just dspx. I'm not sure how
- // the latter works, but it does.
- char device_name[16];
- struct stat dspstat;
- int dsplink = -1;
- int i = 0;
- if (lstat(DAC_NAME, &dspstat) == 0) {
- if (S_ISLNK(dspstat.st_mode)) {
- i = readlink(DAC_NAME, device_name, sizeof(device_name));
- if (i > 0) {
- device_name[i] = '\0';
- if (i > 8) { // check for "/dev/dspx"
- if (!strncmp(DAC_NAME, device_name, 8))
- dsplink = atoi(&device_name[8]);
- }
- else if (i > 3) { // check for "dspx"
- if (!strncmp("dsp", device_name, 3))
- dsplink = atoi(&device_name[3]);
- }
- }
- else {
- sprintf(message, "RtAudio: cannot read value of symbolic link %s.", DAC_NAME);
- error(RtError::SYSTEM_ERROR);
- }
- }
- }
- else {
- sprintf(message, "RtAudio: cannot stat %s.", DAC_NAME);
- error(RtError::SYSTEM_ERROR);
- }
-
- // The OSS API doesn't provide a routine for determining the number
- // of devices. Thus, we'll just pursue a brute force method. The
- // idea is to start with /dev/dsp(0) and continue with higher device
- // numbers until we reach MAX_DSP_DEVICES. This should tell us how
- // many devices we have ... it is not a fullproof scheme, but hopefully
- // it will work most of the time.
-
- int fd = 0;
- char names[MAX_DEVICES][16];
- for (i=-1; i<MAX_DEVICES; i++) {
-
- // Probe /dev/dsp first, since it is supposed to be the default device.
- if (i == -1)
- sprintf(device_name, "%s", DAC_NAME);
- else if (i == dsplink)
- continue; // We've aready probed this device via /dev/dsp link ... try next device.
- else
- sprintf(device_name, "%s%d", DAC_NAME, i);
-
- // First try to open the device for playback, then record mode.
- fd = open(device_name, O_WRONLY | O_NONBLOCK);
- if (fd == -1) {
- // Open device for playback failed ... either busy or doesn't exist.
- if (errno != EBUSY && errno != EAGAIN) {
- // Try to open for capture
- fd = open(device_name, O_RDONLY | O_NONBLOCK);
- if (fd == -1) {
- // Open device for record failed.
- if (errno != EBUSY && errno != EAGAIN)
- continue;
- else {
- sprintf(message, "RtAudio: OSS record device (%s) is busy.", device_name);
- error(RtError::WARNING);
- // still count it for now
- }
- }
- }
- else {
- sprintf(message, "RtAudio: OSS playback device (%s) is busy.", device_name);
- error(RtError::WARNING);
- // still count it for now
- }
- }
-
- if (fd >= 0) close(fd);
- strncpy(names[nDevices], device_name, 16);
- nDevices++;
- }
-
- if (nDevices == 0) return;
-
- // Allocate the RTAUDIO_DEVICE structures.
- devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE));
- if (devices == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
+ // ASIO cannot run on a multi-threaded appartment. You can call
+ // CoInitialize beforehand, but it must be for appartment threading
+ // (in which case, CoInitilialize will return S_FALSE here).
+ coInitialized_ = false;
+ HRESULT hr = CoInitialize( NULL );
+ if ( FAILED(hr) ) {
+ errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
+ error( RtError::WARNING );
}
+ coInitialized_ = true;
- // Write device ascii identifiers to device control structure and then probe capabilities.
- for (i=0; i<nDevices; i++) {
- strncpy(devices[i].name, names[i], 16);
- //probeDeviceInfo(&devices[i]);
- }
+ drivers.removeCurrentDriver();
+ driverInfo.asioVersion = 2;
- return;
+ // See note in DirectSound implementation about GetDesktopWindow().
+ driverInfo.sysRef = GetForegroundWindow();
}
-int RtAudio :: getDefaultInputDevice(void)
+RtApiAsio :: ~RtApiAsio()
{
- // No OSS API functions for default devices.
- return 0;
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+ if ( coInitialized_ ) CoUninitialize();
}
-int RtAudio :: getDefaultOutputDevice(void)
+unsigned int RtApiAsio :: getDeviceCount( void )
{
- // No OSS API functions for default devices.
- return 0;
+ return (unsigned int) drivers.asioGetNumDev();
}
-void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info)
+RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
{
- int i, fd, channels, mask;
-
- // The OSS API doesn't provide a means for probing the capabilities
- // of devices. Thus, we'll just pursue a brute force method.
-
- // First try for playback
- fd = open(info->name, O_WRONLY | O_NONBLOCK);
- if (fd == -1) {
- // Open device failed ... either busy or doesn't exist
- if (errno == EBUSY || errno == EAGAIN)
- sprintf(message, "RtAudio: OSS playback device (%s) is busy and cannot be probed.",
- info->name);
- else
- sprintf(message, "RtAudio: OSS playback device (%s) open error.", info->name);
- error(RtError::DEBUG_WARNING);
- goto capture_probe;
- }
-
- // We have an open device ... see how many channels it can handle
- for (i=MAX_CHANNELS; i>0; i--) {
- channels = i;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1) {
- // This would normally indicate some sort of hardware error, but under ALSA's
- // OSS emulation, it sometimes indicates an invalid channel value. Further,
- // the returned channel value is not changed. So, we'll ignore the possible
- // hardware error.
- continue; // try next channel number
- }
- // Check to see whether the device supports the requested number of channels
- if (channels != i ) continue; // try next channel number
- // If here, we found the largest working channel value
- break;
- }
- info->maxOutputChannels = i;
+ RtAudio::DeviceInfo info;
+ info.probed = false;
- // Now find the minimum number of channels it can handle
- for (i=1; i<=info->maxOutputChannels; i++) {
- channels = i;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i)
- continue; // try next channel number
- // If here, we found the smallest working channel value
- break;
+ // Get device ID
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices == 0 ) {
+ errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
+ error( RtError::INVALID_USE );
}
- info->minOutputChannels = i;
- close(fd);
- capture_probe:
- // Now try for capture
- fd = open(info->name, O_RDONLY | O_NONBLOCK);
- if (fd == -1) {
- // Open device for capture failed ... either busy or doesn't exist
- if (errno == EBUSY || errno == EAGAIN)
- sprintf(message, "RtAudio: OSS capture device (%s) is busy and cannot be probed.",
- info->name);
- else
- sprintf(message, "RtAudio: OSS capture device (%s) open error.", info->name);
- error(RtError::DEBUG_WARNING);
- if (info->maxOutputChannels == 0)
- // didn't open for playback either ... device invalid
- return;
- goto probe_parameters;
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
+ error( RtError::INVALID_USE );
}
- // We have the device open for capture ... see how many channels it can handle
- for (i=MAX_CHANNELS; i>0; i--) {
- channels = i;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) {
- continue; // as above
+ // If a stream is already open, we cannot probe other devices. Thus, use the saved results.
+ if ( stream_.state != STREAM_CLOSED ) {
+ if ( device >= devices_.size() ) {
+ errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
+ error( RtError::WARNING );
+ return info;
}
- // If here, we found a working channel value
- break;
- }
- info->maxInputChannels = i;
-
- // Now find the minimum number of channels it can handle
- for (i=1; i<=info->maxInputChannels; i++) {
- channels = i;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i)
- continue; // try next channel number
- // If here, we found the smallest working channel value
- break;
+ return devices_[ device ];
}
- info->minInputChannels = i;
- close(fd);
- if (info->maxOutputChannels == 0 && info->maxInputChannels == 0) {
- sprintf(message, "RtAudio: OSS device (%s) reports zero channels for input and output.",
- info->name);
- error(RtError::DEBUG_WARNING);
- return;
+ char driverName[32];
+ ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
}
- // If device opens for both playback and capture, we determine the channels.
- if (info->maxOutputChannels == 0 || info->maxInputChannels == 0)
- goto probe_parameters;
-
- fd = open(info->name, O_RDWR | O_NONBLOCK);
- if (fd == -1)
- goto probe_parameters;
-
- ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
- ioctl(fd, SNDCTL_DSP_GETCAPS, &mask);
- if (mask & DSP_CAP_DUPLEX) {
- info->hasDuplexSupport = true;
- // We have the device open for duplex ... see how many channels it can handle
- for (i=MAX_CHANNELS; i>0; i--) {
- channels = i;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i)
- continue; // as above
- // If here, we found a working channel value
- break;
- }
- info->maxDuplexChannels = i;
+ info.name = driverName;
- // Now find the minimum number of channels it can handle
- for (i=1; i<=info->maxDuplexChannels; i++) {
- channels = i;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i)
- continue; // try next channel number
- // If here, we found the smallest working channel value
- break;
- }
- info->minDuplexChannels = i;
+ if ( !drivers.loadDriver( driverName ) ) {
+ errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
}
- close(fd);
-
- probe_parameters:
- // At this point, we need to figure out the supported data formats
- // and sample rates. We'll proceed by openning the device in the
- // direction with the maximum number of channels, or playback if
- // they are equal. This might limit our sample rate options, but so
- // be it.
- if (info->maxOutputChannels >= info->maxInputChannels) {
- fd = open(info->name, O_WRONLY | O_NONBLOCK);
- channels = info->maxOutputChannels;
- }
- else {
- fd = open(info->name, O_RDONLY | O_NONBLOCK);
- channels = info->maxInputChannels;
+ result = ASIOInit( &driverInfo );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
}
- if (fd == -1) {
- // We've got some sort of conflict ... abort
- sprintf(message, "RtAudio: OSS device (%s) won't reopen during probe.",
- info->name);
- error(RtError::DEBUG_WARNING);
- return;
+ // Determine the device channel information.
+ long inputChannels, outputChannels;
+ result = ASIOGetChannels( &inputChannels, &outputChannels );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
}
- // We have an open device ... set to maximum channels.
- i = channels;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1 || channels != i) {
- // We've got some sort of conflict ... abort
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) won't revert to previous channel setting.",
- info->name);
- error(RtError::DEBUG_WARNING);
- return;
- }
+ info.outputChannels = outputChannels;
+ info.inputChannels = inputChannels;
+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
- if (ioctl(fd, SNDCTL_DSP_GETFMTS, &mask) == -1) {
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) can't get supported audio formats.",
- info->name);
- error(RtError::DEBUG_WARNING);
- return;
+ // Determine the supported sample rates.
+ info.sampleRates.clear();
+ for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
+ result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
+ if ( result == ASE_OK )
+ info.sampleRates.push_back( SAMPLE_RATES[i] );
}
- // Probe the supported data formats ... we don't care about endian-ness just yet.
- int format;
- info->nativeFormats = 0;
-#if defined (AFMT_S32_BE)
- // This format does not seem to be in the 2.4 kernel version of OSS soundcard.h
- if (mask & AFMT_S32_BE) {
- format = AFMT_S32_BE;
- info->nativeFormats |= RTAUDIO_SINT32;
- }
-#endif
-#if defined (AFMT_S32_LE)
- /* This format is not in the 2.4.4 kernel version of OSS soundcard.h */
- if (mask & AFMT_S32_LE) {
- format = AFMT_S32_LE;
- info->nativeFormats |= RTAUDIO_SINT32;
- }
-#endif
- if (mask & AFMT_S8) {
- format = AFMT_S8;
- info->nativeFormats |= RTAUDIO_SINT8;
- }
- if (mask & AFMT_S16_BE) {
- format = AFMT_S16_BE;
- info->nativeFormats |= RTAUDIO_SINT16;
- }
- if (mask & AFMT_S16_LE) {
- format = AFMT_S16_LE;
- info->nativeFormats |= RTAUDIO_SINT16;
+ // Determine supported data types ... just check first channel and assume rest are the same.
+ ASIOChannelInfo channelInfo;
+ channelInfo.channel = 0;
+ channelInfo.isInput = true;
+ if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
+ result = ASIOGetChannelInfo( &channelInfo );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
}
- // Check that we have at least one supported format
- if (info->nativeFormats == 0) {
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) data format not supported by RtAudio.",
- info->name);
- error(RtError::DEBUG_WARNING);
- return;
+ info.nativeFormats = 0;
+ if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
+ info.nativeFormats |= RTAUDIO_SINT16;
+ else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
+ info.nativeFormats |= RTAUDIO_SINT32;
+ else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+ else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
+ info.nativeFormats |= RTAUDIO_FLOAT64;
+
+ if ( getDefaultOutputDevice() == device )
+ info.isDefaultOutput = true;
+ if ( getDefaultInputDevice() == device )
+ info.isDefaultInput = true;
+
+ info.probed = true;
+ drivers.removeCurrentDriver();
+ return info;
+}
+
+void bufferSwitch( long index, ASIOBool processNow )
+{
+ RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
+ object->callbackEvent( index );
+}
+
+void RtApiAsio :: saveDeviceInfo( void )
+{
+ devices_.clear();
+
+ unsigned int nDevices = getDeviceCount();
+ devices_.resize( nDevices );
+ for ( unsigned int i=0; i<nDevices; i++ )
+ devices_[i] = getDeviceInfo( i );
+}
+
+bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ // For ASIO, a duplex stream MUST use the same driver.
+ if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
+ return FAILURE;
}
- // Set the format
- i = format;
- if (ioctl(fd, SNDCTL_DSP_SETFMT, &format) == -1 || format != i) {
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) error setting data format.",
- info->name);
- error(RtError::DEBUG_WARNING);
- return;
+ char driverName[32];
+ ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: unable to get driver name (" << getAsioErrorString( result ) << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- // Probe the supported sample rates ... first get lower limit
- int speed = 1;
- if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) {
- // If we get here, we're probably using an ALSA driver with OSS-emulation,
- // which doesn't conform to the OSS specification. In this case,
- // we'll probe our predefined list of sample rates for working values.
- info->nSampleRates = 0;
- for (i=0; i<MAX_SAMPLE_RATES; i++) {
- speed = SAMPLE_RATES[i];
- if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) != -1) {
- info->sampleRates[info->nSampleRates] = SAMPLE_RATES[i];
- info->nSampleRates++;
- }
+ // The getDeviceInfo() function will not work when a stream is open
+ // because ASIO does not allow multiple devices to run at the same
+ // time. Thus, we'll probe the system before opening a stream and
+ // save the results for use by getDeviceInfo().
+ this->saveDeviceInfo();
+
+ // Only load the driver once for duplex stream.
+ if ( mode != INPUT || stream_.mode != OUTPUT ) {
+ if ( !drivers.loadDriver( driverName ) ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- if (info->nSampleRates == 0) {
- close(fd);
- return;
+
+ result = ASIOInit( &driverInfo );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- goto finished;
}
- info->sampleRates[0] = speed;
- // Now get upper limit
- speed = 1000000;
- if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) {
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) error setting sample rate.",
- info->name);
- error(RtError::DEBUG_WARNING);
- return;
+ // Check the device channel count.
+ long inputChannels, outputChannels;
+ result = ASIOGetChannels( &inputChannels, &outputChannels );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- info->sampleRates[1] = speed;
- info->nSampleRates = -1;
- finished: // That's all ... close the device and return
- close(fd);
- info->probed = true;
- return;
-}
+ if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
+ ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.nDeviceChannels[mode] = channels;
+ stream_.nUserChannels[mode] = channels;
+ stream_.channelOffset[mode] = firstChannel;
-bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream,
- STREAM_MODE mode, int channels,
- int sampleRate, RTAUDIO_FORMAT format,
- int *bufferSize, int numberOfBuffers)
-{
- int buffers, buffer_bytes, device_channels, device_format;
- int srate, temp, fd;
+ // Verify the sample rate is supported.
+ result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- const char *name = devices[device].name;
+ // Get the current sample rate
+ ASIOSampleRate currentRate;
+ result = ASIOGetSampleRate( ¤tRate );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- if (mode == OUTPUT)
- fd = open(name, O_WRONLY | O_NONBLOCK);
- else { // mode == INPUT
- if (stream->mode == OUTPUT && stream->device[0] == device) {
- // We just set the same device for playback ... close and reopen for duplex (OSS only).
- close(stream->handle[0]);
- stream->handle[0] = 0;
- // First check that the number previously set channels is the same.
- if (stream->nUserChannels[0] != channels) {
- sprintf(message, "RtAudio: input/output channels must be equal for OSS duplex device (%s).", name);
- goto error;
- }
- fd = open(name, O_RDWR | O_NONBLOCK);
+ // Set the sample rate only if necessary
+ if ( currentRate != sampleRate ) {
+ result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- else
- fd = open(name, O_RDONLY | O_NONBLOCK);
}
- if (fd == -1) {
- if (errno == EBUSY || errno == EAGAIN)
- sprintf(message, "RtAudio: OSS device (%s) is busy and cannot be opened.",
- name);
- else
- sprintf(message, "RtAudio: OSS device (%s) cannot be opened.", name);
- goto error;
+ // Determine the driver data type.
+ ASIOChannelInfo channelInfo;
+ channelInfo.channel = 0;
+ if ( mode == OUTPUT ) channelInfo.isInput = false;
+ else channelInfo.isInput = true;
+ result = ASIOGetChannelInfo( &channelInfo );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- // Now reopen in blocking mode.
- close(fd);
- if (mode == OUTPUT)
- fd = open(name, O_WRONLY | O_SYNC);
- else { // mode == INPUT
- if (stream->mode == OUTPUT && stream->device[0] == device)
- fd = open(name, O_RDWR | O_SYNC);
- else
- fd = open(name, O_RDONLY | O_SYNC);
+ // Assuming WINDOWS host is always little-endian.
+ stream_.doByteSwap[mode] = false;
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = 0;
+ if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
+ }
+ else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
+ }
+ else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
+ }
+ else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
+ if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
}
- if (fd == -1) {
- sprintf(message, "RtAudio: OSS device (%s) cannot be opened.", name);
- goto error;
+ if ( stream_.deviceFormat[mode] == 0 ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- // Get the sample format mask
- int mask;
- if (ioctl(fd, SNDCTL_DSP_GETFMTS, &mask) == -1) {
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) can't get supported audio formats.",
- name);
- goto error;
+ // Set the buffer size. For a duplex stream, this will end up
+ // setting the buffer size based on the input constraints, which
+ // should be ok.
+ long minSize, maxSize, preferSize, granularity;
+ result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- // Determine how to set the device format.
- stream->userFormat = format;
- device_format = -1;
- stream->doByteSwap[mode] = false;
- if (format == RTAUDIO_SINT8) {
- if (mask & AFMT_S8) {
- device_format = AFMT_S8;
- stream->deviceFormat[mode] = RTAUDIO_SINT8;
- }
- }
- else if (format == RTAUDIO_SINT16) {
- if (mask & AFMT_S16_NE) {
- device_format = AFMT_S16_NE;
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- }
-#if BYTE_ORDER == LITTLE_ENDIAN
- else if (mask & AFMT_S16_BE) {
- device_format = AFMT_S16_BE;
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- stream->doByteSwap[mode] = true;
- }
-#else
- else if (mask & AFMT_S16_LE) {
- device_format = AFMT_S16_LE;
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- stream->doByteSwap[mode] = true;
- }
-#endif
+ if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
+ else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
+ else if ( granularity == -1 ) {
+ // Make sure bufferSize is a power of two.
+ double power = std::log10( (double) *bufferSize ) / log10( 2.0 );
+ *bufferSize = (int) pow( 2.0, floor(power+0.5) );
+ if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
+ else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
+ else *bufferSize = preferSize;
}
-#if defined (AFMT_S32_NE) && defined (AFMT_S32_LE) && defined (AFMT_S32_BE)
- else if (format == RTAUDIO_SINT32) {
- if (mask & AFMT_S32_NE) {
- device_format = AFMT_S32_NE;
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- }
-#if BYTE_ORDER == LITTLE_ENDIAN
- else if (mask & AFMT_S32_BE) {
- device_format = AFMT_S32_BE;
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- stream->doByteSwap[mode] = true;
- }
-#else
- else if (mask & AFMT_S32_LE) {
- device_format = AFMT_S32_LE;
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- stream->doByteSwap[mode] = true;
- }
-#endif
+ else if ( granularity != 0 ) {
+ // Set to an even multiple of granularity, rounding up.
+ *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
}
-#endif
- if (device_format == -1) {
- // The user requested format is not natively supported by the device.
- if (mask & AFMT_S16_NE) {
- device_format = AFMT_S16_NE;
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- }
-#if BYTE_ORDER == LITTLE_ENDIAN
- else if (mask & AFMT_S16_BE) {
- device_format = AFMT_S16_BE;
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- stream->doByteSwap[mode] = true;
- }
-#else
- else if (mask & AFMT_S16_LE) {
- device_format = AFMT_S16_LE;
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- stream->doByteSwap[mode] = true;
- }
-#endif
-#if defined (AFMT_S32_NE) && defined (AFMT_S32_LE) && defined (AFMT_S32_BE)
- else if (mask & AFMT_S32_NE) {
- device_format = AFMT_S32_NE;
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- }
-#if BYTE_ORDER == LITTLE_ENDIAN
- else if (mask & AFMT_S32_BE) {
- device_format = AFMT_S32_BE;
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- stream->doByteSwap[mode] = true;
- }
-#else
- else if (mask & AFMT_S32_LE) {
- device_format = AFMT_S32_LE;
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- stream->doByteSwap[mode] = true;
- }
-#endif
-#endif
- else if (mask & AFMT_S8) {
- device_format = AFMT_S8;
- stream->deviceFormat[mode] = RTAUDIO_SINT8;
- }
+ if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) {
+ drivers.removeCurrentDriver();
+ errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
+ return FAILURE;
}
- if (stream->deviceFormat[mode] == 0) {
- // This really shouldn't happen ...
- close(fd);
- sprintf(message, "RtAudio: OSS device (%s) data format not supported by RtAudio.",
- name);
- goto error;
- }
+ stream_.bufferSize = *bufferSize;
+ stream_.nBuffers = 2;
- // Determine the number of channels for this device. Note that the
- // channel value requested by the user might be < min_X_Channels.
- stream->nUserChannels[mode] = channels;
- device_channels = channels;
- if (mode == OUTPUT) {
- if (channels < devices[device].minOutputChannels)
- device_channels = devices[device].minOutputChannels;
- }
- else { // mode == INPUT
- if (stream->mode == OUTPUT && stream->device[0] == device) {
- // We're doing duplex setup here.
- if (channels < devices[device].minDuplexChannels)
- device_channels = devices[device].minDuplexChannels;
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
+
+ // ASIO always uses non-interleaved buffers.
+ stream_.deviceInterleaved[mode] = false;
+
+ // Allocate, if necessary, our AsioHandle structure for the stream.
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ if ( handle == 0 ) {
+ try {
+ handle = new AsioHandle;
}
- else {
- if (channels < devices[device].minInputChannels)
- device_channels = devices[device].minInputChannels;
+ catch ( std::bad_alloc& ) {
+ //if ( handle == NULL ) {
+ drivers.removeCurrentDriver();
+ errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
+ return FAILURE;
}
- }
- stream->nDeviceChannels[mode] = device_channels;
+ handle->bufferInfos = 0;
- // Attempt to set the buffer size. According to OSS, the minimum
- // number of buffers is two. The supposed minimum buffer size is 16
- // bytes, so that will be our lower bound. The argument to this
- // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
- // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
- // We'll check the actual value used near the end of the setup
- // procedure.
- buffer_bytes = *bufferSize * formatBytes(stream->deviceFormat[mode]) * device_channels;
- if (buffer_bytes < 16) buffer_bytes = 16;
- buffers = numberOfBuffers;
- if (buffers < 2) buffers = 2;
- temp = ((int) buffers << 16) + (int)(log10((double)buffer_bytes)/log10(2.0));
- if (ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &temp)) {
- close(fd);
- sprintf(message, "RtAudio: OSS error setting fragment size for device (%s).",
- name);
- goto error;
+ // Create a manual-reset event.
+ handle->condition = CreateEvent( NULL, // no security
+ TRUE, // manual-reset
+ FALSE, // non-signaled initially
+ NULL ); // unnamed
+ stream_.apiHandle = (void *) handle;
}
- stream->nBuffers = buffers;
- // Set the data format.
- temp = device_format;
- if (ioctl(fd, SNDCTL_DSP_SETFMT, &device_format) == -1 || device_format != temp) {
- close(fd);
- sprintf(message, "RtAudio: OSS error setting data format for device (%s).",
- name);
- goto error;
+ // Create the ASIO internal buffers. Since RtAudio sets up input
+ // and output separately, we'll have to dispose of previously
+ // created output buffers for a duplex stream.
+ long inputLatency, outputLatency;
+ if ( mode == INPUT && stream_.mode == OUTPUT ) {
+ ASIODisposeBuffers();
+ if ( handle->bufferInfos ) free( handle->bufferInfos );
}
- // Set the number of channels.
- temp = device_channels;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &device_channels) == -1 || device_channels != temp) {
- close(fd);
- sprintf(message, "RtAudio: OSS error setting %d channels on device (%s).",
- temp, name);
+ // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
+ bool buffersAllocated = false;
+ unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+ handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
+ if ( handle->bufferInfos == NULL ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
goto error;
}
- // Set the sample rate.
- srate = sampleRate;
- temp = srate;
- if (ioctl(fd, SNDCTL_DSP_SPEED, &srate) == -1) {
- close(fd);
- sprintf(message, "RtAudio: OSS error setting sample rate = %d on device (%s).",
- temp, name);
- goto error;
+ ASIOBufferInfo *infos;
+ infos = handle->bufferInfos;
+ for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
+ infos->isInput = ASIOFalse;
+ infos->channelNum = i + stream_.channelOffset[0];
+ infos->buffers[0] = infos->buffers[1] = 0;
}
-
- // Verify the sample rate setup worked.
- if (abs(srate - temp) > 100) {
- close(fd);
- sprintf(message, "RtAudio: OSS error ... audio device (%s) doesn't support sample rate of %d.",
- name, temp);
- goto error;
+ for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
+ infos->isInput = ASIOTrue;
+ infos->channelNum = i + stream_.channelOffset[1];
+ infos->buffers[0] = infos->buffers[1] = 0;
}
- stream->sampleRate = sampleRate;
- if (ioctl(fd, SNDCTL_DSP_GETBLKSIZE, &buffer_bytes) == -1) {
- close(fd);
- sprintf(message, "RtAudio: OSS error getting buffer size for device (%s).",
- name);
+ // Set up the ASIO callback structure and create the ASIO data buffers.
+ asioCallbacks.bufferSwitch = &bufferSwitch;
+ asioCallbacks.sampleRateDidChange = &sampleRateChanged;
+ asioCallbacks.asioMessage = &asioMessages;
+ asioCallbacks.bufferSwitchTimeInfo = NULL;
+ result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
+ errorText_ = errorStream_.str();
goto error;
}
+ buffersAllocated = true;
- // Save buffer size (in sample frames).
- *bufferSize = buffer_bytes / (formatBytes(stream->deviceFormat[mode]) * device_channels);
- stream->bufferSize = *bufferSize;
-
- if (mode == INPUT && stream->mode == OUTPUT &&
- stream->device[0] == device) {
- // We're doing duplex setup here.
