-/************************************************************************/
+/************************************************************************/
/*! \class RtAudio
\brief Realtime audio i/o C++ classes.
RtAudio provides a common API (Application Programming Interface)
for realtime audio input/output across Linux (native ALSA, Jack,
- and OSS), SGI, Macintosh OS X (CoreAudio), and Windows
- (DirectSound and ASIO) operating systems.
+ and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
+ (DirectSound, ASIO and WASAPI) operating systems.
- RtAudio WWW site: http://music.mcgill.ca/~gary/rtaudio/
+ RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
- RtAudio: a realtime audio i/o C++ class
- Copyright (c) 2001-2004 Gary P. Scavone
+ RtAudio: realtime audio i/o C++ classes
+ Copyright (c) 2001-2017 Gary P. Scavone
Permission is hereby granted, free of charge, to any person
obtaining a copy of this software and associated documentation files
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 3.0.1, 22 March 2004
+// RtAudio: Version 5.0.0
#include "RtAudio.h"
#include <iostream>
+#include <cstdlib>
+#include <cstring>
+#include <climits>
+#include <cmath>
+#include <algorithm>
// Static variable definitions.
const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
32000, 44100, 48000, 88200, 96000, 176400, 192000
};
-#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__)
+#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__) || defined(__WINDOWS_WASAPI__)
#define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
- #define MUTEX_DESTROY(A) DeleteCriticalSection(A);
- #define MUTEX_LOCK(A) EnterCriticalSection(A)
+ #define MUTEX_DESTROY(A) DeleteCriticalSection(A)
+ #define MUTEX_LOCK(A) EnterCriticalSection(A)
#define MUTEX_UNLOCK(A) LeaveCriticalSection(A)
-#else // pthread API
+
+ #include "tchar.h"
+
+ static std::string convertCharPointerToStdString(const char *text)
+ {
+ return std::string(text);
+ }
+
+ static std::string convertCharPointerToStdString(const wchar_t *text)
+ {
+ int length = WideCharToMultiByte(CP_UTF8, 0, text, -1, NULL, 0, NULL, NULL);
+ std::string s( length-1, '\0' );
+ WideCharToMultiByte(CP_UTF8, 0, text, -1, &s[0], length, NULL, NULL);
+ return s;
+ }
+
+#elif defined(__LINUX_ALSA__) || defined(__LINUX_PULSE__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
+ // pthread API
#define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
- #define MUTEX_DESTROY(A) pthread_mutex_destroy(A);
+ #define MUTEX_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( RtAudioApi api )
+std::string RtAudio :: getVersion( void )
{
- initialize( api );
-}
-
-RtAudio :: RtAudio( int outputDevice, int outputChannels,
- int inputDevice, int inputChannels,
- RtAudioFormat format, int sampleRate,
- int *bufferSize, int numberOfBuffers, RtAudioApi api )
-{
- initialize( api );
-
- try {
- rtapi_->openStream( outputDevice, outputChannels,
- inputDevice, inputChannels,
- format, sampleRate,
- bufferSize, numberOfBuffers );
- }
- catch (RtError &exception) {
- // Deallocate the RtApi instance.
- delete rtapi_;
- throw exception;
- }
+ return RTAUDIO_VERSION;
}
-RtAudio :: ~RtAudio()
+void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis )
{
- delete rtapi_;
-}
+ apis.clear();
-void RtAudio :: openStream( int outputDevice, int outputChannels,
- int inputDevice, int inputChannels,
- RtAudioFormat format, int sampleRate,
- int *bufferSize, int numberOfBuffers )
-{
- rtapi_->openStream( outputDevice, outputChannels, inputDevice,
- inputChannels, format, sampleRate,
- bufferSize, numberOfBuffers );
+ // 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_PULSE__)
+ apis.push_back( LINUX_PULSE );
+#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_WASAPI__)
+ apis.push_back( WINDOWS_WASAPI );
+#endif
+#if defined(__WINDOWS_DS__)
+ apis.push_back( WINDOWS_DS );
+#endif
+#if defined(__MACOSX_CORE__)
+ apis.push_back( MACOSX_CORE );
+#endif
+#if defined(__RTAUDIO_DUMMY__)
+ apis.push_back( RTAUDIO_DUMMY );
+#endif
}
-void RtAudio::initialize( RtAudioApi api )
+void RtAudio :: openRtApi( RtAudio::Api api )
{
+ if ( rtapi_ )
+ delete rtapi_;
rtapi_ = 0;
- // First look for a compiled match to a specified API value. If one
- // of these constructors throws an error, it will be passed up the
- // inheritance chain.
-#if defined(__LINUX_JACK__)
- if ( api == LINUX_JACK )
+#if defined(__UNIX_JACK__)
+ if ( api == UNIX_JACK )
rtapi_ = new RtApiJack();
#endif
#if defined(__LINUX_ALSA__)
if ( api == LINUX_ALSA )
rtapi_ = new RtApiAlsa();
#endif
+#if defined(__LINUX_PULSE__)
+ if ( api == LINUX_PULSE )
+ rtapi_ = new RtApiPulse();
+#endif
#if defined(__LINUX_OSS__)
if ( api == LINUX_OSS )
rtapi_ = new RtApiOss();
if ( api == WINDOWS_ASIO )
rtapi_ = new RtApiAsio();
#endif
+#if defined(__WINDOWS_WASAPI__)
+ if ( api == WINDOWS_WASAPI )
+ rtapi_ = new RtApiWasapi();
+#endif
#if defined(__WINDOWS_DS__)
if ( api == WINDOWS_DS )
rtapi_ = new RtApiDs();
#endif
-#if defined(__IRIX_AL__)
- if ( api == IRIX_AL )
- rtapi_ = new RtApiAl();
-#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
+}
- if ( rtapi_ ) return;
- if ( api > 0 ) {
- // No compiled support for specified API value.
- throw RtError( "RtAudio: no compiled support for specified API argument!", RtError::INVALID_PARAMETER );
- }
+RtAudio :: RtAudio( RtAudio::Api api )
+{
+ rtapi_ = 0;
- // No specified API ... search for "best" option.
- try {
-#if defined(__LINUX_JACK__)
- rtapi_ = new RtApiJack();
-#elif defined(__WINDOWS_ASIO__)
- rtapi_ = new RtApiAsio();
-#elif defined(__IRIX_AL__)
- rtapi_ = new RtApiAl();
-#elif defined(__MACOSX_CORE__)
- rtapi_ = new RtApiCore();
-#else
- ;
-#endif
- }
- catch (RtError &) {
-#if defined(__RTAUDIO_DEBUG__)
- fprintf(stderr, "\nRtAudio: no devices found for first api option (JACK, ASIO, Al, or CoreAudio).\n\n");
-#endif
- rtapi_ = 0;
- }
+ if ( api != UNSPECIFIED ) {
+ // Attempt to open the specified API.
+ openRtApi( api );
+ if ( rtapi_ ) return;
- if ( rtapi_ ) return;
+ // No compiled support for specified API value. Issue a debug
+ // warning and continue as if no API was specified.
+ std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
+ }
-// Try second API support
- if ( rtapi_ == 0 ) {
- try {
-#if defined(__LINUX_ALSA__)
- rtapi_ = new RtApiAlsa();
-#elif defined(__WINDOWS_DS__)
- rtapi_ = new RtApiDs();
-#else
- ;
-#endif
- }
- catch (RtError &) {
-#if defined(__RTAUDIO_DEBUG__)
- fprintf(stderr, "\nRtAudio: no devices found for second api option (Alsa or DirectSound).\n\n");
-#endif
- rtapi_ = 0;
- }
+ // Iterate through the compiled APIs and return as soon as we find
+ // one with at least one device or we reach the end of the list.
+ std::vector< RtAudio::Api > apis;
+ getCompiledApi( apis );
+ for ( unsigned int i=0; i<apis.size(); i++ ) {
+ openRtApi( apis[i] );
+ if ( rtapi_ && rtapi_->getDeviceCount() ) break;
}
if ( rtapi_ ) return;
- // Try third API support
- if ( rtapi_ == 0 ) {
-#if defined(__LINUX_OSS__)
- try {
- rtapi_ = new RtApiOss();
- }
- catch (RtError &error) {
- rtapi_ = 0;
- }
-#else
- ;
-#endif
- }
+ // It should not be possible to get here because the preprocessor
+ // definition __RTAUDIO_DUMMY__ is automatically defined if no
+ // API-specific definitions are passed to the compiler. But just in
+ // case something weird happens, we'll thow an error.
+ std::string errorText = "\nRtAudio: no compiled API support found ... critical error!!\n\n";
+ throw( RtAudioError( errorText, RtAudioError::UNSPECIFIED ) );
+}
- if ( rtapi_ == 0 ) {
- // No devices found.
- throw RtError( "RtAudio: no devices found for compiled audio APIs!", RtError::NO_DEVICES_FOUND );
- }
+RtAudio :: ~RtAudio()
+{
+ if ( rtapi_ )
+ delete rtapi_;
+}
+
+void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
+ RtAudio::StreamParameters *inputParameters,
+ RtAudioFormat format, unsigned int sampleRate,
+ unsigned int *bufferFrames,
+ RtAudioCallback callback, void *userData,
+ RtAudio::StreamOptions *options,
+ RtAudioErrorCallback errorCallback )
+{
+ return rtapi_->openStream( outputParameters, inputParameters, format,
+ sampleRate, bufferFrames, callback,
+ userData, options, errorCallback );
}
+// *************************************************** //
+//
+// Public RtApi definitions (see end of file for
+// private or protected utility functions).
+//
+// *************************************************** //
+
RtApi :: RtApi()
{
+ stream_.state = STREAM_CLOSED;
stream_.mode = UNINITIALIZED;
stream_.apiHandle = 0;
- MUTEX_INITIALIZE(&stream_.mutex);
+ stream_.userBuffer[0] = 0;
+ stream_.userBuffer[1] = 0;
+ MUTEX_INITIALIZE( &stream_.mutex );
+ showWarnings_ = true;
+ firstErrorOccurred_ = false;
}
RtApi :: ~RtApi()
{
- MUTEX_DESTROY(&stream_.mutex);
+ MUTEX_DESTROY( &stream_.mutex );
}
-void RtApi :: openStream( int outputDevice, int outputChannels,
- int inputDevice, int inputChannels,
- RtAudioFormat format, int sampleRate,
- int *bufferSize, int numberOfBuffers )
+void RtApi :: openStream( RtAudio::StreamParameters *oParams,
+ RtAudio::StreamParameters *iParams,
+ RtAudioFormat format, unsigned int sampleRate,
+ unsigned int *bufferFrames,
+ RtAudioCallback callback, void *userData,
+ RtAudio::StreamOptions *options,
+ RtAudioErrorCallback errorCallback )
{
- if ( stream_.mode != UNINITIALIZED ) {
- sprintf(message_, "RtApi: only one open stream allowed per class instance.");
- error(RtError::INVALID_STREAM);
+ if ( stream_.state != STREAM_CLOSED ) {
+ errorText_ = "RtApi::openStream: a stream is already open!";
+ error( RtAudioError::INVALID_USE );
+ return;
}
- if (outputChannels < 1 && inputChannels < 1) {
- sprintf(message_,"RtApi: one or both 'channel' parameters must be greater than zero.");
- error(RtError::INVALID_PARAMETER);
- }
+ // Clear stream information potentially left from a previously open stream.
+ clearStreamInfo();
- if ( formatBytes(format) == 0 ) {
- sprintf(message_,"RtApi: '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( RtAudioError::INVALID_USE );
+ return;
}
- if ( outputChannels > 0 ) {
- if (outputDevice > nDevices_ || outputDevice < 0) {
- sprintf(message_,"RtApi: '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( RtAudioError::INVALID_USE );
+ return;
}
- if ( inputChannels > 0 ) {
- if (inputDevice > nDevices_ || inputDevice < 0) {
- sprintf(message_,"RtApi: '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( RtAudioError::INVALID_USE );
+ return;
}
- clearStreamInfo();
- bool result = FAILURE;
- int device, defaultDevice = 0;
- StreamMode mode;
- int channels;
- if ( outputChannels > 0 ) {
-
- mode = OUTPUT;
- channels = outputChannels;
+ if ( formatBytes(format) == 0 ) {
+ errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
- 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( RtAudioError::INVALID_USE );
+ return;
}
- 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, mode, channels, sampleRate,
- format, bufferSize, numberOfBuffers);
- if ( result == SUCCESS ) break;
- if ( outputDevice > 0 ) break;
- clearStreamInfo();
+ unsigned int iChannels = 0;
+ if ( iParams ) {
+ iChannels = iParams->nChannels;
+ if ( iParams->deviceId >= nDevices ) {
+ errorText_ = "RtApi::openStream: input device parameter value is invalid.";
+ error( RtAudioError::INVALID_USE );
+ return;
}
}
- if ( inputChannels > 0 && ( result == SUCCESS || outputChannels <= 0 ) ) {
+ bool result;
- mode = INPUT;
- channels = inputChannels;
+ if ( oChannels > 0 ) {
- if ( inputDevice == 0 ) { // Try default device first.
- defaultDevice = getDefaultInputDevice();
- device = defaultDevice;
+ result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
+ sampleRate, format, bufferFrames, options );
+ if ( result == false ) {
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
}
- else
- device = inputDevice - 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, 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( RtAudioError::SYSTEM_ERROR );
+ return;
}
}
- if ( result == SUCCESS )
- return;
+ stream_.callbackInfo.callback = (void *) callback;
+ stream_.callbackInfo.userData = userData;
+ stream_.callbackInfo.errorCallback = (void *) errorCallback;
- // If we get here, all attempted probes failed. Close any opened
- // devices and clear the stream structure.
- if ( stream_.mode != UNINITIALIZED ) closeStream();
- clearStreamInfo();
- if ( ( outputDevice == 0 && outputChannels > 0 )
- || ( inputDevice == 0 && inputChannels > 0 ) )
- sprintf(message_,"RtApi: no devices found for given stream parameters.");
- else
- sprintf(message_,"RtApi: unable to open specified device(s) with given stream parameters.");
- error(RtError::INVALID_PARAMETER);
+ if ( options ) options->numberOfBuffers = stream_.nBuffers;
+ stream_.state = STREAM_STOPPED;
+}
- return;
+unsigned int RtApi :: getDefaultInputDevice( void )
+{
+ // Should be implemented in subclasses if possible.
+ return 0;
}
-int RtApi :: getDeviceCount(void)
+unsigned int RtApi :: getDefaultOutputDevice( void )
{
- return devices_.size();
+ // Should be implemented in subclasses if possible.
+ return 0;
}
-RtAudioDeviceInfo RtApi :: getDeviceInfo( int device )
+void RtApi :: closeStream( void )
{
- if (device > (int) devices_.size() || device < 1) {
- sprintf(message_, "RtApi: invalid device specifier (%d)!", device);
- error(RtError::INVALID_DEVICE);
- }
+ // MUST be implemented in subclasses!
+ return;
+}
- RtAudioDeviceInfo info;
- int deviceIndex = device - 1;
+bool RtApi :: probeDeviceOpen( unsigned int /*device*/, StreamMode /*mode*/, unsigned int /*channels*/,
+ unsigned int /*firstChannel*/, unsigned int /*sampleRate*/,
+ RtAudioFormat /*format*/, unsigned int * /*bufferSize*/,
+ RtAudio::StreamOptions * /*options*/ )
+{
+ // MUST be implemented in subclasses!
+ return FAILURE;
+}
- // If the device wasn't successfully probed before, try it now (or again).
- if (devices_[deviceIndex].probed == false) {
- clearDeviceInfo(&devices_[deviceIndex]);
- probeDeviceInfo(&devices_[deviceIndex]);
- }
+void RtApi :: tickStreamTime( void )
+{
+ // Subclasses that do not provide their own implementation of
+ // getStreamTime should call this function once per buffer I/O to
+ // provide basic stream time support.
- info.name.append( devices_[deviceIndex].name );
- info.probed = devices_[deviceIndex].probed;
- if ( info.probed == true ) {
- info.outputChannels = devices_[deviceIndex].maxOutputChannels;
- info.inputChannels = devices_[deviceIndex].maxInputChannels;
- info.duplexChannels = devices_[deviceIndex].maxDuplexChannels;
- for (unsigned int i=0; i<devices_[deviceIndex].sampleRates.size(); i++)
- info.sampleRates.push_back( devices_[deviceIndex].sampleRates[i] );
- info.nativeFormats = devices_[deviceIndex].nativeFormats;
- if ( (deviceIndex == getDefaultOutputDevice()) ||
- (deviceIndex == getDefaultInputDevice()) )
- info.isDefault = true;
- }
+ stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
- return info;
+#if defined( HAVE_GETTIMEOFDAY )
+ gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
}
-char * const RtApi :: getStreamBuffer(void)
+long RtApi :: getStreamLatency( void )
{
verifyStream();
- return stream_.userBuffer;
-}
-int RtApi :: getDefaultInputDevice(void)
-{
- // Should be implemented in subclasses if appropriate.
- return 0;
-}
+ 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];
-int RtApi :: getDefaultOutputDevice(void)
-{
- // Should be implemented in subclasses if appropriate.
- return 0;
+ return totalLatency;
}
-void RtApi :: closeStream(void)
+double RtApi :: getStreamTime( void )
{
- // MUST be implemented in subclasses!
+ verifyStream();
+
+#if defined( HAVE_GETTIMEOFDAY )
+ // Return a very accurate estimate of the stream time by
+ // adding in the elapsed time since the last tick.
+ struct timeval then;
+ struct timeval now;
+
+ if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
+ return stream_.streamTime;
+
+ gettimeofday( &now, NULL );
+ then = stream_.lastTickTimestamp;
+ return stream_.streamTime +
+ ((now.tv_sec + 0.000001 * now.tv_usec) -
+ (then.tv_sec + 0.000001 * then.tv_usec));
+#else
+ return stream_.streamTime;
+#endif
}
-void RtApi :: probeDeviceInfo( RtApiDevice *info )
+void RtApi :: setStreamTime( double time )
{
- // MUST be implemented in subclasses!
+ verifyStream();
+
+ if ( time >= 0.0 )
+ stream_.streamTime = time;
+#if defined( HAVE_GETTIMEOFDAY )
+ gettimeofday( &stream_.lastTickTimestamp, NULL );
+#endif
}
-bool RtApi :: probeDeviceOpen( int device, StreamMode mode, int channels,
- int sampleRate, RtAudioFormat format,
- int *bufferSize, int numberOfBuffers )
+unsigned int RtApi :: getStreamSampleRate( void )
{
- // MUST be implemented in subclasses!
- return FAILURE;
+ verifyStream();
+
+ return stream_.sampleRate;
}
//
// *************************************************** //
-#if defined(__LINUX_OSS__)
-
-#include <unistd.h>
-#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>
+#if defined(__MACOSX_CORE__)
-#define DAC_NAME "/dev/dsp"
-#define MAX_DEVICES 16
-#define MAX_CHANNELS 16
+// The OS X CoreAudio API is designed to use a separate callback
+// procedure for each of its audio devices. A single RtAudio duplex
+// stream using two different devices is supported here, though it
+// cannot be guaranteed to always behave correctly because we cannot
+// synchronize these two callbacks.
+//
+// A property listener is installed for over/underrun information.
+// However, no functionality is currently provided to allow property
+// listeners to trigger user handlers because it is unclear what could
+// be done if a critical stream parameter (buffer size, sample rate,
+// device disconnect) notification arrived. The listeners entail
+// quite a bit of extra code and most likely, a user program wouldn't
+// be prepared for the result anyway. However, we do provide a flag
+// to the client callback function to inform of an over/underrun.
-extern "C" void *ossCallbackHandler(void * ptr);
+// A structure to hold various information related to the CoreAudio API
+// implementation.
+struct CoreHandle {
+ AudioDeviceID id[2]; // device ids
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ AudioDeviceIOProcID procId[2];
+#endif
+ UInt32 iStream[2]; // device stream index (or first if using multiple)
+ UInt32 nStreams[2]; // number of streams to use
+ bool xrun[2];
+ char *deviceBuffer;
+ pthread_cond_t condition;
+ int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
-RtApiOss :: RtApiOss()
-{
- this->initialize();
+ CoreHandle()
+ :deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
+};
- if (nDevices_ <= 0) {
- sprintf(message_, "RtApiOss: no Linux OSS audio devices found!");
- error(RtError::NO_DEVICES_FOUND);
- }
+RtApiCore:: RtApiCore()
+{
+#if defined( AVAILABLE_MAC_OS_X_VERSION_10_6_AND_LATER )
+ // This is a largely undocumented but absolutely necessary
+ // requirement starting with OS-X 10.6. If not called, queries and
+ // updates to various audio device properties are not handled
+ // correctly.
+ CFRunLoopRef theRunLoop = NULL;
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyRunLoop,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectSetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, sizeof(CFRunLoopRef), &theRunLoop);
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::RtApiCore: error setting run loop property!";
+ error( RtAudioError::WARNING );
+ }
+#endif
}
-RtApiOss :: ~RtApiOss()
+RtApiCore :: ~RtApiCore()
{
- if ( stream_.mode != UNINITIALIZED )
- closeStream();
+ // 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();
}
-void RtApiOss :: initialize(void)
+unsigned int RtApiCore :: getDeviceCount( void )
{
- // 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_, "RtApiOss: cannot read value of symbolic link %s.", DAC_NAME);
- error(RtError::SYSTEM_ERROR);
- }
- }
+ // Find out how many audio devices there are, if any.
+ UInt32 dataSize;
+ AudioObjectPropertyAddress propertyAddress = { kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyDataSize( kAudioObjectSystemObject, &propertyAddress, 0, NULL, &dataSize );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
+ error( RtAudioError::WARNING );
+ return 0;
}
- else {
- sprintf(message_, "RtApiOss: 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;
- RtApiDevice device;
- 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_, "RtApiOss: OSS record device (%s) is busy.", device_name);
- error(RtError::WARNING);
- // still count it for now
- }
- }
- }
- else {
- sprintf(message_, "RtApiOss: OSS playback device (%s) is busy.", device_name);
- error(RtError::WARNING);
- // still count it for now
- }
- }
- if (fd >= 0) close(fd);
- device.name.erase();
- device.name.append( (const char *)device_name, strlen(device_name)+1);
- devices_.push_back(device);
- nDevices_++;
- }
+ return dataSize / sizeof( AudioDeviceID );
}
-void RtApiOss :: probeDeviceInfo(RtApiDevice *info)
+unsigned int RtApiCore :: getDefaultInputDevice( void )
{
- 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.