- stream->deviceFormat[0] = stream->deviceFormat[1];
- stream->nDeviceChannels[0] = device_channels;
- }
-
- // Set flags for buffer conversion
- stream->doConvertBuffer[mode] = false;
- if (stream->userFormat != stream->deviceFormat[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode])
- stream->doConvertBuffer[mode] = true;
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
// Allocate necessary internal buffers
- if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) {
-
- long buffer_bytes;
- if (stream->nUserChannels[0] >= stream->nUserChannels[1])
- buffer_bytes = stream->nUserChannels[0];
- else
- buffer_bytes = stream->nUserChannels[1];
-
- buffer_bytes *= *bufferSize * formatBytes(stream->userFormat);
- if (stream->userBuffer) free(stream->userBuffer);
- stream->userBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->userBuffer == NULL) {
- close(fd);
- sprintf(message, "RtAudio: OSS error allocating user buffer memory (%s).",
- name);
- goto error;
- }
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
}
- if ( stream->doConvertBuffer[mode] ) {
+ if ( stream_.doConvertBuffer[mode] ) {
- long buffer_bytes;
bool makeBuffer = true;
- if ( mode == OUTPUT )
- buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- else { // mode == INPUT
- buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]);
- if ( stream->mode == OUTPUT && stream->deviceBuffer ) {
- long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- if ( buffer_bytes < bytes_out ) makeBuffer = false;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
}
}
if ( makeBuffer ) {
- buffer_bytes *= *bufferSize;
- if (stream->deviceBuffer) free(stream->deviceBuffer);
- stream->deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->deviceBuffer == NULL) {
- close(fd);
- free(stream->userBuffer);
- sprintf(message, "RtAudio: OSS error allocating device buffer memory (%s).",
- name);
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
goto error;
}
}
}
- stream->device[mode] = device;
- stream->handle[mode] = fd;
- stream->state = STREAM_STOPPED;
- if ( stream->mode == OUTPUT && mode == INPUT ) {
- stream->mode = DUPLEX;
- if (stream->device[0] == device)
- stream->handle[0] = fd;
- }
+ stream_.sampleRate = sampleRate;
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ asioCallbackInfo = &stream_.callbackInfo;
+ stream_.callbackInfo.object = (void *) this;
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ // We had already set up an output stream.
+ stream_.mode = DUPLEX;
else
- stream->mode = mode;
+ stream_.mode = mode;
+
+ // Determine device latencies
+ result = ASIOGetLatencies( &inputLatency, &outputLatency );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING); // warn but don't fail
+ }
+ else {
+ stream_.latency[0] = outputLatency;
+ stream_.latency[1] = inputLatency;
+ }
+
+ // Setup the buffer conversion information structure. We don't use
+ // buffers to do channel offsets, so we override that parameter
+ // here.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
return SUCCESS;
error:
- if (stream->handle[0]) {
- close(stream->handle[0]);
- stream->handle[0] = 0;
+ if ( buffersAllocated )
+ ASIODisposeBuffers();
+ drivers.removeCurrentDriver();
+
+ if ( handle ) {
+ CloseHandle( handle->condition );
+ if ( handle->bufferInfos )
+ free( handle->bufferInfos );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
}
- error(RtError::WARNING);
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
return FAILURE;
}
-void RtAudio :: closeStream(int streamId)
+void RtApiAsio :: closeStream()
{
- // We don't want an exception to be thrown here because this
- // function is called by our class destructor. So, do our own
- // streamId check.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
- error(RtError::WARNING);
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
+ error( RtError::WARNING );
return;
}
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId];
-
- if (stream->callbackInfo.usingCallback) {
- pthread_cancel(stream->callbackInfo.thread);
- pthread_join(stream->callbackInfo.thread, NULL);
+ if ( stream_.state == STREAM_RUNNING ) {
+ stream_.state = STREAM_STOPPED;
+ ASIOStop();
}
+ ASIODisposeBuffers();
+ drivers.removeCurrentDriver();
- if (stream->state == STREAM_RUNNING) {
- if (stream->mode == OUTPUT || stream->mode == DUPLEX)
- ioctl(stream->handle[0], SNDCTL_DSP_RESET, 0);
- if (stream->mode == INPUT || stream->mode == DUPLEX)
- ioctl(stream->handle[1], SNDCTL_DSP_RESET, 0);
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ if ( handle ) {
+ CloseHandle( handle->condition );
+ if ( handle->bufferInfos )
+ free( handle->bufferInfos );
+ delete handle;
+ stream_.apiHandle = 0;
}
- pthread_mutex_destroy(&stream->mutex);
-
- if (stream->handle[0])
- close(stream->handle[0]);
-
- if (stream->handle[1])
- close(stream->handle[1]);
-
- if (stream->userBuffer)
- free(stream->userBuffer);
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
- if (stream->deviceBuffer)
- free(stream->deviceBuffer);
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
- free(stream);
- streams.erase(streamId);
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
}
-void RtAudio :: startStream(int streamId)
+void RtApiAsio :: startStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiAsio::startStream(): the stream is already running!";
+ error( RtError::WARNING );
+ return;
+ }
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK( &stream_.mutex );
- stream->state = STREAM_RUNNING;
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ ASIOError result = ASIOStart();
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::startStream: error (" << getAsioErrorString( result ) << ") starting device.";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
- // No need to do anything else here ... OSS automatically starts
- // when fed samples.
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ stream_.state = STREAM_RUNNING;
+ asioXRun = false;
+
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
- MUTEX_UNLOCK(&stream->mutex);
+ if ( result == ASE_OK ) return;
+ error( RtError::SYSTEM_ERROR );
}
-void RtAudio :: stopStream(int streamId)
+void RtApiAsio :: stopStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
+ }
- if (stream->state == STREAM_STOPPED)
- goto unlock;
+ MUTEX_LOCK( &stream_.mutex );
- int err;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- err = ioctl(stream->handle[0], SNDCTL_DSP_SYNC, 0);
- if (err < -1) {
- sprintf(message, "RtAudio: OSS error stopping device (%s).",
- devices[stream->device[0]].name);
- error(RtError::DRIVER_ERROR);
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 1;
+ MUTEX_UNLOCK( &stream_.mutex );
+ WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled
+ ResetEvent( handle->condition );
+ MUTEX_LOCK( &stream_.mutex );
}
}
- else {
- err = ioctl(stream->handle[1], SNDCTL_DSP_SYNC, 0);
- if (err < -1) {
- sprintf(message, "RtAudio: OSS error stopping device (%s).",
- devices[stream->device[1]].name);
- error(RtError::DRIVER_ERROR);
- }
+
+ ASIOError result = ASIOStop();
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
+ errorText_ = errorStream_.str();
}
- stream->state = STREAM_STOPPED;
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result == ASE_OK ) return;
+ error( RtError::SYSTEM_ERROR );
}
-void RtAudio :: abortStream(int streamId)
+void RtApiAsio :: abortStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- int err;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- err = ioctl(stream->handle[0], SNDCTL_DSP_RESET, 0);
- if (err < -1) {
- sprintf(message, "RtAudio: OSS error aborting device (%s).",
- devices[stream->device[0]].name);
- error(RtError::DRIVER_ERROR);
- }
- }
- else {
- err = ioctl(stream->handle[1], SNDCTL_DSP_RESET, 0);
- if (err < -1) {
- sprintf(message, "RtAudio: OSS error aborting device (%s).",
- devices[stream->device[1]].name);
- error(RtError::DRIVER_ERROR);
- }
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
}
- stream->state = STREAM_STOPPED;
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ // The following lines were commented-out because some behavior was
+ // noted where the device buffers need to be zeroed to avoid
+ // continuing sound, even when the device buffers are completed
+ // disposed. So now, calling abort is the same as calling stop.
+ //AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ //handle->drainCounter = 1;
+ stopStream();
}
-int RtAudio :: streamWillBlock(int streamId)
+bool RtApiAsio :: callbackEvent( long bufferIndex )
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- int bytes = 0, channels = 0, frames = 0;
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- audio_buf_info info;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- ioctl(stream->handle[0], SNDCTL_DSP_GETOSPACE, &info);
- bytes = info.bytes;
- channels = stream->nDeviceChannels[0];
- }
-
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- ioctl(stream->handle[1], SNDCTL_DSP_GETISPACE, &info);
- if (stream->mode == DUPLEX ) {
- bytes = (bytes < info.bytes) ? bytes : info.bytes;
- channels = stream->nDeviceChannels[0];
- }
- else {
- bytes = info.bytes;
- channels = stream->nDeviceChannels[1];
- }
+ if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtError::WARNING );
+ return FAILURE;
}
- frames = (int) (bytes / (channels * formatBytes(stream->deviceFormat[0])));
- frames -= stream->bufferSize;
- if (frames < 0) frames = 0;
-
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
- return frames;
-}
-
-void RtAudio :: tickStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
- int stopStream = 0;
- if (stream->state == STREAM_STOPPED) {
- if (stream->callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds
- return;
- }
- else if (stream->callbackInfo.usingCallback) {
- RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback;
- stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData);
+ // Check if we were draining the stream and signal is finished.
+ if ( handle->drainCounter > 3 ) {
+ if ( handle->internalDrain == false )
+ SetEvent( handle->condition );
+ else
+ stopStream();
+ return SUCCESS;
}
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK( &stream_.mutex );
// The state might change while waiting on a mutex.
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- int result;
- char *buffer;
- int samples;
- RTAUDIO_FORMAT format;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
+ if ( stream_.state == STREAM_STOPPED ) goto unlock;
- // Setup parameters and do buffer conversion if necessary.
- if (stream->doConvertBuffer[0]) {
- convertStreamBuffer(stream, OUTPUT);
- buffer = stream->deviceBuffer;
- samples = stream->bufferSize * stream->nDeviceChannels[0];
- format = stream->deviceFormat[0];
+ // Invoke user callback to get fresh output data UNLESS we are
+ // draining stream.
+ if ( handle->drainCounter == 0 ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && asioXRun == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ asioXRun = false;
}
- else {
- buffer = stream->userBuffer;
- samples = stream->bufferSize * stream->nUserChannels[0];
- format = stream->userFormat;
+ if ( stream_.mode != OUTPUT && asioXRun == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ asioXRun = false;
}
-
- // Do byte swapping if necessary.
- if (stream->doByteSwap[0])
- byteSwapBuffer(buffer, samples, format);
-
- // Write samples to device.
- result = write(stream->handle[0], buffer, samples * formatBytes(format));
-
- if (result == -1) {
- // This could be an underrun, but the basic OSS API doesn't provide a means for determining that.
- sprintf(message, "RtAudio: OSS audio write error for device (%s).",
- devices[stream->device[0]].name);
- error(RtError::DRIVER_ERROR);
+ handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( handle->drainCounter == 2 ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ abortStream();
+ return SUCCESS;
}
+ else if ( handle->drainCounter == 1 )
+ handle->internalDrain = true;
}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
-
- // Setup parameters.
- if (stream->doConvertBuffer[1]) {
- buffer = stream->deviceBuffer;
- samples = stream->bufferSize * stream->nDeviceChannels[1];
- format = stream->deviceFormat[1];
- }
- else {
- buffer = stream->userBuffer;
- samples = stream->bufferSize * stream->nUserChannels[1];
- format = stream->userFormat;
- }
-
- // Read samples from device.
- result = read(stream->handle[1], buffer, samples * formatBytes(format));
-
- if (result == -1) {
- // This could be an overrun, but the basic OSS API doesn't provide a means for determining that.
- sprintf(message, "RtAudio: OSS audio read error for device (%s).",
- devices[stream->device[1]].name);
- error(RtError::DRIVER_ERROR);
- }
-
- // Do byte swapping if necessary.
- if (stream->doByteSwap[1])
- byteSwapBuffer(buffer, samples, format);
-
- // Do buffer conversion if necessary.
- if (stream->doConvertBuffer[1])
- convertStreamBuffer(stream, INPUT);
- }
+ unsigned int nChannels, bufferBytes, i, j;
+ nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
- if (stream->callbackInfo.usingCallback && stopStream)
- this->stopStream(streamId);
-}
+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
-extern "C" void *callbackHandler(void *ptr)
-{
- CALLBACK_INFO *info = (CALLBACK_INFO *) ptr;
- RtAudio *object = (RtAudio *) info->object;
- int stream = info->streamId;
- bool *usingCallback = &info->usingCallback;
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput != ASIOTrue )
+ memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
+ }
- while ( *usingCallback ) {
- pthread_testcancel();
- try {
- object->tickStream(stream);
- }
- catch (RtError &exception) {
- fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n",
- exception.getMessage());
- break;
}
- }
-
- return 0;
-}
-
-
-//******************** End of __LINUX_OSS__ *********************//
-
-#elif defined(__WINDOWS_ASIO__) // ASIO API on Windows
-
-// The ASIO API is designed around a callback scheme, so this
-// implementation is similar to that used for OS X CoreAudio. The
-// primary constraint with ASIO is that it only allows access to a
-// single driver at a time. Thus, it is not possible to have more
-// than one simultaneous RtAudio stream.
-//
-// This implementation also requires a number of external ASIO files
-// and a few global variables. The ASIO callback scheme does not
-// allow for the passing of user data, so we must create a global
-// pointer to our callbackInfo structure.
+ else if ( stream_.doConvertBuffer[0] ) {
-#include "asio/asiosys.h"
-#include "asio/asio.h"
-#include "asio/asiodrivers.h"
-#include <math.h>
+ convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer( stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nDeviceChannels[0],
+ stream_.deviceFormat[0] );
-AsioDrivers drivers;
-ASIOCallbacks asioCallbacks;
-CALLBACK_INFO *asioCallbackInfo;
-ASIODriverInfo driverInfo;
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput != ASIOTrue )
+ memcpy( handle->bufferInfos[i].buffers[bufferIndex],
+ &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
+ }
-void RtAudio :: initialize(void)
-{
- nDevices = drivers.asioGetNumDev();
- if (nDevices <= 0) return;
-
- // Allocate the RTAUDIO_DEVICE structures.
- devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE));
- if (devices == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
- }
-
- // Write device driver names to device structures and then probe the
- // device capabilities.
- for (int i=0; i<nDevices; i++) {
- if ( drivers.asioGetDriverName( i, devices[i].name, 128 ) == 0 )
- //probeDeviceInfo(&devices[i]);
- ;
- else {
- sprintf(message, "RtAudio: error getting ASIO driver name for device index %d!", i);
- error(RtError::WARNING);
}
- }
-
- drivers.removeCurrentDriver();
- driverInfo.asioVersion = 2;
- // See note in DirectSound implementation about GetDesktopWindow().
- driverInfo.sysRef = GetForegroundWindow();
-}
-
-int RtAudio :: getDefaultInputDevice(void)
-{
- return 0;
-}
-
-int RtAudio :: getDefaultOutputDevice(void)
-{
- return 0;
-}
+ else {
-void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info)
-{
- // Don't probe if a stream is already open.
- if ( streams.size() > 0 ) {
- sprintf(message, "RtAudio: unable to probe ASIO driver while a stream is open.");
- error(RtError::DEBUG_WARNING);
- return;
- }
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer( stream_.userBuffer[0],
+ stream_.bufferSize * stream_.nUserChannels[0],
+ stream_.userFormat );
- if ( !drivers.loadDriver( info->name ) ) {
- sprintf(message, "RtAudio: ASIO error loading driver (%s).", info->name);
- error(RtError::DEBUG_WARNING);
- return;
- }
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput != ASIOTrue )
+ memcpy( handle->bufferInfos[i].buffers[bufferIndex],
+ &stream_.userBuffer[0][bufferBytes*j++], bufferBytes );
+ }
- ASIOError result = ASIOInit( &driverInfo );
- if ( result != ASE_OK ) {
- char details[32];
- if ( result == ASE_HWMalfunction )
- sprintf(details, "hardware malfunction");
- else if ( result == ASE_NoMemory )
- sprintf(details, "no memory");
- else if ( result == ASE_NotPresent )
- sprintf(details, "driver/hardware not present");
- else
- sprintf(details, "unspecified");
- sprintf(message, "RtAudio: ASIO error (%s) initializing driver (%s).", details, info->name);
- error(RtError::DEBUG_WARNING);
- return;
- }
+ }
- // Determine the device channel information.
- long inputChannels, outputChannels;
- result = ASIOGetChannels( &inputChannels, &outputChannels );
- if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO error getting input/output channel count (%s).", info->name);
- error(RtError::DEBUG_WARNING);
- return;
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
+ }
}
- info->maxOutputChannels = outputChannels;
- if ( outputChannels > 0 ) info->minOutputChannels = 1;
-
- info->maxInputChannels = inputChannels;
- if ( inputChannels > 0 ) info->minInputChannels = 1;
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
- // If device opens for both playback and capture, we determine the channels.
- if (info->maxOutputChannels > 0 && info->maxInputChannels > 0) {
- info->hasDuplexSupport = true;
- info->maxDuplexChannels = (info->maxOutputChannels > info->maxInputChannels) ?
- info->maxInputChannels : info->maxOutputChannels;
- info->minDuplexChannels = (info->minOutputChannels > info->minInputChannels) ?
- info->minInputChannels : info->minOutputChannels;
- }
+ bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
- // Determine the supported sample rates.
- info->nSampleRates = 0;
- for (int i=0; i<MAX_SAMPLE_RATES; i++) {
- result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
- if ( result == ASE_OK )
- info->sampleRates[info->nSampleRates++] = SAMPLE_RATES[i];
- }
+ if (stream_.doConvertBuffer[1]) {
- if (info->nSampleRates == 0) {
- drivers.removeCurrentDriver();
- sprintf( message, "RtAudio: No supported sample rates found for ASIO driver (%s).", info->name );
- error(RtError::DEBUG_WARNING);
- return;
- }
+ // Always interleave ASIO input data.
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput == ASIOTrue )
+ memcpy( &stream_.deviceBuffer[j++*bufferBytes],
+ handle->bufferInfos[i].buffers[bufferIndex],
+ bufferBytes );
+ }
- // Determine supported data types ... just check first channel and assume rest are the same.
- ASIOChannelInfo channelInfo;
- channelInfo.channel = 0;
- channelInfo.isInput = true;
- if ( info->maxInputChannels <= 0 ) channelInfo.isInput = false;
- result = ASIOGetChannelInfo( &channelInfo );
- if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO error getting driver (%s) channel information.", info->name);
- error(RtError::DEBUG_WARNING);
- return;
- }
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nDeviceChannels[1],
+ stream_.deviceFormat[1] );
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
- if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
- info->nativeFormats |= RTAUDIO_SINT16;
- else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
- info->nativeFormats |= RTAUDIO_SINT32;
- else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
- info->nativeFormats |= RTAUDIO_FLOAT32;
- else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
- info->nativeFormats |= RTAUDIO_FLOAT64;
+ }
+ else {
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
+ memcpy( &stream_.userBuffer[1][bufferBytes*j++],
+ handle->bufferInfos[i].buffers[bufferIndex],
+ bufferBytes );
+ }
+ }
- // Check that we have at least one supported format.
- if (info->nativeFormats == 0) {
- drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO driver (%s) data format not supported by RtAudio.",
- info->name);
- error(RtError::DEBUG_WARNING);
- return;
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( stream_.userBuffer[1],
+ stream_.bufferSize * stream_.nUserChannels[1],
+ stream_.userFormat );
+ }
}
- info->probed = true;
- drivers.removeCurrentDriver();
-}
+ unlock:
+ // The following call was suggested by Malte Clasen. While the API
+ // documentation indicates it should not be required, some device
+ // drivers apparently do not function correctly without it.
+ ASIOOutputReady();
-void bufferSwitch(long index, ASIOBool processNow)
-{
- RtAudio *object = (RtAudio *) asioCallbackInfo->object;
- try {
- object->callbackEvent( asioCallbackInfo->streamId, index, (void *)NULL, (void *)NULL );
- }
- catch (RtError &exception) {
- fprintf(stderr, "\nCallback handler error (%s)!\n\n", exception.getMessage());
- return;
- }
+ MUTEX_UNLOCK( &stream_.mutex );
- return;
+ RtApi::tickStreamTime();
+ return SUCCESS;
}
-void sampleRateChanged(ASIOSampleRate sRate)
+void sampleRateChanged( ASIOSampleRate sRate )
{
// The ASIO documentation says that this usually only happens during
// external sync. Audio processing is not stopped by the driver,
// sample rate status of an AES/EBU or S/PDIF digital input at the
// audio device.
- RtAudio *object = (RtAudio *) asioCallbackInfo->object;
+ RtApi *object = (RtApi *) asioCallbackInfo->object;
try {
- object->stopStream( asioCallbackInfo->streamId );
+ object->stopStream();
}
- catch (RtError &exception) {
- fprintf(stderr, "\nRtAudio: sampleRateChanged() error (%s)!\n\n", exception.getMessage());
+ catch ( RtError &exception ) {
+ std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
return;
}
- fprintf(stderr, "\nRtAudio: ASIO driver reports sample rate changed to %d ... stream stopped!!!", (int) sRate);
+ std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
}
-long asioMessages(long selector, long value, void* message, double* opt)
+long asioMessages( long selector, long value, void* message, double* opt )
{
long ret = 0;
- switch(selector) {
+
+ switch( selector ) {
case kAsioSelectorSupported:
- if(value == kAsioResetRequest
- || value == kAsioEngineVersion
- || value == kAsioResyncRequest
- || value == kAsioLatenciesChanged
- // The following three were added for ASIO 2.0, you don't
- // necessarily have to support them.
- || value == kAsioSupportsTimeInfo
- || value == kAsioSupportsTimeCode
- || value == kAsioSupportsInputMonitor)
+ if ( value == kAsioResetRequest
+ || value == kAsioEngineVersion
+ || value == kAsioResyncRequest
+ || value == kAsioLatenciesChanged
+ // The following three were added for ASIO 2.0, you don't
+ // necessarily have to support them.
+ || value == kAsioSupportsTimeInfo
+ || value == kAsioSupportsTimeCode
+ || value == kAsioSupportsInputMonitor)
ret = 1L;
break;
case kAsioResetRequest:
// done by completely destruct is. I.e. ASIOStop(),
// ASIODisposeBuffers(), Destruction Afterwards you initialize the
// driver again.
- fprintf(stderr, "\nRtAudio: ASIO driver reset requested!!!");
+ std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
ret = 1L;
break;
case kAsioResyncRequest:
// which could lose data because the Mutex was held too long by
// another thread. However a driver can issue it in other
// situations, too.
- fprintf(stderr, "\nRtAudio: ASIO driver resync requested!!!");
+ // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
+ asioXRun = true;
ret = 1L;
break;
case kAsioLatenciesChanged:
// latencies changed. Beware, it this does not mean that the
// buffer sizes have changed! You might need to update internal
// delay data.
- fprintf(stderr, "\nRtAudio: ASIO driver latency may have changed!!!");
+ std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
ret = 1L;
break;
case kAsioEngineVersion:
ret = 0;
break;
case kAsioSupportsTimeCode:
- // Informs the driver wether application is interested in time
+ // Informs the driver whether application is interested in time
// code info. If an application does not need to know about time
// code, the driver has less work to do.
ret = 0;
return ret;
}
-bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream,
- STREAM_MODE mode, int channels,
- int sampleRate, RTAUDIO_FORMAT format,
- int *bufferSize, int numberOfBuffers)
+static const char* getAsioErrorString( ASIOError result )
{
- // Don't attempt to load another driver if a stream is already open.
- if ( streams.size() > 0 ) {
- sprintf(message, "RtAudio: unable to load ASIO driver while a stream is open.");
- error(RtError::WARNING);
- return FAILURE;
- }
+ struct Messages
+ {
+ ASIOError value;
+ const char*message;
+ };
+
+ static Messages m[] =
+ {
+ { ASE_NotPresent, "Hardware input or output is not present or available." },
+ { ASE_HWMalfunction, "Hardware is malfunctioning." },
+ { ASE_InvalidParameter, "Invalid input parameter." },
+ { ASE_InvalidMode, "Invalid mode." },
+ { ASE_SPNotAdvancing, "Sample position not advancing." },
+ { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },
+ { ASE_NoMemory, "Not enough memory to complete the request." }
+ };
+
+ for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
+ if ( m[i].value == result ) return m[i].message;
+
+ return "Unknown error.";
+}
+//******************** End of __WINDOWS_ASIO__ *********************//
+#endif
- // For ASIO, a duplex stream MUST use the same driver.
- if ( mode == INPUT && stream->mode == OUTPUT && stream->device[0] != device ) {
- sprintf(message, "RtAudio: ASIO duplex stream must use the same device for input and output.");
- error(RtError::WARNING);
- return FAILURE;
- }
- // Only load the driver once for duplex stream.
- ASIOError result;
- if ( mode != INPUT || stream->mode != OUTPUT ) {
- if ( !drivers.loadDriver( devices[device].name ) ) {
- sprintf(message, "RtAudio: ASIO error loading driver (%s).", devices[device].name);
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
+#if defined(__WINDOWS_DS__) // Windows DirectSound API
- result = ASIOInit( &driverInfo );
- if ( result != ASE_OK ) {
- char details[32];
- if ( result == ASE_HWMalfunction )
- sprintf(details, "hardware malfunction");
- else if ( result == ASE_NoMemory )
- sprintf(details, "no memory");
- else if ( result == ASE_NotPresent )
- sprintf(details, "driver/hardware not present");
- else
- sprintf(details, "unspecified");
- sprintf(message, "RtAudio: ASIO error (%s) initializing driver (%s).", details, devices[device].name);
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
- }
+// Modified by Robin Davies, October 2005
+// - Improvements to DirectX pointer chasing.
+// - Backdoor RtDsStatistics hook provides DirectX performance information.
+// - 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
- // Check the device channel count.
- long inputChannels, outputChannels;
- result = ASIOGetChannels( &inputChannels, &outputChannels );
- if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO error getting input/output channel count (%s).",
- devices[device].name);
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
+#include <dsound.h>
+#include <assert.h>
+
+#if defined(__MINGW32__)
+// missing from latest mingw winapi
+#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
+#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
+#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
+#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
+#endif
- if ( ( mode == OUTPUT && channels > outputChannels) ||
- ( mode == INPUT && channels > inputChannels) ) {
- drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO driver (%s) does not support requested channel count (%d).",
- devices[device].name, channels);
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
- stream->nDeviceChannels[mode] = channels;
- stream->nUserChannels[mode] = channels;
+#define MINIMUM_DEVICE_BUFFER_SIZE 32768
- // Verify the sample rate is supported.
- result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
- if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO driver (%s) does not support requested sample rate (%d).",
- devices[device].name, sampleRate);
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
+#ifdef _MSC_VER // if Microsoft Visual C++
+#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
+#endif
- // Set the sample rate.
- result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
- if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO driver (%s) error setting sample rate (%d).",
- devices[device].name, sampleRate);
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
+static inline DWORD dsPointerDifference( DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
+{
+ if (laterPointer > earlierPointer)
+ return laterPointer - earlierPointer;
+ else
+ return laterPointer - earlierPointer + bufferSize;
+}
- // Determine the driver data type.
- ASIOChannelInfo channelInfo;
- channelInfo.channel = 0;
- if ( mode == OUTPUT ) channelInfo.isInput = false;
- else channelInfo.isInput = true;
- result = ASIOGetChannelInfo( &channelInfo );
- if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO driver (%s) error getting data format.",
- devices[device].name);
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
+static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
+{
+ if ( pointer > bufferSize ) pointer -= bufferSize;
+ if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
+ if ( pointer < earlierPointer ) pointer += bufferSize;
+ return pointer >= earlierPointer && pointer < laterPointer;
+}
- // Assuming WINDOWS host is always little-endian.
- stream->doByteSwap[mode] = false;
- stream->userFormat = format;
- stream->deviceFormat[mode] = 0;
- if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- if ( channelInfo.type == ASIOSTInt16MSB ) stream->doByteSwap[mode] = true;
- }
- else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
- stream->deviceFormat[mode] = RTAUDIO_SINT32;
- if ( channelInfo.type == ASIOSTInt32MSB ) stream->doByteSwap[mode] = true;
- }
- else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
- stream->deviceFormat[mode] = RTAUDIO_FLOAT32;
- if ( channelInfo.type == ASIOSTFloat32MSB ) stream->doByteSwap[mode] = true;
- }
- else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
- stream->deviceFormat[mode] = RTAUDIO_FLOAT64;
- if ( channelInfo.type == ASIOSTFloat64MSB ) stream->doByteSwap[mode] = true;
- }
+// A structure to hold various information related to the DirectSound
+// API implementation.
+struct DsHandle {
+ unsigned int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
+ void *id[2];
+ void *buffer[2];
+ bool xrun[2];
+ UINT bufferPointer[2];
+ DWORD dsBufferSize[2];
+ DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
+ HANDLE condition;
+
+ DsHandle()
+ :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; }
+};
- if ( stream->deviceFormat[mode] == 0 ) {
- drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO driver (%s) data format not supported by RtAudio.",
- devices[device].name);
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
+/*
+RtApiDs::RtDsStatistics RtApiDs::statistics;
- // Set the buffer size. For a duplex stream, this will end up
- // setting the buffer size based on the input constraints, which
- // should be ok.