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices <= 1 ) return 0;
- // First try for playback
- fd = open(info->name.c_str(), O_WRONLY | O_NONBLOCK);
- if (fd == -1) {
- // Open device failed ... either busy or doesn't exist
- if (errno == EBUSY || errno == EAGAIN)
- sprintf(message_, "RtApiOss: OSS playback device (%s) is busy and cannot be probed.",
- info->name.c_str());
- else
- sprintf(message_, "RtApiOss: OSS playback device (%s) open error.", info->name.c_str());
- 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;
+ AudioDeviceID id;
+ UInt32 dataSize = sizeof( AudioDeviceID );
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultInputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
+ error( RtAudioError::WARNING );
+ return 0;
}
- info->maxOutputChannels = i;
- // 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;
- }
- info->minOutputChannels = i;
- close(fd);
-
- capture_probe:
- // Now try for capture
- fd = open(info->name.c_str(), 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_, "RtApiOss: OSS capture device (%s) is busy and cannot be probed.",
- info->name.c_str());
- else
- sprintf(message_, "RtApiOss: OSS capture device (%s) open error.", info->name.c_str());
- error(RtError::DEBUG_WARNING);
- if (info->maxOutputChannels == 0)
- // didn't open for playback either ... device invalid
- return;
- goto probe_parameters;
+ dataSize *= nDevices;
+ AudioDeviceID deviceList[ nDevices ];
+ property.mSelector = kAudioHardwarePropertyDevices;
+ result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
+ error( RtAudioError::WARNING );
+ return 0;
}
- // 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 here, we found a working channel value
- break;
- }
- info->maxInputChannels = i;
+ for ( unsigned int i=0; i<nDevices; i++ )
+ if ( id == deviceList[i] ) return 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;
+ errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
+ error( RtAudioError::WARNING );
+ return 0;
+}
+
+unsigned int RtApiCore :: getDefaultOutputDevice( void )
+{
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices <= 1 ) return 0;
+
+ AudioDeviceID id;
+ UInt32 dataSize = sizeof( AudioDeviceID );
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDefaultOutputDevice, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, &id );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
+ error( RtAudioError::WARNING );
+ return 0;
}
- info->minInputChannels = i;
- close(fd);
- if (info->maxOutputChannels == 0 && info->maxInputChannels == 0) {
- sprintf(message_, "RtApiOss: device (%s) reports zero channels for input and output.",
- info->name.c_str());
- error(RtError::DEBUG_WARNING);
- return;
+ dataSize = sizeof( AudioDeviceID ) * nDevices;
+ AudioDeviceID deviceList[ nDevices ];
+ property.mSelector = kAudioHardwarePropertyDevices;
+ result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property, 0, NULL, &dataSize, (void *) &deviceList );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
+ error( RtAudioError::WARNING );
+ return 0;
}
- // 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.c_str(), 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;
+ for ( unsigned int i=0; i<nDevices; i++ )
+ if ( id == deviceList[i] ) return i;
- // 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;
- }
- close(fd);
+ errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
+ error( RtAudioError::WARNING );
+ return 0;
+}
- 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.
+RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
- if (info->maxOutputChannels >= info->maxInputChannels) {
- fd = open(info->name.c_str(), O_WRONLY | O_NONBLOCK);
- channels = info->maxOutputChannels;
- }
- else {
- fd = open(info->name.c_str(), O_RDONLY | O_NONBLOCK);
- channels = info->maxInputChannels;
+ // Get device ID
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices == 0 ) {
+ errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
+ error( RtAudioError::INVALID_USE );
+ return info;
}
- if (fd == -1) {
- // We've got some sort of conflict ... abort
- sprintf(message_, "RtApiOss: device (%s) won't reopen during probe.",
- info->name.c_str());
- error(RtError::DEBUG_WARNING);
- return;
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ 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_, "RtApiOss: device (%s) won't revert to previous channel setting.",
- info->name.c_str());
- error(RtError::DEBUG_WARNING);
- return;
+ AudioDeviceID deviceList[ nDevices ];
+ UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
+ 0, NULL, &dataSize, (void *) &deviceList );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
+ error( RtAudioError::WARNING );
+ return info;
}
- if (ioctl(fd, SNDCTL_DSP_GETFMTS, &mask) == -1) {
- close(fd);
- sprintf(message_, "RtApiOss: device (%s) can't get supported audio formats.",
- info->name.c_str());
- error(RtError::DEBUG_WARNING);
- return;
- }
+ AudioDeviceID id = deviceList[ device ];
- // 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;
+ // Get the device name.
+ info.name.erase();
+ CFStringRef cfname;
+ dataSize = sizeof( CFStringRef );
+ property.mSelector = kAudioObjectPropertyManufacturer;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
}
+
+ //const char *mname = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
+ int length = CFStringGetLength(cfname);
+ char *mname = (char *)malloc(length * 3 + 1);
+#if defined( UNICODE ) || defined( _UNICODE )
+ CFStringGetCString(cfname, mname, length * 3 + 1, kCFStringEncodingUTF8);
+#else
+ CFStringGetCString(cfname, mname, length * 3 + 1, CFStringGetSystemEncoding());
#endif
- 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;
+ info.name.append( (const char *)mname, strlen(mname) );
+ info.name.append( ": " );
+ CFRelease( cfname );
+ free(mname);
+
+ property.mSelector = kAudioObjectPropertyName;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &cfname );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ //const char *name = CFStringGetCStringPtr( cfname, CFStringGetSystemEncoding() );
+ length = CFStringGetLength(cfname);
+ char *name = (char *)malloc(length * 3 + 1);
+#if defined( UNICODE ) || defined( _UNICODE )
+ CFStringGetCString(cfname, name, length * 3 + 1, kCFStringEncodingUTF8);
+#else
+ CFStringGetCString(cfname, name, length * 3 + 1, CFStringGetSystemEncoding());
+#endif
+ info.name.append( (const char *)name, strlen(name) );
+ CFRelease( cfname );
+ free(name);
+
+ // Get the output stream "configuration".
+ AudioBufferList *bufferList = nil;
+ property.mSelector = kAudioDevicePropertyStreamConfiguration;
+ property.mScope = kAudioDevicePropertyScopeOutput;
+ // property.mElement = kAudioObjectPropertyElementWildcard;
+ dataSize = 0;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+ if ( result != noErr || dataSize == 0 ) {
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Allocate the AudioBufferList.
+ bufferList = (AudioBufferList *) malloc( dataSize );
+ if ( bufferList == NULL ) {
+ errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+ if ( result != noErr || dataSize == 0 ) {
+ free( bufferList );
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
}
- // Check that we have at least one supported format
- if (info->nativeFormats == 0) {
- close(fd);
- sprintf(message_, "RtApiOss: device (%s) data format not supported by RtAudio.",
- info->name.c_str());
- error(RtError::DEBUG_WARNING);
- return;
+ // Get output channel information.
+ unsigned int i, nStreams = bufferList->mNumberBuffers;
+ for ( i=0; i<nStreams; i++ )
+ info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
+ free( bufferList );
+
+ // Get the input stream "configuration".
+ property.mScope = kAudioDevicePropertyScopeInput;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+ if ( result != noErr || dataSize == 0 ) {
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Allocate the AudioBufferList.
+ bufferList = (AudioBufferList *) malloc( dataSize );
+ if ( bufferList == NULL ) {
+ errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+ if (result != noErr || dataSize == 0) {
+ free( bufferList );
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
}
- // Set the format
- i = format;
- if (ioctl(fd, SNDCTL_DSP_SETFMT, &format) == -1 || format != i) {
- close(fd);
- sprintf(message_, "RtApiOss: device (%s) error setting data format.",
- info->name.c_str());
- error(RtError::DEBUG_WARNING);
- return;
+ // Get input channel information.
+ nStreams = bufferList->mNumberBuffers;
+ for ( i=0; i<nStreams; i++ )
+ info.inputChannels += bufferList->mBuffers[i].mNumberChannels;
+ free( bufferList );
+
+ // If device opens for both playback and capture, we determine the channels.
+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+
+ // Probe the device sample rates.
+ bool isInput = false;
+ if ( info.outputChannels == 0 ) isInput = true;
+
+ // Determine the supported sample rates.
+ property.mSelector = kAudioDevicePropertyAvailableNominalSampleRates;
+ if ( isInput == false ) property.mScope = kAudioDevicePropertyScopeOutput;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+ if ( result != kAudioHardwareNoError || dataSize == 0 ) {
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ UInt32 nRanges = dataSize / sizeof( AudioValueRange );
+ AudioValueRange rangeList[ nRanges ];
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &rangeList );
+ if ( result != kAudioHardwareNoError ) {
+ errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // The sample rate reporting mechanism is a bit of a mystery. It
+ // seems that it can either return individual rates or a range of
+ // rates. I assume that if the min / max range values are the same,
+ // then that represents a single supported rate and if the min / max
+ // range values are different, the device supports an arbitrary
+ // range of values (though there might be multiple ranges, so we'll
+ // use the most conservative range).
+ Float64 minimumRate = 1.0, maximumRate = 10000000000.0;
+ bool haveValueRange = false;
+ info.sampleRates.clear();
+ for ( UInt32 i=0; i<nRanges; i++ ) {
+ if ( rangeList[i].mMinimum == rangeList[i].mMaximum ) {
+ unsigned int tmpSr = (unsigned int) rangeList[i].mMinimum;
+ info.sampleRates.push_back( tmpSr );
+
+ if ( !info.preferredSampleRate || ( tmpSr <= 48000 && tmpSr > info.preferredSampleRate ) )
+ info.preferredSampleRate = tmpSr;
+
+ } else {
+ haveValueRange = true;
+ if ( rangeList[i].mMinimum > minimumRate ) minimumRate = rangeList[i].mMinimum;
+ if ( rangeList[i].mMaximum < maximumRate ) maximumRate = rangeList[i].mMaximum;
+ }
+ }
+
+ if ( haveValueRange ) {
+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+ if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate ) {
+ info.sampleRates.push_back( SAMPLE_RATES[k] );
+
+ if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
+ info.preferredSampleRate = SAMPLE_RATES[k];
+ }
+ }
}
- // Probe the supported sample rates.
- info->sampleRates.clear();
- for (unsigned int k=0; k<MAX_SAMPLE_RATES; k++) {
- int speed = SAMPLE_RATES[k];
- if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) != -1 && speed == (int)SAMPLE_RATES[k])
- info->sampleRates.push_back(speed);
- }
-
- if (info->sampleRates.size() == 0) {
- close(fd);
- sprintf(message_, "RtApiOss: no supported sample rates found for device (%s).",
- info->name.c_str());
- error(RtError::DEBUG_WARNING);
- return;
+ // Sort and remove any redundant values
+ std::sort( info.sampleRates.begin(), info.sampleRates.end() );
+ info.sampleRates.erase( unique( info.sampleRates.begin(), info.sampleRates.end() ), info.sampleRates.end() );
+
+ if ( info.sampleRates.size() == 0 ) {
+ errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
}
- // That's all ... close the device and return
- close(fd);
- info->probed = true;
- return;
+ // CoreAudio always uses 32-bit floating point data for PCM streams.
+ // Thus, any other "physical" formats supported by the device are of
+ // no interest to the client.
+ info.nativeFormats = RTAUDIO_FLOAT32;
+
+ if ( info.outputChannels > 0 )
+ if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
+ if ( info.inputChannels > 0 )
+ if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
+
+ info.probed = true;
+ return info;
}
-bool RtApiOss :: probeDeviceOpen(int device, StreamMode mode, int channels,
- int sampleRate, RtAudioFormat format,
- int *bufferSize, int numberOfBuffers)
+static OSStatus callbackHandler( AudioDeviceID inDevice,
+ const AudioTimeStamp* /*inNow*/,
+ const AudioBufferList* inInputData,
+ const AudioTimeStamp* /*inInputTime*/,
+ AudioBufferList* outOutputData,
+ const AudioTimeStamp* /*inOutputTime*/,
+ void* infoPointer )
{
- int buffers, buffer_bytes, device_channels, device_format;
- int srate, temp, fd;
- int *handle = (int *) stream_.apiHandle;
+ CallbackInfo *info = (CallbackInfo *) infoPointer;
- const char *name = devices_[device].name.c_str();
+ RtApiCore *object = (RtApiCore *) info->object;
+ if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
+ return kAudioHardwareUnspecifiedError;
+ else
+ return kAudioHardwareNoError;
+}
- 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(handle[0]);
- handle[0] = 0;
- // First check that the number previously set channels is the same.
- if (stream_.nUserChannels[0] != channels) {
- sprintf(message_, "RtApiOss: input/output channels must be equal for OSS duplex device (%s).", name);
- goto error;
- }
- fd = open(name, O_RDWR | O_NONBLOCK);
+static OSStatus xrunListener( AudioObjectID /*inDevice*/,
+ UInt32 nAddresses,
+ const AudioObjectPropertyAddress properties[],
+ void* handlePointer )
+{
+ CoreHandle *handle = (CoreHandle *) handlePointer;
+ for ( UInt32 i=0; i<nAddresses; i++ ) {
+ if ( properties[i].mSelector == kAudioDeviceProcessorOverload ) {
+ if ( properties[i].mScope == kAudioDevicePropertyScopeInput )
+ handle->xrun[1] = true;
+ else
+ handle->xrun[0] = true;
}
- else
- fd = open(name, O_RDONLY | O_NONBLOCK);
}
- if (fd == -1) {
- if (errno == EBUSY || errno == EAGAIN)
- sprintf(message_, "RtApiOss: device (%s) is busy and cannot be opened.",
- name);
- else
- sprintf(message_, "RtApiOss: device (%s) cannot be opened.", name);
- goto error;
+ return kAudioHardwareNoError;
+}
+
+static OSStatus rateListener( AudioObjectID inDevice,
+ UInt32 /*nAddresses*/,
+ const AudioObjectPropertyAddress /*properties*/[],
+ void* ratePointer )
+{
+ Float64 *rate = (Float64 *) ratePointer;
+ UInt32 dataSize = sizeof( Float64 );
+ AudioObjectPropertyAddress property = { kAudioDevicePropertyNominalSampleRate,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ AudioObjectGetPropertyData( inDevice, &property, 0, NULL, &dataSize, rate );
+ return kAudioHardwareNoError;
+}
+
+bool RtApiCore :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ // Get device ID
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices == 0 ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiCore::probeDeviceOpen: no devices found!";
+ return FAILURE;
}
- // 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);
+ 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;
}
- if (fd == -1) {
- sprintf(message_, "RtApiOss: device (%s) cannot be opened.", name);
- goto error;
+ AudioDeviceID deviceList[ nDevices ];
+ UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+ OSStatus result = AudioObjectGetPropertyData( kAudioObjectSystemObject, &property,
+ 0, NULL, &dataSize, (void *) &deviceList );
+ if ( result != noErr ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";
+ return FAILURE;
}
- // Get the sample format mask
- int mask;
- if (ioctl(fd, SNDCTL_DSP_GETFMTS, &mask) == -1) {
- close(fd);
- sprintf(message_, "RtApiOss: device (%s) can't get supported audio formats.",
- name);
- goto error;
+ AudioDeviceID id = deviceList[ device ];
+
+ // Setup for stream mode.
+ bool isInput = false;
+ if ( mode == INPUT ) {
+ isInput = true;
+ property.mScope = kAudioDevicePropertyScopeInput;
}
+ else
+ property.mScope = kAudioDevicePropertyScopeOutput;
- // 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;
- }
+ // Get the stream "configuration".
+ AudioBufferList *bufferList = nil;
+ dataSize = 0;
+ property.mSelector = kAudioDevicePropertyStreamConfiguration;
+ result = AudioObjectGetPropertyDataSize( id, &property, 0, NULL, &dataSize );
+ if ( result != noErr || dataSize == 0 ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- 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
+
+ // Allocate the AudioBufferList.
+ bufferList = (AudioBufferList *) malloc( dataSize );
+ if ( bufferList == NULL ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";
+ return FAILURE;
}
-#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
+
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, bufferList );
+ if (result != noErr || dataSize == 0) {
+ free( bufferList );
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
-#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;
+ // Search for one or more streams that contain the desired number of
+ // channels. CoreAudio devices can have an arbitrary number of
+ // streams and each stream can have an arbitrary number of channels.
+ // For each stream, a single buffer of interleaved samples is
+ // provided. RtAudio prefers the use of one stream of interleaved
+ // data or multiple consecutive single-channel streams. However, we
+ // now support multiple consecutive multi-channel streams of
+ // interleaved data as well.
+ UInt32 iStream, offsetCounter = firstChannel;
+ UInt32 nStreams = bufferList->mNumberBuffers;
+ bool monoMode = false;
+ bool foundStream = false;
+
+ // First check that the device supports the requested number of
+ // channels.
+ UInt32 deviceChannels = 0;
+ for ( iStream=0; iStream<nStreams; iStream++ )
+ deviceChannels += bufferList->mBuffers[iStream].mNumberChannels;
+
+ if ( deviceChannels < ( channels + firstChannel ) ) {
+ free( bufferList );
+ errorStream_ << "RtApiCore::probeDeviceOpen: the device (" << device << ") does not support the requested channel count.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Look for a single stream meeting our needs.
+ UInt32 firstStream, streamCount = 1, streamChannels = 0, channelOffset = 0;
+ for ( iStream=0; iStream<nStreams; iStream++ ) {
+ streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
+ if ( streamChannels >= channels + offsetCounter ) {
+ firstStream = iStream;
+ channelOffset = offsetCounter;
+ foundStream = true;
+ break;
}
-#else
- else if (mask & AFMT_S32_LE) {
- device_format = AFMT_S32_LE;
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;
- stream_.doByteSwap[mode] = true;
+ if ( streamChannels > offsetCounter ) break;
+ offsetCounter -= streamChannels;
+ }
+
+ // If we didn't find a single stream above, then we should be able
+ // to meet the channel specification with multiple streams.
+ if ( foundStream == false ) {
+ monoMode = true;
+ offsetCounter = firstChannel;
+ for ( iStream=0; iStream<nStreams; iStream++ ) {
+ streamChannels = bufferList->mBuffers[iStream].mNumberChannels;
+ if ( streamChannels > offsetCounter ) break;
+ offsetCounter -= streamChannels;
}
-#endif
-#endif
- else if (mask & AFMT_S8) {
- device_format = AFMT_S8;
- stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+
+ firstStream = iStream;
+ channelOffset = offsetCounter;
+ Int32 channelCounter = channels + offsetCounter - streamChannels;
+
+ if ( streamChannels > 1 ) monoMode = false;
+ while ( channelCounter > 0 ) {
+ streamChannels = bufferList->mBuffers[++iStream].mNumberChannels;
+ if ( streamChannels > 1 ) monoMode = false;
+ channelCounter -= streamChannels;
+ streamCount++;
}
}
- if (stream_.deviceFormat[mode] == 0) {
- // This really shouldn't happen ...
- close(fd);
- sprintf(message_, "RtApiOss: device (%s) data format not supported by RtAudio.",
- name);
- goto error;
+ free( bufferList );
+
+ // Determine the buffer size.
+ AudioValueRange bufferRange;
+ dataSize = sizeof( AudioValueRange );
+ property.mSelector = kAudioDevicePropertyBufferFrameSizeRange;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &bufferRange );
+
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- // 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;
+ if ( bufferRange.mMinimum > *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMinimum;
+ else if ( bufferRange.mMaximum < *bufferSize ) *bufferSize = (unsigned long) bufferRange.mMaximum;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) *bufferSize = (unsigned long) bufferRange.mMinimum;
+
+ // Set the buffer size. For multiple streams, I'm assuming we only
+ // need to make this setting for the master channel.
+ UInt32 theSize = (UInt32) *bufferSize;
+ dataSize = sizeof( UInt32 );
+ property.mSelector = kAudioDevicePropertyBufferFrameSize;
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &theSize );
+
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- 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 attempting to setup a duplex stream, the bufferSize parameter
+ // MUST be the same in both directions!
+ *bufferSize = theSize;
+ if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ stream_.bufferSize = *bufferSize;
+ stream_.nBuffers = 1;
+
+ // Try to set "hog" mode ... it's not clear to me this is working.
+ if ( options && options->flags & RTAUDIO_HOG_DEVICE ) {
+ pid_t hog_pid;
+ dataSize = sizeof( hog_pid );
+ property.mSelector = kAudioDevicePropertyHogMode;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &hog_pid );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting 'hog' state!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- else {
- if (channels < devices_[device].minInputChannels)
- device_channels = devices_[device].minInputChannels;
+
+ if ( hog_pid != getpid() ) {
+ hog_pid = getpid();
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &hog_pid );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
}
}
- stream_.nDeviceChannels[mode] = device_channels;
- // 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_, "RtApiOss: error setting fragment size for device (%s).",
- name);
- goto error;
+ // Check and if necessary, change the sample rate for the device.
+ Float64 nominalRate;
+ dataSize = sizeof( Float64 );
+ property.mSelector = kAudioDevicePropertyNominalSampleRate;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &nominalRate );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting current sample rate.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- 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_, "RtApiOss: error setting data format for device (%s).",
- name);
- goto error;
- }
+ // Only change the sample rate if off by more than 1 Hz.
+ if ( fabs( nominalRate - (double)sampleRate ) > 1.0 ) {
- // Set the number of channels.
- temp = device_channels;
- if (ioctl(fd, SNDCTL_DSP_CHANNELS, &device_channels) == -1 || device_channels != temp) {
- close(fd);
- sprintf(message_, "RtApiOss: error setting %d channels on device (%s).",
- temp, name);
- goto error;
- }
+ // Set a property listener for the sample rate change
+ Float64 reportedRate = 0.0;
+ AudioObjectPropertyAddress tmp = { kAudioDevicePropertyNominalSampleRate, kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyElementMaster };
+ result = AudioObjectAddPropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate property listener for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // Set the sample rate.