- long minSize, maxSize, preferSize, granularity;
- result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
- if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO driver (%s) error getting buffer size.",
- devices[device].name);
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
+// Provides a backdoor hook to monitor for DirectSound read overruns and write underruns.
+RtApiDs::RtDsStatistics RtApiDs::getDsStatistics()
+{
+ RtDsStatistics s = statistics;
- if ( *bufferSize < minSize ) *bufferSize = minSize;
- else if ( *bufferSize > maxSize ) *bufferSize = maxSize;
- else if ( granularity == -1 ) {
- // Make sure bufferSize is a power of two.
- double power = log10( *bufferSize ) / log10( 2.0 );
- *bufferSize = pow( 2.0, floor(power+0.5) );
- if ( *bufferSize < minSize ) *bufferSize = minSize;
- else if ( *bufferSize > maxSize ) *bufferSize = maxSize;
- else *bufferSize = preferSize;
- }
+ // update the calculated fields.
+ if ( s.inputFrameSize != 0 )
+ s.latency += s.readDeviceSafeLeadBytes * 1.0 / s.inputFrameSize / s.sampleRate;
- if ( mode == INPUT && stream->mode == OUTPUT && stream->bufferSize != *bufferSize )
- cout << "possible input/output buffersize discrepancy" << endl;
+ if ( s.outputFrameSize != 0 )
+ s.latency += (s.writeDeviceSafeLeadBytes + s.writeDeviceBufferLeadBytes) * 1.0 / s.outputFrameSize / s.sampleRate;
- stream->bufferSize = *bufferSize;
- stream->nBuffers = 2;
+ return s;
+}
+*/
- // ASIO always uses deinterleaved channels.
- stream->deInterleave[mode] = true;
+// Declarations for utility functions, callbacks, and structures
+// specific to the DirectSound implementation.
+static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
+ LPCTSTR description,
+ LPCTSTR module,
+ LPVOID lpContext );
- // Create the ASIO internal buffers. Since RtAudio sets up input
- // and output separately, we'll have to dispose of previously
- // created output buffers for a duplex stream.
- if ( mode == INPUT && stream->mode == OUTPUT ) {
- free(stream->callbackInfo.buffers);
- result = ASIODisposeBuffers();
- if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO driver (%s) error disposing previously allocated buffers.",
- devices[device].name);
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
- }
+static char* getErrorString( int code );
- // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
- int i, nChannels = stream->nDeviceChannels[0] + stream->nDeviceChannels[1];
- stream->callbackInfo.buffers = 0;
- ASIOBufferInfo *bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
- stream->callbackInfo.buffers = (void *) bufferInfos;
- ASIOBufferInfo *infos = bufferInfos;
- for ( i=0; i<stream->nDeviceChannels[1]; i++, infos++ ) {
- infos->isInput = ASIOTrue;
- infos->channelNum = i;
- infos->buffers[0] = infos->buffers[1] = 0;
- }
+extern "C" unsigned __stdcall callbackHandler( void *ptr );
- for ( i=0; i<stream->nDeviceChannels[0]; i++, infos++ ) {
- infos->isInput = ASIOFalse;
- infos->channelNum = i;
- infos->buffers[0] = infos->buffers[1] = 0;
- }
+struct EnumInfo {
+ bool isInput;
+ bool getDefault;
+ bool findIndex;
+ unsigned int counter;
+ unsigned int index;
+ LPGUID id;
+ std::string name;
- // Set up the ASIO callback structure and create the ASIO data buffers.
- asioCallbacks.bufferSwitch = &bufferSwitch;
- asioCallbacks.sampleRateDidChange = &sampleRateChanged;
- asioCallbacks.asioMessage = &asioMessages;
- asioCallbacks.bufferSwitchTimeInfo = NULL;
- result = ASIOCreateBuffers( bufferInfos, nChannels, stream->bufferSize, &asioCallbacks);
- if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- sprintf(message, "RtAudio: ASIO driver (%s) error creating buffers.",
- devices[device].name);
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
-
- // Set flags for buffer conversion.
- stream->doConvertBuffer[mode] = false;
- if (stream->userFormat != stream->deviceFormat[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] > 1 && stream->deInterleave[mode])
- stream->doConvertBuffer[mode] = true;
-
- // Allocate necessary internal buffers
- if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) {
-
- long buffer_bytes;
- if (stream->nUserChannels[0] >= stream->nUserChannels[1])
- buffer_bytes = stream->nUserChannels[0];
- else
- buffer_bytes = stream->nUserChannels[1];
-
- buffer_bytes *= *bufferSize * formatBytes(stream->userFormat);
- if (stream->userBuffer) free(stream->userBuffer);
- stream->userBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->userBuffer == NULL)
- goto memory_error;
- }
-
- if ( stream->doConvertBuffer[mode] ) {
-
- long buffer_bytes;
- bool makeBuffer = true;
- if ( mode == OUTPUT )
- buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- else { // mode == INPUT
- buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]);
- if ( stream->mode == OUTPUT && stream->deviceBuffer ) {
- long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- if ( buffer_bytes < bytes_out ) makeBuffer = false;
- }
- }
-
- if ( makeBuffer ) {
- buffer_bytes *= *bufferSize;
- if (stream->deviceBuffer) free(stream->deviceBuffer);
- stream->deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->deviceBuffer == NULL)
- goto memory_error;
- }
- }
-
- stream->device[mode] = device;
- stream->state = STREAM_STOPPED;
- if ( stream->mode == OUTPUT && mode == INPUT )
- // We had already set up an output stream.
- stream->mode = DUPLEX;
- else
- stream->mode = mode;
- stream->sampleRate = sampleRate;
- asioCallbackInfo = &stream->callbackInfo;
- stream->callbackInfo.object = (void *) this;
- stream->callbackInfo.waitTime = (unsigned long) (200.0 * stream->bufferSize / stream->sampleRate);
-
- return SUCCESS;
-
- memory_error:
- ASIODisposeBuffers();
- drivers.removeCurrentDriver();
-
- if (stream->callbackInfo.buffers)
- free(stream->callbackInfo.buffers);
- stream->callbackInfo.buffers = 0;
-
- if (stream->userBuffer) {
- free(stream->userBuffer);
- stream->userBuffer = 0;
- }
- sprintf(message, "RtAudio: error allocating buffer memory (%s).",
- devices[device].name);
- error(RtError::WARNING);
- return FAILURE;
-}
-
-void RtAudio :: cancelStreamCallback(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- if (stream->callbackInfo.usingCallback) {
-
- if (stream->state == STREAM_RUNNING)
- stopStream( streamId );
-
- MUTEX_LOCK(&stream->mutex);
-
- stream->callbackInfo.usingCallback = false;
- stream->callbackInfo.userData = NULL;
- stream->state = STREAM_STOPPED;
- stream->callbackInfo.callback = NULL;
-
- MUTEX_UNLOCK(&stream->mutex);
- }
-}
-
-void RtAudio :: closeStream(int streamId)
-{
- // We don't want an exception to be thrown here because this
- // function is called by our class destructor. So, do our own
- // streamId check.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
- error(RtError::WARNING);
- return;
- }
-
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId];
-
- if (stream->state == STREAM_RUNNING)
- ASIOStop();
-
- ASIODisposeBuffers();
- //ASIOExit();
- drivers.removeCurrentDriver();
-
- DeleteCriticalSection(&stream->mutex);
-
- if (stream->callbackInfo.buffers)
- free(stream->callbackInfo.buffers);
-
- if (stream->userBuffer)
- free(stream->userBuffer);
-
- if (stream->deviceBuffer)
- free(stream->deviceBuffer);
-
- free(stream);
- streams.erase(streamId);
-}
-
-void RtAudio :: startStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_RUNNING) {
- MUTEX_UNLOCK(&stream->mutex);
- return;
- }
-
- stream->callbackInfo.blockTick = true;
- stream->callbackInfo.stopStream = false;
- stream->callbackInfo.streamId = streamId;
- ASIOError result = ASIOStart();
- if ( result != ASE_OK ) {
- sprintf(message, "RtAudio: ASIO error starting device (%s).",
- devices[stream->device[0]].name);
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- stream->state = STREAM_RUNNING;
-
- MUTEX_UNLOCK(&stream->mutex);
-}
-
-void RtAudio :: stopStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- if (stream->state == STREAM_STOPPED) {
- MUTEX_UNLOCK(&stream->mutex);
- return;
- }
-
- ASIOError result = ASIOStop();
- if ( result != ASE_OK ) {
- sprintf(message, "RtAudio: ASIO error stopping device (%s).",
- devices[stream->device[0]].name);
- MUTEX_UNLOCK(&stream->mutex);
- error(RtError::DRIVER_ERROR);
- }
- stream->state = STREAM_STOPPED;
-
- MUTEX_UNLOCK(&stream->mutex);
-}
-
-void RtAudio :: abortStream(int streamId)
-{
- stopStream( streamId );
-}
+ EnumInfo()
+ : isInput(false), getDefault(false), findIndex(false), counter(0), index(0) {}
+};
-// I don't know how this function can be implemented.
-int RtAudio :: streamWillBlock(int streamId)
+RtApiDs :: RtApiDs()
{
- sprintf(message, "RtAudio: streamWillBlock() cannot be implemented for ASIO.");
- error(RtError::WARNING);
- return 0;
+ // Dsound will run both-threaded. If CoInitialize fails, then just
+ // accept whatever the mainline chose for a threading model.
+ coInitialized_ = false;
+ HRESULT hr = CoInitialize( NULL );
+ if ( !FAILED( hr ) ) coInitialized_ = true;
}
-void RtAudio :: tickStream(int streamId)
+RtApiDs :: ~RtApiDs()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- if (stream->state == STREAM_STOPPED)
- return;
-
- if (stream->callbackInfo.usingCallback) {
- sprintf(message, "RtAudio: tickStream() should not be used when a callback function is set!");
- error(RtError::WARNING);
- return;
- }
-
- // Block waiting here until the user data is processed in callbackEvent().
- while ( stream->callbackInfo.blockTick )
- Sleep(stream->callbackInfo.waitTime);
-
- MUTEX_LOCK(&stream->mutex);
-
- stream->callbackInfo.blockTick = true;
-
- MUTEX_UNLOCK(&stream->mutex);
+ if ( coInitialized_ ) CoUninitialize(); // balanced call.
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
}
-void RtAudio :: callbackEvent(int streamId, int bufferIndex, void *inData, void *outData)
+unsigned int RtApiDs :: getDefaultInputDevice( void )
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- CALLBACK_INFO *info = asioCallbackInfo;
- if ( !info->usingCallback ) {
- // Block waiting here until we get new user data in tickStream().
- while ( !info->blockTick )
- Sleep(info->waitTime);
- }
- else if ( info->stopStream ) {
- // Check if the stream should be stopped (via the previous user
- // callback return value). We stop the stream here, rather than
- // after the function call, so that output data can first be
- // processed.
- this->stopStream(asioCallbackInfo->streamId);
- return;
- }
-
- MUTEX_LOCK(&stream->mutex);
- int nChannels = stream->nDeviceChannels[0] + stream->nDeviceChannels[1];
- int bufferBytes;
- ASIOBufferInfo *bufferInfos = (ASIOBufferInfo *) info->buffers;
- if ( stream->mode == INPUT || stream->mode == DUPLEX ) {
-
- bufferBytes = stream->bufferSize * formatBytes(stream->deviceFormat[1]);
- if (stream->doConvertBuffer[1]) {
-
- // Always interleave ASIO input data.
- for ( int i=0; i<stream->nDeviceChannels[1]; i++, bufferInfos++ )
- memcpy(&stream->deviceBuffer[i*bufferBytes], bufferInfos->buffers[bufferIndex], bufferBytes );
-
- if ( stream->doByteSwap[1] )
- byteSwapBuffer(stream->deviceBuffer,
- stream->bufferSize * stream->nDeviceChannels[1],
- stream->deviceFormat[1]);
- convertStreamBuffer(stream, INPUT);
-
- }
- else { // single channel only
- memcpy(stream->userBuffer, bufferInfos->buffers[bufferIndex], bufferBytes );
-
- if (stream->doByteSwap[1])
- byteSwapBuffer(stream->userBuffer,
- stream->bufferSize * stream->nUserChannels[1],
- stream->userFormat);
- }
- }
-
- if ( info->usingCallback ) {
- RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) info->callback;
- if ( callback(stream->userBuffer, stream->bufferSize, info->userData) )
- info->stopStream = true;
- }
-
- if ( stream->mode == OUTPUT || stream->mode == DUPLEX ) {
-
- bufferBytes = stream->bufferSize * formatBytes(stream->deviceFormat[0]);
- if (stream->doConvertBuffer[0]) {
-
- convertStreamBuffer(stream, OUTPUT);
- if ( stream->doByteSwap[0] )
- byteSwapBuffer(stream->deviceBuffer,
- stream->bufferSize * stream->nDeviceChannels[0],
- stream->deviceFormat[0]);
-
- // Always de-interleave ASIO output data.
- for ( int i=0; i<stream->nDeviceChannels[0]; i++, bufferInfos++ ) {
- memcpy(bufferInfos->buffers[bufferIndex],
- &stream->deviceBuffer[i*bufferBytes], bufferBytes );
- }
- }
- else { // single channel only
-
- if (stream->doByteSwap[0])
- byteSwapBuffer(stream->userBuffer,
- stream->bufferSize * stream->nUserChannels[0],
- stream->userFormat);
-
- memcpy(bufferInfos->buffers[bufferIndex], stream->userBuffer, bufferBytes );
- }
+ // Count output devices.
+ EnumInfo info;
+ HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") counting output devices!";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return 0;
}
- if ( !info->usingCallback )
- info->blockTick = false;
-
- MUTEX_UNLOCK(&stream->mutex);
-}
-
-void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- stream->callbackInfo.callback = (void *) callback;
- stream->callbackInfo.userData = userData;
- stream->callbackInfo.usingCallback = true;
-}
-
-//******************** End of __WINDOWS_ASIO__ *********************//
-
-#elif defined(__WINDOWS_DS__) // Windows DirectSound API
-
-#include <dsound.h>
-
-// Declarations for utility functions, callbacks, and structures
-// specific to the DirectSound implementation.
-static bool CALLBACK deviceCountCallback(LPGUID lpguid,
- LPCSTR lpcstrDescription,
- LPCSTR lpcstrModule,
- LPVOID lpContext);
-
-static bool CALLBACK deviceInfoCallback(LPGUID lpguid,
- LPCSTR lpcstrDescription,
- LPCSTR lpcstrModule,
- LPVOID lpContext);
-
-static bool CALLBACK defaultDeviceCallback(LPGUID lpguid,
- LPCSTR lpcstrDescription,
- LPCSTR lpcstrModule,
- LPVOID lpContext);
-
-static bool CALLBACK deviceIdCallback(LPGUID lpguid,
- LPCSTR lpcstrDescription,
- LPCSTR lpcstrModule,
- LPVOID lpContext);
-
-static char* getErrorString(int code);
-
-struct enum_info {
- char name[64];
- LPGUID id;
- bool isInput;
- bool isValid;
-};
-
-int RtAudio :: getDefaultInputDevice(void)
-{
- enum_info info;
- info.name[0] = '\0';
-
- // Enumerate through devices to find the default output.
- HRESULT result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)defaultDeviceCallback, &info);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Error performing default input device enumeration: %s.",
- getErrorString(result));
- error(RtError::WARNING);
+ // Now enumerate input devices until we find the id = NULL.
+ info.isInput = true;
+ info.getDefault = true;
+ result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDefaultInputDevice: error (" << getErrorString( result ) << ") enumerating input devices!";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
return 0;
}
- for ( int i=0; i<nDevices; i++ )
- if ( strncmp( devices[i].name, info.name, 64 ) == 0 ) return i;
-
+ if ( info.counter > 0 ) return info.counter - 1;
return 0;
}
-int RtAudio :: getDefaultOutputDevice(void)
+unsigned int RtApiDs :: getDefaultOutputDevice( void )
{
- enum_info info;
- info.name[0] = '\0';
-
- // Enumerate through devices to find the default output.
- HRESULT result = DirectSoundEnumerate((LPDSENUMCALLBACK)defaultDeviceCallback, &info);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Error performing default output device enumeration: %s.",
- getErrorString(result));
- error(RtError::WARNING);
+ // Enumerate output devices until we find the id = NULL.
+ EnumInfo info;
+ info.getDefault = true;
+ HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") enumerating output devices!";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
return 0;
}
- for ( int i=0; i<nDevices; i++ )
- if ( strncmp(devices[i].name, info.name, 64 ) == 0 ) return i;
-
+ if ( info.counter > 0 ) return info.counter - 1;
return 0;
}
-void RtAudio :: initialize(void)
+unsigned int RtApiDs :: getDeviceCount( void )
{
- int i, ins = 0, outs = 0, count = 0;
- HRESULT result;
- nDevices = 0;
-
// Count DirectSound devices.
- result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceCountCallback, &outs);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to enumerate through sound playback devices: %s.",
- getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ EnumInfo info;
+ HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
}
// Count DirectSoundCapture devices.
- result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceCountCallback, &ins);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to enumerate through sound capture devices: %s.",
- getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ info.isInput = true;
+ result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
}
- count = ins + outs;
- if (count == 0) return;
+ return info.counter;
+}
- std::vector<enum_info> info(count);
- for (i=0; i<count; i++) {
- info[i].name[0] = '\0';
- if (i < outs) info[i].isInput = false;
- else info[i].isInput = true;
+RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
+{
+ // Because DirectSound always enumerates input and output devices
+ // separately (and because we don't attempt to combine devices
+ // internally), none of our "devices" will ever be duplex.
+
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ // Enumerate through devices to find the id (if it exists). Note
+ // that we have to do the output enumeration first, even if this is
+ // an input device, in order for the device counter to be correct.
+ EnumInfo dsinfo;
+ dsinfo.findIndex = true;
+ dsinfo.index = device;
+ HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating output devices!";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ }
+
+ if ( dsinfo.name.empty() ) goto probeInput;
+
+ LPDIRECTSOUND output;
+ DSCAPS outCaps;
+ result = DirectSoundCreate( dsinfo.id, &output, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
}
- // Get playback device info and check capabilities.
- result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceInfoCallback, &info[0]);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to enumerate through sound playback devices: %s.",
- getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ outCaps.dwSize = sizeof( outCaps );
+ result = output->GetCaps( &outCaps );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
}
- // Get capture device info and check capabilities.
- result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceInfoCallback, &info[0]);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to enumerate through sound capture devices: %s.",
- getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ // Get output channel information.
+ info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
+
+ // Get sample rate information.
+ info.sampleRates.clear();
+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+ if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
+ SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate )
+ info.sampleRates.push_back( SAMPLE_RATES[k] );
}
- // Parse the devices and check validity. Devices are considered
- // invalid if they cannot be opened, they report < 1 supported
- // channels, or they report no supported data (capture only).
- for (i=0; i<count; i++)
- if ( info[i].isValid ) nDevices++;
+ // Get format information.
+ if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
- if (nDevices == 0) return;
+ output->Release();
- // Allocate the RTAUDIO_DEVICE structures.
- devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE));
- if (devices == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
- }
+ if ( getDefaultOutputDevice() == device )
+ info.isDefaultOutput = true;
- // Copy the names to our devices structures.
- int index = 0;
- for (i=0; i<count; i++) {
- if ( info[i].isValid )
- strncpy(devices[index++].name, info[i].name, 64);
- }
+ // Copy name and return.
+ info.name = dsinfo.name;
- //for (i=0;i<nDevices; i++)
- //probeDeviceInfo(&devices[i]);
+ info.probed = true;
+ return info;
- return;
-}
+ probeInput:
-void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info)
-{
- enum_info dsinfo;
- strncpy( dsinfo.name, info->name, 64 );
- dsinfo.isValid = false;
-
- // Enumerate through input devices to find the id (if it exists).
- HRESULT result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Error performing input device id enumeration: %s.",
- getErrorString(result));
- error(RtError::WARNING);
- return;
+ dsinfo.isInput = true;
+ result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating input devices!";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
}
- // Do capture probe first.
- if ( dsinfo.isValid == false )
- goto playback_probe;
+ if ( dsinfo.name.empty() ) return info;
- LPDIRECTSOUNDCAPTURE input;
+ LPDIRECTSOUNDCAPTURE input;
result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL );
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Could not create DirectSound capture object (%s): %s.",
- info->name, getErrorString(result));
- error(RtError::WARNING);
- goto playback_probe;
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
}
- DSCCAPS in_caps;
- in_caps.dwSize = sizeof(in_caps);
- result = input->GetCaps( &in_caps );
- if ( FAILED(result) ) {
+ DSCCAPS inCaps;
+ inCaps.dwSize = sizeof( inCaps );
+ result = input->GetCaps( &inCaps );
+ if ( FAILED( result ) ) {
input->Release();
- sprintf(message, "RtAudio: Could not get DirectSound capture capabilities (%s): %s.",
- info->name, getErrorString(result));
- error(RtError::WARNING);
- goto playback_probe;
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsinfo.name << ")!";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
}
// Get input channel information.
- info->minInputChannels = 1;
- info->maxInputChannels = in_caps.dwChannels;
+ info.inputChannels = inCaps.dwChannels;
// Get sample rate and format information.
- if( in_caps.dwChannels == 2 ) {
- if( in_caps.dwFormats & WAVE_FORMAT_1S16 ) info->nativeFormats |= RTAUDIO_SINT16;
- if( in_caps.dwFormats & WAVE_FORMAT_2S16 ) info->nativeFormats |= RTAUDIO_SINT16;
- if( in_caps.dwFormats & WAVE_FORMAT_4S16 ) info->nativeFormats |= RTAUDIO_SINT16;
- if( in_caps.dwFormats & WAVE_FORMAT_1S08 ) info->nativeFormats |= RTAUDIO_SINT8;
- if( in_caps.dwFormats & WAVE_FORMAT_2S08 ) info->nativeFormats |= RTAUDIO_SINT8;
- if( in_caps.dwFormats & WAVE_FORMAT_4S08 ) info->nativeFormats |= RTAUDIO_SINT8;
-
- if ( info->nativeFormats & RTAUDIO_SINT16 ) {
- if( in_caps.dwFormats & WAVE_FORMAT_1S16 ) info->sampleRates[info->nSampleRates++] = 11025;
- if( in_caps.dwFormats & WAVE_FORMAT_2S16 ) info->sampleRates[info->nSampleRates++] = 22050;
- if( in_caps.dwFormats & WAVE_FORMAT_4S16 ) info->sampleRates[info->nSampleRates++] = 44100;
- }
- else if ( info->nativeFormats & RTAUDIO_SINT8 ) {
- if( in_caps.dwFormats & WAVE_FORMAT_1S08 ) info->sampleRates[info->nSampleRates++] = 11025;
- if( in_caps.dwFormats & WAVE_FORMAT_2S08 ) info->sampleRates[info->nSampleRates++] = 22050;
- if( in_caps.dwFormats & WAVE_FORMAT_4S08 ) info->sampleRates[info->nSampleRates++] = 44100;
- }
- }
- else if ( in_caps.dwChannels == 1 ) {
- if( in_caps.dwFormats & WAVE_FORMAT_1M16 ) info->nativeFormats |= RTAUDIO_SINT16;
- if( in_caps.dwFormats & WAVE_FORMAT_2M16 ) info->nativeFormats |= RTAUDIO_SINT16;
- if( in_caps.dwFormats & WAVE_FORMAT_4M16 ) info->nativeFormats |= RTAUDIO_SINT16;
- if( in_caps.dwFormats & WAVE_FORMAT_1M08 ) info->nativeFormats |= RTAUDIO_SINT8;
- if( in_caps.dwFormats & WAVE_FORMAT_2M08 ) info->nativeFormats |= RTAUDIO_SINT8;
- if( in_caps.dwFormats & WAVE_FORMAT_4M08 ) info->nativeFormats |= RTAUDIO_SINT8;
-
- if ( info->nativeFormats & RTAUDIO_SINT16 ) {
- if( in_caps.dwFormats & WAVE_FORMAT_1M16 ) info->sampleRates[info->nSampleRates++] = 11025;
- if( in_caps.dwFormats & WAVE_FORMAT_2M16 ) info->sampleRates[info->nSampleRates++] = 22050;
- if( in_caps.dwFormats & WAVE_FORMAT_4M16 ) info->sampleRates[info->nSampleRates++] = 44100;
- }
- else if ( info->nativeFormats & RTAUDIO_SINT8 ) {
- if( in_caps.dwFormats & WAVE_FORMAT_1M08 ) info->sampleRates[info->nSampleRates++] = 11025;
- if( in_caps.dwFormats & WAVE_FORMAT_2M08 ) info->sampleRates[info->nSampleRates++] = 22050;
- if( in_caps.dwFormats & WAVE_FORMAT_4M08 ) info->sampleRates[info->nSampleRates++] = 44100;
- }
- }
- else info->minInputChannels = 0; // technically, this would be an error
+ if ( inCaps.dwChannels == 2 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+
+ if ( info.nativeFormats & RTAUDIO_SINT16 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.sampleRates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.sampleRates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.sampleRates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.sampleRates.push_back( 96000 );
+ }
+ else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.sampleRates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.sampleRates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.sampleRates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.sampleRates.push_back( 44100 );
+ }
+ }
+ else if ( inCaps.dwChannels == 1 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+
+ if ( info.nativeFormats & RTAUDIO_SINT16 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.sampleRates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.sampleRates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.sampleRates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.sampleRates.push_back( 96000 );
+ }
+ else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.sampleRates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.sampleRates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.sampleRates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.sampleRates.push_back( 96000 );
+ }
+ }
+ else info.inputChannels = 0; // technically, this would be an error
input->Release();
- playback_probe:
+ if ( info.inputChannels == 0 ) return info;
- dsinfo.isValid = false;
-
- // Enumerate through output devices to find the id (if it exists).
- result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Error performing output device id enumeration: %s.",
- getErrorString(result));
- error(RtError::WARNING);
- return;
- }
+ if ( getDefaultInputDevice() == device )
+ info.isDefaultInput = true;
- // Now do playback probe.
- if ( dsinfo.isValid == false )
- goto check_parameters;
+ // Copy name and return.
+ info.name = dsinfo.name;
+ info.probed = true;
+ return info;
+}
- LPDIRECTSOUND output;
- DSCAPS out_caps;
- result = DirectSoundCreate( dsinfo.id, &output, NULL );
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Could not create DirectSound playback object (%s): %s.",
- info->name, getErrorString(result));
- error(RtError::WARNING);
- goto check_parameters;
+bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ if ( channels + firstChannel > 2 ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
+ return FAILURE;
}
- out_caps.dwSize = sizeof(out_caps);
- result = output->GetCaps( &out_caps );
- if ( FAILED(result) ) {
- output->Release();
- sprintf(message, "RtAudio: Could not get DirectSound playback capabilities (%s): %s.",
- info->name, getErrorString(result));
- error(RtError::WARNING);
- goto check_parameters;
+ // Enumerate through devices to find the id (if it exists). Note
+ // that we have to do the output enumeration first, even if this is
+ // an input device, in order for the device counter to be correct.
+ EnumInfo dsinfo;
+ dsinfo.findIndex = true;
+ dsinfo.index = device;
+ HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating output devices!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- // Get output channel information.
- info->minOutputChannels = 1;
- info->maxOutputChannels = ( out_caps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
-
- // Get sample rate information. Use capture device rate information
- // if it exists.
- if ( info->nSampleRates == 0 ) {
- info->sampleRates[0] = (int) out_caps.dwMinSecondarySampleRate;
- info->sampleRates[1] = (int) out_caps.dwMaxSecondarySampleRate;
- if ( out_caps.dwFlags & DSCAPS_CONTINUOUSRATE )
- info->nSampleRates = -1;
- else if ( out_caps.dwMinSecondarySampleRate == out_caps.dwMaxSecondarySampleRate ) {
- if ( out_caps.dwMinSecondarySampleRate == 0 ) {
- // This is a bogus driver report ... fake the range and cross
- // your fingers.
- info->sampleRates[0] = 11025;
- info->sampleRates[1] = 48000;
- info->nSampleRates = -1; /* continuous range */
- sprintf(message, "RtAudio: bogus sample rates reported by DirectSound driver ... using defaults (%s).",
- info->name);
- error(RtError::DEBUG_WARNING);
- }
- else {
- info->nSampleRates = 1;
- }
- }
- else if ( (out_caps.dwMinSecondarySampleRate < 1000.0) &&
- (out_caps.dwMaxSecondarySampleRate > 50000.0) ) {
- // This is a bogus driver report ... support for only two
- // distant rates. We'll assume this is a range.
- info->nSampleRates = -1;
- sprintf(message, "RtAudio: bogus sample rates reported by DirectSound driver ... using range (%s).",
- info->name);
- error(RtError::WARNING);
+ if ( mode == OUTPUT ) {
+ if ( dsinfo.name.empty() ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- else info->nSampleRates = 2;
}
- else {
- // Check input rates against output rate range
- for ( int i=info->nSampleRates-1; i>=0; i-- ) {
- if ( info->sampleRates[i] <= out_caps.dwMaxSecondarySampleRate )
- break;
- info->nSampleRates--;
+ else { // mode == INPUT
+ dsinfo.isInput = true;
+ HRESULT result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating input devices!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- while ( info->sampleRates[0] < out_caps.dwMinSecondarySampleRate ) {
- info->nSampleRates--;
- for ( int i=0; i<info->nSampleRates; i++)
- info->sampleRates[i] = info->sampleRates[i+1];
- if ( info->nSampleRates <= 0 ) break;
+ if ( dsinfo.name.empty() ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
}
- // Get format information.
- if ( out_caps.dwFlags & DSCAPS_PRIMARY16BIT ) info->nativeFormats |= RTAUDIO_SINT16;
- if ( out_caps.dwFlags & DSCAPS_PRIMARY8BIT ) info->nativeFormats |= RTAUDIO_SINT8;
-
- output->Release();
-
- check_parameters:
- if ( info->maxInputChannels == 0 && info->maxOutputChannels == 0 )
- return;
- if ( info->nSampleRates == 0 || info->nativeFormats == 0 )
- return;
-
- // Determine duplex status.
- if (info->maxInputChannels < info->maxOutputChannels)
- info->maxDuplexChannels = info->maxInputChannels;
- else
- info->maxDuplexChannels = info->maxOutputChannels;
- if (info->minInputChannels < info->minOutputChannels)
- info->minDuplexChannels = info->minInputChannels;
- else
- info->minDuplexChannels = info->minOutputChannels;
-
- if ( info->maxDuplexChannels > 0 ) info->hasDuplexSupport = true;
- else info->hasDuplexSupport = false;
-
- info->probed = true;
-
- return;
-}
-
-bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream,
- STREAM_MODE mode, int channels,
- int sampleRate, RTAUDIO_FORMAT format,
- int *bufferSize, int numberOfBuffers)
-{
- HRESULT result;
- HWND hWnd = GetForegroundWindow();
// According to a note in PortAudio, using GetDesktopWindow()
// instead of GetForegroundWindow() is supposed to avoid problems
// that occur when the application's window is not the foreground
// window. Also, if the application window closes before the
// DirectSound buffer, DirectSound can crash. However, for console
// applications, no sound was produced when using GetDesktopWindow().
- long buffer_size;
- LPVOID audioPtr;
- DWORD dataLen;
- int nBuffers;
+ HWND hWnd = GetForegroundWindow();
// Check the numberOfBuffers parameter and limit the lowest value to
// two. This is a judgement call and a value of two is probably too
// low for capture, but it should work for playback.
- if (numberOfBuffers < 2)
- nBuffers = 2;
- else
- nBuffers = numberOfBuffers;
+ int nBuffers = 0;
+ if ( options ) nBuffers = options->numberOfBuffers;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
+ if ( nBuffers < 2 ) nBuffers = 3;
- // Define the wave format structure (16-bit PCM, srate, channels)
+ // Create the wave format structure. The data format setting will
+ // be determined later.
WAVEFORMATEX waveFormat;
- ZeroMemory(&waveFormat, sizeof(WAVEFORMATEX));
+ ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
- waveFormat.nChannels = channels;
+ waveFormat.nChannels = channels + firstChannel;
waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
- // Determine the data format.
- if ( devices[device].nativeFormats ) { // 8-bit and/or 16-bit support
- if ( format == RTAUDIO_SINT8 ) {
- if ( devices[device].nativeFormats & RTAUDIO_SINT8 )
- waveFormat.wBitsPerSample = 8;
- else
- waveFormat.wBitsPerSample = 16;
- }
- else {
- if ( devices[device].nativeFormats & RTAUDIO_SINT16 )
- waveFormat.wBitsPerSample = 16;
- else
- waveFormat.wBitsPerSample = 8;
- }
- }
- else {
- sprintf(message, "RtAudio: no reported data formats for DirectSound device (%s).",
- devices[device].name);
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
-
- waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ // Determine the device buffer size. By default, 32k, but we will
+ // grow it to make allowances for very large software buffer sizes.
+ DWORD dsBufferSize = 0;
+ DWORD dsPointerLeadTime = 0;
+ long bufferBytes = MINIMUM_DEVICE_BUFFER_SIZE; // sound cards will always *knock wood* support this
- enum_info dsinfo;
- strncpy( dsinfo.name, devices[device].name, 64 );
- dsinfo.isValid = false;
+ void *ohandle = 0, *bhandle = 0;
if ( mode == OUTPUT ) {
- if ( devices[device].maxOutputChannels < channels )
- return FAILURE;
-
- // Enumerate through output devices to find the id (if it exists).
- result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Error performing output device id enumeration: %s.",
- getErrorString(result));
- error(RtError::DEBUG_WARNING);
+ LPDIRECTSOUND output;
+ result = DirectSoundCreate( dsinfo.id, &output, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!";
+ errorText_ = errorStream_.str();
return FAILURE;
}
- if ( dsinfo.isValid == false ) {
- sprintf(message, "RtAudio: DS output device (%s) id not found!", devices[device].name);
- error(RtError::DEBUG_WARNING);
+ DSCAPS outCaps;
+ outCaps.dwSize = sizeof( outCaps );
+ result = output->GetCaps( &outCaps );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsinfo.name << ")!";
+ errorText_ = errorStream_.str();
return FAILURE;
}
- LPGUID id = dsinfo.id;
- LPDIRECTSOUND object;
- LPDIRECTSOUNDBUFFER buffer;
- DSBUFFERDESC bufferDescription;
-
- result = DirectSoundCreate( id, &object, NULL );
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Could not create DirectSound playback object (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::DEBUG_WARNING);
+ // Check channel information.
+ if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
+ errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsinfo.name << ") does not support stereo playback.";
+ errorText_ = errorStream_.str();
return FAILURE;
}
- // Set cooperative level to DSSCL_EXCLUSIVE
- result = object->SetCooperativeLevel(hWnd, DSSCL_EXCLUSIVE);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message, "RtAudio: Unable to set DirectSound cooperative level (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
+ // Check format information. Use 16-bit format unless not
+ // supported or user requests 8-bit.
+ if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
+ !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
+ waveFormat.wBitsPerSample = 16;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ else {
+ waveFormat.wBitsPerSample = 8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ stream_.userFormat = format;
+
+ // Update wave format structure and buffer information.
+ waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+ waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
+
+ // If the user wants an even bigger buffer, increase the device buffer size accordingly.
+ while ( dsPointerLeadTime * 2U > (DWORD) bufferBytes )
+ bufferBytes *= 2;
+
+ // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
+ //result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
+ // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
+ result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsinfo.name << ")!";
+ errorText_ = errorStream_.str();
return FAILURE;
}
// Even though we will write to the secondary buffer, we need to
- // access the primary buffer to set the correct output format.
- // The default is 8-bit, 22 kHz!
- // Setup the DS primary buffer description.
- ZeroMemory(&bufferDescription, sizeof(DSBUFFERDESC));
- bufferDescription.dwSize = sizeof(DSBUFFERDESC);
+ // access the primary buffer to set the correct output format
+ // (since the default is 8-bit, 22 kHz!). Setup the DS primary
+ // buffer description.
+ DSBUFFERDESC bufferDescription;
+ ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
+ bufferDescription.dwSize = sizeof( DSBUFFERDESC );
bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
+
// Obtain the primary buffer
- result = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message, "RtAudio: Unable to access DS primary buffer (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
+ LPDIRECTSOUNDBUFFER buffer;
+ result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsinfo.name << ")!";
+ errorText_ = errorStream_.str();
return FAILURE;
}
// Set the primary DS buffer sound format.
- result = buffer->SetFormat(&waveFormat);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message, "RtAudio: Unable to set DS primary buffer format (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
+ result = buffer->SetFormat( &waveFormat );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsinfo.name << ")!";
+ errorText_ = errorStream_.str();
return FAILURE;
}
// Setup the secondary DS buffer description.
- buffer_size = channels * *bufferSize * nBuffers * waveFormat.wBitsPerSample / 8;
- ZeroMemory(&bufferDescription, sizeof(DSBUFFERDESC));
- bufferDescription.dwSize = sizeof(DSBUFFERDESC);
+ dsBufferSize = (DWORD) bufferBytes;
+ ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
+ bufferDescription.dwSize = sizeof( DSBUFFERDESC );
bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
+ DSBCAPS_GLOBALFOCUS |
DSBCAPS_GETCURRENTPOSITION2 |
DSBCAPS_LOCHARDWARE ); // Force hardware mixing
- bufferDescription.dwBufferBytes = buffer_size;
+ bufferDescription.dwBufferBytes = bufferBytes;
bufferDescription.lpwfxFormat = &waveFormat;
// Try to create the secondary DS buffer. If that doesn't work,
// try to use software mixing. Otherwise, there's a problem.
- result = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL);
- if ( FAILED(result) ) {
+ result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
+ DSBCAPS_GLOBALFOCUS |
DSBCAPS_GETCURRENTPOSITION2 |
DSBCAPS_LOCSOFTWARE ); // Force software mixing
- result = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message, "RtAudio: Unable to create secondary DS buffer (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
+ result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsinfo.name << ")!";
+ errorText_ = errorStream_.str();
return FAILURE;
}
}
// Get the buffer size ... might be different from what we specified.
DSBCAPS dsbcaps;
- dsbcaps.dwSize = sizeof(DSBCAPS);
- buffer->GetCaps(&dsbcaps);
- buffer_size = dsbcaps.dwBufferBytes;
+ dsbcaps.dwSize = sizeof( DSBCAPS );
+ result = buffer->GetCaps( &dsbcaps );
+ if ( FAILED( result ) ) {
+ output->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsinfo.name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ bufferBytes = dsbcaps.dwBufferBytes;
// Lock the DS buffer
- result = buffer->Lock(0, buffer_size, &audioPtr, &dataLen, NULL, NULL, 0);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message, "RtAudio: Unable to lock DS buffer (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
+ LPVOID audioPtr;
+ DWORD dataLen;
+ result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ output->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsinfo.name << ")!";
+ errorText_ = errorStream_.str();
return FAILURE;
}
// Zero the DS buffer
- ZeroMemory(audioPtr, dataLen);
+ ZeroMemory( audioPtr, dataLen );
// Unlock the DS buffer
- result = buffer->Unlock(audioPtr, dataLen, NULL, 0);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message, "RtAudio: Unable to unlock DS buffer(%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ output->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsinfo.name << ")!";
+ errorText_ = errorStream_.str();
return FAILURE;
}
- stream->handle[0].object = (void *) object;
- stream->handle[0].buffer = (void *) buffer;
- stream->nDeviceChannels[0] = channels;
+ dsBufferSize = bufferBytes;
+ ohandle = (void *) output;
+ bhandle = (void *) buffer;
}
if ( mode == INPUT ) {
- if ( devices[device].maxInputChannels < channels )
- return FAILURE;
-
- // Enumerate through input devices to find the id (if it exists).
- result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Error performing input device id enumeration: %s.",
- getErrorString(result));
- error(RtError::DEBUG_WARNING);
+ LPDIRECTSOUNDCAPTURE input;
+ result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!";
+ errorText_ = errorStream_.str();
return FAILURE;
}
- if ( dsinfo.isValid == false ) {
- sprintf(message, "RtAudio: DS input device (%s) id not found!", devices[device].name);
- error(RtError::DEBUG_WARNING);
+ DSCCAPS inCaps;
+ inCaps.dwSize = sizeof( inCaps );
+ result = input->GetCaps( &inCaps );
+ if ( FAILED( result ) ) {
+ input->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsinfo.name << ")!";
+ errorText_ = errorStream_.str();
return FAILURE;
}
- LPGUID id = dsinfo.id;
- LPDIRECTSOUNDCAPTURE object;
- LPDIRECTSOUNDCAPTUREBUFFER buffer;
- DSCBUFFERDESC bufferDescription;
-
- result = DirectSoundCaptureCreate( id, &object, NULL );
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Could not create DirectSound capture object (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
+ // Check channel information.
+ if ( inCaps.dwChannels < channels + firstChannel ) {
+ errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
return FAILURE;
}
- // Setup the secondary DS buffer description.
- buffer_size = channels * *bufferSize * nBuffers * waveFormat.wBitsPerSample / 8;
- ZeroMemory(&bufferDescription, sizeof(DSCBUFFERDESC));
- bufferDescription.dwSize = sizeof(DSCBUFFERDESC);
- bufferDescription.dwFlags = 0;
- bufferDescription.dwReserved = 0;
- bufferDescription.dwBufferBytes = buffer_size;
- bufferDescription.lpwfxFormat = &waveFormat;
+ // Check format information. Use 16-bit format unless user
+ // requests 8-bit.
+ DWORD deviceFormats;
+ if ( channels + firstChannel == 2 ) {
+ deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
+ if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
+ waveFormat.wBitsPerSample = 8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ else { // assume 16-bit is supported
+ waveFormat.wBitsPerSample = 16;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ }
+ else { // channel == 1
+ deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
+ if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
+ waveFormat.wBitsPerSample = 8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ else { // assume 16-bit is supported
+ waveFormat.wBitsPerSample = 16;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ }
+ stream_.userFormat = format;
+
+ // Update wave format structure and buffer information.
+ waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+ waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+
+ // Setup the secondary DS buffer description.
+ dsBufferSize = bufferBytes;
+ DSCBUFFERDESC bufferDescription;
+ ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
+ bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
+ bufferDescription.dwFlags = 0;
+ bufferDescription.dwReserved = 0;
+ bufferDescription.dwBufferBytes = bufferBytes;
+ bufferDescription.lpwfxFormat = &waveFormat;
// Create the capture buffer.
- result = object->CreateCaptureBuffer(&bufferDescription, &buffer, NULL);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message, "RtAudio: Unable to create DS capture buffer (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
+ LPDIRECTSOUNDCAPTUREBUFFER buffer;
+ result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ input->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsinfo.name << ")!";
+ errorText_ = errorStream_.str();
return FAILURE;
}
// Lock the capture buffer
- result = buffer->Lock(0, buffer_size, &audioPtr, &dataLen, NULL, NULL, 0);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
+ LPVOID audioPtr;
+ DWORD dataLen;
+ result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ input->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsinfo.name << ")!";
+ errorText_ = errorStream_.str();
return FAILURE;
}
// Zero the buffer
- ZeroMemory(audioPtr, dataLen);
+ ZeroMemory( audioPtr, dataLen );
// Unlock the buffer
- result = buffer->Unlock(audioPtr, dataLen, NULL, 0);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.",
- devices[device].name, getErrorString(result));
- error(RtError::WARNING);
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ input->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsinfo.name << ")!";
+ errorText_ = errorStream_.str();
return FAILURE;
}
- stream->handle[1].object = (void *) object;
- stream->handle[1].buffer = (void *) buffer;
- stream->nDeviceChannels[1] = channels;
+ dsBufferSize = bufferBytes;
+ ohandle = (void *) input;
+ bhandle = (void *) buffer;
}
- stream->userFormat = format;
- if ( waveFormat.wBitsPerSample == 8 )
- stream->deviceFormat[mode] = RTAUDIO_SINT8;
- else
- stream->deviceFormat[mode] = RTAUDIO_SINT16;
- stream->nUserChannels[mode] = channels;
- *bufferSize = buffer_size / (channels * nBuffers * waveFormat.wBitsPerSample / 8);
- stream->bufferSize = *bufferSize;
+ // Set various stream parameters
+ DsHandle *handle = 0;
+ stream_.nDeviceChannels[mode] = channels + firstChannel;
+ stream_.nUserChannels[mode] = channels;
+ stream_.bufferSize = *bufferSize;
+ stream_.channelOffset[mode] = firstChannel;
+ stream_.deviceInterleaved[mode] = true;
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
- // Set flags for buffer conversion
- stream->doConvertBuffer[mode] = false;
- if (stream->userFormat != stream->deviceFormat[mode])
- stream->doConvertBuffer[mode] = true;
- if (stream->nUserChannels[mode] < stream->nDeviceChannels[mode])
- stream->doConvertBuffer[mode] = true;
+ // Set flag for buffer conversion
+ stream_.doConvertBuffer[mode] = false;
+ if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if (stream_.userFormat != stream_.deviceFormat[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
// Allocate necessary internal buffers
- if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) {
-
- long buffer_bytes;
- if (stream->nUserChannels[0] >= stream->nUserChannels[1])
- buffer_bytes = stream->nUserChannels[0];
- else
- buffer_bytes = stream->nUserChannels[1];
-
- buffer_bytes *= *bufferSize * formatBytes(stream->userFormat);
- if (stream->userBuffer) free(stream->userBuffer);
- stream->userBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->userBuffer == NULL)
- goto memory_error;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
}
- if ( stream->doConvertBuffer[mode] ) {
+ if ( stream_.doConvertBuffer[mode] ) {
- long buffer_bytes;
bool makeBuffer = true;
- if ( mode == OUTPUT )
- buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- else { // mode == INPUT
- buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]);
- if ( stream->mode == OUTPUT && stream->deviceBuffer ) {
- long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- if ( buffer_bytes < bytes_out ) makeBuffer = false;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
}
}
if ( makeBuffer ) {
- buffer_bytes *= *bufferSize;
- if (stream->deviceBuffer) free(stream->deviceBuffer);
- stream->deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->deviceBuffer == NULL)
- goto memory_error;
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
}
}
- stream->device[mode] = device;
- stream->state = STREAM_STOPPED;
- if ( stream->mode == OUTPUT && mode == INPUT )
+ // Allocate our DsHandle structures for the stream.
+ if ( stream_.apiHandle == 0 ) {
+ try {
+ handle = new DsHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
+ goto error;
+ }
+
+ // Create a manual-reset event.
+ handle->condition = CreateEvent( NULL, // no security
+ TRUE, // manual-reset
+ FALSE, // non-signaled initially
+ NULL ); // unnamed
+ stream_.apiHandle = (void *) handle;
+ }
+ else
+ handle = (DsHandle *) stream_.apiHandle;
+ handle->id[mode] = ohandle;
+ handle->buffer[mode] = bhandle;
+ handle->dsBufferSize[mode] = dsBufferSize;
+ handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
+
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ if ( stream_.mode == OUTPUT && mode == INPUT )
// We had already set up an output stream.
- stream->mode = DUPLEX;
+ stream_.mode = DUPLEX;
else
- stream->mode = mode;
- stream->nBuffers = nBuffers;
- stream->sampleRate = sampleRate;
+ stream_.mode = mode;
+ stream_.nBuffers = nBuffers;
+ stream_.sampleRate = sampleRate;
+
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+
+ // Setup the callback thread.
+ unsigned threadId;
+ stream_.callbackInfo.object = (void *) this;
+ stream_.callbackInfo.isRunning = true;
+ stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
+ &stream_.callbackInfo, 0, &threadId );
+ if ( stream_.callbackInfo.thread == 0 ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
+ goto error;
+ }
+ // Boost DS thread priority
+ SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
return SUCCESS;
- memory_error:
- if (stream->handle[0].object) {
- LPDIRECTSOUND object = (LPDIRECTSOUND) stream->handle[0].object;
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- if (buffer) {
- buffer->Release();
- stream->handle[0].buffer = NULL;
+ error:
+ if ( handle ) {
+ if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
+ LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ if ( buffer ) buffer->Release();
+ object->Release();
}
- object->Release();
- stream->handle[0].object = NULL;
+ if ( handle->buffer[1] ) {
+ LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ if ( buffer ) buffer->Release();
+ object->Release();
+ }
+ CloseHandle( handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
}
- if (stream->handle[1].object) {
- LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) stream->handle[1].object;
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
- if (buffer) {
- buffer->Release();
- stream->handle[1].buffer = NULL;
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
}
- object->Release();
- stream->handle[1].object = NULL;
}
- if (stream->userBuffer) {
- free(stream->userBuffer);
- stream->userBuffer = 0;
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
}
- sprintf(message, "RtAudio: error allocating buffer memory (%s).",
- devices[device].name);
- error(RtError::WARNING);
+
return FAILURE;
}
-void RtAudio :: cancelStreamCallback(int streamId)
+void RtApiDs :: closeStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- if (stream->callbackInfo.usingCallback) {
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiDs::closeStream(): no open stream to close!";
+ error( RtError::WARNING );
+ return;
+ }
- if (stream->state == STREAM_RUNNING)
- stopStream( streamId );
+ // Stop the callback thread.
+ stream_.callbackInfo.isRunning = false;
+ WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
+ CloseHandle( (HANDLE) stream_.callbackInfo.thread );
- MUTEX_LOCK(&stream->mutex);
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ if ( handle ) {
+ if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
+ LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ if ( buffer ) {
+ buffer->Stop();
+ buffer->Release();
+ }
+ object->Release();
+ }
+ if ( handle->buffer[1] ) {
+ LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ if ( buffer ) {
+ buffer->Stop();
+ buffer->Release();
+ }
+ object->Release();
+ }
+ CloseHandle( handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
- stream->callbackInfo.usingCallback = false;
- WaitForSingleObject( (HANDLE)stream->callbackInfo.thread, INFINITE );
- CloseHandle( (HANDLE)stream->callbackInfo.thread );
- stream->callbackInfo.thread = 0;
- stream->callbackInfo.callback = NULL;
- stream->callbackInfo.userData = NULL;
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
- MUTEX_UNLOCK(&stream->mutex);
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
}
+
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
}
-void RtAudio :: closeStream(int streamId)
+void RtApiDs :: startStream()
{
- // We don't want an exception to be thrown here because this
- // function is called by our class destructor. So, do our own
- // streamId check.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
- error(RtError::WARNING);
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiDs::startStream(): the stream is already running!";
+ error( RtError::WARNING );
return;
}
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId];
+ // Increase scheduler frequency on lesser windows (a side-effect of
+ // increasing timer accuracy). On greater windows (Win2K or later),
+ // this is already in effect.
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+
+ timeBeginPeriod( 1 );
+
+ /*
+ memset( &statistics, 0, sizeof( statistics ) );
+ statistics.sampleRate = stream_.sampleRate;
+ statistics.writeDeviceBufferLeadBytes = handle->dsPointerLeadTime[0];
+ */
+
+ buffersRolling = false;
+ duplexPrerollBytes = 0;
- if (stream->callbackInfo.usingCallback) {
- stream->callbackInfo.usingCallback = false;
- WaitForSingleObject( (HANDLE)stream->callbackInfo.thread, INFINITE );
- CloseHandle( (HANDLE)stream->callbackInfo.thread );
+ if ( stream_.mode == DUPLEX ) {
+ // 0.5 seconds of silence in DUPLEX mode while the devices spin up and synchronize.
+ duplexPrerollBytes = (int) ( 0.5 * stream_.sampleRate * formatBytes( stream_.deviceFormat[1] ) * stream_.nDeviceChannels[1] );
}
- DeleteCriticalSection(&stream->mutex);
+ HRESULT result = 0;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ //statistics.outputFrameSize = formatBytes( stream_.deviceFormat[0] ) * stream_.nDeviceChannels[0];
- if (stream->handle[0].object) {
- LPDIRECTSOUND object = (LPDIRECTSOUND) stream->handle[0].object;
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- if (buffer) {
- buffer->Stop();
- buffer->Release();
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ result = buffer->Play( 0, 0, DSBPLAY_LOOPING );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
- object->Release();
}
- if (stream->handle[1].object) {
- LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) stream->handle[1].object;
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
- if (buffer) {
- buffer->Stop();
- buffer->Release();
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ //statistics.inputFrameSize = formatBytes( stream_.deviceFormat[1]) * stream_.nDeviceChannels[1];
+
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ result = buffer->Start( DSCBSTART_LOOPING );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
- object->Release();
}
- if (stream->userBuffer)
- free(stream->userBuffer);
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ stream_.state = STREAM_RUNNING;
- if (stream->deviceBuffer)
- free(stream->deviceBuffer);
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
- free(stream);
- streams.erase(streamId);
+ if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
}
-void RtAudio :: startStream(int streamId)
+void RtApiDs :: stopStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
+ }
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK( &stream_.mutex );
- if (stream->state == STREAM_RUNNING)
- goto unlock;
+ HRESULT result = 0;
+ LPVOID audioPtr;
+ DWORD dataLen;
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 1;
+ MUTEX_UNLOCK( &stream_.mutex );
+ WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled
+ ResetEvent( handle->condition );
+ MUTEX_LOCK( &stream_.mutex );
+ }
+
+ // Stop the buffer and clear memory
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ result = buffer->Stop();
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
- HRESULT result;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- result = buffer->Play(0, 0, DSBPLAY_LOOPING );
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to start DS buffer (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ // Lock the buffer and clear it so that if we start to play again,
+ // we won't have old data playing.
+ result = buffer->Lock( 0, handle->dsBufferSize[0], &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
+
+ // Zero the DS buffer
+ ZeroMemory( audioPtr, dataLen );
+
+ // Unlock the DS buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // If we start playing again, we must begin at beginning of buffer.
+ handle->bufferPointer[0] = 0;
}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
- result = buffer->Start(DSCBSTART_LOOPING );
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to start DS capture buffer (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ audioPtr = NULL;
+ dataLen = 0;
+
+ result = buffer->Stop();
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") stopping input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // Lock the buffer and clear it so that if we start to play again,
+ // we won't have old data playing.
+ result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") locking input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // Zero the DS buffer
+ ZeroMemory( audioPtr, dataLen );
+
+ // Unlock the DS buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::abortStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
+
+ // If we start recording again, we must begin at beginning of buffer.
+ handle->bufferPointer[1] = 0;
}
- stream->state = STREAM_RUNNING;
unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
+ if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
+}
+
+void RtApiDs :: abortStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
+ }
+
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ handle->drainCounter = 1;
+
+ stopStream();
}
-void RtAudio :: stopStream(int streamId)
+void RtApiDs :: callbackEvent()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ if ( stream_.state == STREAM_STOPPED ) {
+ Sleep(50); // sleep 50 milliseconds
+ return;
+ }
+
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtError::WARNING );
+ return;
+ }
- MUTEX_LOCK(&stream->mutex);
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
- if (stream->state == STREAM_STOPPED) {
- MUTEX_UNLOCK(&stream->mutex);
+ // Check if we were draining the stream and signal is finished.
+ if ( handle->drainCounter > stream_.nBuffers + 2 ) {
+ if ( handle->internalDrain == false )
+ SetEvent( handle->condition );
+ else
+ stopStream();
return;
}
- // There is no specific DirectSound API call to "drain" a buffer
- // before stopping. We can hack this for playback by writing zeroes
- // for another bufferSize * nBuffers frames. For capture, the
- // concept is less clear so we'll repeat what we do in the
- // abortStream() case.