- srate = sampleRate;
- temp = srate;
- if (ioctl(fd, SNDCTL_DSP_SPEED, &srate) == -1) {
- close(fd);
- sprintf(message_, "RtApiOss: error setting sample rate = %d on device (%s).",
- temp, name);
- goto error;
+ nominalRate = (Float64) sampleRate;
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &nominalRate );
+ if ( result != noErr ) {
+ AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Now wait until the reported nominal rate is what we just set.
+ UInt32 microCounter = 0;
+ while ( reportedRate != nominalRate ) {
+ microCounter += 5000;
+ if ( microCounter > 5000000 ) break;
+ usleep( 5000 );
+ }
+
+ // Remove the property listener.
+ AudioObjectRemovePropertyListener( id, &tmp, rateListener, (void *) &reportedRate );
+
+ if ( microCounter > 5000000 ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: timeout waiting for sample rate update for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
}
- // Verify the sample rate setup worked.
- if (abs(srate - temp) > 100) {
- close(fd);
- sprintf(message_, "RtApiOss: error ... audio device (%s) doesn't support sample rate of %d.",
- name, temp);
- goto error;
+ // Now set the stream format for all streams. Also, check the
+ // physical format of the device and change that if necessary.
+ AudioStreamBasicDescription description;
+ dataSize = sizeof( AudioStreamBasicDescription );
+ property.mSelector = kAudioStreamPropertyVirtualFormat;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- stream_.sampleRate = sampleRate;
- if (ioctl(fd, SNDCTL_DSP_GETBLKSIZE, &buffer_bytes) == -1) {
- close(fd);
- sprintf(message_, "RtApiOss: error getting buffer size for device (%s).",
- name);
- goto error;
+ // Set the sample rate and data format id. However, only make the
+ // change if the sample rate is not within 1.0 of the desired
+ // rate and the format is not linear pcm.
+ bool updateFormat = false;
+ if ( fabs( description.mSampleRate - (Float64)sampleRate ) > 1.0 ) {
+ description.mSampleRate = (Float64) sampleRate;
+ updateFormat = true;
}
- // Save buffer size (in sample frames).
- *bufferSize = buffer_bytes / (formatBytes(stream_.deviceFormat[mode]) * device_channels);
- stream_.bufferSize = *bufferSize;
+ if ( description.mFormatID != kAudioFormatLinearPCM ) {
+ description.mFormatID = kAudioFormatLinearPCM;
+ updateFormat = true;
+ }
- 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;
+ if ( updateFormat ) {
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &description );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
}
- // Allocate the stream handles if necessary and then save.
- if ( stream_.apiHandle == 0 ) {
- handle = (int *) calloc(2, sizeof(int));
- stream_.apiHandle = (void *) handle;
- handle[0] = 0;
- handle[1] = 0;
+ // Now check the physical format.
+ property.mSelector = kAudioStreamPropertyPhysicalFormat;
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &description );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- else {
- handle = (int *) stream_.apiHandle;
+
+ //std::cout << "Current physical stream format:" << std::endl;
+ //std::cout << " mBitsPerChan = " << description.mBitsPerChannel << std::endl;
+ //std::cout << " aligned high = " << (description.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (description.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
+ //std::cout << " bytesPerFrame = " << description.mBytesPerFrame << std::endl;
+ //std::cout << " sample rate = " << description.mSampleRate << std::endl;
+
+ if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 16 ) {
+ description.mFormatID = kAudioFormatLinearPCM;
+ //description.mSampleRate = (Float64) sampleRate;
+ AudioStreamBasicDescription testDescription = description;
+ UInt32 formatFlags;
+
+ // We'll try higher bit rates first and then work our way down.
+ std::vector< std::pair<UInt32, UInt32> > physicalFormats;
+ formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsFloat) & ~kLinearPCMFormatFlagIsSignedInteger;
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
+ formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 32, formatFlags ) );
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 24, formatFlags ) ); // 24-bit packed
+ formatFlags &= ~( kAudioFormatFlagIsPacked | kAudioFormatFlagIsAlignedHigh );
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 24.2, formatFlags ) ); // 24-bit in 4 bytes, aligned low
+ formatFlags |= kAudioFormatFlagIsAlignedHigh;
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 24.4, formatFlags ) ); // 24-bit in 4 bytes, aligned high
+ formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked) & ~kLinearPCMFormatFlagIsFloat;
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 16, formatFlags ) );
+ physicalFormats.push_back( std::pair<Float32, UInt32>( 8, formatFlags ) );
+
+ bool setPhysicalFormat = false;
+ for( unsigned int i=0; i<physicalFormats.size(); i++ ) {
+ testDescription = description;
+ testDescription.mBitsPerChannel = (UInt32) physicalFormats[i].first;
+ testDescription.mFormatFlags = physicalFormats[i].second;
+ if ( (24 == (UInt32)physicalFormats[i].first) && ~( physicalFormats[i].second & kAudioFormatFlagIsPacked ) )
+ testDescription.mBytesPerFrame = 4 * testDescription.mChannelsPerFrame;
+ else
+ testDescription.mBytesPerFrame = testDescription.mBitsPerChannel/8 * testDescription.mChannelsPerFrame;
+ testDescription.mBytesPerPacket = testDescription.mBytesPerFrame * testDescription.mFramesPerPacket;
+ result = AudioObjectSetPropertyData( id, &property, 0, NULL, dataSize, &testDescription );
+ if ( result == noErr ) {
+ setPhysicalFormat = true;
+ //std::cout << "Updated physical stream format:" << std::endl;
+ //std::cout << " mBitsPerChan = " << testDescription.mBitsPerChannel << std::endl;
+ //std::cout << " aligned high = " << (testDescription.mFormatFlags & kAudioFormatFlagIsAlignedHigh) << ", isPacked = " << (testDescription.mFormatFlags & kAudioFormatFlagIsPacked) << std::endl;
+ //std::cout << " bytesPerFrame = " << testDescription.mBytesPerFrame << std::endl;
+ //std::cout << " sample rate = " << testDescription.mSampleRate << std::endl;
+ break;
+ }
+ }
+
+ if ( !setPhysicalFormat ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ } // done setting virtual/physical formats.
+
+ // Get the stream / device latency.
+ UInt32 latency;
+ dataSize = sizeof( UInt32 );
+ property.mSelector = kAudioDevicePropertyLatency;
+ if ( AudioObjectHasProperty( id, &property ) == true ) {
+ result = AudioObjectGetPropertyData( id, &property, 0, NULL, &dataSize, &latency );
+ if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
+ else {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ }
}
- handle[mode] = fd;
- // Set flags for buffer conversion
+ // Byte-swapping: According to AudioHardware.h, the stream data will
+ // always be presented in native-endian format, so we should never
+ // need to byte swap.
+ stream_.doByteSwap[mode] = false;
+
+ // From the CoreAudio documentation, PCM data must be supplied as
+ // 32-bit floats.
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+
+ if ( streamCount == 1 )
+ stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
+ else // multiple streams
+ stream_.nDeviceChannels[mode] = channels;
+ stream_.nUserChannels[mode] = channels;
+ stream_.channelOffset[mode] = channelOffset; // offset within a CoreAudio stream
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
+ stream_.deviceInterleaved[mode] = true;
+ if ( monoMode == true ) stream_.deviceInterleaved[mode] = false;
+
+ // Set flags for buffer conversion.
stream_.doConvertBuffer[mode] = false;
- if (stream_.userFormat != stream_.deviceFormat[mode])
+ 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] < stream_.nDeviceChannels[mode])
+ if ( streamCount == 1 ) {
+ if ( stream_.nUserChannels[mode] > 1 &&
+ stream_.userInterleaved != stream_.deviceInterleaved[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ }
+ else if ( monoMode && stream_.userInterleaved )
stream_.doConvertBuffer[mode] = true;
- // Allocate necessary internal buffers
- if ( stream_.nUserChannels[0] != stream_.nUserChannels[1] ) {
+ // 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;
+ }
- 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_, "RtApiOss: error allocating user buffer memory (%s).",
- name);
+ if ( pthread_cond_init( &handle->condition, NULL ) ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: error initializing pthread condition variable.";
goto error;
}
+ stream_.apiHandle = (void *) handle;
}
+ else
+ handle = (CoreHandle *) stream_.apiHandle;
+ handle->iStream[mode] = firstStream;
+ handle->nStreams[mode] = streamCount;
+ handle->id[mode] = id;
- if ( stream_.doConvertBuffer[mode] ) {
+ // Allocate necessary internal buffers.
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ // stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ stream_.userBuffer[mode] = (char *) malloc( bufferBytes * sizeof(char) );
+ memset( stream_.userBuffer[mode], 0, bufferBytes * sizeof(char) );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiCore::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+
+ // If possible, we will make use of the CoreAudio stream buffers as
+ // "device buffers". However, we can't do this if using multiple
+ // streams.
+ if ( stream_.doConvertBuffer[mode] && handle->nStreams[mode] > 1 ) {
- 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]);
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
- long bytes_out = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
- if ( buffer_bytes < bytes_out ) makeBuffer = false;
+ 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);
- sprintf(message_, "RtApiOss: 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_ = "RtApiCore::probeDeviceOpen: error allocating device buffer memory.";
goto error;
}
}
}
+ stream_.sampleRate = sampleRate;
stream_.device[mode] = device;
stream_.state = STREAM_STOPPED;
+ stream_.callbackInfo.object = (void *) this;
- if ( stream_.mode == OUTPUT && mode == INPUT ) {
- stream_.mode = DUPLEX;
- if (stream_.device[0] == device)
- handle[0] = fd;
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) {
+ if ( streamCount > 1 ) setConvertInfo( mode, 0 );
+ else setConvertInfo( mode, channelOffset );
}
- else
- stream_.mode = mode;
+
+ if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] == device )
+ // Only one callback procedure per device.
+ stream_.mode = DUPLEX;
+ else {
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ result = AudioDeviceCreateIOProcID( id, callbackHandler, (void *) &stream_.callbackInfo, &handle->procId[mode] );
+#else
+ // deprecated in favor of AudioDeviceCreateIOProcID()
+ result = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
+#endif
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::probeDeviceOpen: system error setting callback for device (" << device << ").";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ stream_.mode = DUPLEX;
+ else
+ stream_.mode = mode;
+ }
+
+ // Setup the device property listener for over/underload.
+ property.mSelector = kAudioDeviceProcessorOverload;
+ property.mScope = kAudioObjectPropertyScopeGlobal;
+ result = AudioObjectAddPropertyListener( id, &property, xrunListener, (void *) handle );
return SUCCESS;
error:
- if (handle) {
- if (handle[0])
- close(handle[0]);
- free(handle);
+ if ( handle ) {
+ pthread_cond_destroy( &handle->condition );
+ delete handle;
stream_.apiHandle = 0;
}
- if (stream_.userBuffer) {
- free(stream_.userBuffer);
- stream_.userBuffer = 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;
}
- error(RtError::WARNING);
+ stream_.state = STREAM_CLOSED;
return FAILURE;
}
-void RtApiOss :: closeStream()
+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
- // stream check.
- if ( stream_.mode == UNINITIALIZED ) {
- sprintf(message_, "RtApiOss::closeStream(): no open stream to close!");
- error(RtError::WARNING);
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiCore::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
return;
}
- int *handle = (int *) stream_.apiHandle;
- if (stream_.state == STREAM_RUNNING) {
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX)
- ioctl(handle[0], SNDCTL_DSP_RESET, 0);
- else
- ioctl(handle[1], SNDCTL_DSP_RESET, 0);
- stream_.state = STREAM_STOPPED;
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if (handle) {
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
+
+ property.mSelector = kAudioDeviceProcessorOverload;
+ property.mScope = kAudioObjectPropertyScopeGlobal;
+ if (AudioObjectRemovePropertyListener( handle->id[0], &property, xrunListener, (void *) handle ) != noErr) {
+ errorText_ = "RtApiCore::closeStream(): error removing property listener!";
+ error( RtAudioError::WARNING );
+ }
+ }
+ if ( stream_.state == STREAM_RUNNING )
+ AudioDeviceStop( handle->id[0], callbackHandler );
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ AudioDeviceDestroyIOProcID( handle->id[0], handle->procId[0] );
+#else
+ // deprecated in favor of AudioDeviceDestroyIOProcID()
+ AudioDeviceRemoveIOProc( handle->id[0], callbackHandler );
+#endif
}
- if (stream_.callbackInfo.usingCallback) {
- stream_.callbackInfo.usingCallback = false;
- pthread_join(stream_.callbackInfo.thread, NULL);
- }
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+ if (handle) {
+ AudioObjectPropertyAddress property = { kAudioHardwarePropertyDevices,
+ kAudioObjectPropertyScopeGlobal,
+ kAudioObjectPropertyElementMaster };
- if (handle) {
- if (handle[0]) close(handle[0]);
- if (handle[1]) close(handle[1]);
- free(handle);
- stream_.apiHandle = 0;
+ property.mSelector = kAudioDeviceProcessorOverload;
+ property.mScope = kAudioObjectPropertyScopeGlobal;
+ if (AudioObjectRemovePropertyListener( handle->id[1], &property, xrunListener, (void *) handle ) != noErr) {
+ errorText_ = "RtApiCore::closeStream(): error removing property listener!";
+ error( RtAudioError::WARNING );
+ }
+ }
+ if ( stream_.state == STREAM_RUNNING )
+ AudioDeviceStop( handle->id[1], callbackHandler );
+#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
+ AudioDeviceDestroyIOProcID( handle->id[1], handle->procId[1] );
+#else
+ // deprecated in favor of AudioDeviceDestroyIOProcID()
+ AudioDeviceRemoveIOProc( handle->id[1], callbackHandler );
+#endif
}
- if (stream_.userBuffer) {
- free(stream_.userBuffer);
- stream_.userBuffer = 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);
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
stream_.deviceBuffer = 0;
}
+ // Destroy pthread condition variable.
+ pthread_cond_destroy( &handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
+
stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
}
-void RtApiOss :: startStream()
+void RtApiCore :: startStream( void )
{
verifyStream();
- if (stream_.state == STREAM_RUNNING) return;
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiCore::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
- MUTEX_LOCK(&stream_.mutex);
+ OSStatus result = noErr;
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- stream_.state = STREAM_RUNNING;
+ 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;
+ }
+ }
- // No need to do anything else here ... OSS automatically starts
- // when fed samples.
+ if ( stream_.mode == INPUT ||
+ ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+
+ result = AudioDeviceStart( handle->id[1], callbackHandler );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::startStream: system error starting input callback procedure on device (" << stream_.device[1] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+ }
+
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ stream_.state = STREAM_RUNNING;
- MUTEX_UNLOCK(&stream_.mutex);
+ unlock:
+ if ( result == noErr ) return;
+ error( RtAudioError::SYSTEM_ERROR );
}
-void RtApiOss :: stopStream()
+void RtApiCore :: stopStream( void )
{
verifyStream();
- if (stream_.state == STREAM_STOPPED) return;
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
- // Change the state before the lock to improve shutdown response
- // when using a callback.
- stream_.state = STREAM_STOPPED;
- MUTEX_LOCK(&stream_.mutex);
+ OSStatus result = noErr;
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 2;
+ pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
+ }
- int err;
- int *handle = (int *) stream_.apiHandle;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
- err = ioctl(handle[0], SNDCTL_DSP_POST, 0);
- //err = ioctl(handle[0], SNDCTL_DSP_SYNC, 0);
- if (err < -1) {
- sprintf(message_, "RtApiOss: error stopping device (%s).",
- devices_[stream_.device[0]].name.c_str());
- 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;
}
}
- else {
- err = ioctl(handle[1], SNDCTL_DSP_POST, 0);
- //err = ioctl(handle[1], SNDCTL_DSP_SYNC, 0);
- if (err < -1) {
- sprintf(message_, "RtApiOss: error stopping device (%s).",
- devices_[stream_.device[1]].name.c_str());
- error(RtError::DRIVER_ERROR);
+
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
+
+ result = AudioDeviceStop( handle->id[1], callbackHandler );
+ if ( result != noErr ) {
+ errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
}
- MUTEX_UNLOCK(&stream_.mutex);
+ stream_.state = STREAM_STOPPED;
+
+ unlock:
+ if ( result == noErr ) return;
+ error( RtAudioError::SYSTEM_ERROR );
}
-void RtApiOss :: abortStream()
+void RtApiCore :: abortStream( void )
{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
+
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ handle->drainCounter = 2;
+
stopStream();
}
-int RtApiOss :: streamWillBlock()
+// This function will be called by a spawned thread when the user
+// callback function signals that the stream should be stopped or
+// aborted. It is better to handle it this way because the
+// callbackEvent() function probably should return before the AudioDeviceStop()
+// function is called.
+static void *coreStopStream( void *ptr )
{
- verifyStream();
- if (stream_.state == STREAM_STOPPED) return 0;
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiCore *object = (RtApiCore *) info->object;
- MUTEX_LOCK(&stream_.mutex);
+ object->stopStream();
+ pthread_exit( NULL );
+}
- int bytes = 0, channels = 0, frames = 0;
- audio_buf_info info;
- int *handle = (int *) stream_.apiHandle;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
- ioctl(handle[0], SNDCTL_DSP_GETOSPACE, &info);
- bytes = info.bytes;
- channels = stream_.nDeviceChannels[0];
+bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
+ const AudioBufferList *inBufferList,
+ const AudioBufferList *outBufferList )
+{
+ if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return FAILURE;
}
- if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
- ioctl(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];
- }
- }
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
- frames = (int) (bytes / (channels * formatBytes(stream_.deviceFormat[0])));
- frames -= stream_.bufferSize;
- if (frames < 0) frames = 0;
+ // Check if we were draining the stream and signal is finished.
+ if ( handle->drainCounter > 3 ) {
+ ThreadHandle threadId;
- MUTEX_UNLOCK(&stream_.mutex);
- return frames;
-}
+ stream_.state = STREAM_STOPPING;
+ if ( handle->internalDrain == true )
+ pthread_create( &threadId, NULL, coreStopStream, info );
+ else // external call to stopStream()
+ pthread_cond_signal( &handle->condition );
+ return SUCCESS;
+ }
-void RtApiOss :: tickStream()
-{
- verifyStream();
+ AudioDeviceID outputDevice = handle->id[0];
- int stopStream = 0;
- if (stream_.state == STREAM_STOPPED) {
- if (stream_.callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds
- return;
- }
- else if (stream_.callbackInfo.usingCallback) {
- RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
- stopStream = callback(stream_.userBuffer, stream_.bufferSize, stream_.callbackInfo.userData);
- }
+ // 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;
+ }
- MUTEX_LOCK(&stream_.mutex);
+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( cbReturnValue == 2 ) {
+ stream_.state = STREAM_STOPPING;
+ handle->drainCounter = 2;
+ abortStream();
+ return SUCCESS;
+ }
+ else if ( cbReturnValue == 1 ) {
+ handle->drainCounter = 1;
+ handle->internalDrain = true;
+ }
+ }
- // The state might change while waiting on a mutex.
- if (stream_.state == STREAM_STOPPED)
- goto unlock;
+ if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
- int result, *handle;
- char *buffer;
- int samples;
- RtAudioFormat format;
- handle = (int *) stream_.apiHandle;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
- // Setup parameters and do buffer conversion if necessary.
- if (stream_.doConvertBuffer[0]) {
- convertStreamBuffer(OUTPUT);
- buffer = stream_.deviceBuffer;
- samples = stream_.bufferSize * stream_.nDeviceChannels[0];
- format = stream_.deviceFormat[0];
+ if ( handle->nStreams[0] == 1 ) {
+ memset( outBufferList->mBuffers[handle->iStream[0]].mData,
+ 0,
+ outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
+ }
+ else { // fill multiple streams with zeros
+ for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
+ memset( outBufferList->mBuffers[handle->iStream[0]+i].mData,
+ 0,
+ outBufferList->mBuffers[handle->iStream[0]+i].mDataByteSize );
+ }
+ }
}
- else {
- buffer = stream_.userBuffer;
- samples = stream_.bufferSize * stream_.nUserChannels[0];
- format = stream_.userFormat;
+ else if ( handle->nStreams[0] == 1 ) {
+ if ( stream_.doConvertBuffer[0] ) { // convert directly to CoreAudio stream buffer
+ convertBuffer( (char *) outBufferList->mBuffers[handle->iStream[0]].mData,
+ stream_.userBuffer[0], stream_.convertInfo[0] );
+ }
+ else { // copy from user buffer
+ memcpy( outBufferList->mBuffers[handle->iStream[0]].mData,
+ stream_.userBuffer[0],
+ outBufferList->mBuffers[handle->iStream[0]].mDataByteSize );
+ }
}
+ else { // fill multiple streams
+ Float32 *inBuffer = (Float32 *) stream_.userBuffer[0];
+ if ( stream_.doConvertBuffer[0] ) {
+ convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ inBuffer = (Float32 *) stream_.deviceBuffer;
+ }
- // Do byte swapping if necessary.
- if (stream_.doByteSwap[0])
- byteSwapBuffer(buffer, samples, format);
+ if ( stream_.deviceInterleaved[0] == false ) { // mono mode
+ UInt32 bufferBytes = outBufferList->mBuffers[handle->iStream[0]].mDataByteSize;
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ memcpy( outBufferList->mBuffers[handle->iStream[0]+i].mData,
+ (void *)&inBuffer[i*stream_.bufferSize], bufferBytes );
+ }
+ }
+ else { // fill multiple multi-channel streams with interleaved data
+ UInt32 streamChannels, channelsLeft, inJump, outJump, inOffset;
+ Float32 *out, *in;
+
+ bool inInterleaved = ( stream_.userInterleaved ) ? true : false;
+ UInt32 inChannels = stream_.nUserChannels[0];
+ if ( stream_.doConvertBuffer[0] ) {
+ inInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
+ inChannels = stream_.nDeviceChannels[0];
+ }
+
+ if ( inInterleaved ) inOffset = 1;
+ else inOffset = stream_.bufferSize;
+
+ channelsLeft = inChannels;
+ for ( unsigned int i=0; i<handle->nStreams[0]; i++ ) {
+ in = inBuffer;
+ out = (Float32 *) outBufferList->mBuffers[handle->iStream[0]+i].mData;
+ streamChannels = outBufferList->mBuffers[handle->iStream[0]+i].mNumberChannels;
+
+ outJump = 0;
+ // Account for possible channel offset in first stream
+ if ( i == 0 && stream_.channelOffset[0] > 0 ) {
+ streamChannels -= stream_.channelOffset[0];
+ outJump = stream_.channelOffset[0];
+ out += outJump;
+ }
+
+ // Account for possible unfilled channels at end of the last stream
+ if ( streamChannels > channelsLeft ) {
+ outJump = streamChannels - channelsLeft;
+ streamChannels = channelsLeft;
+ }
- // Write samples to device.