+ MUTEX_LOCK( &stream_.mutex );
+
+ // Invoke user callback to get fresh output data UNLESS we are
+ // draining stream.
+ if ( handle->drainCounter == 0 ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
+ }
+ handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( handle->drainCounter == 2 ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ abortStream();
+ return;
+ }
+ else if ( handle->drainCounter == 1 )
+ handle->internalDrain = true;
+ }
+
HRESULT result;
- DWORD dsBufferSize;
+ DWORD currentWritePos, safeWritePos;
+ DWORD currentReadPos, safeReadPos;
+ DWORD leadPos;
+ UINT nextWritePos;
+
+#ifdef GENERATE_DEBUG_LOG
+ DWORD writeTime, readTime;
+#endif
+
LPVOID buffer1 = NULL;
LPVOID buffer2 = NULL;
DWORD bufferSize1 = 0;
DWORD bufferSize2 = 0;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- DWORD currentPos, safePos;
- long buffer_bytes = stream->bufferSize * stream->nDeviceChannels[0];
- buffer_bytes *= formatBytes(stream->deviceFormat[0]);
+ char *buffer;
+ long bufferBytes;
+
+ if ( stream_.mode == DUPLEX && !buffersRolling ) {
+ assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
+
+ // It takes a while for the devices to get rolling. As a result,
+ // there's no guarantee that the capture and write device pointers
+ // will move in lockstep. Wait here for both devices to start
+ // rolling, and then set our buffer pointers accordingly.
+ // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
+ // bytes later than the write buffer.
+
+ // Stub: a serious risk of having a pre-emptive scheduling round
+ // take place between the two GetCurrentPosition calls... but I'm
+ // really not sure how to solve the problem. Temporarily boost to
+ // Realtime priority, maybe; but I'm not sure what priority the
+ // DirectSound service threads run at. We *should* be roughly
+ // within a ms or so of correct.
+
+ LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+
+ DWORD initialWritePos, initialSafeWritePos;
+ DWORD initialReadPos, initialSafeReadPos;
+
+ result = dsWriteBuffer->GetCurrentPosition( &initialWritePos, &initialSafeWritePos );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ error( RtError::SYSTEM_ERROR );
+ }
+ result = dsCaptureBuffer->GetCurrentPosition( &initialReadPos, &initialSafeReadPos );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ error( RtError::SYSTEM_ERROR );
+ }
+ while ( true ) {
+ result = dsWriteBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ error( RtError::SYSTEM_ERROR );
+ }
+ result = dsCaptureBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ error( RtError::SYSTEM_ERROR );
+ }
+ if ( safeWritePos != initialSafeWritePos && safeReadPos != initialSafeReadPos ) break;
+ Sleep( 1 );
+ }
- LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- UINT nextWritePos = stream->handle[0].bufferPointer;
- dsBufferSize = buffer_bytes * stream->nBuffers;
+ assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
- // Write zeroes for nBuffer counts.
- for (int i=0; i<stream->nBuffers; i++) {
+ buffersRolling = true;
+ handle->bufferPointer[0] = ( safeWritePos + handle->dsPointerLeadTime[0] );
+ handle->bufferPointer[1] = safeReadPos;
+ }
- // Find out where the read and "safe write" pointers are.
- result = dsBuffer->GetCurrentPosition(¤tPos, &safePos);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
- if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset
- DWORD endWrite = nextWritePos + buffer_bytes;
+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+ bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
+ bufferBytes *= formatBytes( stream_.userFormat );
+ memset( stream_.userBuffer[0], 0, bufferBytes );
+ }
- // Check whether the entire write region is behind the play pointer.
- while ( currentPos < endWrite ) {
- float millis = (endWrite - currentPos) * 900.0;
- millis /= ( formatBytes(stream->deviceFormat[0]) * stream->sampleRate);
- if ( millis < 1.0 ) millis = 1.0;
- Sleep( (DWORD) millis );
+ // Setup parameters and do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[0];
+ bufferBytes *= formatBytes( stream_.deviceFormat[0] );
+ }
+ else {
+ buffer = stream_.userBuffer[0];
+ bufferBytes = stream_.bufferSize * stream_.nUserChannels[0];
+ bufferBytes *= formatBytes( stream_.userFormat );
+ }
- // Wake up, find out where we are now
- result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos );
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
- if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset
- }
+ // No byte swapping necessary in DirectSound implementation.
- // Lock free space in the buffer
- result = dsBuffer->Lock (nextWritePos, buffer_bytes, &buffer1,
- &bufferSize1, &buffer2, &bufferSize2, 0);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to lock DS buffer during playback (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
+ // Ahhh ... windoze. 16-bit data is signed but 8-bit data is
+ // unsigned. So, we need to convert our signed 8-bit data here to
+ // unsigned.
+ if ( stream_.deviceFormat[0] == RTAUDIO_SINT8 )
+ for ( int i=0; i<bufferBytes; i++ ) buffer[i] = (unsigned char) ( buffer[i] + 128 );
- // Zero the free space
- ZeroMemory(buffer1, bufferSize1);
- if (buffer2 != NULL) ZeroMemory(buffer2, bufferSize2);
+ DWORD dsBufferSize = handle->dsBufferSize[0];
+ nextWritePos = handle->bufferPointer[0];
- // Update our buffer offset and unlock sound buffer
- dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to unlock DS buffer during playback (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ DWORD endWrite;
+ while ( true ) {
+ // Find out where the read and "safe write" pointers are.
+ result = dsBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ error( RtError::SYSTEM_ERROR );
}
- nextWritePos = (nextWritePos + bufferSize1 + bufferSize2) % dsBufferSize;
- stream->handle[0].bufferPointer = nextWritePos;
- }
- // If we play again, start at the beginning of the buffer.
- stream->handle[0].bufferPointer = 0;
- }
+ leadPos = safeWritePos + handle->dsPointerLeadTime[0];
+ if ( leadPos > dsBufferSize ) leadPos -= dsBufferSize;
+ if ( leadPos < nextWritePos ) leadPos += dsBufferSize; // unwrap offset
+ endWrite = nextWritePos + bufferBytes;
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
- buffer1 = NULL;
- bufferSize1 = 0;
+ // Check whether the entire write region is behind the play pointer.
+ if ( leadPos >= endWrite ) break;
- result = buffer->Stop();
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to stop DS capture buffer (%s): %s",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ // If we are here, then we must wait until the play pointer gets
+ // beyond the write region. The approach here is to use the
+ // Sleep() function to suspend operation until safePos catches
+ // up. Calculate number of milliseconds to wait as:
+ // time = distance * (milliseconds/second) * fudgefactor /
+ // ((bytes/sample) * (samples/second))
+ // A "fudgefactor" less than 1 is used because it was found
+ // that sleeping too long was MUCH worse than sleeping for
+ // several shorter periods.
+ double millis = ( endWrite - leadPos ) * 900.0;
+ millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
+ if ( millis < 1.0 ) millis = 1.0;
+ if ( millis > 50.0 ) {
+ static int nOverruns = 0;
+ ++nOverruns;
+ }
+ Sleep( (DWORD) millis );
}
- dsBufferSize = stream->bufferSize * stream->nDeviceChannels[1];
- dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers;
+ //if ( statistics.writeDeviceSafeLeadBytes < dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ) ) {
+ // statistics.writeDeviceSafeLeadBytes = dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] );
+ //}
- // Lock the buffer and clear it so that if we start to play again,
- // we won't have old data playing.
- result = buffer->Lock(0, dsBufferSize, &buffer1, &bufferSize1, NULL, NULL, 0);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ if ( dsPointerBetween( nextWritePos, safeWritePos, currentWritePos, dsBufferSize )
+ || dsPointerBetween( endWrite, safeWritePos, currentWritePos, dsBufferSize ) ) {
+ // We've strayed into the forbidden zone ... resync the read pointer.
+ //++statistics.numberOfWriteUnderruns;
+ handle->xrun[0] = true;
+ nextWritePos = safeWritePos + handle->dsPointerLeadTime[0] - bufferBytes + dsBufferSize;
+ while ( nextWritePos >= dsBufferSize ) nextWritePos -= dsBufferSize;
+ handle->bufferPointer[0] = nextWritePos;
+ endWrite = nextWritePos + bufferBytes;
}
- // Zero the DS buffer
- ZeroMemory(buffer1, bufferSize1);
+ // Lock free space in the buffer
+ result = dsBuffer->Lock( nextWritePos, bufferBytes, &buffer1,
+ &bufferSize1, &buffer2, &bufferSize2, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
+ errorText_ = errorStream_.str();
+ error( RtError::SYSTEM_ERROR );
+ }
- // Unlock the DS buffer
- result = buffer->Unlock(buffer1, bufferSize1, NULL, 0);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ // Copy our buffer into the DS buffer
+ CopyMemory( buffer1, buffer, bufferSize1 );
+ if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
+
+ // Update our buffer offset and unlock sound buffer
+ dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
+ errorText_ = errorStream_.str();
+ error( RtError::SYSTEM_ERROR );
}
+ nextWritePos = ( nextWritePos + bufferSize1 + bufferSize2 ) % dsBufferSize;
+ handle->bufferPointer[0] = nextWritePos;
- // If we start recording again, we must begin at beginning of buffer.
- stream->handle[1].bufferPointer = 0;
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
+ }
}
- stream->state = STREAM_STOPPED;
- MUTEX_UNLOCK(&stream->mutex);
-}
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-void RtAudio :: abortStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ // Setup parameters.
+ if ( stream_.doConvertBuffer[1] ) {
+ buffer = stream_.deviceBuffer;
+ bufferBytes = stream_.bufferSize * stream_.nDeviceChannels[1];
+ bufferBytes *= formatBytes( stream_.deviceFormat[1] );
+ }
+ else {
+ buffer = stream_.userBuffer[1];
+ bufferBytes = stream_.bufferSize * stream_.nUserChannels[1];
+ bufferBytes *= formatBytes( stream_.userFormat );
+ }
- MUTEX_LOCK(&stream->mutex);
+ LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ long nextReadPos = handle->bufferPointer[1];
+ DWORD dsBufferSize = handle->dsBufferSize[1];
- if (stream->state == STREAM_STOPPED)
- goto unlock;
+ // Find out where the write and "safe read" pointers are.
+ result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ error( RtError::SYSTEM_ERROR );
+ }
+
+ if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset
+ DWORD endRead = nextReadPos + bufferBytes;
+
+ // Handling depends on whether we are INPUT or DUPLEX.
+ // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
+ // then a wait here will drag the write pointers into the forbidden zone.
+ //
+ // In DUPLEX mode, rather than wait, we will back off the read pointer until
+ // it's in a safe position. This causes dropouts, but it seems to be the only
+ // practical way to sync up the read and write pointers reliably, given the
+ // the very complex relationship between phase and increment of the read and write
+ // pointers.
+ //
+ // In order to minimize audible dropouts in DUPLEX mode, we will
+ // provide a pre-roll period of 0.5 seconds in which we return
+ // zeros from the read buffer while the pointers sync up.
+
+ if ( stream_.mode == DUPLEX ) {
+ if ( safeReadPos < endRead ) {
+ if ( duplexPrerollBytes <= 0 ) {
+ // Pre-roll time over. Be more agressive.
+ int adjustment = endRead-safeReadPos;
+
+ handle->xrun[1] = true;
+ //++statistics.numberOfReadOverruns;
+ // Two cases:
+ // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
+ // and perform fine adjustments later.
+ // - small adjustments: back off by twice as much.
+ if ( adjustment >= 2*bufferBytes )
+ nextReadPos = safeReadPos-2*bufferBytes;
+ else
+ nextReadPos = safeReadPos-bufferBytes-adjustment;
- HRESULT result;
- long dsBufferSize;
- LPVOID audioPtr;
- DWORD dataLen;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- result = buffer->Stop();
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to stop DS buffer (%s): %s",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ //statistics.readDeviceSafeLeadBytes = currentReadPos-nextReadPos;
+ //if ( statistics.readDeviceSafeLeadBytes < 0) statistics.readDeviceSafeLeadBytes += dsBufferSize;
+ if ( nextReadPos < 0 ) nextReadPos += dsBufferSize;
+
+ }
+ else {
+ // In pre=roll time. Just do it.
+ nextReadPos = safeReadPos-bufferBytes;
+ while ( nextReadPos < 0 ) nextReadPos += dsBufferSize;
+ }
+ endRead = nextReadPos + bufferBytes;
+ }
+ }
+ else { // mode == INPUT
+ while ( safeReadPos < endRead ) {
+ // See comments for playback.
+ double millis = (endRead - safeReadPos) * 900.0;
+ millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
+ if ( millis < 1.0 ) millis = 1.0;
+ Sleep( (DWORD) millis );
+
+ // Wake up, find out where we are now
+ result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ error( RtError::SYSTEM_ERROR );
+ }
+
+ if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset
+ }
}
- dsBufferSize = stream->bufferSize * stream->nDeviceChannels[0];
- dsBufferSize *= formatBytes(stream->deviceFormat[0]) * stream->nBuffers;
+ //if (statistics.readDeviceSafeLeadBytes < dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ) )
+ // statistics.readDeviceSafeLeadBytes = dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize );
- // Lock the buffer and clear it so that if we start to play again,
- // we won't have old data playing.
- result = buffer->Lock(0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to lock DS buffer (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ // Lock free space in the buffer
+ result = dsBuffer->Lock( nextReadPos, bufferBytes, &buffer1,
+ &bufferSize1, &buffer2, &bufferSize2, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
+ errorText_ = errorStream_.str();
+ error( RtError::SYSTEM_ERROR );
}
- // Zero the DS buffer
- ZeroMemory(audioPtr, dataLen);
+ if ( duplexPrerollBytes <= 0 ) {
+ // Copy our buffer into the DS buffer
+ CopyMemory( buffer, buffer1, bufferSize1 );
+ if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
+ }
+ else {
+ memset( buffer, 0, bufferSize1 );
+ if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
+ duplexPrerollBytes -= bufferSize1 + bufferSize2;
+ }
- // Unlock the DS buffer
- result = buffer->Unlock(audioPtr, dataLen, NULL, 0);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to unlock DS buffer (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ // Update our buffer offset and unlock sound buffer
+ nextReadPos = ( nextReadPos + bufferSize1 + bufferSize2 ) % dsBufferSize;
+ dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
+ errorText_ = errorStream_.str();
+ error( RtError::SYSTEM_ERROR );
}
+ handle->bufferPointer[1] = nextReadPos;
- // If we start playing again, we must begin at beginning of buffer.
- stream->handle[0].bufferPointer = 0;
+ // No byte swapping necessary in DirectSound implementation.
+
+ // If necessary, convert 8-bit data from unsigned to signed.
+ if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
+ for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
+
+ // Do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[1] )
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+ }
+#ifdef GENERATE_DEBUG_LOG
+ if ( currentDebugLogEntry < debugLog.size() )
+ {
+ TTickRecord &r = debugLog[currentDebugLogEntry++];
+ r.currentReadPointer = currentReadPos;
+ r.safeReadPointer = safeReadPos;
+ r.currentWritePointer = currentWritePos;
+ r.safeWritePointer = safeWritePos;
+ r.readTime = readTime;
+ r.writeTime = writeTime;
+ r.nextReadPointer = handles[1].bufferPointer;
+ r.nextWritePointer = handles[0].bufferPointer;
+ }
+#endif
+
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ RtApi::tickStreamTime();
+}
+
+// Definitions for utility functions and callbacks
+// specific to the DirectSound implementation.
+
+extern "C" unsigned __stdcall callbackHandler( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiDs *object = (RtApiDs *) info->object;
+ bool* isRunning = &info->isRunning;
+
+ while ( *isRunning == true ) {
+ object->callbackEvent();
}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
- audioPtr = NULL;
- dataLen = 0;
+ _endthreadex( 0 );
+ return 0;
+}
- result = buffer->Stop();
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to stop DS capture buffer (%s): %s",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
+#include "tchar.h"
- dsBufferSize = stream->bufferSize * stream->nDeviceChannels[1];
- dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers;
+std::string convertTChar( LPCTSTR name )
+{
+ std::string s;
+
+#if defined( UNICODE ) || defined( _UNICODE )
+ // Yes, this conversion doesn't make sense for two-byte characters
+ // but RtAudio is currently written to return an std::string of
+ // one-byte chars for the device name.
+ for ( unsigned int i=0; i<wcslen( name ); i++ )
+ s.push_back( name[i] );
+#else
+ s.append( std::string( name ) );
+#endif
- // Lock the buffer and clear it so that if we start to play again,
- // we won't have old data playing.
- result = buffer->Lock(0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to lock DS capture buffer (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
+ return s;
+}
- // Zero the DS buffer
- ZeroMemory(audioPtr, dataLen);
+static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
+ LPCTSTR description,
+ LPCTSTR module,
+ LPVOID lpContext )
+{
+ EnumInfo *info = (EnumInfo *) lpContext;
- // Unlock the DS buffer
- result = buffer->Unlock(audioPtr, dataLen, NULL, 0);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to unlock DS capture buffer (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ HRESULT hr;
+ if ( info->isInput == true ) {
+ DSCCAPS caps;
+ LPDIRECTSOUNDCAPTURE object;
+
+ hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
+ if ( hr != DS_OK ) return TRUE;
+
+ caps.dwSize = sizeof(caps);
+ hr = object->GetCaps( &caps );
+ if ( hr == DS_OK ) {
+ if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
+ info->counter++;
+ }
+ object->Release();
+ }
+ else {
+ DSCAPS caps;
+ LPDIRECTSOUND object;
+ hr = DirectSoundCreate( lpguid, &object, NULL );
+ if ( hr != DS_OK ) return TRUE;
+
+ caps.dwSize = sizeof(caps);
+ hr = object->GetCaps( &caps );
+ if ( hr == DS_OK ) {
+ if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
+ info->counter++;
+ }
+ object->Release();
+ }
+
+ if ( info->getDefault && lpguid == NULL ) return FALSE;
+
+ if ( info->findIndex && info->counter > info->index ) {
+ info->id = lpguid;
+ info->name = convertTChar( description );
+ return FALSE;
+ }
+
+ return TRUE;
+}
+
+static char* getErrorString( int code )
+{
+ switch ( code ) {
+
+ case DSERR_ALLOCATED:
+ return "Already allocated";
+
+ case DSERR_CONTROLUNAVAIL:
+ return "Control unavailable";
+
+ case DSERR_INVALIDPARAM:
+ return "Invalid parameter";
+
+ case DSERR_INVALIDCALL:
+ return "Invalid call";
+
+ case DSERR_GENERIC:
+ return "Generic error";
+
+ case DSERR_PRIOLEVELNEEDED:
+ return "Priority level needed";
+
+ case DSERR_OUTOFMEMORY:
+ return "Out of memory";
+
+ case DSERR_BADFORMAT:
+ return "The sample rate or the channel format is not supported";
+
+ case DSERR_UNSUPPORTED:
+ return "Not supported";
+
+ case DSERR_NODRIVER:
+ return "No driver";
+
+ case DSERR_ALREADYINITIALIZED:
+ return "Already initialized";
+
+ case DSERR_NOAGGREGATION:
+ return "No aggregation";
+
+ case DSERR_BUFFERLOST:
+ return "Buffer lost";
+
+ case DSERR_OTHERAPPHASPRIO:
+ return "Another application already has priority";
+
+ case DSERR_UNINITIALIZED:
+ return "Uninitialized";
+
+ default:
+ return "DirectSound unknown error";
+ }
+}
+//******************** End of __WINDOWS_DS__ *********************//
+#endif
+
+
+#if defined(__LINUX_ALSA__)
+
+#include <alsa/asoundlib.h>
+#include <unistd.h>
+
+// A structure to hold various information related to the ALSA API
+// implementation.
+struct AlsaHandle {
+ snd_pcm_t *handles[2];
+ bool synchronized;
+ bool xrun[2];
+
+ AlsaHandle()
+ :synchronized(false) { xrun[0] = false; xrun[1] = false; }
+};
+
+extern "C" void *alsaCallbackHandler( void * ptr );
+
+RtApiAlsa :: RtApiAlsa()
+{
+ // Nothing to do here.
+}
+
+RtApiAlsa :: ~RtApiAlsa()
+{
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+}
+
+unsigned int RtApiAlsa :: getDeviceCount( void )
+{
+ unsigned nDevices = 0;
+ int result, subdevice, card;
+ char name[64];
+ snd_ctl_t *handle;
+
+ // Count cards and devices
+ card = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &handle, name, 0 );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ goto nextcard;
+ }
+ subdevice = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( handle, &subdevice );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ break;
+ }
+ if ( subdevice < 0 )
+ break;
+ nDevices++;
+ }
+ nextcard:
+ snd_ctl_close( handle );
+ snd_card_next( &card );
+ }
+
+ return nDevices;
+}
+
+RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ unsigned nDevices = 0;
+ int result, subdevice, card;
+ char name[64];
+ snd_ctl_t *chandle;
+
+ // Count cards and devices
+ card = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ goto nextcard;
+ }
+ subdevice = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( chandle, &subdevice );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ break;
+ }
+ if ( subdevice < 0 ) break;
+ if ( nDevices == device ) {
+ sprintf( name, "hw:%d,%d", card, subdevice );
+ goto foundDevice;
+ }
+ nDevices++;
+ }
+ nextcard:
+ snd_ctl_close( chandle );
+ snd_card_next( &card );
+ }
+
+ if ( nDevices == 0 ) {
+ errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
+ error( RtError::INVALID_USE );
+ }
+
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
+ error( RtError::INVALID_USE );
+ }
+
+ foundDevice:
+
+ // If a stream is already open, we cannot probe the stream devices.
+ // Thus, use the saved results.
+ if ( stream_.state != STREAM_CLOSED &&
+ ( stream_.device[0] == device || stream_.device[1] == device ) ) {
+ if ( device >= devices_.size() ) {
+ errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
+ error( RtError::WARNING );
+ return info;
+ }
+ return devices_[ device ];
+ }
+
+ int openMode = SND_PCM_ASYNC;
+ snd_pcm_stream_t stream;
+ snd_pcm_info_t *pcminfo;
+ snd_pcm_info_alloca( &pcminfo );
+ snd_pcm_t *phandle;
+ snd_pcm_hw_params_t *params;
+ snd_pcm_hw_params_alloca( ¶ms );
+
+ // First try for playback
+ stream = SND_PCM_STREAM_PLAYBACK;
+ snd_pcm_info_set_device( pcminfo, subdevice );
+ snd_pcm_info_set_subdevice( pcminfo, 0 );
+ snd_pcm_info_set_stream( pcminfo, stream );
+
+ result = snd_ctl_pcm_info( chandle, pcminfo );
+ if ( result < 0 ) {
+ // Device probably doesn't support playback.
+ goto captureProbe;
+ }
+
+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ goto captureProbe;
+ }
+
+ // The device is open ... fill the parameter structure.
+ result = snd_pcm_hw_params_any( phandle, params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ goto captureProbe;
+ }
+
+ // Get output channel information.
+ unsigned int value;
+ result = snd_pcm_hw_params_get_channels_max( params, &value );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ goto captureProbe;
+ }
+ info.outputChannels = value;
+ snd_pcm_close( phandle );
+
+ captureProbe:
+ // Now try for capture
+ stream = SND_PCM_STREAM_CAPTURE;
+ snd_pcm_info_set_stream( pcminfo, stream );
+
+ result = snd_ctl_pcm_info( chandle, pcminfo );
+ snd_ctl_close( chandle );
+ if ( result < 0 ) {
+ // Device probably doesn't support capture.
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
+
+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
+
+ // The device is open ... fill the parameter structure.
+ result = snd_pcm_hw_params_any( phandle, params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
+
+ result = snd_pcm_hw_params_get_channels_max( params, &value );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
+ info.inputChannels = value;
+ snd_pcm_close( phandle );
+
+ // If device opens for both playback and capture, we determine the channels.
+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+
+ // ALSA doesn't provide default devices so we'll use the first available one.
+ if ( device == 0 && info.outputChannels > 0 )
+ info.isDefaultOutput = true;
+ if ( device == 0 && info.inputChannels > 0 )
+ info.isDefaultInput = true;
+
+ probeParameters:
+ // At this point, we just need to figure out the supported data
+ // formats and sample rates. We'll proceed by opening the device in
+ // the direction with the maximum number of channels, or playback if
+ // they are equal. This might limit our sample rate options, but so
+ // be it.
+
+ if ( info.outputChannels >= info.inputChannels )
+ stream = SND_PCM_STREAM_PLAYBACK;
+ else
+ stream = SND_PCM_STREAM_CAPTURE;
+ snd_pcm_info_set_stream( pcminfo, stream );
+
+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
+ }
+
+ // The device is open ... fill the parameter structure.
+ result = snd_pcm_hw_params_any( phandle, params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
+ }
+
+ // Test our discrete set of sample rate values.
+ info.sampleRates.clear();
+ for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
+ if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 )
+ info.sampleRates.push_back( SAMPLE_RATES[i] );
+ }
+ if ( info.sampleRates.size() == 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
+ }
+
+ // Probe the supported data formats ... we don't care about endian-ness just yet
+ snd_pcm_format_t format;
+ info.nativeFormats = 0;
+ format = SND_PCM_FORMAT_S8;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_SINT8;
+ format = SND_PCM_FORMAT_S16;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_SINT16;
+ format = SND_PCM_FORMAT_S24;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_SINT24;
+ format = SND_PCM_FORMAT_S32;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_SINT32;
+ format = SND_PCM_FORMAT_FLOAT;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+ format = SND_PCM_FORMAT_FLOAT64;
+ if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
+ info.nativeFormats |= RTAUDIO_FLOAT64;
+
+ // Check that we have at least one supported format
+ if ( info.nativeFormats == 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
+ }
+
+ // Get the device name
+ char *cardname;
+ result = snd_card_get_name( card, &cardname );
+ if ( result >= 0 )
+ sprintf( name, "hw:%s,%d", cardname, subdevice );
+ info.name = name;
+
+ // That's all ... close the device and return
+ snd_pcm_close( phandle );
+ info.probed = true;
+ return info;
+}
+
+void RtApiAlsa :: saveDeviceInfo( void )
+{
+ devices_.clear();
+
+ unsigned int nDevices = getDeviceCount();
+ devices_.resize( nDevices );
+ for ( unsigned int i=0; i<nDevices; i++ )
+ devices_[i] = getDeviceInfo( i );
+}
+
+bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+
+{
+#if defined(__RTAUDIO_DEBUG__)
+ snd_output_t *out;
+ snd_output_stdio_attach(&out, stderr, 0);
+#endif
+
+ // I'm not using the "plug" interface ... too much inconsistent behavior.
+
+ unsigned nDevices = 0;
+ int result, subdevice, card;
+ char name[64];
+ snd_ctl_t *chandle;
+
+ // Count cards and devices
+ card = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ subdevice = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( chandle, &subdevice );
+ if ( result < 0 ) break;
+ if ( subdevice < 0 ) break;
+ if ( nDevices == device ) {
+ sprintf( name, "hw:%d,%d", card, subdevice );
+ snd_ctl_close( chandle );
+ goto foundDevice;
+ }
+ nDevices++;
+ }
+ snd_ctl_close( chandle );
+ snd_card_next( &card );
+ }
+
+ if ( nDevices == 0 ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
+ return FAILURE;
+ }
+
+ if ( device >= nDevices ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
+
+ foundDevice:
+
+ // The getDeviceInfo() function will not work for a device that is
+ // already open. Thus, we'll probe the system before opening a
+ // stream and save the results for use by getDeviceInfo().
+ if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
+ this->saveDeviceInfo();
+
+ snd_pcm_stream_t stream;
+ if ( mode == OUTPUT )
+ stream = SND_PCM_STREAM_PLAYBACK;
+ else
+ stream = SND_PCM_STREAM_CAPTURE;
+
+ snd_pcm_t *phandle;
+ int openMode = SND_PCM_ASYNC;
+ result = snd_pcm_open( &phandle, name, stream, openMode );
+ if ( result < 0 ) {
+ if ( mode == OUTPUT )
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
+ else
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Fill the parameter structure.
+ snd_pcm_hw_params_t *hw_params;
+ snd_pcm_hw_params_alloca( &hw_params );
+ result = snd_pcm_hw_params_any( phandle, hw_params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+#if defined(__RTAUDIO_DEBUG__)
+ fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
+ snd_pcm_hw_params_dump( hw_params, out );
+#endif
+
+ // Set access ... check user preference.
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
+ stream_.userInterleaved = false;
+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
+ if ( result < 0 ) {
+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
+ stream_.deviceInterleaved[mode] = true;
+ }
+ else
+ stream_.deviceInterleaved[mode] = false;
+ }
+ else {
+ stream_.userInterleaved = true;
+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
+ if ( result < 0 ) {
+ result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
+ stream_.deviceInterleaved[mode] = false;
+ }
+ else
+ stream_.deviceInterleaved[mode] = true;
+ }
+
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Determine how to set the device format.
+ stream_.userFormat = format;
+ snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
+
+ if ( format == RTAUDIO_SINT8 )
+ deviceFormat = SND_PCM_FORMAT_S8;
+ else if ( format == RTAUDIO_SINT16 )
+ deviceFormat = SND_PCM_FORMAT_S16;
+ else if ( format == RTAUDIO_SINT24 )
+ deviceFormat = SND_PCM_FORMAT_S24;
+ else if ( format == RTAUDIO_SINT32 )
+ deviceFormat = SND_PCM_FORMAT_S32;
+ else if ( format == RTAUDIO_FLOAT32 )
+ deviceFormat = SND_PCM_FORMAT_FLOAT;
+ else if ( format == RTAUDIO_FLOAT64 )
+ deviceFormat = SND_PCM_FORMAT_FLOAT64;
+
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
+ stream_.deviceFormat[mode] = format;
+ goto setFormat;
+ }
+
+ // The user requested format is not natively supported by the device.
+ deviceFormat = SND_PCM_FORMAT_FLOAT64;
+ if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
+ goto setFormat;
+ }
+
+ deviceFormat = SND_PCM_FORMAT_FLOAT;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ goto setFormat;
+ }
+
+ deviceFormat = SND_PCM_FORMAT_S32;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ goto setFormat;
+ }
+
+ deviceFormat = SND_PCM_FORMAT_S24;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ goto setFormat;
+ }
+
+ deviceFormat = SND_PCM_FORMAT_S16;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ goto setFormat;
+ }
+
+ deviceFormat = SND_PCM_FORMAT_S8;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ goto setFormat;
+ }
+
+ // If we get here, no supported format was found.
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+
+ setFormat:
+ result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Determine whether byte-swaping is necessary.
+ stream_.doByteSwap[mode] = false;
+ if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
+ result = snd_pcm_format_cpu_endian( deviceFormat );
+ if ( result == 0 )
+ stream_.doByteSwap[mode] = true;
+ else if (result < 0) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // Set the sample rate.
+ result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Determine the number of channels for this device. We support a possible
+ // minimum device channel number > than the value requested by the user.
+ stream_.nUserChannels[mode] = channels;
+ unsigned int value;
+ result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
+ unsigned int deviceChannels = value;
+ if ( result < 0 || deviceChannels < channels + firstChannel ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ deviceChannels = value;
+ if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
+ stream_.nDeviceChannels[mode] = deviceChannels;
+
+ // Set the device channels.
+ result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Set the buffer number, which in ALSA is referred to as the "period".
+ int dir;
+ unsigned int periods = 0;
+ if ( options ) periods = options->numberOfBuffers;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
+ // Even though the hardware might allow 1 buffer, it won't work reliably.
+ if ( periods < 2 ) periods = 2;
+ result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Set the buffer (or period) size.
+ snd_pcm_uframes_t periodSize = *bufferSize;
+ result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ *bufferSize = periodSize;
+
+ // If attempting to setup a duplex stream, the bufferSize parameter
+ // MUST be the same in both directions!
+ if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ stream_.bufferSize = *bufferSize;
+
+ // Install the hardware configuration
+ result = snd_pcm_hw_params( phandle, hw_params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+#if defined(__RTAUDIO_DEBUG__)
+ fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
+ snd_pcm_hw_params_dump( hw_params, out );
+#endif
+
+ // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
+ snd_pcm_sw_params_t *sw_params = NULL;
+ snd_pcm_sw_params_alloca( &sw_params );
+ snd_pcm_sw_params_current( phandle, sw_params );
+ snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
+ snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, 0x7fffffff );
+ snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
+ snd_pcm_sw_params_set_silence_size( phandle, sw_params, INT_MAX );
+ result = snd_pcm_sw_params( phandle, sw_params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+#if defined(__RTAUDIO_DEBUG__)
+ fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
+ snd_pcm_sw_params_dump( sw_params, out );
+#endif
+
+ // Set flags for buffer conversion
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate the ApiHandle if necessary and then save.
+ AlsaHandle *apiInfo = 0;
+ if ( stream_.apiHandle == 0 ) {
+ try {
+ apiInfo = (AlsaHandle *) new AlsaHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
+ goto error;
+ }
+ stream_.apiHandle = (void *) apiInfo;
+ apiInfo->handles[0] = 0;
+ apiInfo->handles[1] = 0;
+ }
+ else {
+ apiInfo = (AlsaHandle *) stream_.apiHandle;
+ }
+ apiInfo->handles[mode] = phandle;
+
+ // Allocate necessary internal buffers.
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
+ }
+ }
+
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
+
+ stream_.sampleRate = sampleRate;
+ stream_.nBuffers = periods;
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+
+ // Setup thread if necessary.
+ if ( stream_.mode == OUTPUT && mode == INPUT ) {
+ // We had already set up an output stream.
+ stream_.mode = DUPLEX;
+ // Link the streams if possible.
+ apiInfo->synchronized = false;
+ if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
+ apiInfo->synchronized = true;
+ else {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
+ error( RtError::WARNING );
+ }
+ }
+ else {
+ stream_.mode = mode;
+
+ // Setup callback thread.
+ stream_.callbackInfo.object = (void *) this;
+
+ // Set the thread attributes for joinable and realtime scheduling
+ // priority. The higher priority will only take affect if the
+ // program is run as root or suid.
+ pthread_attr_t attr;
+ pthread_attr_init( &attr );
+ pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
+#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
+ pthread_attr_setschedpolicy( &attr, SCHED_RR );
+#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;
+ }
+ }
+
+ return SUCCESS;
+
+ error:
+ if ( apiInfo ) {
+ if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
+ if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
+ delete apiInfo;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ return FAILURE;
+}
+
+void RtApiAlsa :: closeStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
+ error( RtError::WARNING );
+ return;
+ }
+
+ stream_.callbackInfo.isRunning = false;
+ pthread_join( stream_.callbackInfo.thread, NULL );
+
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ if ( stream_.state == STREAM_RUNNING ) {
+ stream_.state = STREAM_STOPPED;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
+ snd_pcm_drop( apiInfo->handles[0] );
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
+ snd_pcm_drop( apiInfo->handles[1] );
+ }
+
+ if ( apiInfo ) {
+ if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
+ if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
+ delete apiInfo;
+ stream_.apiHandle = 0;
+ }
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
+
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
+
+void RtApiAlsa :: startStream()
+{
+ // This method calls snd_pcm_prepare if the device isn't already in that state.
+
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
+ error( RtError::WARNING );
+ return;
+ }
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ int result = 0;
+ snd_pcm_state_t state;
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ state = snd_pcm_state( handle[0] );
+ if ( state != SND_PCM_STATE_PREPARED ) {
+ result = snd_pcm_prepare( handle[0] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
}
+ }
- // If we start recording again, we must begin at beginning of buffer.
- stream->handle[1].bufferPointer = 0;
+ if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+ state = snd_pcm_state( handle[1] );
+ if ( state != SND_PCM_STATE_PREPARED ) {
+ result = snd_pcm_prepare( handle[1] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
}
- stream->state = STREAM_STOPPED;
+
+ stream_.state = STREAM_RUNNING;
unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result >= 0 ) return;
+ error( RtError::SYSTEM_ERROR );
}
-int RtAudio :: streamWillBlock(int streamId)
+void RtApiAlsa :: stopStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
-
- int channels;
- int frames = 0;
- if (stream->state == STREAM_STOPPED)
- goto unlock;
-
- HRESULT result;
- DWORD currentPos, safePos;
- channels = 1;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
+ }
- LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- UINT nextWritePos = stream->handle[0].bufferPointer;
- channels = stream->nDeviceChannels[0];
- DWORD dsBufferSize = stream->bufferSize * channels;
- dsBufferSize *= formatBytes(stream->deviceFormat[0]) * stream->nBuffers;
+ // Change the state before the lock to improve shutdown response
+ // when using a callback.
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK( &stream_.mutex );
- // Find out where the read and "safe write" pointers are.
- result = dsBuffer->GetCurrentPosition(¤tPos, &safePos);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ int result = 0;
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( apiInfo->synchronized )
+ result = snd_pcm_drop( handle[0] );
+ else
+ result = snd_pcm_drain( handle[0] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::stopStream: error draining output pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
+ }
- if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset
- frames = currentPos - nextWritePos;
- frames /= channels * formatBytes(stream->deviceFormat[0]);
+ if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+ result = snd_pcm_drop( handle[1] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::stopStream: error stopping input pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
- LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
- UINT nextReadPos = stream->handle[1].bufferPointer;
- channels = stream->nDeviceChannels[1];
- DWORD dsBufferSize = stream->bufferSize * channels;
- dsBufferSize *= formatBytes(stream->deviceFormat[1]) * stream->nBuffers;
+ if ( result >= 0 ) return;
+ error( RtError::SYSTEM_ERROR );
+}
- // Find out where the write and "safe read" pointers are.
- result = dsBuffer->GetCurrentPosition(¤tPos, &safePos);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
+void RtApiAlsa :: abortStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
+ }
- if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset
+ // Change the state before the lock to improve shutdown response
+ // when using a callback.
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK( &stream_.mutex );
- if (stream->mode == DUPLEX ) {
- // Take largest value of the two.
- int temp = safePos - nextReadPos;
- temp /= channels * formatBytes(stream->deviceFormat[1]);
- frames = ( temp > frames ) ? temp : frames;
- }
- else {
- frames = safePos - nextReadPos;
- frames /= channels * formatBytes(stream->deviceFormat[1]);
+ int result = 0;
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ result = snd_pcm_drop( handle[0] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
}
- frames = stream->bufferSize - frames;
- if (frames < 0) frames = 0;
+ if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+ result = snd_pcm_drop( handle[1] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
unlock:
- MUTEX_UNLOCK(&stream->mutex);
- return frames;
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ stream_.state = STREAM_STOPPED;
+ if ( result >= 0 ) return;
+ error( RtError::SYSTEM_ERROR );
}
-void RtAudio :: tickStream(int streamId)
+void RtApiAlsa :: callbackEvent()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ if ( stream_.state == STREAM_STOPPED ) {
+ if ( stream_.callbackInfo.isRunning ) usleep( 50000 ); // sleep 50 milliseconds
+ return;
+ }
- int stopStream = 0;
- if (stream->state == STREAM_STOPPED) {
- if (stream->callbackInfo.usingCallback) Sleep(50); // sleep 50 milliseconds
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtError::WARNING );
return;
}
- else if (stream->callbackInfo.usingCallback) {
- RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback;
- stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData);
+
+ int doStopStream = 0;
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ apiInfo->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ apiInfo->xrun[1] = false;
}
+ doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK( &stream_.mutex );
// The state might change while waiting on a mutex.
- if (stream->state == STREAM_STOPPED) {
- MUTEX_UNLOCK(&stream->mutex);
- return;
- }
+ if ( stream_.state == STREAM_STOPPED ) goto unlock;
- HRESULT result;
- DWORD currentPos, safePos;
- LPVOID buffer1 = NULL;
- LPVOID buffer2 = NULL;
- DWORD bufferSize1 = 0;
- DWORD bufferSize2 = 0;
+ int result;
char *buffer;
- long buffer_bytes;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
+ int channels;
+ snd_pcm_t **handle;
+ snd_pcm_sframes_t frames;
+ RtAudioFormat format;
+ handle = (snd_pcm_t **) apiInfo->handles;
- // Setup parameters and do buffer conversion if necessary.
- if (stream->doConvertBuffer[0]) {
- convertStreamBuffer(stream, OUTPUT);
- buffer = stream->deviceBuffer;
- buffer_bytes = stream->bufferSize * stream->nDeviceChannels[0];
- buffer_bytes *= formatBytes(stream->deviceFormat[0]);
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ // Setup parameters.
+ if ( stream_.doConvertBuffer[1] ) {
+ buffer = stream_.deviceBuffer;
+ channels = stream_.nDeviceChannels[1];
+ format = stream_.deviceFormat[1];
}
else {
- buffer = stream->userBuffer;
- buffer_bytes = stream->bufferSize * stream->nUserChannels[0];
- buffer_bytes *= formatBytes(stream->userFormat);
+ buffer = stream_.userBuffer[1];
+ channels = stream_.nUserChannels[1];
+ format = stream_.userFormat;
}
- // No byte swapping necessary in DirectSound implementation.
-
- LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) stream->handle[0].buffer;
- UINT nextWritePos = stream->handle[0].bufferPointer;
- DWORD dsBufferSize = buffer_bytes * stream->nBuffers;
-
- // Find out where the read and "safe write" pointers are.
- result = dsBuffer->GetCurrentPosition(¤tPos, &safePos);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ // Read samples from device in interleaved/non-interleaved format.
+ if ( stream_.deviceInterleaved[1] )
+ result = snd_pcm_readi( handle[1], buffer, stream_.bufferSize );
+ else {
+ void *bufs[channels];
+ size_t offset = stream_.bufferSize * formatBytes( format );
+ for ( int i=0; i<channels; i++ )
+ bufs[i] = (void *) (buffer + (i * offset));
+ result = snd_pcm_readn( handle[1], bufs, stream_.bufferSize );
}
- if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset
- DWORD endWrite = nextWritePos + buffer_bytes;
-
- // Check whether the entire write region is behind the play pointer.
- while ( currentPos < endWrite ) {
- // If we are here, then we must wait until the play pointer gets
- // beyond the write region. The approach here is to use the
- // Sleep() function to suspend operation until safePos catches
- // up. Calculate number of milliseconds to wait as:
- // time = distance * (milliseconds/second) * fudgefactor /
- // ((bytes/sample) * (samples/second))
- // A "fudgefactor" less than 1 is used because it was found
- // that sleeping too long was MUCH worse than sleeping for
- // several shorter periods.
- float millis = (endWrite - currentPos) * 900.0;
- millis /= ( formatBytes(stream->deviceFormat[0]) * stream->sampleRate);
- if ( millis < 1.0 ) millis = 1.0;
- Sleep( (DWORD) millis );
-
- // Wake up, find out where we are now
- result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos );
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS position (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ if ( result < (int) stream_.bufferSize ) {
+ // Either an error or underrun occured.
+ if ( result == -EPIPE ) {
+ snd_pcm_state_t state = snd_pcm_state( handle[1] );
+ if ( state == SND_PCM_STATE_XRUN ) {
+ apiInfo->xrun[1] = true;
+ result = snd_pcm_prepare( handle[1] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ }
+ else {
+ errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ }
+ else {
+ errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
}
- if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset
+ error( RtError::WARNING );
+ goto unlock;
}
- // Lock free space in the buffer
- result = dsBuffer->Lock (nextWritePos, buffer_bytes, &buffer1,
- &bufferSize1, &buffer2, &bufferSize2, 0);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to lock DS buffer during playback (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
+ // Do byte swapping if necessary.
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
- // Copy our buffer into the DS buffer
- CopyMemory(buffer1, buffer, bufferSize1);
- if (buffer2 != NULL) CopyMemory(buffer2, buffer+bufferSize1, bufferSize2);
+ // Do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[1] )
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
- // Update our buffer offset and unlock sound buffer
- dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to unlock DS buffer during playback (%s): %s.",
- devices[stream->device[0]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
- nextWritePos = (nextWritePos + bufferSize1 + bufferSize2) % dsBufferSize;
- stream->handle[0].bufferPointer = nextWritePos;
+ // Check stream latency
+ result = snd_pcm_delay( handle[1], &frames );
+ if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- // Setup parameters.
- if (stream->doConvertBuffer[1]) {
- buffer = stream->deviceBuffer;
- buffer_bytes = stream->bufferSize * stream->nDeviceChannels[1];
- buffer_bytes *= formatBytes(stream->deviceFormat[1]);
+ // Setup parameters and do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ channels = stream_.nDeviceChannels[0];
+ format = stream_.deviceFormat[0];
}
else {
- buffer = stream->userBuffer;
- buffer_bytes = stream->bufferSize * stream->nUserChannels[1];
- buffer_bytes *= formatBytes(stream->userFormat);
+ buffer = stream_.userBuffer[0];
+ channels = stream_.nUserChannels[0];
+ format = stream_.userFormat;
}
- LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) stream->handle[1].buffer;
- UINT nextReadPos = stream->handle[1].bufferPointer;
- DWORD dsBufferSize = buffer_bytes * stream->nBuffers;
+ // Do byte swapping if necessary.
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
- // Find out where the write and "safe read" pointers are.
- result = dsBuffer->GetCurrentPosition(¤tPos, &safePos);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ // Write samples to device in interleaved/non-interleaved format.
+ if ( stream_.deviceInterleaved[0] )
+ result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
+ else {
+ void *bufs[channels];
+ size_t offset = stream_.bufferSize * formatBytes( format );
+ for ( int i=0; i<channels; i++ )
+ bufs[i] = (void *) (buffer + (i * offset));
+ result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
}
- if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset
- DWORD endRead = nextReadPos + buffer_bytes;
-
- // Check whether the entire write region is behind the play pointer.
- while ( safePos < endRead ) {
- // See comments for playback.
- float millis = (endRead - safePos) * 900.0;
- millis /= ( formatBytes(stream->deviceFormat[1]) * stream->sampleRate);
- if ( millis < 1.0 ) millis = 1.0;
- Sleep( (DWORD) millis );
-
- // Wake up, find out where we are now
- result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos );
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to get current DS capture position (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ if ( result < (int) stream_.bufferSize ) {
+ // Either an error or underrun occured.
+ if ( result == -EPIPE ) {
+ snd_pcm_state_t state = snd_pcm_state( handle[0] );
+ if ( state == SND_PCM_STATE_XRUN ) {
+ apiInfo->xrun[0] = true;
+ result = snd_pcm_prepare( handle[0] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ }
+ else {
+ errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
}
-
- if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset
- }
-
- // Lock free space in the buffer
- result = dsBuffer->Lock (nextReadPos, buffer_bytes, &buffer1,
- &bufferSize1, &buffer2, &bufferSize2, 0);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to lock DS buffer during capture (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- // Copy our buffer into the DS buffer
- CopyMemory(buffer, buffer1, bufferSize1);
- if (buffer2 != NULL) CopyMemory(buffer+bufferSize1, buffer2, bufferSize2);
-
- // Update our buffer offset and unlock sound buffer
- nextReadPos = (nextReadPos + bufferSize1 + bufferSize2) % dsBufferSize;
- dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2);
- if ( FAILED(result) ) {
- sprintf(message, "RtAudio: Unable to unlock DS buffer during capture (%s): %s.",
- devices[stream->device[1]].name, getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ else {
+ errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ error( RtError::WARNING );
+ goto unlock;
}
- stream->handle[1].bufferPointer = nextReadPos;
-
- // No byte swapping necessary in DirectSound implementation.
- // Do buffer conversion if necessary.
- if (stream->doConvertBuffer[1])
- convertStreamBuffer(stream, INPUT);
+ // Check stream latency
+ result = snd_pcm_delay( handle[0], &frames );
+ if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
}
- MUTEX_UNLOCK(&stream->mutex);
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
- if (stream->callbackInfo.usingCallback && stopStream)
- this->stopStream(streamId);
+ RtApi::tickStreamTime();
+ if ( doStopStream == 1 ) this->stopStream();
+ else if ( doStopStream == 2 ) this->abortStream();
}
-// Definitions for utility functions and callbacks
-// specific to the DirectSound implementation.
-
-extern "C" unsigned __stdcall callbackHandler(void *ptr)
+extern "C" void *alsaCallbackHandler( void *ptr )
{
- CALLBACK_INFO *info = (CALLBACK_INFO *) ptr;
- RtAudio *object = (RtAudio *) info->object;
- int stream = info->streamId;
- bool *usingCallback = &info->usingCallback;
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiAlsa *object = (RtApiAlsa *) info->object;
+ bool *isRunning = &info->isRunning;
+
+#ifdef SCHED_RR
+ // Set a higher scheduler priority (P.J. Leonard)
+ struct sched_param param;
+ int min = sched_get_priority_min( SCHED_RR );
+ int max = sched_get_priority_max( SCHED_RR );
+ param.sched_priority = min + ( max - min ) / 2; // Is this the best number?
+ sched_setscheduler( 0, SCHED_RR, ¶m );
+#endif
- while ( *usingCallback ) {
- try {
- object->tickStream(stream);
- }
- catch (RtError &exception) {
- fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n",
- exception.getMessage());
- break;
- }
+ while ( *isRunning == true ) {
+ pthread_testcancel();
+ object->callbackEvent();
}
- _endthreadex( 0 );
- return 0;
+ pthread_exit( NULL );
}
-void RtAudio :: setStreamCallback(int streamId, RTAUDIO_CALLBACK callback, void *userData)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+//******************** End of __LINUX_ALSA__ *********************//
+#endif
- CALLBACK_INFO *info = (CALLBACK_INFO *) &stream->callbackInfo;
- if ( info->usingCallback ) {
- sprintf(message, "RtAudio: A callback is already set for this stream!");
- error(RtError::WARNING);
- return;
- }
- info->callback = (void *) callback;
- info->userData = userData;
- info->usingCallback = true;
- info->object = (void *) this;
- info->streamId = streamId;
+#if defined(__LINUX_OSS__)
- unsigned thread_id;
- info->thread = _beginthreadex(NULL, 0, &callbackHandler,
- &stream->callbackInfo, 0, &thread_id);
- if (info->thread == 0) {
- info->usingCallback = false;
- sprintf(message, "RtAudio: error starting callback thread!");
- error(RtError::THREAD_ERROR);
- }
+#include <unistd.h>
+#include <sys/ioctl.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include "soundcard.h"
+#include <errno.h>
+#include <math.h>
- // When spawning multiple threads in quick succession, it appears to be
- // necessary to wait a bit for each to initialize ... another windoism!
- Sleep(1);
-}
+extern "C" void *ossCallbackHandler(void * ptr);
-static bool CALLBACK deviceCountCallback(LPGUID lpguid,
- LPCSTR lpcstrDescription,
- LPCSTR lpcstrModule,
- LPVOID lpContext)
-{
- int *pointer = ((int *) lpContext);
- (*pointer)++;
+// A structure to hold various information related to the OSS API
+// implementation.
+struct OssHandle {
+ int id[2]; // device ids
+ bool xrun[2];
+ bool triggered;
- return true;
-}
+ OssHandle()
+ :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
+};
-static bool CALLBACK deviceInfoCallback(LPGUID lpguid,
- LPCSTR lpcstrDescription,
- LPCSTR lpcstrModule,
- LPVOID lpContext)
+RtApiOss :: RtApiOss()
{
- enum_info *info = ((enum_info *) lpContext);
- while (strlen(info->name) > 0) info++;
-
- strncpy(info->name, lpcstrDescription, 64);
- info->id = lpguid;
-
- HRESULT hr;
- info->isValid = false;
- if (info->isInput == true) {
- DSCCAPS caps;
- LPDIRECTSOUNDCAPTURE object;
-
- hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
- if( hr != DS_OK ) return true;
-
- caps.dwSize = sizeof(caps);
- hr = object->GetCaps( &caps );
- if( hr == DS_OK ) {
- if (caps.dwChannels > 0 && caps.dwFormats > 0)
- info->isValid = true;
- }
- object->Release();
- }
- else {
- DSCAPS caps;
- LPDIRECTSOUND object;
- hr = DirectSoundCreate( lpguid, &object, NULL );
- if( hr != DS_OK ) return true;
-
- caps.dwSize = sizeof(caps);
- hr = object->GetCaps( &caps );
- if( hr == DS_OK ) {
- if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
- info->isValid = true;
- }
- object->Release();
- }
-
- return true;
+ // Nothing to do here.
}
-static bool CALLBACK defaultDeviceCallback(LPGUID lpguid,
- LPCSTR lpcstrDescription,
- LPCSTR lpcstrModule,
- LPVOID lpContext)
+RtApiOss :: ~RtApiOss()
{
- enum_info *info = ((enum_info *) lpContext);
-
- if ( lpguid == NULL ) {
- strncpy(info->name, lpcstrDescription, 64);
- return false;
- }
-
- return true;
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
}
-static bool CALLBACK deviceIdCallback(LPGUID lpguid,
- LPCSTR lpcstrDescription,
- LPCSTR lpcstrModule,
- LPVOID lpContext)
+unsigned int RtApiOss :: getDeviceCount( void )
{
- enum_info *info = ((enum_info *) lpContext);
+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+ if ( mixerfd == -1 ) {
+ errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
+ error( RtError::WARNING );
+ return 0;
+ }
- if ( strncmp( info->name, lpcstrDescription, 64 ) == 0 ) {
- info->id = lpguid;
- info->isValid = true;
- return false;
+ oss_sysinfo sysinfo;
+ if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
+ error( RtError::WARNING );
+ return 0;
}
- return true;
+ close( mixerfd );
+ return sysinfo.numaudios;
}
-static char* getErrorString(int code)
+RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
{
- switch (code) {
-
- case DSERR_ALLOCATED:
- return "Direct Sound already allocated";
-
- case DSERR_CONTROLUNAVAIL:
- return "Direct Sound control unavailable";
+ RtAudio::DeviceInfo info;
+ info.probed = false;
- case DSERR_INVALIDPARAM:
- return "Direct Sound invalid parameter";
-
- case DSERR_INVALIDCALL:
- return "Direct Sound invalid call";
-
- case DSERR_GENERIC:
- return "Direct Sound generic error";
-
- case DSERR_PRIOLEVELNEEDED:
- return "Direct Sound Priority level needed";
-
- case DSERR_OUTOFMEMORY:
- return "Direct Sound out of memory";
-
- case DSERR_BADFORMAT:
- return "Direct Sound bad format";
-
- case DSERR_UNSUPPORTED:
- return "Direct Sound unsupported error";
+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+ if ( mixerfd == -1 ) {
+ errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
+ error( RtError::WARNING );
+ return info;
+ }
- case DSERR_NODRIVER:
- return "Direct Sound no driver error";
+ oss_sysinfo sysinfo;
+ int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
+ if ( result == -1 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
+ error( RtError::WARNING );
+ return info;
+ }
- case DSERR_ALREADYINITIALIZED:
- return "Direct Sound already initialized";
+ unsigned nDevices = sysinfo.numaudios;
+ if ( nDevices == 0 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
+ error( RtError::INVALID_USE );
+ }
- case DSERR_NOAGGREGATION:
- return "Direct Sound no aggregation";
+ if ( device >= nDevices ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
+ error( RtError::INVALID_USE );
+ }
- case DSERR_BUFFERLOST:
- return "Direct Sound buffer lost";
+ oss_audioinfo ainfo;
+ ainfo.dev = device;
+ result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
+ close( mixerfd );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
+ }
+
+ // Probe channels
+ if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
+ if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
+ if ( ainfo.caps & PCM_CAP_DUPLEX ) {
+ if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+ }
+
+ // Probe data formats ... do for input
+ unsigned long mask = ainfo.iformats;
+ if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
+ info.nativeFormats |= RTAUDIO_SINT16;
+ if ( mask & AFMT_S8 )
+ info.nativeFormats |= RTAUDIO_SINT8;
+ if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
+ info.nativeFormats |= RTAUDIO_SINT32;
+ if ( mask & AFMT_FLOAT )
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+ if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
+ info.nativeFormats |= RTAUDIO_SINT24;
- case DSERR_OTHERAPPHASPRIO:
- return "Direct Sound other app has priority";
+ // Check that we have at least one supported format
+ if ( info.nativeFormats == 0 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ return info;
+ }
+
+ // Probe the supported sample rates.