- result = write(handle[0], buffer, samples * formatBytes(format));
+ // Determine input buffer offsets and skips
+ if ( inInterleaved ) {
+ inJump = inChannels;
+ in += inChannels - channelsLeft;
+ }
+ else {
+ inJump = 1;
+ in += (inChannels - channelsLeft) * inOffset;
+ }
- if (result == -1) {
- // This could be an underrun, but the basic OSS API doesn't provide a means for determining that.
- sprintf(message_, "RtApiOss: audio write error for device (%s).",
- devices_[stream_.device[0]].name.c_str());
- error(RtError::DRIVER_ERROR);
+ for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
+ for ( unsigned int j=0; j<streamChannels; j++ ) {
+ *out++ = in[j*inOffset];
+ }
+ out += outJump;
+ in += inJump;
+ }
+ channelsLeft -= streamChannels;
+ }
+ }
}
}
- if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
+ // Don't bother draining input
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
+ }
- // 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;
+ AudioDeviceID inputDevice;
+ inputDevice = handle->id[1];
+ if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == inputDevice ) ) {
+
+ if ( handle->nStreams[1] == 1 ) {
+ if ( stream_.doConvertBuffer[1] ) { // convert directly from CoreAudio stream buffer
+ convertBuffer( stream_.userBuffer[1],
+ (char *) inBufferList->mBuffers[handle->iStream[1]].mData,
+ stream_.convertInfo[1] );
+ }
+ else { // copy to user buffer
+ memcpy( stream_.userBuffer[1],
+ inBufferList->mBuffers[handle->iStream[1]].mData,
+ inBufferList->mBuffers[handle->iStream[1]].mDataByteSize );
+ }
}
+ else { // read from multiple streams
+ Float32 *outBuffer = (Float32 *) stream_.userBuffer[1];
+ if ( stream_.doConvertBuffer[1] ) outBuffer = (Float32 *) stream_.deviceBuffer;
- // Read samples from device.
- result = read(handle[1], buffer, samples * formatBytes(format));
+ if ( stream_.deviceInterleaved[1] == false ) { // mono mode
+ UInt32 bufferBytes = inBufferList->mBuffers[handle->iStream[1]].mDataByteSize;
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ memcpy( (void *)&outBuffer[i*stream_.bufferSize],
+ inBufferList->mBuffers[handle->iStream[1]+i].mData, bufferBytes );
+ }
+ }
+ else { // read from multiple multi-channel streams
+ UInt32 streamChannels, channelsLeft, inJump, outJump, outOffset;
+ Float32 *out, *in;
+
+ bool outInterleaved = ( stream_.userInterleaved ) ? true : false;
+ UInt32 outChannels = stream_.nUserChannels[1];
+ if ( stream_.doConvertBuffer[1] ) {
+ outInterleaved = true; // device buffer will always be interleaved for nStreams > 1 and not mono mode
+ outChannels = stream_.nDeviceChannels[1];
+ }
- if (result == -1) {
- // This could be an overrun, but the basic OSS API doesn't provide a means for determining that.
- sprintf(message_, "RtApiOss: audio read error for device (%s).",
- devices_[stream_.device[1]].name.c_str());
- error(RtError::DRIVER_ERROR);
- }
+ if ( outInterleaved ) outOffset = 1;
+ else outOffset = stream_.bufferSize;
+
+ channelsLeft = outChannels;
+ for ( unsigned int i=0; i<handle->nStreams[1]; i++ ) {
+ out = outBuffer;
+ in = (Float32 *) inBufferList->mBuffers[handle->iStream[1]+i].mData;
+ streamChannels = inBufferList->mBuffers[handle->iStream[1]+i].mNumberChannels;
+
+ inJump = 0;
+ // Account for possible channel offset in first stream
+ if ( i == 0 && stream_.channelOffset[1] > 0 ) {
+ streamChannels -= stream_.channelOffset[1];
+ inJump = stream_.channelOffset[1];
+ in += inJump;
+ }
- // Do byte swapping if necessary.
- if (stream_.doByteSwap[1])
- byteSwapBuffer(buffer, samples, format);
+ // Account for possible unread channels at end of the last stream
+ if ( streamChannels > channelsLeft ) {
+ inJump = streamChannels - channelsLeft;
+ streamChannels = channelsLeft;
+ }
- // Do buffer conversion if necessary.
- if (stream_.doConvertBuffer[1])
- convertStreamBuffer(INPUT);
+ // Determine output buffer offsets and skips
+ if ( outInterleaved ) {
+ outJump = outChannels;
+ out += outChannels - channelsLeft;
+ }
+ else {
+ outJump = 1;
+ out += (outChannels - channelsLeft) * outOffset;
+ }
+
+ for ( unsigned int i=0; i<stream_.bufferSize; i++ ) {
+ for ( unsigned int j=0; j<streamChannels; j++ ) {
+ out[j*outOffset] = *in++;
+ }
+ out += outJump;
+ in += inJump;
+ }
+ channelsLeft -= streamChannels;
+ }
+ }
+
+ if ( stream_.doConvertBuffer[1] ) { // convert from our internal "device" buffer
+ convertBuffer( stream_.userBuffer[1],
+ stream_.deviceBuffer,
+ stream_.convertInfo[1] );
+ }
+ }
}
unlock:
- MUTEX_UNLOCK(&stream_.mutex);
+ //MUTEX_UNLOCK( &stream_.mutex );
- if (stream_.callbackInfo.usingCallback && stopStream)
- this->stopStream();
+ RtApi::tickStreamTime();
+ return SUCCESS;
}
-void RtApiOss :: setStreamCallback(RtAudioCallback callback, void *userData)
+const char* RtApiCore :: getErrorCode( OSStatus code )
{
- verifyStream();
+ switch( code ) {
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
- if ( info->usingCallback ) {
- sprintf(message_, "RtApiOss: A callback is already set for this stream!");
- error(RtError::WARNING);
- return;
- }
+ case kAudioHardwareNotRunningError:
+ return "kAudioHardwareNotRunningError";
- info->callback = (void *) callback;
- info->userData = userData;
- info->usingCallback = true;
- info->object = (void *) this;
+ case kAudioHardwareUnspecifiedError:
+ return "kAudioHardwareUnspecifiedError";
- // 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);
- pthread_attr_setschedpolicy(&attr, SCHED_RR);
+ case kAudioHardwareUnknownPropertyError:
+ return "kAudioHardwareUnknownPropertyError";
- int err = pthread_create(&(info->thread), &attr, ossCallbackHandler, &stream_.callbackInfo);
- pthread_attr_destroy(&attr);
- if (err) {
- info->usingCallback = false;
- sprintf(message_, "RtApiOss: error starting callback thread!");
- error(RtError::THREAD_ERROR);
- }
-}
+ case kAudioHardwareBadPropertySizeError:
+ return "kAudioHardwareBadPropertySizeError";
-void RtApiOss :: cancelStreamCallback()
-{
- verifyStream();
+ case kAudioHardwareIllegalOperationError:
+ return "kAudioHardwareIllegalOperationError";
- if (stream_.callbackInfo.usingCallback) {
+ case kAudioHardwareBadObjectError:
+ return "kAudioHardwareBadObjectError";
- if (stream_.state == STREAM_RUNNING)
- stopStream();
+ case kAudioHardwareBadDeviceError:
+ return "kAudioHardwareBadDeviceError";
- MUTEX_LOCK(&stream_.mutex);
+ case kAudioHardwareBadStreamError:
+ return "kAudioHardwareBadStreamError";
- stream_.callbackInfo.usingCallback = false;
- pthread_join(stream_.callbackInfo.thread, NULL);
- stream_.callbackInfo.thread = 0;
- stream_.callbackInfo.callback = NULL;
- stream_.callbackInfo.userData = NULL;
+ case kAudioHardwareUnsupportedOperationError:
+ return "kAudioHardwareUnsupportedOperationError";
- MUTEX_UNLOCK(&stream_.mutex);
- }
-}
+ case kAudioDeviceUnsupportedFormatError:
+ return "kAudioDeviceUnsupportedFormatError";
-extern "C" void *ossCallbackHandler(void *ptr)
-{
- CallbackInfo *info = (CallbackInfo *) ptr;
- RtApiOss *object = (RtApiOss *) info->object;
- bool *usingCallback = &info->usingCallback;
+ case kAudioDevicePermissionsError:
+ return "kAudioDevicePermissionsError";
- while ( *usingCallback ) {
- pthread_testcancel();
- try {
- object->tickStream();
- }
- catch (RtError &exception) {
- fprintf(stderr, "\nRtApiOss: callback thread error (%s) ... closing thread.\n\n",
- exception.getMessageString());
- break;
- }
+ default:
+ return "CoreAudio unknown error";
}
-
- return 0;
}
-//******************** End of __LINUX_OSS__ *********************//
+ //******************** End of __MACOSX_CORE__ *********************//
#endif
-#if defined(__MACOSX_CORE__)
-
+#if defined(__UNIX_JACK__)
-// The OS X CoreAudio API is designed to use a separate callback
-// procedure for each of its audio devices. A single RtAudio duplex
-// stream using two different devices is supported here, though it
-// cannot be guaranteed to always behave correctly because we cannot
-// synchronize these two callbacks. 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()).
+// 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.
//
-// 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.
+// When using JACK with RtAudio, "devices" refer to JACK clients that
+// have ports connected to the server. The JACK server is typically
+// started in a terminal as follows:
+//
+// .jackd -d alsa -d hw:0
+//
+// or through an interface program such as qjackctl. Many of the
+// parameters normally set for a stream are fixed by the JACK server
+// and can be specified when the JACK server is started. In
+// particular,
+//
+// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
+//
+// specifies a sample rate of 44100 Hz, a buffer size of 512 sample
+// frames, and number of buffers = 4. Once the server is running, it
+// is not possible to override these values. If the values are not
+// specified in the command-line, the JACK server uses default values.
+//
+// The JACK server does not have to be running when an instance of
+// RtApiJack is created, though the function getDeviceCount() will
+// report 0 devices found until JACK has been started. When no
+// devices are available (i.e., the JACK server is not running), a
+// stream cannot be opened.
+
+#include <jack/jack.h>
+#include <unistd.h>
+#include <cstdio>
-// A structure to hold various information related to the CoreAuio API
+// A structure to hold various information related to the Jack API
// implementation.
-struct CoreHandle {
- UInt32 index[2];
- bool stopStream;
- bool xrun;
- char *deviceBuffer;
+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.
- CoreHandle()
- :stopStream(false), xrun(false), deviceBuffer(0) {}
+ JackHandle()
+ :client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
};
-RtApiCore :: RtApiCore()
-{
- this->initialize();
+#if !defined(__RTAUDIO_DEBUG__)
+static void jackSilentError( const char * ) {};
+#endif
- if (nDevices_ <= 0) {
- sprintf(message_, "RtApiCore: no Macintosh OS-X Core Audio devices found!");
- error(RtError::NO_DEVICES_FOUND);
- }
+RtApiJack :: RtApiJack()
+ :shouldAutoconnect_(true) {
+ // Nothing to do here.
+#if !defined(__RTAUDIO_DEBUG__)
+ // Turn off Jack's internal error reporting.
+ jack_set_error_function( &jackSilentError );
+#endif
}
-RtApiCore :: ~RtApiCore()
+RtApiJack :: ~RtApiJack()
{
- // The subclass destructor gets called before the base class
- // destructor, so close an existing stream before deallocating
- // apiDeviceId memory.
- if ( stream_.mode != UNINITIALIZED ) closeStream();
-
- // Free our allocated apiDeviceId memory.
- AudioDeviceID *id;
- for ( unsigned int i=0; i<devices_.size(); i++ ) {
- id = (AudioDeviceID *) devices_[i].apiDeviceId;
- if (id) free(id);
- }
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
}
-void RtApiCore :: initialize(void)
+unsigned int RtApiJack :: getDeviceCount( void )
{
- OSStatus err = noErr;
- UInt32 dataSize;
- AudioDeviceID *deviceList = NULL;
- nDevices_ = 0;
+ // See if we can become a jack client.
+ jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
+ jack_status_t *status = NULL;
+ jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
+ if ( client == 0 ) return 0;
- // Find out how many audio devices there are, if any.
- err = AudioHardwareGetPropertyInfo(kAudioHardwarePropertyDevices, &dataSize, NULL);
- if (err != noErr) {
- sprintf(message_, "RtApiCore: OS-X error getting device info!");
- error(RtError::SYSTEM_ERROR);
+ const char **ports;
+ std::string port, previousPort;
+ unsigned int nChannels = 0, nDevices = 0;
+ ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
+ if ( ports ) {
+ // Parse the port names up to the first colon (:).
+ size_t iColon = 0;
+ 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 );
}
- nDevices_ = dataSize / sizeof(AudioDeviceID);
- if (nDevices_ == 0) return;
+ jack_client_close( client );
+ return nDevices;
+}
+
+RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
- // Make space for the devices we are about to get.
- deviceList = (AudioDeviceID *) malloc( dataSize );
- if (deviceList == NULL) {
- sprintf(message_, "RtApiCore: memory allocation error during initialization!");
- error(RtError::MEMORY_ERROR);
+ jack_options_t options = (jack_options_t) ( JackNoStartServer ); //JackNullOption
+ jack_status_t *status = NULL;
+ jack_client_t *client = jack_client_open( "RtApiJackInfo", options, status );
+ if ( client == 0 ) {
+ errorText_ = "RtApiJack::getDeviceInfo: Jack server not found or connection error!";
+ error( RtAudioError::WARNING );
+ return info;
}
- // Get the array of AudioDeviceIDs.
- err = AudioHardwareGetProperty(kAudioHardwarePropertyDevices, &dataSize, (void *) deviceList);
- if (err != noErr) {
- free(deviceList);
- sprintf(message_, "RtApiCore: OS-X error getting device properties!");
- error(RtError::SYSTEM_ERROR);
+ const char **ports;
+ std::string port, previousPort;
+ unsigned int nPorts = 0, nDevices = 0;
+ ports = jack_get_ports( client, NULL, JACK_DEFAULT_AUDIO_TYPE, 0 );
+ if ( ports ) {
+ // Parse the port names up to the first colon (:).
+ size_t iColon = 0;
+ 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 );
}
- // Create list of device structures and write device identifiers.
- RtApiDevice device;
- AudioDeviceID *id;
- for (int i=0; i<nDevices_; i++) {
- devices_.push_back(device);
- id = (AudioDeviceID *) malloc( sizeof(AudioDeviceID) );
- *id = deviceList[i];
- devices_[i].apiDeviceId = (void *) id;
+ if ( device >= nDevices ) {
+ jack_client_close( client );
+ errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
}
- free(deviceList);
-}
-
-int RtApiCore :: getDefaultInputDevice(void)
-{
- AudioDeviceID id, *deviceId;
- UInt32 dataSize = sizeof( AudioDeviceID );
+ // Get the current jack server sample rate.
+ info.sampleRates.clear();
- OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultInputDevice,
- &dataSize, &id );
+ info.preferredSampleRate = jack_get_sample_rate( client );
+ info.sampleRates.push_back( info.preferredSampleRate );
- if (result != noErr) {
- sprintf( message_, "RtApiCore: OS-X error getting default input device." );
- error(RtError::WARNING);
- return 0;
+ // 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(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput );
+ if ( ports ) {
+ while ( ports[ nChannels ] ) nChannels++;
+ free( ports );
+ info.outputChannels = nChannels;
}
- for ( int i=0; i<nDevices_; i++ ) {
- deviceId = (AudioDeviceID *) devices_[i].apiDeviceId;
- if ( id == *deviceId ) return i;
+ // Jack "output ports" equal RtAudio input channels.
+ nChannels = 0;
+ ports = jack_get_ports( client, info.name.c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput );
+ if ( ports ) {
+ while ( ports[ nChannels ] ) nChannels++;
+ free( ports );
+ info.inputChannels = nChannels;
}
- return 0;
-}
-
-int RtApiCore :: getDefaultOutputDevice(void)
-{
- AudioDeviceID id, *deviceId;
- UInt32 dataSize = sizeof( AudioDeviceID );
+ if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
+ jack_client_close(client);
+ errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
+ error( RtAudioError::WARNING );
+ return info;
+ }
- OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultOutputDevice,
- &dataSize, &id );
+ // If device opens for both playback and capture, we determine the channels.
+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
- if (result != noErr) {
- sprintf( message_, "RtApiCore: OS-X error getting default output device." );
- error(RtError::WARNING);
- return 0;
- }
+ // Jack always uses 32-bit floats.
+ info.nativeFormats = RTAUDIO_FLOAT32;
- for ( int i=0; i<nDevices_; i++ ) {
- deviceId = (AudioDeviceID *) devices_[i].apiDeviceId;
- if ( id == *deviceId ) return i;
- }
+ // 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;
- return 0;
+ jack_client_close(client);
+ info.probed = true;
+ return info;
}
-static bool deviceSupportsFormat( AudioDeviceID id, bool isInput,
- AudioStreamBasicDescription *desc, bool isDuplex )
+static int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
{
- 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 );
-
+ CallbackInfo *info = (CallbackInfo *) infoPointer;
- if (result != kAudioHardwareNoError)
- return false;
- }
- return true;
- }
+ RtApiJack *object = (RtApiJack *) info->object;
+ if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
- return false;
+ return 0;
}
-void RtApiCore :: probeDeviceInfo( RtApiDevice *info )
+// This function will be called by a spawned thread when the Jack
+// server signals that it is shutting down. It is necessary to handle
+// it this way because the jackShutdown() function must return before
+// the jack_deactivate() function (in closeStream()) will return.
+static void *jackCloseStream( void *ptr )
{
- OSStatus err = noErr;
-
- // Get the device manufacturer and name.
- char name[256];
- char fullname[512];
- UInt32 dataSize = 256;
- AudioDeviceID *id = (AudioDeviceID *) info->apiDeviceId;
- err = AudioDeviceGetProperty( *id, 0, false,
- kAudioDevicePropertyDeviceManufacturer,
- &dataSize, name );
- if (err != noErr) {
- sprintf( message_, "RtApiCore: OS-X error getting device manufacturer." );
- error(RtError::DEBUG_WARNING);
- return;
- }
- strncpy(fullname, name, 256);
- strcat(fullname, ": " );
-
- dataSize = 256;
- err = AudioDeviceGetProperty( *id, 0, false,
- kAudioDevicePropertyDeviceName,
- &dataSize, name );
- if (err != noErr) {
- sprintf( message_, "RtApiCore: OS-X error getting device name." );
- error(RtError::DEBUG_WARNING);
- return;
- }
- strncat(fullname, name, 254);
- info->name.erase();
- info->name.append( (const char *)fullname, strlen(fullname)+1);
-
- // Get output channel information.
- unsigned int i, minChannels = 0, maxChannels = 0, nStreams = 0;
- AudioBufferList *bufferList = nil;
- err = AudioDeviceGetPropertyInfo( *id, 0, false,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, NULL );
- if (err == noErr && dataSize > 0) {
- bufferList = (AudioBufferList *) malloc( dataSize );
- if (bufferList == NULL) {
- sprintf(message_, "RtApiCore: memory allocation error!");
- error(RtError::DEBUG_WARNING);
- return;
- }
-
- err = AudioDeviceGetProperty( *id, 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;
- }
- }
- }
- free (bufferList);
-
- if (err != noErr || dataSize <= 0) {
- sprintf( message_, "RtApiCore: OS-X error getting output channels for device (%s).",
- info->name.c_str() );
- error(RtError::DEBUG_WARNING);
- return;
- }
-
- if ( nStreams ) {
- if ( maxChannels > 0 )
- info->maxOutputChannels = maxChannels;
- if ( minChannels > 0 )
- info->minOutputChannels = minChannels;
- }
-
- // Get input channel information.
- bufferList = nil;
- err = AudioDeviceGetPropertyInfo( *id, 0, true,
- kAudioDevicePropertyStreamConfiguration,
- &dataSize, NULL );
- if (err == noErr && dataSize > 0) {
- bufferList = (AudioBufferList *) malloc( dataSize );
- if (bufferList == NULL) {
- sprintf(message_, "RtApiCore: memory allocation error!");
- error(RtError::DEBUG_WARNING);
- return;
- }
- err = AudioDeviceGetProperty( *id, 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;
- }
- }
- }
- free (bufferList);
-
- if (err != noErr || dataSize <= 0) {
- sprintf( message_, "RtApiCore: OS-X error getting input channels for device (%s).",
- info->name.c_str() );
- error(RtError::DEBUG_WARNING);
- return;
- }
-
- if ( nStreams ) {
- if ( maxChannels > 0 )
- info->maxInputChannels = maxChannels;
- if ( minChannels > 0 )
- info->minInputChannels = minChannels;
- }
-
- // 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.
-
- // Create a stream description structure.
- AudioStreamBasicDescription description;
- dataSize = sizeof( AudioStreamBasicDescription );
- memset(&description, 0, sizeof(AudioStreamBasicDescription));
- bool isInput = false;
- if ( info->maxOutputChannels == 0 ) isInput = true;
- bool isDuplex = false;
- if ( info->maxDuplexChannels > 0 ) isDuplex = true;
-
- // Determine the supported sample rates.
- info->sampleRates.clear();
- for (unsigned int k=0; k<MAX_SAMPLE_RATES; k++) {
- description.mSampleRate = (double) SAMPLE_RATES[k];
- if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
- info->sampleRates.push_back( SAMPLE_RATES[k] );
- }
-
- if (info->sampleRates.size() == 0) {
- sprintf( message_, "RtApiCore: No supported sample rates found for OS-X device (%s).",
- info->name.c_str() );
- error(RtError::DEBUG_WARNING);
- return;
- }
-
- // Determine the supported data formats.
- info->nativeFormats = 0;
- description.mFormatID = kAudioFormatLinearPCM;
- description.mBitsPerChannel = 8;
- description.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_SINT8;
- else {
- description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_SINT8;
- }
-
- description.mBitsPerChannel = 16;
- description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_SINT16;
- else {
- description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_SINT16;
- }
-
- description.mBitsPerChannel = 32;
- description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_SINT32;
- else {
- description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_SINT32;
- }
-
- description.mBitsPerChannel = 24;
- description.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsAlignedHigh | kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_SINT24;
- else {
- description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_SINT24;
- }
-
- description.mBitsPerChannel = 32;
- description.mFormatFlags = kLinearPCMFormatFlagIsFloat | kLinearPCMFormatFlagIsPacked | kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_FLOAT32;
- else {
- description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_FLOAT32;
- }
-
- description.mBitsPerChannel = 64;
- description.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_FLOAT64;
- else {
- description.mFormatFlags &= ~kLinearPCMFormatFlagIsBigEndian;
- if ( deviceSupportsFormat( *id, isInput, &description, isDuplex ) )
- info->nativeFormats |= RTAUDIO_FLOAT64;
- }
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiJack *object = (RtApiJack *) info->object;
- // Check that we have at least one supported format.
- if (info->nativeFormats == 0) {
- sprintf(message_, "RtApiCore: OS-X device (%s) data format not supported by RtAudio.",
- info->name.c_str());
- error(RtError::DEBUG_WARNING);
- return;
- }
+ object->closeStream();
- info->probed = true;
+ pthread_exit( NULL );
}
-
-OSStatus callbackHandler(AudioDeviceID inDevice,
- const AudioTimeStamp* inNow,
- const AudioBufferList* inInputData,
- const AudioTimeStamp* inInputTime,
- AudioBufferList* outOutputData,
- const AudioTimeStamp* inOutputTime,
- void* infoPointer)
+static void jackShutdown( void *infoPointer )
{
CallbackInfo *info = (CallbackInfo *) infoPointer;
+ RtApiJack *object = (RtApiJack *) info->object;
- RtApiCore *object = (RtApiCore *) info->object;
- try {
- object->callbackEvent( inDevice, (void *)inInputData, (void *)outOutputData );
- }
- catch (RtError &exception) {
- fprintf(stderr, "\nRtApiCore: callback handler error (%s)!\n\n", exception.getMessageString());
- return kAudioHardwareUnspecifiedError;
- }
+ // 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;
- return kAudioHardwareNoError;
+ ThreadHandle threadId;
+ pthread_create( &threadId, NULL, jackCloseStream, info );
+ std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
}
-OSStatus deviceListener(AudioDeviceID inDevice,
- UInt32 channel,
- Boolean isInput,
- AudioDevicePropertyID propertyID,
- void* handlePointer)
+static int jackXrun( void *infoPointer )
{
- CoreHandle *handle = (CoreHandle *) handlePointer;
- if ( propertyID == kAudioDeviceProcessorOverload ) {
- if ( isInput )
- fprintf(stderr, "\nRtApiCore: OS-X audio input overrun detected!\n");
- else
- fprintf(stderr, "\nRtApiCore: OS-X audio output underrun detected!\n");
- handle->xrun = true;
- }
+ JackHandle *handle = *((JackHandle **) infoPointer);
- return kAudioHardwareNoError;
+ if ( handle->ports[0] ) handle->xrun[0] = true;
+ if ( handle->ports[1] ) handle->xrun[1] = true;
+
+ return 0;
}
-bool RtApiCore :: probeDeviceOpen( int device, StreamMode mode, int channels,
- int sampleRate, RtAudioFormat format,
- int *bufferSize, int numberOfBuffers )
+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 )
{
- // Setup for stream mode.
- bool isInput = false;
- AudioDeviceID id = *((AudioDeviceID *) devices_[device].apiDeviceId);
- if ( mode == INPUT ) isInput = true;
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
- // Search for a stream which contains the desired number of channels.
- OSStatus err = noErr;
- UInt32 dataSize;
- unsigned int deviceChannels, 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_, "RtApiCore: memory allocation error in probeDeviceOpen()!");
- error(RtError::DEBUG_WARNING);
+ // Look for jack server and try to become a client (only do once per stream).
+ jack_client_t *client = 0;
+ if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) {
+ jack_options_t jackoptions = (jack_options_t) ( JackNoStartServer ); //JackNullOption;
+ jack_status_t *status = NULL;
+ if ( options && !options->streamName.empty() )
+ client = jack_client_open( options->streamName.c_str(), jackoptions, status );
+ else
+ client = jack_client_open( "RtApiJack", jackoptions, status );
+ if ( client == 0 ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: Jack server not found or connection error!";
+ error( RtAudioError::WARNING );
return FAILURE;
}
- 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 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 {
+ // 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, JACK_DEFAULT_AUDIO_TYPE, 0 );
+ if ( ports ) {
+ // Parse the port names up to the first colon (:).
+ size_t iColon = 0;
+ do {
+ port = (char *) ports[ nPorts ];
+ iColon = port.find(":");
+ if ( iColon != std::string::npos ) {
+ port = port.substr( 0, iColon );
+ if ( port != previousPort ) {
+ if ( nDevices == device ) deviceName = port;
+ nDevices++;
+ previousPort = port;
}
}
- }
- }
- if (err != noErr || dataSize <= 0) {
- if ( bufferList ) free( bufferList );
- sprintf( message_, "RtApiCore: OS-X error getting channels for device (%s).",
- devices_[device].name.c_str() );
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ } while ( ports[++nPorts] );
+ free( ports );
}
- if (iStream >= nStreams) {
- free (bufferList);
- sprintf( message_, "RtApiCore: unable to find OS-X audio stream on device (%s) for requested channels (%d).",
- devices_[device].name.c_str(), channels );
- error(RtError::DEBUG_WARNING);
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
return FAILURE;
}
- // This is ok even for mono mode ... it gets updated later.
- deviceChannels = bufferList->mBuffers[iStream].mNumberChannels;
- free (bufferList);
+ unsigned long flag = JackPortIsInput;
+ if ( mode == INPUT ) flag = JackPortIsOutput;
- // Determine the buffer size.
- AudioValueRange bufferRange;
- dataSize = sizeof(AudioValueRange);
- err = AudioDeviceGetProperty( id, 0, isInput,
- kAudioDevicePropertyBufferSizeRange,
- &dataSize, &bufferRange);
- if (err != noErr) {
- sprintf( message_, "RtApiCore: OS-X error getting buffer size range for device (%s).",
- devices_[device].name.c_str() );
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ if ( ! (options && (options->flags & RTAUDIO_JACK_DONT_CONNECT)) ) {
+ // Count the available ports containing the client name as device
+ // channels. Jack "input ports" equal RtAudio output channels.
+ unsigned int nChannels = 0;
+ ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag );
+ if ( ports ) {
+ while ( ports[ nChannels ] ) nChannels++;
+ free( ports );
+ }
+ // Compare the jack ports for specified client to the requested number of channels.
+ if ( nChannels < (channels + firstChannel) ) {
+ errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
}
- long bufferBytes = *bufferSize * deviceChannels * formatBytes(RTAUDIO_FLOAT32);
- if (bufferRange.mMinimum > bufferBytes) bufferBytes = (int) bufferRange.mMinimum;
- else if (bufferRange.mMaximum < bufferBytes) bufferBytes = (int) bufferRange.mMaximum;
-
- // 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_, "RtApiCore: OS-X error setting the buffer size for device (%s).",
- devices_[device].name.c_str() );
- error(RtError::DEBUG_WARNING);
+ // 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;
- // 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_, "RtApiCore: OS-X error setting buffer size for duplex stream on device (%s).",
- devices_[device].name.c_str() );
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ // Get the latency of the JACK port.
+ ports = jack_get_ports( client, deviceName.c_str(), JACK_DEFAULT_AUDIO_TYPE, flag );
+ if ( ports[ firstChannel ] ) {
+ // Added by Ge Wang
+ jack_latency_callback_mode_t cbmode = (mode == INPUT ? JackCaptureLatency : JackPlaybackLatency);
+ // the range (usually the min and max are equal)
+ jack_latency_range_t latrange; latrange.min = latrange.max = 0;
+ // get the latency range
+ jack_port_get_latency_range( jack_port_by_name( client, ports[firstChannel] ), cbmode, &latrange );
+ // be optimistic, use the min!
+ stream_.latency[mode] = latrange.min;
+ //stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
}
+ free( ports );
- stream_.bufferSize = *bufferSize;
- stream_.nBuffers = 1;
-
- // Set the stream format description. Do for each channel in mono mode.
- 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_, "RtApiCore: OS-X error getting stream format for device (%s).",
- devices_[device].name.c_str() );
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
+ // The jack server always uses 32-bit floating-point data.
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ stream_.userFormat = format;
- // 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_, "RtApiCore: OS-X error setting sample rate or data format for device (%s).",
- devices_[device].name.c_str() );
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
- }
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
- // 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_, "RtApiCore: OS-X error getting stream format for device (%s).", devices_[device].name.c_str() );
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
+ // Jack always uses non-interleaved buffers.
+ stream_.deviceInterleaved[mode] = false;
+ // Jack always provides host byte-ordered data.
stream_.doByteSwap[mode] = false;
- if ( !description.mFormatFlags & kLinearPCMFormatFlagIsBigEndian )
- stream_.doByteSwap[mode] = true;
- // From the CoreAudio documentation, PCM data must be supplied as
- // 32-bit floats.
- stream_.userFormat = format;
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ // 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;
- if ( stream_.deInterleave[mode] ) // mono mode
- stream_.nDeviceChannels[mode] = channels;
- else
- stream_.nDeviceChannels[mode] = description.mChannelsPerFrame;
+ 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_.nUserChannels[mode] < stream_.nDeviceChannels[mode])
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
stream_.doConvertBuffer[mode] = true;
- if (stream_.nUserChannels[mode] > 1 && stream_.deInterleave[mode])
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
stream_.doConvertBuffer[mode] = true;
- // Allocate our CoreHandle structure for the stream.
- CoreHandle *handle;
- if ( stream_.apiHandle == 0 ) {
- handle = (CoreHandle *) calloc(1, sizeof(CoreHandle));
- if ( handle == NULL ) {
- sprintf(message_, "RtApiCore: OS-X error allocating coreHandle memory (%s).",
- devices_[device].name.c_str());
+ // 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;
}
- handle->index[0] = 0;
- handle->index[1] = 0;
+
if ( pthread_cond_init(&handle->condition, NULL) ) {
- sprintf(message_, "RtApiCore: error initializing pthread condition variable (%s).",
- devices_[device].name.c_str());
+ errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
goto error;
}
stream_.apiHandle = (void *) handle;
+ handle->client = client;
}
- else
- handle = (CoreHandle *) stream_.apiHandle;
- handle->index[mode] = iStream;
+ handle->deviceName[mode] = deviceName;
// 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) {
- sprintf(message_, "RtApiCore: OS-X error allocating user buffer memory (%s).",
- devices_[device].name.c_str());
- 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_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
}
- if ( stream_.deInterleave[mode] ) {
+ if ( stream_.doConvertBuffer[mode] ) {
- long buffer_bytes;
bool makeBuffer = true;
if ( mode == OUTPUT )
- buffer_bytes = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
+ bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
else { // mode == INPUT
- buffer_bytes = stream_.nDeviceChannels[1] * formatBytes(stream_.deviceFormat[1]);
+ bufferBytes = 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;
+ 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) {
- sprintf(message_, "RtApiCore: error allocating device buffer memory (%s).",
- devices_[device].name.c_str());
+ 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 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 CallbackInfo structure.
- handle->deviceBuffer = stream_.deviceBuffer;
}
}
- stream_.sampleRate = sampleRate;
+ // Allocate memory for the Jack ports (channels) identifiers.
+ handle->ports[mode] = (jack_port_t **) malloc ( sizeof (jack_port_t *) * channels );
+ if ( handle->ports[mode] == NULL ) {
+ errorText_ = "RtApiJack::probeDeviceOpen: error allocating port memory.";
+ goto error;
+ }
+
stream_.device[mode] = device;
+ stream_.channelOffset[mode] = firstChannel;
stream_.state = STREAM_STOPPED;
stream_.callbackInfo.object = (void *) this;
- if ( stream_.mode == OUTPUT && mode == INPUT && stream_.device[0] == device )
- // Only one callback procedure per device.
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ // We had already set up the stream for output.
stream_.mode = DUPLEX;
else {
- err = AudioDeviceAddIOProc( id, callbackHandler, (void *) &stream_.callbackInfo );
- if (err != noErr) {
- sprintf( message_, "RtApiCore: OS-X error setting callback for device (%s).", devices_[device].name.c_str() );
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ stream_.mode = mode;
+ jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
+ jack_set_xrun_callback( handle->client, jackXrun, (void *) &stream_.apiHandle );
+ jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
+ }
+
+ // Register our ports.
+ char label[64];
+ if ( mode == OUTPUT ) {
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ snprintf( label, 64, "outport %d", i );
+ handle->ports[0][i] = jack_port_register( handle->client, (const char *)label,
+ JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0 );
+ }
+ }
+ else {
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ snprintf( label, 64, "inport %d", i );
+ handle->ports[1][i] = jack_port_register( handle->client, (const char *)label,
+ JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0 );
}
- if ( stream_.mode == OUTPUT && mode == INPUT )
- stream_.mode = DUPLEX;
- else
- stream_.mode = mode;
}
- // Setup the device property listener for over/underload.
- err = AudioDeviceAddPropertyListener( id, iChannel, isInput,
- kAudioDeviceProcessorOverload,
- deviceListener, (void *) handle );
+ // Setup the buffer conversion information structure. We don't use
+ // buffers to do channel offsets, so we override that parameter
+ // here.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
+
+ if ( options && options->flags & RTAUDIO_JACK_DONT_CONNECT ) shouldAutoconnect_ = false;
return SUCCESS;
error:
if ( handle ) {
- pthread_cond_destroy(&handle->condition);
- free(handle);
+ pthread_cond_destroy( &handle->condition );
+ jack_client_close( handle->client );
+
+ if ( handle->ports[0] ) free( handle->ports[0] );
+ if ( handle->ports[1] ) free( handle->ports[1] );
+
+ delete handle;
stream_.apiHandle = 0;
}
- if (stream_.userBuffer) {
- free(stream_.userBuffer);
- stream_.userBuffer = 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;
}
- error(RtError::WARNING);
return FAILURE;
}
-void RtApiCore :: closeStream()
+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
- // stream check.
- if ( stream_.mode == UNINITIALIZED ) {
- sprintf(message_, "RtApiCore::closeStream(): no open stream to close!");
- error(RtError::WARNING);
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiJack::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
return;
}
- AudioDeviceID id = *( (AudioDeviceID *) devices_[stream_.device[0]].apiDeviceId );
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
- if (stream_.state == STREAM_RUNNING)
- AudioDeviceStop( id, callbackHandler );
- AudioDeviceRemoveIOProc( id, callbackHandler );
- }
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ if ( handle ) {
- id = *( (AudioDeviceID *) devices_[stream_.device[1]].apiDeviceId );
- if (stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1]) ) {
- if (stream_.state == STREAM_RUNNING)
- AudioDeviceStop( id, callbackHandler );
- AudioDeviceRemoveIOProc( id, callbackHandler );
- }
+ if ( stream_.state == STREAM_RUNNING )
+ jack_deactivate( handle->client );
- if (stream_.userBuffer) {
- free(stream_.userBuffer);
- stream_.userBuffer = 0;
+ jack_client_close( handle->client );
}
- if ( stream_.deInterleave[0] || stream_.deInterleave[1] ) {
- free(stream_.deviceBuffer);
- stream_.deviceBuffer = 0;
+ 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;
}
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
- // Destroy pthread condition variable and free the CoreHandle structure.
- if ( handle ) {
- pthread_cond_destroy(&handle->condition);
- free( handle );
- stream_.apiHandle = 0;
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
}
stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
}
-void RtApiCore :: startStream()
+void RtApiJack :: startStream( void )
{
verifyStream();
- if (stream_.state == STREAM_RUNNING) return;
-
- MUTEX_LOCK(&stream_.mutex);
-
- OSStatus err;
- AudioDeviceID id;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiJack::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
- id = *( (AudioDeviceID *) devices_[stream_.device[0]].apiDeviceId );
- err = AudioDeviceStart(id, callbackHandler);
- if (err != noErr) {
- sprintf(message_, "RtApiCore: OS-X error starting callback procedure on device (%s).",
- devices_[stream_.device[0]].name.c_str());
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
- }
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ int result = jack_activate( handle->client );
+ if ( result ) {
+ errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
+ goto unlock;
}
- if (stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1]) ) {
+ const char **ports;
- id = *( (AudioDeviceID *) devices_[stream_.device[1]].apiDeviceId );
- err = AudioDeviceStart(id, callbackHandler);
- if (err != noErr) {
- sprintf(message_, "RtApiCore: OS-X error starting input callback procedure on device (%s).",
- devices_[stream_.device[0]].name.c_str());
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
+ // Get the list of available ports.
+ if ( shouldAutoconnect_ && (stream_.mode == OUTPUT || stream_.mode == DUPLEX) ) {
+ result = 1;
+ ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput);
+ if ( ports == NULL) {
+ errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
+ goto unlock;
}
- }
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
- handle->stopStream = false;
+ // Now make the port connections. Since RtAudio wasn't designed to
+ // allow the user to select particular channels of a device, we'll
+ // just open the first "nChannels" ports with offset.
+ for ( unsigned int i=0; i<stream_.nUserChannels[0]; i++ ) {
+ result = 1;
+ if ( ports[ stream_.channelOffset[0] + i ] )
+ result = jack_connect( handle->client, jack_port_name( handle->ports[0][i] ), ports[ stream_.channelOffset[0] + i ] );
+ if ( result ) {
+ free( ports );
+ errorText_ = "RtApiJack::startStream(): error connecting output ports!";
+ goto unlock;
+ }
+ }
+ free(ports);
+ }
+
+ if ( shouldAutoconnect_ && (stream_.mode == INPUT || stream_.mode == DUPLEX) ) {
+ result = 1;
+ ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput );
+ if ( ports == NULL) {
+ errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
+ goto unlock;
+ }
+
+ // Now make the port connections. See note above.
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ result = 1;
+ if ( ports[ stream_.channelOffset[1] + i ] )
+ result = jack_connect( handle->client, ports[ stream_.channelOffset[1] + i ], jack_port_name( handle->ports[1][i] ) );
+ if ( result ) {
+ free( ports );
+ errorText_ = "RtApiJack::startStream(): error connecting input ports!";
+ goto unlock;
+ }
+ }
+ free(ports);
+ }
+
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
stream_.state = STREAM_RUNNING;
- MUTEX_UNLOCK(&stream_.mutex);
+ unlock:
+ if ( result == 0 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
}
-void RtApiCore :: stopStream()
+void RtApiJack :: stopStream( void )
{
verifyStream();
- if (stream_.state == STREAM_STOPPED) return;
-
- // Change the state before the lock to improve shutdown response
- // when using a callback.
- stream_.state = STREAM_STOPPED;
- MUTEX_LOCK(&stream_.mutex);
-
- OSStatus err;
- AudioDeviceID id;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
-
- id = *( (AudioDeviceID *) devices_[stream_.device[0]].apiDeviceId );
- err = AudioDeviceStop(id, callbackHandler);
- if (err != noErr) {
- sprintf(message_, "RtApiCore: OS-X error stopping callback procedure on device (%s).",
- devices_[stream_.device[0]].name.c_str());
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
- }
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
}
- if (stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1]) ) {
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- id = *( (AudioDeviceID *) devices_[stream_.device[1]].apiDeviceId );
- err = AudioDeviceStop(id, callbackHandler);
- if (err != noErr) {
- sprintf(message_, "RtApiCore: OS-X error stopping input callback procedure on device (%s).",
- devices_[stream_.device[0]].name.c_str());
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 2;
+ pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
}
}
- MUTEX_UNLOCK(&stream_.mutex);
-}
-
-void RtApiCore :: abortStream()
-{
- stopStream();
+ jack_deactivate( handle->client );
+ stream_.state = STREAM_STOPPED;
}
-void RtApiCore :: tickStream()
+void RtApiJack :: abortStream( void )
{
verifyStream();
-
- if (stream_.state == STREAM_STOPPED) return;
-
- if (stream_.callbackInfo.usingCallback) {
- sprintf(message_, "RtApiCore: tickStream() should not be used when a callback function is set!");
- error(RtError::WARNING);
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
return;
}
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ handle->drainCounter = 2;
- MUTEX_LOCK(&stream_.mutex);
+ stopStream();
+}
- pthread_cond_wait(&handle->condition, &stream_.mutex);
+// This function will be called by a spawned thread when the user
+// callback function signals that the stream should be stopped or
+// aborted. It is necessary to handle it this way because the
+// callbackEvent() function must return before the jack_deactivate()
+// function will return.
+static void *jackStopStream( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiJack *object = (RtApiJack *) info->object;
- MUTEX_UNLOCK(&stream_.mutex);
+ object->stopStream();
+ pthread_exit( NULL );
}
-void RtApiCore :: callbackEvent( AudioDeviceID deviceId, void *inData, void *outData )
+bool RtApiJack :: callbackEvent( unsigned long nframes )
{
- verifyStream();
-
- if (stream_.state == STREAM_STOPPED) return;
+ if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return FAILURE;
+ }
+ if ( stream_.bufferSize != nframes ) {
+ errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
+ error( RtAudioError::WARNING );
+ return FAILURE;
+ }
CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
- CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
- AudioBufferList *inBufferList = (AudioBufferList *) inData;
- AudioBufferList *outBufferList = (AudioBufferList *) outData;
-
- if ( info->usingCallback && handle->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();
- return;
- }
+ JackHandle *handle = (JackHandle *) stream_.apiHandle;
- MUTEX_LOCK(&stream_.mutex);
+ // Check if we were draining the stream and signal is finished.
+ if ( handle->drainCounter > 3 ) {
+ ThreadHandle threadId;
- // Invoke user callback first, to get fresh output data. Don't
- // invoke the user callback if duplex mode AND the input/output devices
- // are different AND this function is called for the input device.
- AudioDeviceID id = *( (AudioDeviceID *) devices_[stream_.device[0]].apiDeviceId );
- if ( info->usingCallback && (stream_.mode != DUPLEX || deviceId == id ) ) {
+ stream_.state = STREAM_STOPPING;
+ if ( handle->internalDrain == true )
+ pthread_create( &threadId, NULL, jackStopStream, info );
+ else
+ pthread_cond_signal( &handle->condition );
+ return SUCCESS;
+ }
+
+ // Invoke user callback first, to get fresh output data.