+ info.sampleRates.clear();
+ if ( ainfo.nrates ) {
+ for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+ if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
+ info.sampleRates.push_back( SAMPLE_RATES[k] );
+ break;
+ }
+ }
+ }
+ }
+ else {
+ // Check min and max rate values;
+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+ if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] )
+ info.sampleRates.push_back( SAMPLE_RATES[k] );
+ }
+ }
- case DSERR_UNINITIALIZED:
- return "Direct Sound uninitialized";
+ if ( info.sampleRates.size() == 0 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ error( RtError::WARNING );
+ }
+ else {
+ info.probed = true;
+ info.name = ainfo.name;
+ }
- default:
- return "Direct Sound unknown error";
- }
+ return info;
}
-//******************** End of __WINDOWS_DS__ *********************//
-
-#elif defined(__IRIX_AL__) // SGI's AL API for IRIX
-#include <unistd.h>
-#include <errno.h>
-
-void RtAudio :: initialize(void)
+bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
{
- // Count cards and devices
- nDevices = 0;
-
- // Determine the total number of input and output devices.
- nDevices = alQueryValues(AL_SYSTEM, AL_DEVICES, 0, 0, 0, 0);
- if (nDevices < 0) {
- sprintf(message, "RtAudio: AL error counting devices: %s.",
- alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+ if ( mixerfd == -1 ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
+ return FAILURE;
}
- if (nDevices <= 0) return;
-
- ALvalue *vls = (ALvalue *) new ALvalue[nDevices];
-
- // Allocate the RTAUDIO_DEVICE structures.
- devices = (RTAUDIO_DEVICE *) calloc(nDevices, sizeof(RTAUDIO_DEVICE));
- if (devices == NULL) {
- sprintf(message, "RtAudio: memory allocation error!");
- error(RtError::MEMORY_ERROR);
+ oss_sysinfo sysinfo;
+ int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
+ if ( result == -1 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::probeDeviceOpen: error getting sysinfo, OSS version >= 4.0 is required.";
+ return FAILURE;
}
- // Write device ascii identifiers and resource ids to device info
- // structure.
- char name[32];
- int outs, ins, i;
- ALpv pvs[1];
- pvs[0].param = AL_NAME;
- pvs[0].value.ptr = name;
- pvs[0].sizeIn = 32;
-
- outs = alQueryValues(AL_SYSTEM, AL_DEFAULT_OUTPUT, vls, nDevices, 0, 0);
- if (outs < 0) {
- sprintf(message, "RtAudio: AL error getting output devices: %s.",
- alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
+ unsigned nDevices = sysinfo.numaudios;
+ if ( nDevices == 0 ) {
+ // This should not happen because a check is made before this function is called.
+ close( mixerfd );
+ errorText_ = "RtApiOss::probeDeviceOpen: no devices found!";
+ return FAILURE;
}
- for (i=0; i<outs; i++) {
- if (alGetParams(vls[i].i, pvs, 1) < 0) {
- sprintf(message, "RtAudio: AL error querying output devices: %s.",
- alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
- }
- strncpy(devices[i].name, name, 32);
- devices[i].id[0] = vls[i].i;
+ if ( device >= nDevices ) {
+ // This should not happen because a check is made before this function is called.
+ close( mixerfd );
+ errorText_ = "RtApiOss::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
}
- ins = alQueryValues(AL_SYSTEM, AL_DEFAULT_INPUT, &vls[outs], nDevices-outs, 0, 0);
- if (ins < 0) {
- sprintf(message, "RtAudio: AL error getting input devices: %s.",
- alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
+ oss_audioinfo ainfo;
+ ainfo.dev = device;
+ result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
+ close( mixerfd );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- for (i=outs; i<ins+outs; i++) {
- if (alGetParams(vls[i].i, pvs, 1) < 0) {
- sprintf(message, "RtAudio: AL error querying input devices: %s.",
- alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
- }
- strncpy(devices[i].name, name, 32);
- devices[i].id[1] = vls[i].i;
+ // Check if device supports input or output
+ if ( ( mode == OUTPUT && !( ainfo.caps & PCM_CAP_OUTPUT ) ) ||
+ ( mode == INPUT && !( ainfo.caps & PCM_CAP_INPUT ) ) ) {
+ if ( mode == OUTPUT )
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support output.";
+ else
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support input.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- delete [] vls;
-
- return;
-}
-
-int RtAudio :: getDefaultInputDevice(void)
-{
- ALvalue value;
- int result = alQueryValues(AL_SYSTEM, AL_DEFAULT_INPUT, &value, 1, 0, 0);
- if (result < 0) {
- sprintf(message, "RtAudio: AL error getting default input device id: %s.",
- alGetErrorString(oserror()));
- error(RtError::WARNING);
- }
- else {
- for ( int i=0; i<nDevices; i++ )
- if ( devices[i].id[1] == value.i ) return i;
+ int flags = 0;
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( mode == OUTPUT )
+ flags |= O_WRONLY;
+ else { // mode == INPUT
+ if (stream_.mode == OUTPUT && stream_.device[0] == device) {
+ // We just set the same device for playback ... close and reopen for duplex (OSS only).
+ close( handle->id[0] );
+ handle->id[0] = 0;
+ if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ // Check that the number previously set channels is the same.
+ if ( stream_.nUserChannels[0] != channels ) {
+ errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ flags |= O_RDWR;
+ }
+ else
+ flags |= O_RDONLY;
}
- return 0;
-}
+ // Set exclusive access if specified.
+ if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
-int RtAudio :: getDefaultOutputDevice(void)
-{
- ALvalue value;
- int result = alQueryValues(AL_SYSTEM, AL_DEFAULT_OUTPUT, &value, 1, 0, 0);
- if (result < 0) {
- sprintf(message, "RtAudio: AL error getting default output device id: %s.",
- alGetErrorString(oserror()));
- error(RtError::WARNING);
+ // Try to open the device.
+ int fd;
+ fd = open( ainfo.devnode, flags, 0 );
+ if ( fd == -1 ) {
+ if ( errno == EBUSY )
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
+ else
+ errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- else {
- for ( int i=0; i<nDevices; i++ )
- if ( devices[i].id[0] == value.i ) return i;
+
+ // For duplex operation, specifically set this mode (this doesn't seem to work).
+ /*
+ if ( flags | O_RDWR ) {
+ result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
+ if ( result == -1) {
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
}
+ */
- return 0;
-}
+ // Check the device channel support.
+ stream_.nUserChannels[mode] = channels;
+ if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
-void RtAudio :: probeDeviceInfo(RTAUDIO_DEVICE *info)
-{
- int resource, result, i;
- ALvalue value;
- ALparamInfo pinfo;
+ // Set the number of channels.
+ int deviceChannels = channels + firstChannel;
+ result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
+ if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.nDeviceChannels[mode] = deviceChannels;
- // Get output resource ID if it exists.
- resource = info->id[0];
- if (resource > 0) {
+ // Get the data format mask
+ int mask;
+ result = ioctl( fd, SNDCTL_DSP_GETFMTS, &mask );
+ if ( result == -1 ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error getting device (" << ainfo.name << ") data formats.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // Probe output device parameters.
- result = alQueryValues(resource, AL_CHANNELS, &value, 1, 0, 0);
- if (result < 0) {
- sprintf(message, "RtAudio: AL error getting device (%s) channels: %s.",
- info->name, alGetErrorString(oserror()));
- error(RtError::WARNING);
+ // Determine how to set the device format.
+ stream_.userFormat = format;
+ int deviceFormat = -1;
+ stream_.doByteSwap[mode] = false;
+ if ( format == RTAUDIO_SINT8 ) {
+ if ( mask & AFMT_S8 ) {
+ deviceFormat = AFMT_S8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
}
- else {
- info->maxOutputChannels = value.i;
- info->minOutputChannels = 1;
+ }
+ else if ( format == RTAUDIO_SINT16 ) {
+ if ( mask & AFMT_S16_NE ) {
+ deviceFormat = AFMT_S16_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
}
-
- result = alGetParamInfo(resource, AL_RATE, &pinfo);
- if (result < 0) {
- sprintf(message, "RtAudio: AL error getting device (%s) rates: %s.",
- info->name, alGetErrorString(oserror()));
- error(RtError::WARNING);
+ else if ( mask & AFMT_S16_OE ) {
+ deviceFormat = AFMT_S16_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ stream_.doByteSwap[mode] = true;
}
- else {
- info->nSampleRates = 0;
- for (i=0; i<MAX_SAMPLE_RATES; i++) {
- if ( SAMPLE_RATES[i] >= pinfo.min.i && SAMPLE_RATES[i] <= pinfo.max.i ) {
- info->sampleRates[info->nSampleRates] = SAMPLE_RATES[i];
- info->nSampleRates++;
- }
- }
+ }
+ else if ( format == RTAUDIO_SINT24 ) {
+ if ( mask & AFMT_S24_NE ) {
+ deviceFormat = AFMT_S24_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ }
+ else if ( mask & AFMT_S24_OE ) {
+ deviceFormat = AFMT_S24_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ stream_.doByteSwap[mode] = true;
}
-
- // The AL library supports all our formats, except 24-bit and 32-bit ints.
- info->nativeFormats = (RTAUDIO_FORMAT) 51;
}
-
- // Now get input resource ID if it exists.
- resource = info->id[1];
- if (resource > 0) {
-
- // Probe input device parameters.
- result = alQueryValues(resource, AL_CHANNELS, &value, 1, 0, 0);
- if (result < 0) {
- sprintf(message, "RtAudio: AL error getting device (%s) channels: %s.",
- info->name, alGetErrorString(oserror()));
- error(RtError::WARNING);
+ else if ( format == RTAUDIO_SINT32 ) {
+ if ( mask & AFMT_S32_NE ) {
+ deviceFormat = AFMT_S32_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
}
- else {
- info->maxInputChannels = value.i;
- info->minInputChannels = 1;
+ else if ( mask & AFMT_S32_OE ) {
+ deviceFormat = AFMT_S32_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ stream_.doByteSwap[mode] = true;
}
+ }
- result = alGetParamInfo(resource, AL_RATE, &pinfo);
- if (result < 0) {
- sprintf(message, "RtAudio: AL error getting device (%s) rates: %s.",
- info->name, alGetErrorString(oserror()));
- error(RtError::WARNING);
+ if ( deviceFormat == -1 ) {
+ // The user requested format is not natively supported by the device.
+ if ( mask & AFMT_S16_NE ) {
+ deviceFormat = AFMT_S16_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
}
- else {
- // In the case of the default device, these values will
- // overwrite the rates determined for the output device. Since
- // the input device is most likely to be more limited than the
- // output device, this is ok.
- info->nSampleRates = 0;
- for (i=0; i<MAX_SAMPLE_RATES; i++) {
- if ( SAMPLE_RATES[i] >= pinfo.min.i && SAMPLE_RATES[i] <= pinfo.max.i ) {
- info->sampleRates[info->nSampleRates] = SAMPLE_RATES[i];
- info->nSampleRates++;
- }
- }
+ else if ( mask & AFMT_S32_NE ) {
+ deviceFormat = AFMT_S32_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ }
+ else if ( mask & AFMT_S24_NE ) {
+ deviceFormat = AFMT_S24_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ }
+ else if ( mask & AFMT_S16_OE ) {
+ deviceFormat = AFMT_S16_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ stream_.doByteSwap[mode] = true;
+ }
+ else if ( mask & AFMT_S32_OE ) {
+ deviceFormat = AFMT_S32_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ stream_.doByteSwap[mode] = true;
+ }
+ else if ( mask & AFMT_S24_OE ) {
+ deviceFormat = AFMT_S24_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ stream_.doByteSwap[mode] = true;
+ }
+ else if ( mask & AFMT_S8) {
+ deviceFormat = AFMT_S8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
}
-
- // The AL library supports all our formats, except 24-bit and 32-bit ints.
- info->nativeFormats = (RTAUDIO_FORMAT) 51;
}
- if ( info->maxInputChannels == 0 && info->maxOutputChannels == 0 )
- return;
- if ( info->nSampleRates == 0 )
- return;
-
- // Determine duplex status.
- if (info->maxInputChannels < info->maxOutputChannels)
- info->maxDuplexChannels = info->maxInputChannels;
- else
- info->maxDuplexChannels = info->maxOutputChannels;
- if (info->minInputChannels < info->minOutputChannels)
- info->minDuplexChannels = info->minInputChannels;
- else
- info->minDuplexChannels = info->minOutputChannels;
-
- if ( info->maxDuplexChannels > 0 ) info->hasDuplexSupport = true;
- else info->hasDuplexSupport = false;
-
- info->probed = true;
-
- return;
-}
-
-bool RtAudio :: probeDeviceOpen(int device, RTAUDIO_STREAM *stream,
- STREAM_MODE mode, int channels,
- int sampleRate, RTAUDIO_FORMAT format,
- int *bufferSize, int numberOfBuffers)
-{
- int result, resource, nBuffers;
- ALconfig al_config;
- ALport port;
- ALpv pvs[2];
-
- // Get a new ALconfig structure.
- al_config = alNewConfig();
- if ( !al_config ) {
- sprintf(message,"RtAudio: can't get AL config: %s.",
- alGetErrorString(oserror()));
- error(RtError::WARNING);
+ if ( stream_.deviceFormat[mode] == 0 ) {
+ // This really shouldn't happen ...
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
return FAILURE;
}
- // Set the channels.
- result = alSetChannels(al_config, channels);
- if ( result < 0 ) {
- sprintf(message,"RtAudio: can't set %d channels in AL config: %s.",
- channels, alGetErrorString(oserror()));
- error(RtError::WARNING);
+ // Set the data format.
+ int temp = deviceFormat;
+ result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
+ if ( result == -1 || deviceFormat != temp ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
return FAILURE;
}
- // Attempt to set the queue size. The al API doesn't provide a
- // means for querying the minimum/maximum buffer size of a device,
- // so if the specified size doesn't work, take whatever the
- // al_config structure returns.
- if ( numberOfBuffers < 1 )
- nBuffers = 1;
- else
- nBuffers = numberOfBuffers;
- long buffer_size = *bufferSize * nBuffers;
- result = alSetQueueSize(al_config, buffer_size); // in sample frames
- if ( result < 0 ) {
- // Get the buffer size specified by the al_config and try that.
- buffer_size = alGetQueueSize(al_config);
- result = alSetQueueSize(al_config, buffer_size);
- if ( result < 0 ) {
- sprintf(message,"RtAudio: can't set buffer size (%ld) in AL config: %s.",
- buffer_size, alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
- }
- *bufferSize = buffer_size / nBuffers;
+ // Attempt to set the buffer size. According to OSS, the minimum
+ // number of buffers is two. The supposed minimum buffer size is 16
+ // bytes, so that will be our lower bound. The argument to this
+ // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
+ // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
+ // We'll check the actual value used near the end of the setup
+ // procedure.
+ int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
+ if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
+ int buffers = 0;
+ if ( options ) buffers = options->numberOfBuffers;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
+ if ( buffers < 2 ) buffers = 3;
+ temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
+ result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
+ if ( result == -1 ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
+ stream_.nBuffers = buffers;
- // Set the data format.
- stream->userFormat = format;
- stream->deviceFormat[mode] = format;
- if (format == RTAUDIO_SINT8) {
- result = alSetSampFmt(al_config, AL_SAMPFMT_TWOSCOMP);
- result = alSetWidth(al_config, AL_SAMPLE_8);
- }
- else if (format == RTAUDIO_SINT16) {
- result = alSetSampFmt(al_config, AL_SAMPFMT_TWOSCOMP);
- result = alSetWidth(al_config, AL_SAMPLE_16);
- }
- else if (format == RTAUDIO_SINT24) {
- // Our 24-bit format assumes the upper 3 bytes of a 4 byte word.
- // The AL library uses the lower 3 bytes, so we'll need to do our
- // own conversion.
- result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT);
- stream->deviceFormat[mode] = RTAUDIO_FLOAT32;
- }
- else if (format == RTAUDIO_SINT32) {
- // The AL library doesn't seem to support the 32-bit integer
- // format, so we'll need to do our own conversion.
- result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT);
- stream->deviceFormat[mode] = RTAUDIO_FLOAT32;
- }
- else if (format == RTAUDIO_FLOAT32)
- result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT);
- else if (format == RTAUDIO_FLOAT64)
- result = alSetSampFmt(al_config, AL_SAMPFMT_DOUBLE);
+ // Save buffer size (in sample frames).
+ *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
+ stream_.bufferSize = *bufferSize;
+ // Set the sample rate.
+ int srate = sampleRate;
+ result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
if ( result == -1 ) {
- sprintf(message,"RtAudio: AL error setting sample format in AL config: %s.",
- alGetErrorString(oserror()));
- error(RtError::WARNING);
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
return FAILURE;
}
- if (mode == OUTPUT) {
-
- // Set our device.
- if (device == 0)
- resource = AL_DEFAULT_OUTPUT;
- else
- resource = devices[device].id[0];
- result = alSetDevice(al_config, resource);
- if ( result == -1 ) {
- sprintf(message,"RtAudio: AL error setting device (%s) in AL config: %s.",
- devices[device].name, alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
- }
-
- // Open the port.
- port = alOpenPort("RtAudio Output Port", "w", al_config);
- if( !port ) {
- sprintf(message,"RtAudio: AL error opening output port: %s.",
- alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
- }
+ // Verify the sample rate setup worked.
+ if ( abs( srate - sampleRate ) > 100 ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.sampleRate = sampleRate;
- // Set the sample rate
- pvs[0].param = AL_MASTER_CLOCK;
- pvs[0].value.i = AL_CRYSTAL_MCLK_TYPE;
- pvs[1].param = AL_RATE;
- pvs[1].value.ll = alDoubleToFixed((double)sampleRate);
- result = alSetParams(resource, pvs, 2);
- if ( result < 0 ) {
- alClosePort(port);
- sprintf(message,"RtAudio: AL error setting sample rate (%d) for device (%s): %s.",
- sampleRate, devices[device].name, alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
- }
+ if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device) {
+ // We're doing duplex setup here.
+ stream_.deviceFormat[0] = stream_.deviceFormat[1];
+ stream_.nDeviceChannels[0] = deviceChannels;
}
- else { // mode == INPUT
- // Set our device.
- if (device == 0)
- resource = AL_DEFAULT_INPUT;
- else
- resource = devices[device].id[1];
- result = alSetDevice(al_config, resource);
- if ( result == -1 ) {
- sprintf(message,"RtAudio: AL error setting device (%s) in AL config: %s.",
- devices[device].name, alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
- }
+ // Set interleaving parameters.
+ stream_.userInterleaved = true;
+ stream_.deviceInterleaved[mode] = true;
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
+ stream_.userInterleaved = false;
- // Open the port.
- port = alOpenPort("RtAudio Output Port", "r", al_config);
- if( !port ) {
- sprintf(message,"RtAudio: AL error opening input port: %s.",
- alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
+ // Set flags for buffer conversion
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate the stream handles if necessary and then save.
+ if ( stream_.apiHandle == 0 ) {
+ try {
+ handle = new OssHandle;
}
-
- // Set the sample rate
- pvs[0].param = AL_MASTER_CLOCK;
- pvs[0].value.i = AL_CRYSTAL_MCLK_TYPE;
- pvs[1].param = AL_RATE;
- pvs[1].value.ll = alDoubleToFixed((double)sampleRate);
- result = alSetParams(resource, pvs, 2);
- if ( result < 0 ) {
- alClosePort(port);
- sprintf(message,"RtAudio: AL error setting sample rate (%d) for device (%s): %s.",
- sampleRate, devices[device].name, alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
+ goto error;
}
- }
-
- alFreeConfig(al_config);
-
- stream->nUserChannels[mode] = channels;
- stream->nDeviceChannels[mode] = channels;
- // Set handle and flags for buffer conversion
- stream->handle[mode] = port;
- stream->doConvertBuffer[mode] = false;
- if (stream->userFormat != stream->deviceFormat[mode])
- stream->doConvertBuffer[mode] = true;
-
- // Allocate necessary internal buffers
- if ( stream->nUserChannels[0] != stream->nUserChannels[1] ) {
-
- long buffer_bytes;
- if (stream->nUserChannels[0] >= stream->nUserChannels[1])
- buffer_bytes = stream->nUserChannels[0];
- else
- buffer_bytes = stream->nUserChannels[1];
+ stream_.apiHandle = (void *) handle;
+ }
+ else {
+ handle = (OssHandle *) stream_.apiHandle;
+ }
+ handle->id[mode] = fd;
- buffer_bytes *= *bufferSize * formatBytes(stream->userFormat);
- if (stream->userBuffer) free(stream->userBuffer);
- stream->userBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->userBuffer == NULL)
- goto memory_error;
+ // Allocate necessary internal buffers.
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
}
- if ( stream->doConvertBuffer[mode] ) {
+ if ( stream_.doConvertBuffer[mode] ) {
- long buffer_bytes;
bool makeBuffer = true;
- if ( mode == OUTPUT )
- buffer_bytes = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- else { // mode == INPUT
- buffer_bytes = stream->nDeviceChannels[1] * formatBytes(stream->deviceFormat[1]);
- if ( stream->mode == OUTPUT && stream->deviceBuffer ) {
- long bytes_out = stream->nDeviceChannels[0] * formatBytes(stream->deviceFormat[0]);
- if ( buffer_bytes < bytes_out ) makeBuffer = false;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
}
}
if ( makeBuffer ) {
- buffer_bytes *= *bufferSize;
- if (stream->deviceBuffer) free(stream->deviceBuffer);
- stream->deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream->deviceBuffer == NULL)
- goto memory_error;
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
}
}
- stream->device[mode] = device;
- stream->state = STREAM_STOPPED;
- if ( stream->mode == OUTPUT && mode == INPUT )
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+
+ // Setup thread if necessary.
+ if ( stream_.mode == OUTPUT && mode == INPUT ) {
// We had already set up an output stream.
- stream->mode = DUPLEX;
- else
- stream->mode = mode;
- stream->nBuffers = nBuffers;
- stream->bufferSize = *bufferSize;
- stream->sampleRate = sampleRate;
+ stream_.mode = DUPLEX;
+ if ( stream_.device[0] == device ) handle->id[0] = fd;
+ }
+ else {
+ stream_.mode = mode;
+
+ // Setup callback thread.
+ stream_.callbackInfo.object = (void *) this;
+
+ // Set the thread attributes for joinable and realtime scheduling
+ // priority. The higher priority will only take affect if the
+ // program is run as root or suid.
+ pthread_attr_t attr;
+ pthread_attr_init( &attr );
+ pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
+#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
+ pthread_attr_setschedpolicy( &attr, SCHED_RR );
+#else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#endif
+
+ stream_.callbackInfo.isRunning = true;
+ result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
+ pthread_attr_destroy( &attr );
+ if ( result ) {
+ stream_.callbackInfo.isRunning = false;
+ errorText_ = "RtApiOss::error creating callback thread!";
+ goto error;
+ }
+ }
return SUCCESS;
- memory_error:
- if (stream->handle[0]) {
- alClosePort(stream->handle[0]);
- stream->handle[0] = 0;
+ error:
+ if ( handle ) {
+ if ( handle->id[0] ) close( handle->id[0] );
+ if ( handle->id[1] ) close( handle->id[1] );
+ delete handle;
+ stream_.apiHandle = 0;
}
- if (stream->handle[1]) {
- alClosePort(stream->handle[1]);
- stream->handle[1] = 0;
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
}
- if (stream->userBuffer) {
- free(stream->userBuffer);
- stream->userBuffer = 0;
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
}
- sprintf(message, "RtAudio: ALSA error allocating buffer memory for device (%s).",
- devices[device].name);
- error(RtError::WARNING);
+
return FAILURE;
}
-void RtAudio :: closeStream(int streamId)
+void RtApiOss :: closeStream()
{
- // We don't want an exception to be thrown here because this
- // function is called by our class destructor. So, do our own
- // streamId check.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
- error(RtError::WARNING);
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiOss::closeStream(): no open stream to close!";
+ error( RtError::WARNING );
return;
}
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) streams[streamId];
+ stream_.callbackInfo.isRunning = false;
+ pthread_join( stream_.callbackInfo.thread, NULL );
- if (stream->callbackInfo.usingCallback) {
- pthread_cancel(stream->callbackInfo.thread);
- pthread_join(stream->callbackInfo.thread, NULL);
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( stream_.state == STREAM_RUNNING ) {
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
+ ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+ else
+ ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+ stream_.state = STREAM_STOPPED;
}
- pthread_mutex_destroy(&stream->mutex);
-
- if (stream->handle[0])
- alClosePort(stream->handle[0]);
-
- if (stream->handle[1])
- alClosePort(stream->handle[1]);
+ if ( handle ) {
+ if ( handle->id[0] ) close( handle->id[0] );
+ if ( handle->id[1] ) close( handle->id[1] );
+ delete handle;
+ stream_.apiHandle = 0;
+ }
- if (stream->userBuffer)
- free(stream->userBuffer);
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
- if (stream->deviceBuffer)
- free(stream->deviceBuffer);
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
- free(stream);
- streams.erase(streamId);
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
}
-void RtAudio :: startStream(int streamId)
+void RtApiOss :: startStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- if (stream->state == STREAM_RUNNING)
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiOss::startStream(): the stream is already running!";
+ error( RtError::WARNING );
return;
+ }
- // The AL port is ready as soon as it is opened.
- stream->state = STREAM_RUNNING;
-}
-
-void RtAudio :: stopStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK( &stream_.mutex );
- if (stream->state == STREAM_STOPPED)
- goto unlock;
+ stream_.state = STREAM_RUNNING;
- int result;
- int buffer_size = stream->bufferSize * stream->nBuffers;
+ // No need to do anything else here ... OSS automatically starts
+ // when fed samples.
- if (stream->mode == OUTPUT || stream->mode == DUPLEX)
- alZeroFrames(stream->handle[0], buffer_size);
+ MUTEX_UNLOCK( &stream_.mutex );
+}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- result = alDiscardFrames(stream->handle[1], buffer_size);
- if (result == -1) {
- sprintf(message, "RtAudio: AL error draining stream device (%s): %s.",
- devices[stream->device[1]].name, alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
- }
+void RtApiOss :: stopStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
}
- stream->state = STREAM_STOPPED;
- unlock:
- MUTEX_UNLOCK(&stream->mutex);
-}
+ // Change the state before the lock to improve shutdown response
+ // when using a callback.
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK( &stream_.mutex );
-void RtAudio :: abortStream(int streamId)
-{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ int result = 0;
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- MUTEX_LOCK(&stream->mutex);
+ // Flush the output with zeros a few times.
+ char *buffer;
+ int samples;
+ RtAudioFormat format;
- if (stream->state == STREAM_STOPPED)
- goto unlock;
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ samples = stream_.bufferSize * stream_.nDeviceChannels[0];
+ format = stream_.deviceFormat[0];
+ }
+ else {
+ buffer = stream_.userBuffer[0];
+ samples = stream_.bufferSize * stream_.nUserChannels[0];
+ format = stream_.userFormat;
+ }
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
+ memset( buffer, 0, samples * formatBytes(format) );
+ for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
+ result = write( handle->id[0], buffer, samples * formatBytes(format) );
+ if ( result == -1 ) {
+ errorText_ = "RtApiOss::stopStream: audio write error.";
+ error( RtError::WARNING );
+ }
+ }
- int buffer_size = stream->bufferSize * stream->nBuffers;
- int result = alDiscardFrames(stream->handle[0], buffer_size);
- if (result == -1) {
- sprintf(message, "RtAudio: AL error aborting stream device (%s): %s.",
- devices[stream->device[0]].name, alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
+ result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::stopStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
+ handle->triggered = false;
}
- // There is no clear action to take on the input stream, since the
- // port will continue to run in any event.