+ if ( handle->drainCounter == 0 ) {
RtAudioCallback callback = (RtAudioCallback) info->callback;
- handle->stopStream = callback(stream_.userBuffer, stream_.bufferSize, info->userData);
- if ( handle->xrun == true ) {
- handle->xrun = false;
- MUTEX_UNLOCK(&stream_.mutex);
- return;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
+ }
+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( cbReturnValue == 2 ) {
+ stream_.state = STREAM_STOPPING;
+ handle->drainCounter = 2;
+ ThreadHandle id;
+ pthread_create( &id, NULL, jackStopStream, info );
+ return SUCCESS;
+ }
+ else if ( cbReturnValue == 1 ) {
+ handle->drainCounter = 1;
+ handle->internalDrain = true;
}
}
- if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == id ) ) {
-
- if (stream_.doConvertBuffer[0]) {
+ jack_default_audio_sample_t *jackbuffer;
+ unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- if ( !stream_.deInterleave[0] )
- stream_.deviceBuffer = (char *) outBufferList->mBuffers[handle->index[0]].mData;
- else
- stream_.deviceBuffer = handle->deviceBuffer;
+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
- convertStreamBuffer(OUTPUT);
- if ( stream_.doByteSwap[0] )
- byteSwapBuffer(stream_.deviceBuffer,
- stream_.bufferSize * stream_.nDeviceChannels[0],
- stream_.deviceFormat[0]);
-
- if ( stream_.deInterleave[0] ) {
- int bufferBytes = outBufferList->mBuffers[handle->index[0]].mDataByteSize;
- for ( int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
- memcpy(outBufferList->mBuffers[handle->index[0]+i].mData,
- &stream_.deviceBuffer[i*bufferBytes], bufferBytes );
- }
+ 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_.doByteSwap[0])
- byteSwapBuffer(stream_.userBuffer,
- stream_.bufferSize * stream_.nUserChannels[0],
- stream_.userFormat);
-
- memcpy(outBufferList->mBuffers[handle->index[0]].mData,
- stream_.userBuffer,
- outBufferList->mBuffers[handle->index[0]].mDataByteSize );
- }
- }
+ else if ( stream_.doConvertBuffer[0] ) {
- if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && deviceId == id ) ) {
-
- if (stream_.doConvertBuffer[1]) {
+ convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
- if ( stream_.deInterleave[1] ) {
- stream_.deviceBuffer = (char *) handle->deviceBuffer;
- int bufferBytes = inBufferList->mBuffers[handle->index[1]].mDataByteSize;
- for ( int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
- memcpy(&stream_.deviceBuffer[i*bufferBytes],
- inBufferList->mBuffers[handle->index[1]+i].mData, bufferBytes );
- }
+ 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
- stream_.deviceBuffer = (char *) inBufferList->mBuffers[handle->index[1]].mData;
-
- if ( stream_.doByteSwap[1] )
- byteSwapBuffer(stream_.deviceBuffer,
- stream_.bufferSize * stream_.nDeviceChannels[1],
- stream_.deviceFormat[1]);
- convertStreamBuffer(INPUT);
-
}
- else {
- memcpy(stream_.userBuffer,
- inBufferList->mBuffers[handle->index[1]].mData,
- inBufferList->mBuffers[handle->index[1]].mDataByteSize );
-
- if (stream_.doByteSwap[1])
- byteSwapBuffer(stream_.userBuffer,
- stream_.bufferSize * stream_.nUserChannels[1],
- 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 );
+ }
}
}
- if ( !info->usingCallback && (stream_.mode != DUPLEX || deviceId == id ) )
- pthread_cond_signal(&handle->condition);
-
- MUTEX_UNLOCK(&stream_.mutex);
-}
-
-void RtApiCore :: setStreamCallback(RtAudioCallback callback, void *userData)
-{
- verifyStream();
-
- if ( stream_.callbackInfo.usingCallback ) {
- sprintf(message_, "RtApiCore: A callback is already set for this stream!");
- error(RtError::WARNING);
- return;
+ // Don't bother draining input
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
}
- stream_.callbackInfo.callback = (void *) callback;
- stream_.callbackInfo.userData = userData;
- stream_.callbackInfo.usingCallback = true;
-}
-
-void RtApiCore :: cancelStreamCallback()
-{
- verifyStream();
-
- if (stream_.callbackInfo.usingCallback) {
-
- if (stream_.state == STREAM_RUNNING)
- stopStream();
-
- MUTEX_LOCK(&stream_.mutex);
-
- stream_.callbackInfo.usingCallback = false;
- stream_.callbackInfo.userData = NULL;
- stream_.state = STREAM_STOPPED;
- stream_.callbackInfo.callback = NULL;
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
- MUTEX_UNLOCK(&stream_.mutex);
+ if ( stream_.doConvertBuffer[1] ) {
+ for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
+ memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
+ }
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
+ }
+ else { // no buffer conversion
+ for ( unsigned int i=0; i<stream_.nUserChannels[1]; i++ ) {
+ jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
+ memcpy( &stream_.userBuffer[1][i*bufferBytes], jackbuffer, bufferBytes );
+ }
+ }
}
-}
-
-//******************** End of __MACOSX_CORE__ *********************//
+ unlock:
+ RtApi::tickStreamTime();
+ return SUCCESS;
+}
+ //******************** End of __UNIX_JACK__ *********************//
#endif
-#if defined(__LINUX_JACK__)
+#if defined(__WINDOWS_ASIO__) // ASIO API on Windows
-// JACK is a low-latency audio server, written primarily for the
-// GNU/Linux operating system. It can connect a number of different
-// applications to an audio device, as well as allowing them to share
-// audio between themselves.
-//
-// The JACK server must be running before RtApiJack can be instantiated.
-// RtAudio will report just a single "device", which is the JACK audio
-// server. The JACK server is typically started in a terminal as follows:
-//
-// .jackd -d alsa -d hw:0
-//
-// 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,
+// 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.
//
-// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
+// 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.
//
-// 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.
+// On unix systems, we make use of a pthread condition variable.
+// Since there is no equivalent in Windows, I hacked something based
+// on information found in
+// http://www.cs.wustl.edu/~schmidt/win32-cv-1.html.
-#include <jack/jack.h>
-#include <unistd.h>
+#include "asiosys.h"
+#include "asio.h"
+#include "iasiothiscallresolver.h"
+#include "asiodrivers.h"
+#include <cmath>
-// A structure to hold various information related to the Jack API
-// implementation.
-struct JackHandle {
- jack_client_t *client;
- jack_port_t **ports[2];
- bool clientOpen;
- bool stopStream;
- pthread_cond_t condition;
+static AsioDrivers drivers;
+static ASIOCallbacks asioCallbacks;
+static ASIODriverInfo driverInfo;
+static CallbackInfo *asioCallbackInfo;
+static bool asioXRun;
- JackHandle()
- :client(0), clientOpen(false), stopStream(false) {}
+struct AsioHandle {
+ int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
+ ASIOBufferInfo *bufferInfos;
+ HANDLE condition;
+
+ AsioHandle()
+ :drainCounter(0), internalDrain(false), bufferInfos(0) {}
};
-std::string jackmsg;
+// Function declarations (definitions at end of section)
+static const char* getAsioErrorString( ASIOError result );
+static void sampleRateChanged( ASIOSampleRate sRate );
+static long asioMessages( long selector, long value, void* message, double* opt );
-static void jackerror (const char *desc)
+RtApiAsio :: RtApiAsio()
{
- jackmsg.erase();
- jackmsg.append( desc, strlen(desc)+1 );
-}
+ // ASIO cannot run on a multi-threaded appartment. You can call
+ // CoInitialize beforehand, but it must be for appartment threading
+ // (in which case, CoInitilialize will return S_FALSE here).
+ coInitialized_ = false;
+ HRESULT hr = CoInitialize( NULL );
+ if ( FAILED(hr) ) {
+ errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
+ error( RtAudioError::WARNING );
+ }
+ coInitialized_ = true;
-RtApiJack :: RtApiJack()
-{
- this->initialize();
+ drivers.removeCurrentDriver();
+ driverInfo.asioVersion = 2;
- if (nDevices_ <= 0) {
- sprintf(message_, "RtApiJack: no Linux Jack server found or connection error (jack: %s)!",
- jackmsg.c_str());
- error(RtError::NO_DEVICES_FOUND);
- }
+ // See note in DirectSound implementation about GetDesktopWindow().
+ driverInfo.sysRef = GetForegroundWindow();
}
-RtApiJack :: ~RtApiJack()
+RtApiAsio :: ~RtApiAsio()
{
- if ( stream_.mode != UNINITIALIZED ) closeStream();
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+ if ( coInitialized_ ) CoUninitialize();
}
-void RtApiJack :: initialize(void)
+unsigned int RtApiAsio :: getDeviceCount( void )
{
- nDevices_ = 0;
-
- // Tell the jack server to call jackerror() when it experiences an
- // error. This function saves the error message for subsequent
- // reporting via the normal RtAudio error function.
- jack_set_error_function( jackerror );
-
- // Look for jack server and try to become a client.
- jack_client_t *client;
- if ( (client = jack_client_new( "RtApiJack" )) == 0)
- return;
-
- RtApiDevice device;
- // Determine the name of the device.
- device.name = "Jack Server";
- devices_.push_back(device);
- nDevices_++;
-
- jack_client_close(client);
+ return (unsigned int) drivers.asioGetNumDev();
}
-void RtApiJack :: probeDeviceInfo(RtApiDevice *info)
+RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
{
- // Look for jack server and try to become a client.
- jack_client_t *client;
- if ( (client = jack_client_new( "RtApiJack" )) == 0) {
- sprintf(message_, "RtApiJack: error connecting to Linux Jack server in probeDeviceInfo() (jack: %s)!",
- jackmsg.c_str());
- error(RtError::WARNING);
- return;
- }
-
- // Get the current jack server sample rate.
- info->sampleRates.clear();
- info->sampleRates.push_back( jack_get_sample_rate(client) );
+ RtAudio::DeviceInfo info;
+ info.probed = false;
- // Count the available ports as device channels. Jack "input ports"
- // equal RtAudio output channels.
- const char **ports;
- char *port;
- unsigned int nChannels = 0;
- ports = jack_get_ports( client, NULL, NULL, JackPortIsInput );
- if ( ports ) {
- port = (char *) ports[nChannels];
- while ( port )
- port = (char *) ports[++nChannels];
- free( ports );
- info->maxOutputChannels = nChannels;
- info->minOutputChannels = 1;
+ // Get device ID
+ unsigned int nDevices = getDeviceCount();
+ if ( nDevices == 0 ) {
+ errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
+ error( RtAudioError::INVALID_USE );
+ return info;
}
- // Jack "output ports" equal RtAudio input channels.
- nChannels = 0;
- ports = jack_get_ports( client, NULL, NULL, JackPortIsOutput );
- if ( ports ) {
- port = (char *) ports[nChannels];
- while ( port )
- port = (char *) ports[++nChannels];
- free( ports );
- info->maxInputChannels = nChannels;
- info->minInputChannels = 1;
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
}
- if (info->maxOutputChannels == 0 && info->maxInputChannels == 0) {
- jack_client_close(client);
- sprintf(message_, "RtApiJack: error determining jack input/output channels!");
- error(RtError::WARNING);
- return;
+ // If a stream is already open, we cannot probe other devices. Thus, use the saved results.
+ if ( stream_.state != STREAM_CLOSED ) {
+ if ( device >= devices_.size() ) {
+ errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
+ return devices_[ device ];
}
- 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;
+ char driverName[32];
+ ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
}
- // Get the jack data format type. There isn't much documentation
- // regarding supported data formats in jack. I'm assuming here that
- // the default type will always be a floating-point type, of length
- // equal to either 4 or 8 bytes.
- int sample_size = sizeof( jack_default_audio_sample_t );
- if ( sample_size == 4 )
- info->nativeFormats = RTAUDIO_FLOAT32;
- else if ( sample_size == 8 )
- info->nativeFormats = RTAUDIO_FLOAT64;
+ info.name = driverName;
- // Check that we have a supported format
- if (info->nativeFormats == 0) {
- jack_client_close(client);
- sprintf(message_, "RtApiJack: error determining jack server data format!");
- error(RtError::WARNING);
- return;
+ if ( !drivers.loadDriver( driverName ) ) {
+ errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
}
- jack_client_close(client);
- info->probed = true;
-}
-
-int jackCallbackHandler(jack_nframes_t nframes, void *infoPointer)
-{
- CallbackInfo *info = (CallbackInfo *) infoPointer;
- RtApiJack *object = (RtApiJack *) info->object;
- try {
- object->callbackEvent( (unsigned long) nframes );
+ result = ASIOInit( &driverInfo );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
}
- catch (RtError &exception) {
- fprintf(stderr, "\nRtApiJack: callback handler error (%s)!\n\n", exception.getMessageString());
- return 0;
+
+ // Determine the device channel information.
+ long inputChannels, outputChannels;
+ result = ASIOGetChannels( &inputChannels, &outputChannels );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
}
- return 0;
-}
+ info.outputChannels = outputChannels;
+ info.inputChannels = inputChannels;
+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
-void jackShutdown(void *infoPointer)
-{
- CallbackInfo *info = (CallbackInfo *) infoPointer;
- JackHandle *handle = (JackHandle *) info->apiInfo;
- handle->clientOpen = false;
- RtApiJack *object = (RtApiJack *) info->object;
- try {
- object->closeStream();
+ // Determine the supported sample rates.
+ info.sampleRates.clear();
+ for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
+ result = ASIOCanSampleRate( (ASIOSampleRate) SAMPLE_RATES[i] );
+ if ( result == ASE_OK ) {
+ info.sampleRates.push_back( SAMPLE_RATES[i] );
+
+ if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
+ info.preferredSampleRate = SAMPLE_RATES[i];
+ }
}
- catch (RtError &exception) {
- fprintf(stderr, "\nRtApiJack: jackShutdown error (%s)!\n\n", exception.getMessageString());
- return;
+
+ // Determine supported data types ... just check first channel and assume rest are the same.
+ ASIOChannelInfo channelInfo;
+ channelInfo.channel = 0;
+ channelInfo.isInput = true;
+ if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
+ result = ASIOGetChannelInfo( &channelInfo );
+ if ( result != ASE_OK ) {
+ drivers.removeCurrentDriver();
+ errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
}
- fprintf(stderr, "\nRtApiJack: the Jack server is shutting down ... stream stopped and closed!!!\n\n");
+ info.nativeFormats = 0;
+ if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
+ info.nativeFormats |= RTAUDIO_SINT16;
+ else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
+ info.nativeFormats |= RTAUDIO_SINT32;
+ else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+ else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
+ info.nativeFormats |= RTAUDIO_FLOAT64;
+ else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB )
+ info.nativeFormats |= RTAUDIO_SINT24;
+
+ if ( info.outputChannels > 0 )
+ if ( getDefaultOutputDevice() == device ) info.isDefaultOutput = true;
+ if ( info.inputChannels > 0 )
+ if ( getDefaultInputDevice() == device ) info.isDefaultInput = true;
+
+ info.probed = true;
+ drivers.removeCurrentDriver();
+ return info;
}
-int jackXrun( void * )
+static void bufferSwitch( long index, ASIOBool /*processNow*/ )
{
- fprintf(stderr, "\nRtApiJack: audio overrun/underrun reported!\n");
- return 0;
+ RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
+ object->callbackEvent( index );
}
-bool RtApiJack :: probeDeviceOpen(int device, StreamMode mode, int channels,
- int sampleRate, RtAudioFormat format,
- int *bufferSize, int numberOfBuffers)
+void RtApiAsio :: saveDeviceInfo( void )
{
- // Compare the jack server channels to the requested number of channels.
- if ( (mode == OUTPUT && devices_[device].maxOutputChannels < channels ) ||
- (mode == INPUT && devices_[device].maxInputChannels < channels ) ) {
- sprintf(message_, "RtApiJack: the Jack server does not support requested channels!");
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
+ devices_.clear();
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
+ unsigned int nDevices = getDeviceCount();
+ devices_.resize( nDevices );
+ for ( unsigned int i=0; i<nDevices; i++ )
+ devices_[i] = getDeviceInfo( i );
+}
- // Look for jack server and try to become a client (only do once per stream).
- char label[32];
- jack_client_t *client = 0;
- if ( mode == OUTPUT || (mode == INPUT && stream_.mode != OUTPUT) ) {
- snprintf(label, 32, "RtApiJack");
- if ( (client = jack_client_new( (const char *) label )) == 0) {
- sprintf(message_, "RtApiJack: cannot connect to Linux Jack server in probeDeviceOpen() (jack: %s)!",
- jackmsg.c_str());
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
- }
- else {
- // The handle must have been created on an earlier pass.
- client = handle->client;
- }
+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 )
+{////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
- // First, check the jack server sample rate.
- int jack_rate;
- jack_rate = (int) jack_get_sample_rate(client);
- if ( sampleRate != jack_rate ) {
- jack_client_close(client);
- sprintf( message_, "RtApiJack: the requested sample rate (%d) is different than the JACK server rate (%d).",
- sampleRate, jack_rate );
- error(RtError::DEBUG_WARNING);
+ bool isDuplexInput = mode == INPUT && stream_.mode == OUTPUT;
+
+ // For ASIO, a duplex stream MUST use the same driver.
+ if ( isDuplexInput && stream_.device[0] != device ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
return FAILURE;
}
- stream_.sampleRate = jack_rate;
- // The jack server seems to support just a single floating-point
- // data type. Since we already checked it before, just use what we
- // found then.
- stream_.deviceFormat[mode] = devices_[device].nativeFormats;
- stream_.userFormat = format;
+ 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;
+ }
- // Jack always uses non-interleaved buffers. We'll need to
- // de-interleave if we have more than one channel.
- stream_.deInterleave[mode] = false;
- if ( channels > 1 )
- stream_.deInterleave[mode] = true;
+ // Only load the driver once for duplex stream.
+ if ( !isDuplexInput ) {
+ // The getDeviceInfo() function will not work when a stream is open
+ // because ASIO does not allow multiple devices to run at the same
+ // time. Thus, we'll probe the system before opening a stream and
+ // save the results for use by getDeviceInfo().
+ this->saveDeviceInfo();
+
+ if ( !drivers.loadDriver( driverName ) ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // Jack always provides host byte-ordered data.
- stream_.doByteSwap[mode] = false;
+ result = ASIOInit( &driverInfo );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ 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;
+ // keep them before any "goto error", they are used for error cleanup + goto device boundary checks
+ bool buffersAllocated = false;
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ unsigned int nChannels;
+
+
+ // Check the device channel count.
+ long inputChannels, outputChannels;
+ result = ASIOGetChannels( &inputChannels, &outputChannels );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
+ if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
+ ( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
stream_.nDeviceChannels[mode] = channels;
stream_.nUserChannels[mode] = channels;
+ stream_.channelOffset[mode] = firstChannel;
- stream_.doConvertBuffer[mode] = false;
- if (stream_.userFormat != stream_.deviceFormat[mode])
- stream_.doConvertBuffer[mode] = true;
- if (stream_.deInterleave[mode])
- stream_.doConvertBuffer[mode] = true;
-
- // Allocate our JackHandle structure for the stream.
- if ( handle == 0 ) {
- handle = (JackHandle *) calloc(1, sizeof(JackHandle));
- if ( handle == NULL ) {
- sprintf(message_, "RtApiJack: error allocating JackHandle memory (%s).",
- devices_[device].name.c_str());
- goto error;
- }
- handle->ports[0] = 0;
- handle->ports[1] = 0;
- if ( pthread_cond_init(&handle->condition, NULL) ) {
- sprintf(message_, "RtApiJack: error initializing pthread condition variable!");
- goto error;
- }
- stream_.apiHandle = (void *) handle;
- handle->client = client;
- handle->clientOpen = true;
+ // Verify the sample rate is supported.
+ result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
+ errorText_ = errorStream_.str();
+ goto error;
}
- // Allocate necessary internal buffers.
- if ( stream_.nUserChannels[0] != stream_.nUserChannels[1] ) {
+ // Get the current sample rate
+ ASIOSampleRate currentRate;
+ result = ASIOGetSampleRate( ¤tRate );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
- 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) {
- sprintf(message_, "RtApiJack: error allocating user buffer memory (%s).",
- devices_[device].name.c_str());
+ // Set the sample rate only if necessary
+ if ( currentRate != sampleRate ) {
+ result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
+ errorText_ = errorStream_.str();
goto error;
}
}
- if ( stream_.doConvertBuffer[mode] ) {
+ // Determine the driver data type.
+ ASIOChannelInfo channelInfo;
+ channelInfo.channel = 0;
+ if ( mode == OUTPUT ) channelInfo.isInput = false;
+ else channelInfo.isInput = true;
+ result = ASIOGetChannelInfo( &channelInfo );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
+ errorText_ = errorStream_.str();
+ goto error;
+ }
- 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;
- }
- }
+ // Assuming WINDOWS host is always little-endian.
+ stream_.doByteSwap[mode] = false;
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = 0;
+ if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
+ }
+ else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
+ }
+ else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
+ }
+ else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
+ if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
+ }
+ else if ( channelInfo.type == ASIOSTInt24MSB || channelInfo.type == ASIOSTInt24LSB ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ if ( channelInfo.type == ASIOSTInt24MSB ) stream_.doByteSwap[mode] = true;
+ }
- if ( makeBuffer ) {
- buffer_bytes *= *bufferSize;
- if (stream_.deviceBuffer) free(stream_.deviceBuffer);
- stream_.deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream_.deviceBuffer == NULL) {
- sprintf(message_, "RtApiJack: error allocating device buffer memory (%s).",
- devices_[device].name.c_str());
- goto error;
- }
- }
+ if ( stream_.deviceFormat[mode] == 0 ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ goto error;
}
- // Allocate memory for the Jack ports (channels) identifiers.
- handle->ports[mode] = (jack_port_t **) malloc (sizeof (jack_port_t *) * channels);
- if ( handle->ports[mode] == NULL ) {
- sprintf(message_, "RtApiJack: error allocating port handle memory (%s).",
- devices_[device].name.c_str());
+ // Set the buffer size. For a duplex stream, this will end up
+ // setting the buffer size based on the input constraints, which
+ // should be ok.
+ long minSize, maxSize, preferSize, granularity;
+ result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
+ errorText_ = errorStream_.str();
goto error;
}
- stream_.device[mode] = device;
- stream_.state = STREAM_STOPPED;
- stream_.callbackInfo.usingCallback = false;
- stream_.callbackInfo.object = (void *) this;
- stream_.callbackInfo.apiInfo = (void *) handle;
+ if ( isDuplexInput ) {
+ // When this is the duplex input (output was opened before), then we have to use the same
+ // buffersize as the output, because it might use the preferred buffer size, which most
+ // likely wasn't passed as input to this. The buffer sizes have to be identically anyway,
+ // So instead of throwing an error, make them equal. The caller uses the reference
+ // to the "bufferSize" param as usual to set up processing buffers.
- 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, NULL );
- jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
- }
+ *bufferSize = stream_.bufferSize;
- return SUCCESS;
+ } else {
+ if ( *bufferSize == 0 ) *bufferSize = preferSize;
+ else if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
+ else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
+ else if ( granularity == -1 ) {
+ // Make sure bufferSize is a power of two.
+ int log2_of_min_size = 0;
+ int log2_of_max_size = 0;
- error:
- if ( handle ) {
- pthread_cond_destroy(&handle->condition);
- if ( handle->clientOpen == true )
- jack_client_close(handle->client);
+ for ( unsigned int i = 0; i < sizeof(long) * 8; i++ ) {
+ if ( minSize & ((long)1 << i) ) log2_of_min_size = i;
+ if ( maxSize & ((long)1 << i) ) log2_of_max_size = i;
+ }
- if ( handle->ports[0] ) free(handle->ports[0]);
- if ( handle->ports[1] ) free(handle->ports[1]);
+ long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
+ int min_delta_num = log2_of_min_size;
- free( handle );
- stream_.apiHandle = 0;
+ for (int i = log2_of_min_size + 1; i <= log2_of_max_size; i++) {
+ long current_delta = std::abs( (long)*bufferSize - ((long)1 << i) );
+ if (current_delta < min_delta) {
+ min_delta = current_delta;
+ min_delta_num = i;
+ }
+ }
+
+ *bufferSize = ( (unsigned int)1 << min_delta_num );
+ if ( *bufferSize < (unsigned int) minSize ) *bufferSize = (unsigned int) minSize;
+ else if ( *bufferSize > (unsigned int) maxSize ) *bufferSize = (unsigned int) maxSize;
+ }
+ else if ( granularity != 0 ) {
+ // Set to an even multiple of granularity, rounding up.
+ *bufferSize = (*bufferSize + granularity-1) / granularity * granularity;
+ }
}
- if (stream_.userBuffer) {
- free(stream_.userBuffer);
- stream_.userBuffer = 0;
+ /*
+ // we don't use it anymore, see above!
+ // Just left it here for the case...
+ if ( isDuplexInput && stream_.bufferSize != *bufferSize ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
+ goto error;
}
+ */
- error(RtError::WARNING);
- return FAILURE;
-}
+ stream_.bufferSize = *bufferSize;
+ stream_.nBuffers = 2;
-void RtApiJack :: closeStream()
-{
- // We don't want an exception to be thrown here because this
- // function is called by our class destructor. So, do our own
- // stream check.
- if ( stream_.mode == UNINITIALIZED ) {
- sprintf(message_, "RtApiJack::closeStream(): no open stream to close!");
- error(RtError::WARNING);
- return;
- }
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
- if ( handle && handle->clientOpen == true ) {
- if (stream_.state == STREAM_RUNNING)
- jack_deactivate(handle->client);
+ // ASIO always uses non-interleaved buffers.
+ stream_.deviceInterleaved[mode] = false;
- jack_client_close(handle->client);
- }
+ // Allocate, if necessary, our AsioHandle structure for the stream.
+ if ( handle == 0 ) {
+ try {
+ handle = new AsioHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
+ goto error;
+ }
+ handle->bufferInfos = 0;
- if ( handle ) {
- if ( handle->ports[0] ) free(handle->ports[0]);
- if ( handle->ports[1] ) free(handle->ports[1]);
- pthread_cond_destroy(&handle->condition);
- free( handle );
- stream_.apiHandle = 0;
+ // Create a manual-reset event.
+ handle->condition = CreateEvent( NULL, // no security
+ TRUE, // manual-reset
+ FALSE, // non-signaled initially
+ NULL ); // unnamed
+ stream_.apiHandle = (void *) handle;
}
- if (stream_.userBuffer) {
- free(stream_.userBuffer);
- stream_.userBuffer = 0;
+ // Create the ASIO internal buffers. Since RtAudio sets up input
+ // and output separately, we'll have to dispose of previously
+ // created output buffers for a duplex stream.
+ if ( mode == INPUT && stream_.mode == OUTPUT ) {
+ ASIODisposeBuffers();
+ if ( handle->bufferInfos ) free( handle->bufferInfos );
}
- if (stream_.deviceBuffer) {
- free(stream_.deviceBuffer);
- stream_.deviceBuffer = 0;
+ // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
+ unsigned int i;
+ nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+ handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
+ if ( handle->bufferInfos == NULL ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
+ errorText_ = errorStream_.str();
+ goto error;
}
- stream_.mode = UNINITIALIZED;
-}
-
+ ASIOBufferInfo *infos;
+ infos = handle->bufferInfos;
+ for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
+ infos->isInput = ASIOFalse;
+ infos->channelNum = i + stream_.channelOffset[0];
+ infos->buffers[0] = infos->buffers[1] = 0;
+ }
+ for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
+ infos->isInput = ASIOTrue;
+ infos->channelNum = i + stream_.channelOffset[1];
+ infos->buffers[0] = infos->buffers[1] = 0;
+ }
-void RtApiJack :: startStream()
-{
- verifyStream();
- if (stream_.state == STREAM_RUNNING) return;
+ // prepare for callbacks
+ stream_.sampleRate = sampleRate;
+ stream_.device[mode] = device;
+ stream_.mode = isDuplexInput ? DUPLEX : mode;
- MUTEX_LOCK(&stream_.mutex);
+ // store this class instance before registering callbacks, that are going to use it
+ asioCallbackInfo = &stream_.callbackInfo;
+ stream_.callbackInfo.object = (void *) this;
- char label[64];
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- for ( 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);
- }
+ // Set up the ASIO callback structure and create the ASIO data buffers.
+ asioCallbacks.bufferSwitch = &bufferSwitch;
+ asioCallbacks.sampleRateDidChange = &sampleRateChanged;
+ asioCallbacks.asioMessage = &asioMessages;
+ asioCallbacks.bufferSwitchTimeInfo = NULL;
+ result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
+ if ( result != ASE_OK ) {
+ // Standard method failed. This can happen with strict/misbehaving drivers that return valid buffer size ranges
+ // but only accept the preferred buffer size as parameter for ASIOCreateBuffers (e.g. Creative's ASIO driver).
+ // In that case, let's be naïve and try that instead.
+ *bufferSize = preferSize;
+ stream_.bufferSize = *bufferSize;
+ result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
}
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
- for ( 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);
- }
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
+ errorText_ = errorStream_.str();
+ goto error;
}
+ buffersAllocated = true;
+ stream_.state = STREAM_STOPPED;
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
- if (jack_activate(handle->client)) {
- sprintf(message_, "RtApiJack: unable to activate JACK client!");
- error(RtError::SYSTEM_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_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
}
- const char **ports;
- int result;
- // Get the list of available ports.
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- ports = jack_get_ports(handle->client, NULL, NULL, JackPortIsPhysical|JackPortIsInput);
- if ( ports == NULL) {
- sprintf(message_, "RtApiJack: error determining available jack input ports!");
- error(RtError::SYSTEM_ERROR);
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( isDuplexInput && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= bytesOut ) makeBuffer = false;
}
- // 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.
- for ( int i=0; i<stream_.nUserChannels[0]; i++ ) {
- result = 1;
- if ( ports[i] )
- result = jack_connect( handle->client, jack_port_name(handle->ports[0][i]), ports[i] );
- if ( result ) {
- free(ports);
- sprintf(message_, "RtApiJack: error connecting output ports!");
- error(RtError::SYSTEM_ERROR);
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
}
}
- free(ports);
}
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
- ports = jack_get_ports( handle->client, NULL, NULL, JackPortIsPhysical|JackPortIsOutput );
- if ( ports == NULL) {
- sprintf(message_, "RtApiJack: error determining available jack output ports!");
- error(RtError::SYSTEM_ERROR);
- }
-
- // Now make the port connections. See note above.
- for ( int i=0; i<stream_.nUserChannels[1]; i++ ) {
- result = 1;
- if ( ports[i] )
- result = jack_connect( handle->client, ports[i], jack_port_name(handle->ports[1][i]) );
- if ( result ) {
- free(ports);
- sprintf(message_, "RtApiJack: error connecting input ports!");
- error(RtError::SYSTEM_ERROR);
- }
- }
- free(ports);
+ // Determine device latencies
+ long inputLatency, outputLatency;
+ result = ASIOGetLatencies( &inputLatency, &outputLatency );
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING); // warn but don't fail
+ }
+ else {
+ stream_.latency[0] = outputLatency;
+ stream_.latency[1] = inputLatency;
}
- handle->stopStream = false;
- stream_.state = STREAM_RUNNING;
+ // 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 );
- MUTEX_UNLOCK(&stream_.mutex);
-}
+ return SUCCESS;
-void RtApiJack :: stopStream()
-{
- verifyStream();
- if (stream_.state == STREAM_STOPPED) return;
+ error:
+ if ( !isDuplexInput ) {
+ // the cleanup for error in the duplex input, is done by RtApi::openStream
+ // So we clean up for single channel only
- // Change the state before the lock to improve shutdown response
- // when using a callback.
- stream_.state = STREAM_STOPPED;
- MUTEX_LOCK(&stream_.mutex);
+ if ( buffersAllocated )
+ ASIODisposeBuffers();
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
- jack_deactivate(handle->client);
+ drivers.removeCurrentDriver();
- MUTEX_UNLOCK(&stream_.mutex);
-}
+ if ( handle ) {
+ CloseHandle( handle->condition );
+ if ( handle->bufferInfos )
+ free( handle->bufferInfos );
-void RtApiJack :: abortStream()
-{
- stopStream();
-}
+ delete handle;
+ stream_.apiHandle = 0;
+ }
-void RtApiJack :: tickStream()
-{
- verifyStream();
- if (stream_.state == STREAM_STOPPED) return;
+ if ( stream_.userBuffer[mode] ) {
+ free( stream_.userBuffer[mode] );
+ stream_.userBuffer[mode] = 0;
+ }
- if (stream_.callbackInfo.usingCallback) {
- sprintf(message_, "RtApiJack: tickStream() should not be used when a callback function is set!");
- error(RtError::WARNING);
- return;
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
}
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
-
- MUTEX_LOCK(&stream_.mutex);
-
- pthread_cond_wait(&handle->condition, &stream_.mutex);
-
- MUTEX_UNLOCK(&stream_.mutex);
-}
+ return FAILURE;
+}////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
-void RtApiJack :: callbackEvent( unsigned long nframes )
+void RtApiAsio :: closeStream()
{
- verifyStream();
-
- if (stream_.state == STREAM_STOPPED) return;
-
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
- JackHandle *handle = (JackHandle *) stream_.apiHandle;
- if ( info->usingCallback && handle->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();
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
return;
}
- MUTEX_LOCK(&stream_.mutex);
-
- // Invoke user callback first, to get fresh output data.
- if ( info->usingCallback ) {
- RtAudioCallback callback = (RtAudioCallback) info->callback;
- handle->stopStream = callback(stream_.userBuffer, stream_.bufferSize, info->userData);
+ if ( stream_.state == STREAM_RUNNING ) {
+ stream_.state = STREAM_STOPPED;
+ ASIOStop();
}
+ ASIODisposeBuffers();
+ drivers.removeCurrentDriver();
- jack_default_audio_sample_t *jackbuffer;
- long bufferBytes = nframes * sizeof (jack_default_audio_sample_t);
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-
- if (stream_.doConvertBuffer[0]) {
- convertStreamBuffer(OUTPUT);
-
- for ( 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 { // single channel only
- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer(handle->ports[0][0],
- (jack_nframes_t) nframes);
- memcpy(jackbuffer, stream_.userBuffer, bufferBytes );
- }
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ if ( handle ) {
+ CloseHandle( handle->condition );
+ if ( handle->bufferInfos )
+ free( handle->bufferInfos );
+ delete handle;
+ stream_.apiHandle = 0;
}
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
-
- if (stream_.doConvertBuffer[1]) {
- for ( 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 );
- }
- convertStreamBuffer(INPUT);
- }
- else { // single channel only
- jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer(handle->ports[1][0],
- (jack_nframes_t) nframes);
- memcpy(stream_.userBuffer, jackbuffer, bufferBytes );
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
}
}
- if ( !info->usingCallback )
- pthread_cond_signal(&handle->condition);
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
- MUTEX_UNLOCK(&stream_.mutex);
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
}
-void RtApiJack :: setStreamCallback(RtAudioCallback callback, void *userData)
+bool stopThreadCalled = false;
+
+void RtApiAsio :: startStream()
{
verifyStream();
-
- if ( stream_.callbackInfo.usingCallback ) {
- sprintf(message_, "RtApiJack: A callback is already set for this stream!");
- error(RtError::WARNING);
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiAsio::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
return;
}
- stream_.callbackInfo.callback = (void *) callback;
- stream_.callbackInfo.userData = userData;
- stream_.callbackInfo.usingCallback = true;
-}
-
-void RtApiJack :: cancelStreamCallback()
-{
- verifyStream();
-
- if (stream_.callbackInfo.usingCallback) {
+ 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;
+ }
- if (stream_.state == STREAM_RUNNING)
- stopStream();
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ ResetEvent( handle->condition );
+ stream_.state = STREAM_RUNNING;
+ asioXRun = false;
- MUTEX_LOCK(&stream_.mutex);
+ unlock:
+ stopThreadCalled = false;
- stream_.callbackInfo.usingCallback = false;
- stream_.callbackInfo.userData = NULL;
- stream_.state = STREAM_STOPPED;
- stream_.callbackInfo.callback = NULL;
+ if ( result == ASE_OK ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
- MUTEX_UNLOCK(&stream_.mutex);
+void RtApiAsio :: stopStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
}
-}
-#endif
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 2;
+ WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
+ }
+ }
-#if defined(__LINUX_ALSA__)
+ stream_.state = STREAM_STOPPED;
-#include <alsa/asoundlib.h>
-#include <unistd.h>
-#include <ctype.h>
+ ASIOError result = ASIOStop();
+ if ( result != ASE_OK ) {
+ errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
+ errorText_ = errorStream_.str();
+ }
-extern "C" void *alsaCallbackHandler(void * ptr);
+ if ( result == ASE_OK ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
-RtApiAlsa :: RtApiAlsa()
+void RtApiAsio :: abortStream()
{
- this->initialize();
-
- if (nDevices_ <= 0) {
- sprintf(message_, "RtApiAlsa: no Linux ALSA audio devices found!");
- error(RtError::NO_DEVICES_FOUND);
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
}
-}
-RtApiAlsa :: ~RtApiAlsa()
-{
- if ( stream_.mode != UNINITIALIZED )
- closeStream();
+ // The following lines were commented-out because some behavior was
+ // noted where the device buffers need to be zeroed to avoid
+ // continuing sound, even when the device buffers are completely
+ // disposed. So now, calling abort is the same as calling stop.
+ // AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ // handle->drainCounter = 2;
+ stopStream();
}
-void RtApiAlsa :: initialize(void)
+// This function will be called by a spawned thread when the user
+// callback function signals that the stream should be stopped or
+// aborted. It is necessary to handle it this way because the
+// callbackEvent() function must return before the ASIOStop()
+// function will return.
+static unsigned __stdcall asioStopStream( void *ptr )
{
- int card, subdevice, result;
- char name[64];
- const char *cardId;
- snd_ctl_t *handle;
- snd_ctl_card_info_t *info;
- snd_ctl_card_info_alloca(&info);
- RtApiDevice device;
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiAsio *object = (RtApiAsio *) info->object;
- // 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_, "RtApiAlsa: 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_, "RtApiAlsa: control hardware info (%i): %s.", card, snd_strerror(result));
- error(RtError::DEBUG_WARNING);
- goto next_card;
- }
- cardId = snd_ctl_card_info_get_id(info);
- subdevice = -1;
- while (1) {
- result = snd_ctl_pcm_next_device(handle, &subdevice);
- if (result < 0) {
- sprintf(message_, "RtApiAlsa: control next device (%i): %s.", card, snd_strerror(result));
- error(RtError::DEBUG_WARNING);
- break;
- }
- if (subdevice < 0)
- break;
- sprintf( name, "hw:%d,%d", card, subdevice );
- // If a cardId exists and it contains at least one non-numeric
- // character, use it to identify the device. This avoids a bug
- // in ALSA such that a numeric string is interpreted as a device
- // number.
- for ( unsigned int i=0; i<strlen(cardId); i++ ) {
- if ( !isdigit( cardId[i] ) ) {
- sprintf( name, "hw:%s,%d", cardId, subdevice );
- break;
- }
- }
- device.name.erase();
- device.name.append( (const char *)name, strlen(name)+1 );
- devices_.push_back(device);
- nDevices_++;
- }
- next_card:
- snd_ctl_close(handle);
- snd_card_next(&card);
- }
+ object->stopStream();
+ _endthreadex( 0 );
+ return 0;
}
-void RtApiAlsa :: probeDeviceInfo(RtApiDevice *info)
+bool RtApiAsio :: callbackEvent( long bufferIndex )
{
- 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[64];
- char *card;
-
- // Open the control interface for this card.
- strncpy( name, info->name.c_str(), 64 );
- card = strtok(name, ",");
- err = snd_ctl_open(&chandle, card, SND_CTL_NONBLOCK);
- if (err < 0) {
- sprintf(message_, "RtApiAlsa: control open (%s): %s.", card, snd_strerror(err));
- error(RtError::DEBUG_WARNING);
- return;
+ if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) return SUCCESS;
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return FAILURE;
}
- 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);
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
- if ((err = snd_ctl_pcm_info(chandle, pcminfo)) < 0) {
- if (err == -ENOENT) {
- sprintf(message_, "RtApiAlsa: pcm device (%s) doesn't handle output!", info->name.c_str());
- error(RtError::DEBUG_WARNING);
- }
- else {
- sprintf(message_, "RtApiAlsa: snd_ctl_pcm_info error for device (%s) output: %s",
- info->name.c_str(), snd_strerror(err));
- error(RtError::DEBUG_WARNING);
+ // Check if we were draining the stream and signal if finished.
+ if ( handle->drainCounter > 3 ) {
+
+ stream_.state = STREAM_STOPPING;
+ if ( handle->internalDrain == false )
+ SetEvent( handle->condition );
+ else { // spawn a thread to stop the stream
+ unsigned threadId;
+ stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
+ &stream_.callbackInfo, 0, &threadId );
}
- goto capture_probe;
+ return SUCCESS;
}
- err = snd_pcm_open(&handle, info->name.c_str(), stream, open_mode | SND_PCM_NONBLOCK );
- if (err < 0) {
- if ( err == EBUSY )
- sprintf(message_, "RtApiAlsa: pcm playback device (%s) is busy: %s.",
- info->name.c_str(), snd_strerror(err));
- else
- sprintf(message_, "RtApiAlsa: pcm playback open (%s) error: %s.",
- info->name.c_str(), snd_strerror(err));
- error(RtError::DEBUG_WARNING);
- goto capture_probe;
- }
+ // Invoke user callback to get fresh output data UNLESS we are
+ // draining stream.
+ if ( handle->drainCounter == 0 ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && asioXRun == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ asioXRun = false;
+ }
+ if ( stream_.mode != OUTPUT && asioXRun == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ asioXRun = false;
+ }
+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( cbReturnValue == 2 ) {
+ stream_.state = STREAM_STOPPING;
+ handle->drainCounter = 2;
+ unsigned threadId;
+ stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &asioStopStream,
+ &stream_.callbackInfo, 0, &threadId );
+ return SUCCESS;
+ }
+ else if ( cbReturnValue == 1 ) {
+ handle->drainCounter = 1;
+ handle->internalDrain = true;
+ }
+ }
+
+ unsigned int nChannels, bufferBytes, i, j;
+ nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- // 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_, "RtApiAlsa: hardware probe error (%s): %s.",
- info->name.c_str(), snd_strerror(err));
- error(RtError::WARNING);
- goto capture_probe;
- }
+ bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
- // Get output channel information.
- unsigned int value;
- err = snd_pcm_hw_params_get_channels_min(params, &value);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message_, "RtApiAlsa: hardware minimum channel probe error (%s): %s.",
- info->name.c_str(), snd_strerror(err));
- error(RtError::WARNING);
- goto capture_probe;
- }
- info->minOutputChannels = value;
-
- err = snd_pcm_hw_params_get_channels_max(params, &value);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message_, "RtApiAlsa: hardware maximum channel probe error (%s): %s.",
- info->name.c_str(), snd_strerror(err));
- error(RtError::WARNING);
- goto capture_probe;
- }
- info->maxOutputChannels = value;
-
- snd_pcm_close(handle);
-
- capture_probe:
- // Now try for capture
- stream = SND_PCM_STREAM_CAPTURE;
- snd_pcm_info_set_stream(pcminfo, stream);
+ if ( handle->drainCounter > 1 ) { // write zeros to the output stream
+
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput != ASIOTrue )
+ memset( handle->bufferInfos[i].buffers[bufferIndex], 0, bufferBytes );
+ }
- err = snd_ctl_pcm_info(chandle, pcminfo);
- snd_ctl_close(chandle);
- if ( err < 0 ) {
- if (err == -ENOENT) {
- sprintf(message_, "RtApiAlsa: pcm device (%s) doesn't handle input!", info->name.c_str());
- error(RtError::DEBUG_WARNING);
}
- else {
- sprintf(message_, "RtApiAlsa: snd_ctl_pcm_info error for device (%s) input: %s",
- info->name.c_str(), snd_strerror(err));
- error(RtError::DEBUG_WARNING);
+ else if ( stream_.doConvertBuffer[0] ) {
+
+ convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer( stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nDeviceChannels[0],
+ stream_.deviceFormat[0] );
+
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput != ASIOTrue )
+ memcpy( handle->bufferInfos[i].buffers[bufferIndex],
+ &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
+ }
+
}
- if (info->maxOutputChannels == 0)
- // didn't open for playback either ... device invalid
- return;
- goto probe_parameters;
- }
+ else {
- err = snd_pcm_open(&handle, info->name.c_str(), stream, open_mode | SND_PCM_NONBLOCK);
- if (err < 0) {
- if ( err == EBUSY )
- sprintf(message_, "RtApiAlsa: pcm capture device (%s) is busy: %s.",
- info->name.c_str(), snd_strerror(err));
- else
- sprintf(message_, "RtApiAlsa: pcm capture open (%s) error: %s.",
- info->name.c_str(), 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_, "RtApiAlsa: hardware probe error (%s): %s.",
- info->name.c_str(), snd_strerror(err));
- error(RtError::WARNING);
- if (info->maxOutputChannels > 0)
- goto probe_parameters;
- else
- return;
- }
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer( stream_.userBuffer[0],
+ stream_.bufferSize * stream_.nUserChannels[0],
+ stream_.userFormat );
- // Get input channel information.