- stream->state = STREAM_STOPPED;
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
+ result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ stream_.state = STREAM_STOPPED;
+ if ( result != -1 ) return;
+ error( RtError::SYSTEM_ERROR );
}
-int RtAudio :: streamWillBlock(int streamId)
+void RtApiOss :: abortStream()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
-
- MUTEX_LOCK(&stream->mutex);
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
+ error( RtError::WARNING );
+ return;
+ }
- int frames = 0;
- if (stream->state == STREAM_STOPPED)
- goto unlock;
+ // Change the state before the lock to improve shutdown response
+ // when using a callback.
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK( &stream_.mutex );
- int err = 0;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
- err = alGetFillable(stream->handle[0]);
- if (err < 0) {
- sprintf(message, "RtAudio: AL error getting available frames for stream (%s): %s.",
- devices[stream->device[0]].name, alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
+ int result = 0;
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ result = ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::abortStream: system error stopping callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
+ handle->triggered = false;
}
- frames = err;
-
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
- err = alGetFilled(stream->handle[1]);
- if (err < 0) {
- sprintf(message, "RtAudio: AL error getting available frames for stream (%s): %s.",
- devices[stream->device[1]].name, alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
+ result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::abortStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
- if (frames > err) frames = err;
}
- frames = stream->bufferSize - frames;
- if (frames < 0) frames = 0;
-
unlock:
- MUTEX_UNLOCK(&stream->mutex);
- return frames;
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ stream_.state = STREAM_STOPPED;
+ if ( result != -1 ) return;
+ error( RtError::SYSTEM_ERROR );
}
-void RtAudio :: tickStream(int streamId)
+void RtApiOss :: callbackEvent()
{
- RTAUDIO_STREAM *stream = (RTAUDIO_STREAM *) verifyStream(streamId);
+ if ( stream_.state == STREAM_STOPPED ) {
+ if ( stream_.callbackInfo.isRunning ) usleep( 50000 ); // sleep 50 milliseconds
+ return;
+ }
- int stopStream = 0;
- if (stream->state == STREAM_STOPPED) {
- if (stream->callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtError::WARNING );
return;
}
- else if (stream->callbackInfo.usingCallback) {
- RTAUDIO_CALLBACK callback = (RTAUDIO_CALLBACK) stream->callbackInfo.callback;
- stopStream = callback(stream->userBuffer, stream->bufferSize, stream->callbackInfo.userData);
+
+ // Invoke user callback to get fresh output data.
+ int doStopStream = 0;
+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
}
+ doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
- MUTEX_LOCK(&stream->mutex);
+ MUTEX_LOCK( &stream_.mutex );
// The state might change while waiting on a mutex.
- if (stream->state == STREAM_STOPPED)
- goto unlock;
+ if ( stream_.state == STREAM_STOPPED ) goto unlock;
+ int result;
char *buffer;
- int channels;
- RTAUDIO_FORMAT format;
- if (stream->mode == OUTPUT || stream->mode == DUPLEX) {
+ int samples;
+ RtAudioFormat format;
+
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
// Setup parameters and do buffer conversion if necessary.
- if (stream->doConvertBuffer[0]) {
- convertStreamBuffer(stream, OUTPUT);
- buffer = stream->deviceBuffer;
- channels = stream->nDeviceChannels[0];
- format = stream->deviceFormat[0];
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ samples = stream_.bufferSize * stream_.nDeviceChannels[0];
+ format = stream_.deviceFormat[0];
}
else {
- buffer = stream->userBuffer;
- channels = stream->nUserChannels[0];
- format = stream->userFormat;
+ buffer = stream_.userBuffer[0];
+ samples = stream_.bufferSize * stream_.nUserChannels[0];
+ format = stream_.userFormat;
}
// Do byte swapping if necessary.
- if (stream->doByteSwap[0])
- byteSwapBuffer(buffer, stream->bufferSize * channels, format);
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer( buffer, samples, format );
+
+ if ( stream_.mode == DUPLEX && handle->triggered == false ) {
+ int trig = 0;
+ ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
+ result = write( handle->id[0], buffer, samples * formatBytes(format) );
+ trig = PCM_ENABLE_INPUT|PCM_ENABLE_OUTPUT;
+ ioctl( handle->id[0], SNDCTL_DSP_SETTRIGGER, &trig );
+ handle->triggered = true;
+ }
+ else
+ // Write samples to device.
+ result = write( handle->id[0], buffer, samples * formatBytes(format) );
- // Write interleaved samples to device.
- alWriteFrames(stream->handle[0], buffer, stream->bufferSize);
+ if ( result == -1 ) {
+ // We'll assume this is an underrun, though there isn't a
+ // specific means for determining that.
+ handle->xrun[0] = true;
+ errorText_ = "RtApiOss::callbackEvent: audio write error.";
+ error( RtError::WARNING );
+ goto unlock;
+ }
}
- if (stream->mode == INPUT || stream->mode == DUPLEX) {
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
// Setup parameters.
- if (stream->doConvertBuffer[1]) {
- buffer = stream->deviceBuffer;
- channels = stream->nDeviceChannels[1];
- format = stream->deviceFormat[1];
+ if ( stream_.doConvertBuffer[1] ) {
+ buffer = stream_.deviceBuffer;
+ samples = stream_.bufferSize * stream_.nDeviceChannels[1];
+ format = stream_.deviceFormat[1];
}
else {
- buffer = stream->userBuffer;
- channels = stream->nUserChannels[1];
- format = stream->userFormat;
+ buffer = stream_.userBuffer[1];
+ samples = stream_.bufferSize * stream_.nUserChannels[1];
+ format = stream_.userFormat;
}
- // Read interleaved samples from device.
- alReadFrames(stream->handle[1], buffer, stream->bufferSize);
+ // Read samples from device.
+ result = read( handle->id[1], buffer, samples * formatBytes(format) );
+
+ if ( result == -1 ) {
+ // We'll assume this is an overrun, though there isn't a
+ // specific means for determining that.
+ handle->xrun[1] = true;
+ errorText_ = "RtApiOss::callbackEvent: audio read error.";
+ error( RtError::WARNING );
+ goto unlock;
+ }
// Do byte swapping if necessary.
- if (stream->doByteSwap[1])
- byteSwapBuffer(buffer, stream->bufferSize * channels, format);
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( buffer, samples, format );
// Do buffer conversion if necessary.
- if (stream->doConvertBuffer[1])
- convertStreamBuffer(stream, INPUT);
+ if ( stream_.doConvertBuffer[1] )
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
}
unlock:
- MUTEX_UNLOCK(&stream->mutex);
+ MUTEX_UNLOCK( &stream_.mutex );
- if (stream->callbackInfo.usingCallback && stopStream)
- this->stopStream(streamId);
+ RtApi::tickStreamTime();
+ if ( doStopStream == 1 ) this->stopStream();
+ else if ( doStopStream == 2 ) this->abortStream();
}
-extern "C" void *callbackHandler(void *ptr)
+extern "C" void *ossCallbackHandler( void *ptr )
{
- CALLBACK_INFO *info = (CALLBACK_INFO *) ptr;
- RtAudio *object = (RtAudio *) info->object;
- int stream = info->streamId;
- bool *usingCallback = &info->usingCallback;
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiOss *object = (RtApiOss *) info->object;
+ bool *isRunning = &info->isRunning;
+
+#ifdef SCHED_RR
+ // Set a higher scheduler priority (P.J. Leonard)
+ struct sched_param param;
+ param.sched_priority = 39; // Is this the best number?
+ sched_setscheduler( 0, SCHED_RR, ¶m );
+#endif
- while ( *usingCallback ) {
+ while ( *isRunning == true ) {
pthread_testcancel();
- try {
- object->tickStream(stream);
- }
- catch (RtError &exception) {
- fprintf(stderr, "\nRtAudio: Callback thread error (%s) ... closing thread.\n\n",
- exception.getMessage());
- break;
- }
+ object->callbackEvent();
}
- return 0;
+ pthread_exit( NULL );
}
-//******************** End of __IRIX_AL__ *********************//
-
+//******************** End of __LINUX_OSS__ *********************//
#endif
// *************************************************** //
//
-// Private common (OS-independent) RtAudio methods.
+// Protected common (OS-independent) RtAudio methods.
//
// *************************************************** //
// This method can be modified to control the behavior of error
-// message reporting and throwing.
-void RtAudio :: error(RtError::TYPE type)
+// message printing.
+void RtApi :: error( RtError::Type type )
{
- if (type == RtError::WARNING) {
- fprintf(stderr, "\n%s\n\n", message);
- }
- else if (type == RtError::DEBUG_WARNING) {
-#if defined(__RTAUDIO_DEBUG__)
- fprintf(stderr, "\n%s\n\n", message);
-#endif
- }
- else {
- fprintf(stderr, "\n%s\n\n", message);
- throw RtError(message, type);
- }
+ errorStream_.str(""); // clear the ostringstream
+ if ( type == RtError::WARNING && showWarnings_ == true )
+ std::cerr << '\n' << errorText_ << "\n\n";
+ else
+ throw( RtError( errorText_, type ) );
}
-void *RtAudio :: verifyStream(int streamId)
+void RtApi :: verifyStream()
{
- // Verify the stream key.
- if ( streams.find( streamId ) == streams.end() ) {
- sprintf(message, "RtAudio: invalid stream identifier!");
- error(RtError::INVALID_STREAM);
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApi:: a stream is not open!";
+ error( RtError::INVALID_USE );
}
-
- return streams[streamId];
}
-void RtAudio :: clearDeviceInfo(RTAUDIO_DEVICE *info)
+void RtApi :: clearStreamInfo()
{
- // Don't clear the name or DEVICE_ID fields here ... they are
- // typically set prior to a call of this function.
- info->probed = false;
- info->maxOutputChannels = 0;
- info->maxInputChannels = 0;
- info->maxDuplexChannels = 0;
- info->minOutputChannels = 0;
- info->minInputChannels = 0;
- info->minDuplexChannels = 0;
- info->hasDuplexSupport = false;
- info->nSampleRates = 0;
- for (int i=0; i<MAX_SAMPLE_RATES; i++)
- info->sampleRates[i] = 0;
- info->nativeFormats = 0;
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+ stream_.sampleRate = 0;
+ stream_.bufferSize = 0;
+ stream_.nBuffers = 0;
+ stream_.userFormat = 0;
+ stream_.userInterleaved = true;
+ stream_.streamTime = 0.0;
+ stream_.apiHandle = 0;
+ stream_.deviceBuffer = 0;
+ stream_.callbackInfo.callback = 0;
+ stream_.callbackInfo.userData = 0;
+ stream_.callbackInfo.isRunning = false;
+ for ( int i=0; i<2; i++ ) {
+ stream_.device[i] = 11111;
+ stream_.doConvertBuffer[i] = false;
+ stream_.deviceInterleaved[i] = true;
+ stream_.doByteSwap[i] = false;
+ stream_.nUserChannels[i] = 0;
+ stream_.nDeviceChannels[i] = 0;
+ stream_.channelOffset[i] = 0;
+ stream_.deviceFormat[i] = 0;
+ stream_.latency[i] = 0;
+ stream_.userBuffer[i] = 0;
+ stream_.convertInfo[i].channels = 0;
+ stream_.convertInfo[i].inJump = 0;
+ stream_.convertInfo[i].outJump = 0;
+ stream_.convertInfo[i].inFormat = 0;
+ stream_.convertInfo[i].outFormat = 0;
+ stream_.convertInfo[i].inOffset.clear();
+ stream_.convertInfo[i].outOffset.clear();
+ }
}
-int RtAudio :: formatBytes(RTAUDIO_FORMAT format)
+unsigned int RtApi :: formatBytes( RtAudioFormat format )
{
- if (format == RTAUDIO_SINT16)
+ if ( format == RTAUDIO_SINT16 )
return 2;
- else if (format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 ||
- format == RTAUDIO_FLOAT32)
+ else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 ||
+ format == RTAUDIO_FLOAT32 )
return 4;
- else if (format == RTAUDIO_FLOAT64)
+ else if ( format == RTAUDIO_FLOAT64 )
return 8;
- else if (format == RTAUDIO_SINT8)
+ else if ( format == RTAUDIO_SINT8 )
return 1;
- sprintf(message,"RtAudio: undefined format in formatBytes().");
- error(RtError::WARNING);
+ errorText_ = "RtApi::formatBytes: undefined format.";
+ error( RtError::WARNING );
return 0;
}
-void RtAudio :: convertStreamBuffer(RTAUDIO_STREAM *stream, STREAM_MODE mode)
+void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
{
- // This method does format conversion, input/output channel compensation, and
- // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
- // the upper three bytes of a 32-bit integer.
-
- int j, jump_in, jump_out, channels;
- RTAUDIO_FORMAT format_in, format_out;
- char *input, *output;
-
- if (mode == INPUT) { // convert device to user buffer
- input = stream->deviceBuffer;
- output = stream->userBuffer;
- jump_in = stream->nDeviceChannels[1];
- jump_out = stream->nUserChannels[1];
- format_in = stream->deviceFormat[1];
- format_out = stream->userFormat;
+ if ( mode == INPUT ) { // convert device to user buffer
+ stream_.convertInfo[mode].inJump = stream_.nDeviceChannels[1];
+ stream_.convertInfo[mode].outJump = stream_.nUserChannels[1];
+ stream_.convertInfo[mode].inFormat = stream_.deviceFormat[1];
+ stream_.convertInfo[mode].outFormat = stream_.userFormat;
}
else { // convert user to device buffer
- input = stream->userBuffer;
- output = stream->deviceBuffer;
- jump_in = stream->nUserChannels[0];
- jump_out = stream->nDeviceChannels[0];
- format_in = stream->userFormat;
- format_out = stream->deviceFormat[0];
-
- // clear our device buffer when in/out duplex device channels are different
- if ( stream->mode == DUPLEX &&
- stream->nDeviceChannels[0] != stream->nDeviceChannels[1] )
- memset(output, 0, stream->bufferSize * jump_out * formatBytes(format_out));
+ stream_.convertInfo[mode].inJump = stream_.nUserChannels[0];
+ stream_.convertInfo[mode].outJump = stream_.nDeviceChannels[0];
+ stream_.convertInfo[mode].inFormat = stream_.userFormat;
+ stream_.convertInfo[mode].outFormat = stream_.deviceFormat[0];
}
- channels = (jump_in < jump_out) ? jump_in : jump_out;
-
- // Set up the interleave/deinterleave offsets
- std::vector<int> offset_in(channels);
- std::vector<int> offset_out(channels);
- if (mode == INPUT && stream->deInterleave[1]) {
- for (int k=0; k<channels; k++) {
- offset_in[k] = k * stream->bufferSize;
- offset_out[k] = k;
- jump_in = 1;
+ if ( stream_.convertInfo[mode].inJump < stream_.convertInfo[mode].outJump )
+ stream_.convertInfo[mode].channels = stream_.convertInfo[mode].inJump;
+ else
+ stream_.convertInfo[mode].channels = stream_.convertInfo[mode].outJump;
+
+ // Set up the interleave/deinterleave offsets.
+ if ( stream_.deviceInterleaved[mode] != stream_.userInterleaved ) {
+ if ( ( mode == OUTPUT && stream_.deviceInterleaved[mode] ) ||
+ ( mode == INPUT && stream_.userInterleaved ) ) {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+ stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
+ stream_.convertInfo[mode].outOffset.push_back( k );
+ stream_.convertInfo[mode].inJump = 1;
+ }
+ }
+ else {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+ stream_.convertInfo[mode].inOffset.push_back( k );
+ stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
+ stream_.convertInfo[mode].outJump = 1;
+ }
}
}
- else if (mode == OUTPUT && stream->deInterleave[0]) {
- for (int k=0; k<channels; k++) {
- offset_in[k] = k;
- offset_out[k] = k * stream->bufferSize;
- jump_out = 1;
+ else { // no (de)interleaving
+ if ( stream_.userInterleaved ) {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+ stream_.convertInfo[mode].inOffset.push_back( k );
+ stream_.convertInfo[mode].outOffset.push_back( k );
+ }
+ }
+ else {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ ) {
+ stream_.convertInfo[mode].inOffset.push_back( k * stream_.bufferSize );
+ stream_.convertInfo[mode].outOffset.push_back( k * stream_.bufferSize );
+ stream_.convertInfo[mode].inJump = 1;
+ stream_.convertInfo[mode].outJump = 1;
+ }
}
}
- else {
- for (int k=0; k<channels; k++) {
- offset_in[k] = k;
- offset_out[k] = k;
+
+ // Add channel offset.
+ if ( firstChannel > 0 ) {
+ if ( stream_.deviceInterleaved[mode] ) {
+ if ( mode == OUTPUT ) {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+ stream_.convertInfo[mode].outOffset[k] += firstChannel;
+ }
+ else {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+ stream_.convertInfo[mode].inOffset[k] += firstChannel;
+ }
+ }
+ else {
+ if ( mode == OUTPUT ) {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+ stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
+ }
+ else {
+ for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
+ stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
+ }
}
}
+}
+
+void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
+{
+ // This function does format conversion, input/output channel compensation, and
+ // data interleaving/deinterleaving. 24-bit integers are assumed to occupy
+ // the upper three bytes of a 32-bit integer.
- if (format_out == RTAUDIO_FLOAT64) {
- FLOAT64 scale;
- FLOAT64 *out = (FLOAT64 *)output;
+ // Clear our device buffer when in/out duplex device channels are different
+ if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
+ ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
+ memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
- if (format_in == RTAUDIO_SINT8) {
- signed char *in = (signed char *)input;
+ int j;
+ if (info.outFormat == RTAUDIO_FLOAT64) {
+ Float64 scale;
+ Float64 *out = (Float64 *)outBuffer;
+
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
scale = 1.0 / 128.0;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT64) in[offset_in[j]];
- out[offset_out[j]] *= scale;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+ out[info.outOffset[j]] *= scale;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT16) {
- INT16 *in = (INT16 *)input;
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
scale = 1.0 / 32768.0;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT64) in[offset_in[j]];
- out[offset_out[j]] *= scale;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+ out[info.outOffset[j]] *= scale;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT24) {
- INT32 *in = (INT32 *)input;
- scale = 1.0 / 2147483648.0;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT64) (in[offset_in[j]] & 0xffffff00);
- out[offset_out[j]] *= scale;
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int32 *in = (Int32 *)inBuffer;
+ scale = 1.0 / 8388608.0;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]] & 0x00ffffff);
+ out[info.outOffset[j]] *= scale;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT32) {
- INT32 *in = (INT32 *)input;
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
scale = 1.0 / 2147483648.0;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT64) in[offset_in[j]];
- out[offset_out[j]] *= scale;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+ out[info.outOffset[j]] *= scale;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_FLOAT32) {
- FLOAT32 *in = (FLOAT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT64) in[offset_in[j]];
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_FLOAT64) {
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
// Channel compensation and/or (de)interleaving only.
- FLOAT64 *in = (FLOAT64 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = in[offset_in[j]];
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
}
- else if (format_out == RTAUDIO_FLOAT32) {
- FLOAT32 scale;
- FLOAT32 *out = (FLOAT32 *)output;
+ else if (info.outFormat == RTAUDIO_FLOAT32) {
+ Float32 scale;
+ Float32 *out = (Float32 *)outBuffer;
- if (format_in == RTAUDIO_SINT8) {
- signed char *in = (signed char *)input;
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
scale = 1.0 / 128.0;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT32) in[offset_in[j]];
- out[offset_out[j]] *= scale;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+ out[info.outOffset[j]] *= scale;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT16) {
- INT16 *in = (INT16 *)input;
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
scale = 1.0 / 32768.0;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT32) in[offset_in[j]];
- out[offset_out[j]] *= scale;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+ out[info.outOffset[j]] *= scale;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT24) {
- INT32 *in = (INT32 *)input;
- scale = 1.0 / 2147483648.0;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT32) (in[offset_in[j]] & 0xffffff00);
- out[offset_out[j]] *= scale;
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int32 *in = (Int32 *)inBuffer;
+ scale = 1.0 / 8388608.0;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]] & 0x00ffffff);
+ out[info.outOffset[j]] *= scale;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT32) {
- INT32 *in = (INT32 *)input;
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
scale = 1.0 / 2147483648.0;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT32) in[offset_in[j]];
- out[offset_out[j]] *= scale;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+ out[info.outOffset[j]] *= scale;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_FLOAT32) {
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
// Channel compensation and/or (de)interleaving only.
- FLOAT32 *in = (FLOAT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = in[offset_in[j]];
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_FLOAT64) {
- FLOAT64 *in = (FLOAT64 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (FLOAT32) in[offset_in[j]];
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
}
- else if (format_out == RTAUDIO_SINT32) {
- INT32 *out = (INT32 *)output;
- if (format_in == RTAUDIO_SINT8) {
- signed char *in = (signed char *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) in[offset_in[j]];
- out[offset_out[j]] <<= 24;
+ else if (info.outFormat == RTAUDIO_SINT32) {
+ Int32 *out = (Int32 *)outBuffer;
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+ out[info.outOffset[j]] <<= 24;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT16) {
- INT16 *in = (INT16 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) in[offset_in[j]];
- out[offset_out[j]] <<= 16;
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+ out[info.outOffset[j]] <<= 16;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT24) {
- INT32 *in = (INT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) in[offset_in[j]];
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+ out[info.outOffset[j]] <<= 8;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT32) {
+ else if (info.inFormat == RTAUDIO_SINT32) {
// Channel compensation and/or (de)interleaving only.
- INT32 *in = (INT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = in[offset_in[j]];
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_FLOAT32) {
- FLOAT32 *in = (FLOAT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) (in[offset_in[j]] * 2147483647.0);
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.0);
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_FLOAT64) {
- FLOAT64 *in = (FLOAT64 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) (in[offset_in[j]] * 2147483647.0);
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.0);
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
}
- else if (format_out == RTAUDIO_SINT24) {
- INT32 *out = (INT32 *)output;
- if (format_in == RTAUDIO_SINT8) {
- signed char *in = (signed char *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) in[offset_in[j]];
- out[offset_out[j]] <<= 24;
+ else if (info.outFormat == RTAUDIO_SINT24) {
+ Int32 *out = (Int32 *)outBuffer;
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+ out[info.outOffset[j]] <<= 16;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT16) {
- INT16 *in = (INT16 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) in[offset_in[j]];
- out[offset_out[j]] <<= 16;
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+ out[info.outOffset[j]] <<= 8;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT24) {
+ else if (info.inFormat == RTAUDIO_SINT24) {
// Channel compensation and/or (de)interleaving only.
- INT32 *in = (INT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = in[offset_in[j]];
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT32) {
- INT32 *in = (INT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) (in[offset_in[j]] & 0xffffff00);
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
+ out[info.outOffset[j]] >>= 8;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_FLOAT32) {
- FLOAT32 *in = (FLOAT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) (in[offset_in[j]] * 2147483647.0);
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 8388608.0);
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_FLOAT64) {
- FLOAT64 *in = (FLOAT64 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT32) (in[offset_in[j]] * 2147483647.0);
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] * 2147483647.0);
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
}
- else if (format_out == RTAUDIO_SINT16) {
- INT16 *out = (INT16 *)output;
- if (format_in == RTAUDIO_SINT8) {
- signed char *in = (signed char *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT16) in[offset_in[j]];
- out[offset_out[j]] <<= 8;
+ else if (info.outFormat == RTAUDIO_SINT16) {
+ Int16 *out = (Int16 *)outBuffer;
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) in[info.inOffset[j]];
+ out[info.outOffset[j]] <<= 8;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT16) {
+ else if (info.inFormat == RTAUDIO_SINT16) {
// Channel compensation and/or (de)interleaving only.
- INT16 *in = (INT16 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = in[offset_in[j]];
+ Int16 *in = (Int16 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT24) {
- INT32 *in = (INT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT16) ((in[offset_in[j]] >> 16) & 0x0000ffff);
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 8) & 0x0000ffff);
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT32) {
- INT32 *in = (INT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT16) ((in[offset_in[j]] >> 16) & 0x0000ffff);
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 16) & 0x0000ffff);
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_FLOAT32) {
- FLOAT32 *in = (FLOAT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT16) (in[offset_in[j]] * 32767.0);
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.0);
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_FLOAT64) {
- FLOAT64 *in = (FLOAT64 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (INT16) (in[offset_in[j]] * 32767.0);
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]] * 32767.0);
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
}
- else if (format_out == RTAUDIO_SINT8) {
- signed char *out = (signed char *)output;
- if (format_in == RTAUDIO_SINT8) {
+ else if (info.outFormat == RTAUDIO_SINT8) {
+ signed char *out = (signed char *)outBuffer;
+ if (info.inFormat == RTAUDIO_SINT8) {
// Channel compensation and/or (de)interleaving only.
- signed char *in = (signed char *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = in[offset_in[j]];
+ signed char *in = (signed char *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- if (format_in == RTAUDIO_SINT16) {
- INT16 *in = (INT16 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (signed char) ((in[offset_in[j]] >> 8) & 0x00ff);
+ if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 8) & 0x00ff);
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT24) {
- INT32 *in = (INT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (signed char) ((in[offset_in[j]] >> 24) & 0x000000ff);
+ else if (info.inFormat == RTAUDIO_SINT24) {
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 16) & 0x000000ff);
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT32) {
- INT32 *in = (INT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (signed char) ((in[offset_in[j]] >> 24) & 0x000000ff);
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 24) & 0x000000ff);
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_FLOAT32) {
- FLOAT32 *in = (FLOAT32 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (signed char) (in[offset_in[j]] * 127.0);
+ else if (info.inFormat == RTAUDIO_FLOAT32) {
+ Float32 *in = (Float32 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.0);
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_FLOAT64) {
- FLOAT64 *in = (FLOAT64 *)input;
- for (int i=0; i<stream->bufferSize; i++) {
- for (j=0; j<channels; j++) {
- out[offset_out[j]] = (signed char) (in[offset_in[j]] * 127.0);
+ else if (info.inFormat == RTAUDIO_FLOAT64) {
+ Float64 *in = (Float64 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]] * 127.0);
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
}
}
-void RtAudio :: byteSwapBuffer(char *buffer, int samples, RTAUDIO_FORMAT format)
+void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
{
register char val;
register char *ptr;
ptr = buffer;
- if (format == RTAUDIO_SINT16) {
- for (int i=0; i<samples; i++) {
+ if ( format == RTAUDIO_SINT16 ) {
+ for ( unsigned int i=0; i<samples; i++ ) {
// Swap 1st and 2nd bytes.
val = *(ptr);
*(ptr) = *(ptr+1);
ptr += 2;
}
}
- else if (format == RTAUDIO_SINT24 ||
- format == RTAUDIO_SINT32 ||
- format == RTAUDIO_FLOAT32) {
- for (int i=0; i<samples; i++) {
+ else if ( format == RTAUDIO_SINT24 ||
+ format == RTAUDIO_SINT32 ||
+ format == RTAUDIO_FLOAT32 ) {
+ for ( unsigned int i=0; i<samples; i++ ) {
// Swap 1st and 4th bytes.
val = *(ptr);
*(ptr) = *(ptr+3);
ptr += 4;
}
}
- else if (format == RTAUDIO_FLOAT64) {
- for (int i=0; i<samples; i++) {
+ else if ( format == RTAUDIO_FLOAT64 ) {
+ for ( unsigned int i=0; i<samples; i++ ) {
// Swap 1st and 8th bytes
val = *(ptr);
*(ptr) = *(ptr+7);
}
}
-
-// *************************************************** //
+// Indentation settings for Vim and Emacs
//
-// RtError class definition.
+// Local Variables:
+// c-basic-offset: 2
+// indent-tabs-mode: nil
+// End:
//
-// *************************************************** //
+// vim: et sts=2 sw=2
-RtError :: RtError(const char *p, TYPE tipe)
-{
- type = tipe;
- strncpy(error_message, p, 256);
-}
-
-RtError :: ~RtError()
-{
-}
-
-void RtError :: printMessage()
-{
- printf("\n%s\n\n", error_message);
-}
projects / rtaudio-cdist.git / blobdiff