- err = snd_pcm_hw_params_get_channels_min(params, &value);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message_, "RtApiAlsa: hardware minimum in channel probe error (%s): %s.",
- info->name.c_str(), snd_strerror(err));
- error(RtError::WARNING);
- if (info->maxOutputChannels > 0)
- goto probe_parameters;
- else
- 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 );
+ }
+
+ }
}
- info->minInputChannels = value;
- err = snd_pcm_hw_params_get_channels_max(params, &value);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message_, "RtApiAlsa: hardware maximum in channel probe error (%s): %s.",
- info->name.c_str(), snd_strerror(err));
- error(RtError::WARNING);
- if (info->maxOutputChannels > 0)
- goto probe_parameters;
- else
- return;
+ // Don't bother draining input
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
}
- info->maxInputChannels = value;
- snd_pcm_close(handle);
+ 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)
- goto probe_parameters;
+ bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
- info->hasDuplexSupport = true;
- info->maxDuplexChannels = (info->maxOutputChannels > info->maxInputChannels) ?
- info->maxInputChannels : info->maxOutputChannels;
- info->minDuplexChannels = (info->minOutputChannels > info->minInputChannels) ?
- info->minInputChannels : info->minOutputChannels;
+ if (stream_.doConvertBuffer[1]) {
- 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.
+ // Always interleave ASIO input data.
+ for ( i=0, j=0; i<nChannels; i++ ) {
+ if ( handle->bufferInfos[i].isInput == ASIOTrue )
+ memcpy( &stream_.deviceBuffer[j++*bufferBytes],
+ handle->bufferInfos[i].buffers[bufferIndex],
+ bufferBytes );
+ }
- if (info->maxOutputChannels >= info->maxInputChannels)
- stream = SND_PCM_STREAM_PLAYBACK;
- else
- stream = SND_PCM_STREAM_CAPTURE;
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nDeviceChannels[1],
+ stream_.deviceFormat[1] );
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
- err = snd_pcm_open(&handle, info->name.c_str(), stream, open_mode);
- if (err < 0) {
- sprintf(message_, "RtApiAlsa: pcm (%s) won't reopen during probe: %s.",
- info->name.c_str(), snd_strerror(err));
- error(RtError::WARNING);
- return;
- }
+ }
+ 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 );
+ }
+ }
- // 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_, "RtApiAlsa: hardware reopen probe error (%s): %s.",
- info->name.c_str(), snd_strerror(err));
- error(RtError::WARNING);
- return;
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( stream_.userBuffer[1],
+ stream_.bufferSize * stream_.nUserChannels[1],
+ stream_.userFormat );
+ }
}
- // Test our discrete set of sample rate values.
- int dir = 0;
- info->sampleRates.clear();
- for (unsigned int i=0; i<MAX_SAMPLE_RATES; i++) {
- if (snd_pcm_hw_params_test_rate(handle, params, SAMPLE_RATES[i], dir) == 0)
- info->sampleRates.push_back(SAMPLE_RATES[i]);
- }
- if (info->sampleRates.size() == 0) {
- snd_pcm_close(handle);
- sprintf(message_, "RtApiAlsa: no supported sample rates found for device (%s).",
- info->name.c_str());
- error(RtError::DEBUG_WARNING);
- return;
- }
+ 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();
- // 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;
+ RtApi::tickStreamTime();
+ return SUCCESS;
+}
- // Check that we have at least one supported format
- if (info->nativeFormats == 0) {
- snd_pcm_close(handle);
- sprintf(message_, "RtApiAlsa: pcm device (%s) data format not supported by RtAudio.",
- info->name.c_str());
- error(RtError::WARNING);
+static void sampleRateChanged( ASIOSampleRate sRate )
+{
+ // The ASIO documentation says that this usually only happens during
+ // external sync. Audio processing is not stopped by the driver,
+ // actual sample rate might not have even changed, maybe only the
+ // sample rate status of an AES/EBU or S/PDIF digital input at the
+ // audio device.
+
+ RtApi *object = (RtApi *) asioCallbackInfo->object;
+ try {
+ object->stopStream();
+ }
+ catch ( RtAudioError &exception ) {
+ std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
return;
}
- // That's all ... close the device and return
- snd_pcm_close(handle);
- info->probed = true;
- return;
+ std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
}
-bool RtApiAlsa :: probeDeviceOpen( int device, StreamMode mode, int channels,
- int sampleRate, RtAudioFormat format,
- int *bufferSize, int numberOfBuffers )
+static long asioMessages( long selector, long value, void* /*message*/, double* /*opt*/ )
{
-#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.c_str();
+ long ret = 0;
- 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_,"RtApiAlsa: pcm device (%s) won't open: %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+ switch( selector ) {
+ case kAsioSelectorSupported:
+ if ( value == kAsioResetRequest
+ || value == kAsioEngineVersion
+ || value == kAsioResyncRequest
+ || value == kAsioLatenciesChanged
+ // The following three were added for ASIO 2.0, you don't
+ // necessarily have to support them.
+ || value == kAsioSupportsTimeInfo
+ || value == kAsioSupportsTimeCode
+ || value == kAsioSupportsInputMonitor)
+ ret = 1L;
+ break;
+ case kAsioResetRequest:
+ // Defer the task and perform the reset of the driver during the
+ // next "safe" situation. You cannot reset the driver right now,
+ // as this code is called from the driver. Reset the driver is
+ // done by completely destruct is. I.e. ASIOStop(),
+ // ASIODisposeBuffers(), Destruction Afterwards you initialize the
+ // driver again.
+ std::cerr << "\nRtApiAsio: driver reset requested!!!" << std::endl;
+ ret = 1L;
+ break;
+ case kAsioResyncRequest:
+ // This informs the application that the driver encountered some
+ // non-fatal data loss. It is used for synchronization purposes
+ // of different media. Added mainly to work around the Win16Mutex
+ // problems in Windows 95/98 with the Windows Multimedia system,
+ // which could lose data because the Mutex was held too long by
+ // another thread. However a driver can issue it in other
+ // situations, too.
+ // std::cerr << "\nRtApiAsio: driver resync requested!!!" << std::endl;
+ asioXRun = true;
+ ret = 1L;
+ break;
+ case kAsioLatenciesChanged:
+ // This will inform the host application that the drivers were
+ // latencies changed. Beware, it this does not mean that the
+ // buffer sizes have changed! You might need to update internal
+ // delay data.
+ std::cerr << "\nRtApiAsio: driver latency may have changed!!!" << std::endl;
+ ret = 1L;
+ break;
+ case kAsioEngineVersion:
+ // Return the supported ASIO version of the host application. If
+ // a host application does not implement this selector, ASIO 1.0
+ // is assumed by the driver.
+ ret = 2L;
+ break;
+ case kAsioSupportsTimeInfo:
+ // Informs the driver whether the
+ // asioCallbacks.bufferSwitchTimeInfo() callback is supported.
+ // For compatibility with ASIO 1.0 drivers the host application
+ // should always support the "old" bufferSwitch method, too.
+ ret = 0;
+ break;
+ case kAsioSupportsTimeCode:
+ // Informs the driver whether application is interested in time
+ // code info. If an application does not need to know about time
+ // code, the driver has less work to do.
+ ret = 0;
+ break;
}
+ return ret;
+}
- // 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_, "RtApiAlsa: error getting parameter handle (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
+static const char* getAsioErrorString( ASIOError result )
+{
+ struct Messages
+ {
+ ASIOError value;
+ const char*message;
+ };
-#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
+ static const Messages m[] =
+ {
+ { ASE_NotPresent, "Hardware input or output is not present or available." },
+ { ASE_HWMalfunction, "Hardware is malfunctioning." },
+ { ASE_InvalidParameter, "Invalid input parameter." },
+ { ASE_InvalidMode, "Invalid mode." },
+ { ASE_SPNotAdvancing, "Sample position not advancing." },
+ { ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },
+ { ASE_NoMemory, "Not enough memory to complete the request." }
+ };
- // 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;
- }
- else {
- snd_pcm_close(handle);
- sprintf(message_, "RtApiAlsa: device (%s) access not supported by RtAudio.", name);
- error(RtError::WARNING);
- return FAILURE;
+ for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
+ if ( m[i].value == result ) return m[i].message;
+
+ return "Unknown error.";
+}
+
+//******************** End of __WINDOWS_ASIO__ *********************//
+#endif
+
+
+#if defined(__WINDOWS_WASAPI__) // Windows WASAPI API
+
+// Authored by Marcus Tomlinson <themarcustomlinson@gmail.com>, April 2014
+// - Introduces support for the Windows WASAPI API
+// - Aims to deliver bit streams to and from hardware at the lowest possible latency, via the absolute minimum buffer sizes required
+// - Provides flexible stream configuration to an otherwise strict and inflexible WASAPI interface
+// - Includes automatic internal conversion of sample rate and buffer size between hardware and the user
+
+#ifndef INITGUID
+ #define INITGUID
+#endif
+#include <audioclient.h>
+#include <avrt.h>
+#include <mmdeviceapi.h>
+#include <functiondiscoverykeys_devpkey.h>
+#include <sstream>
+
+//=============================================================================
+
+#define SAFE_RELEASE( objectPtr )\
+if ( objectPtr )\
+{\
+ objectPtr->Release();\
+ objectPtr = NULL;\
+}
+
+typedef HANDLE ( __stdcall *TAvSetMmThreadCharacteristicsPtr )( LPCWSTR TaskName, LPDWORD TaskIndex );
+
+//-----------------------------------------------------------------------------
+
+// WASAPI dictates stream sample rate, format, channel count, and in some cases, buffer size.
+// Therefore we must perform all necessary conversions to user buffers in order to satisfy these
+// requirements. WasapiBuffer ring buffers are used between HwIn->UserIn and UserOut->HwOut to
+// provide intermediate storage for read / write synchronization.
+class WasapiBuffer
+{
+public:
+ WasapiBuffer()
+ : buffer_( NULL ),
+ bufferSize_( 0 ),
+ inIndex_( 0 ),
+ outIndex_( 0 ) {}
+
+ ~WasapiBuffer() {
+ free( buffer_ );
}
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message_, "RtApiAlsa: error setting access ( (%s): %s.", name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+ // sets the length of the internal ring buffer
+ void setBufferSize( unsigned int bufferSize, unsigned int formatBytes ) {
+ free( buffer_ );
+
+ buffer_ = ( char* ) calloc( bufferSize, formatBytes );
+
+ bufferSize_ = bufferSize;
+ inIndex_ = 0;
+ outIndex_ = 0;
}
- // Determine how to set the device format.
- stream_.userFormat = format;
- snd_pcm_format_t device_format = SND_PCM_FORMAT_UNKNOWN;
-
- 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;
+ // attempt to push a buffer into the ring buffer at the current "in" index
+ bool pushBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
+ {
+ if ( !buffer || // incoming buffer is NULL
+ bufferSize == 0 || // incoming buffer has no data
+ bufferSize > bufferSize_ ) // incoming buffer too large
+ {
+ return false;
+ }
+
+ unsigned int relOutIndex = outIndex_;
+ unsigned int inIndexEnd = inIndex_ + bufferSize;
+ if ( relOutIndex < inIndex_ && inIndexEnd >= bufferSize_ ) {
+ relOutIndex += bufferSize_;
+ }
+
+ // "in" index can end on the "out" index but cannot begin at it
+ if ( inIndex_ <= relOutIndex && inIndexEnd > relOutIndex ) {
+ return false; // not enough space between "in" index and "out" index
+ }
+
+ // copy buffer from external to internal
+ int fromZeroSize = inIndex_ + bufferSize - bufferSize_;
+ fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
+ int fromInSize = bufferSize - fromZeroSize;
+
+ switch( format )
+ {
+ case RTAUDIO_SINT8:
+ memcpy( &( ( char* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( char ) );
+ memcpy( buffer_, &( ( char* ) buffer )[fromInSize], fromZeroSize * sizeof( char ) );
+ break;
+ case RTAUDIO_SINT16:
+ memcpy( &( ( short* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( short ) );
+ memcpy( buffer_, &( ( short* ) buffer )[fromInSize], fromZeroSize * sizeof( short ) );
+ break;
+ case RTAUDIO_SINT24:
+ memcpy( &( ( S24* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( S24 ) );
+ memcpy( buffer_, &( ( S24* ) buffer )[fromInSize], fromZeroSize * sizeof( S24 ) );
+ break;
+ case RTAUDIO_SINT32:
+ memcpy( &( ( int* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( int ) );
+ memcpy( buffer_, &( ( int* ) buffer )[fromInSize], fromZeroSize * sizeof( int ) );
+ break;
+ case RTAUDIO_FLOAT32:
+ memcpy( &( ( float* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( float ) );
+ memcpy( buffer_, &( ( float* ) buffer )[fromInSize], fromZeroSize * sizeof( float ) );
+ break;
+ case RTAUDIO_FLOAT64:
+ memcpy( &( ( double* ) buffer_ )[inIndex_], buffer, fromInSize * sizeof( double ) );
+ memcpy( buffer_, &( ( double* ) buffer )[fromInSize], fromZeroSize * sizeof( double ) );
+ break;
+ }
+
+ // update "in" index
+ inIndex_ += bufferSize;
+ inIndex_ %= bufferSize_;
+
+ return true;
}
- // 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;
+ // attempt to pull a buffer from the ring buffer from the current "out" index
+ bool pullBuffer( char* buffer, unsigned int bufferSize, RtAudioFormat format )
+ {
+ if ( !buffer || // incoming buffer is NULL
+ bufferSize == 0 || // incoming buffer has no data
+ bufferSize > bufferSize_ ) // incoming buffer too large
+ {
+ return false;
+ }
+
+ unsigned int relInIndex = inIndex_;
+ unsigned int outIndexEnd = outIndex_ + bufferSize;
+ if ( relInIndex < outIndex_ && outIndexEnd >= bufferSize_ ) {
+ relInIndex += bufferSize_;
+ }
+
+ // "out" index can begin at and end on the "in" index
+ if ( outIndex_ < relInIndex && outIndexEnd > relInIndex ) {
+ return false; // not enough space between "out" index and "in" index
+ }
+
+ // copy buffer from internal to external
+ int fromZeroSize = outIndex_ + bufferSize - bufferSize_;
+ fromZeroSize = fromZeroSize < 0 ? 0 : fromZeroSize;
+ int fromOutSize = bufferSize - fromZeroSize;
+
+ switch( format )
+ {
+ case RTAUDIO_SINT8:
+ memcpy( buffer, &( ( char* ) buffer_ )[outIndex_], fromOutSize * sizeof( char ) );
+ memcpy( &( ( char* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( char ) );
+ break;
+ case RTAUDIO_SINT16:
+ memcpy( buffer, &( ( short* ) buffer_ )[outIndex_], fromOutSize * sizeof( short ) );
+ memcpy( &( ( short* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( short ) );
+ break;
+ case RTAUDIO_SINT24:
+ memcpy( buffer, &( ( S24* ) buffer_ )[outIndex_], fromOutSize * sizeof( S24 ) );
+ memcpy( &( ( S24* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( S24 ) );
+ break;
+ case RTAUDIO_SINT32:
+ memcpy( buffer, &( ( int* ) buffer_ )[outIndex_], fromOutSize * sizeof( int ) );
+ memcpy( &( ( int* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( int ) );
+ break;
+ case RTAUDIO_FLOAT32:
+ memcpy( buffer, &( ( float* ) buffer_ )[outIndex_], fromOutSize * sizeof( float ) );
+ memcpy( &( ( float* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( float ) );
+ break;
+ case RTAUDIO_FLOAT64:
+ memcpy( buffer, &( ( double* ) buffer_ )[outIndex_], fromOutSize * sizeof( double ) );
+ memcpy( &( ( double* ) buffer )[fromOutSize], buffer_, fromZeroSize * sizeof( double ) );
+ break;
+ }
+
+ // update "out" index
+ outIndex_ += bufferSize;
+ outIndex_ %= bufferSize_;
+
+ return true;
}
- 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;
+private:
+ char* buffer_;
+ unsigned int bufferSize_;
+ unsigned int inIndex_;
+ unsigned int outIndex_;
+};
+
+//-----------------------------------------------------------------------------
+
+// A structure to hold various information related to the WASAPI implementation.
+struct WasapiHandle
+{
+ IAudioClient* captureAudioClient;
+ IAudioClient* renderAudioClient;
+ IAudioCaptureClient* captureClient;
+ IAudioRenderClient* renderClient;
+ HANDLE captureEvent;
+ HANDLE renderEvent;
+
+ WasapiHandle()
+ : captureAudioClient( NULL ),
+ renderAudioClient( NULL ),
+ captureClient( NULL ),
+ renderClient( NULL ),
+ captureEvent( NULL ),
+ renderEvent( NULL ) {}
+};
+
+//=============================================================================
+
+RtApiWasapi::RtApiWasapi()
+ : coInitialized_( false ), deviceEnumerator_( NULL )
+{
+ // WASAPI can run either apartment or multi-threaded
+ HRESULT hr = CoInitialize( NULL );
+ if ( !FAILED( hr ) )
+ coInitialized_ = true;
+
+ // Instantiate device enumerator
+ hr = CoCreateInstance( __uuidof( MMDeviceEnumerator ), NULL,
+ CLSCTX_ALL, __uuidof( IMMDeviceEnumerator ),
+ ( void** ) &deviceEnumerator_ );
+
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::RtApiWasapi: Unable to instantiate device enumerator";
+ error( RtAudioError::DRIVER_ERROR );
}
+}
- 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;
+//-----------------------------------------------------------------------------
+
+RtApiWasapi::~RtApiWasapi()
+{
+ if ( stream_.state != STREAM_CLOSED )
+ closeStream();
+
+ SAFE_RELEASE( deviceEnumerator_ );
+
+ // If this object previously called CoInitialize()
+ if ( coInitialized_ )
+ CoUninitialize();
+}
+
+//=============================================================================
+
+unsigned int RtApiWasapi::getDeviceCount( void )
+{
+ unsigned int captureDeviceCount = 0;
+ unsigned int renderDeviceCount = 0;
+
+ IMMDeviceCollection* captureDevices = NULL;
+ IMMDeviceCollection* renderDevices = NULL;
+
+ // Count capture devices
+ errorText_.clear();
+ HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device collection.";
+ goto Exit;
}
- 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;
+ hr = captureDevices->GetCount( &captureDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve capture device count.";
+ goto Exit;
}
- 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;
+ // Count render devices
+ hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device collection.";
+ goto Exit;
}
- 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;
+ hr = renderDevices->GetCount( &renderDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceCount: Unable to retrieve render device count.";
+ goto Exit;
}
- // If we get here, no supported format was found.
- sprintf(message_,"RtApiAlsa: pcm device (%s) data format not supported by RtAudio.", name);
- snd_pcm_close(handle);
- error(RtError::WARNING);
- return FAILURE;
+Exit:
+ // release all references
+ SAFE_RELEASE( captureDevices );
+ SAFE_RELEASE( renderDevices );
- set_format:
- err = snd_pcm_hw_params_set_format(handle, hw_params, device_format);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message_, "RtApiAlsa: error setting format (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
+ if ( errorText_.empty() )
+ return captureDeviceCount + renderDeviceCount;
- // 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_, "RtApiAlsa: error getting format endian-ness (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
- }
+ error( RtAudioError::DRIVER_ERROR );
+ return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+RtAudio::DeviceInfo RtApiWasapi::getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ unsigned int captureDeviceCount = 0;
+ unsigned int renderDeviceCount = 0;
+ std::string defaultDeviceName;
+ bool isCaptureDevice = false;
+
+ PROPVARIANT deviceNameProp;
+ PROPVARIANT defaultDeviceNameProp;
+
+ IMMDeviceCollection* captureDevices = NULL;
+ IMMDeviceCollection* renderDevices = NULL;
+ IMMDevice* devicePtr = NULL;
+ IMMDevice* defaultDevicePtr = NULL;
+ IAudioClient* audioClient = NULL;
+ IPropertyStore* devicePropStore = NULL;
+ IPropertyStore* defaultDevicePropStore = NULL;
+
+ WAVEFORMATEX* deviceFormat = NULL;
+ WAVEFORMATEX* closestMatchFormat = NULL;
+
+ // probed
+ info.probed = false;
+
+ // Count capture devices
+ errorText_.clear();
+ RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
+ HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device collection.";
+ goto Exit;
}
- // 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_, "RtApiAlsa: error setting sample rate (%d) on device (%s): %s.",
- sampleRate, name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+ hr = captureDevices->GetCount( &captureDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device count.";
+ goto Exit;
}
- // 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;
- err = snd_pcm_hw_params_get_channels_max(hw_params, &value);
- int device_channels = value;
- if (err < 0 || device_channels < channels) {
- snd_pcm_close(handle);
- sprintf(message_, "RtApiAlsa: channels (%d) not supported by device (%s).",
- channels, name);
- error(RtError::WARNING);
- return FAILURE;
+ // Count render devices
+ hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device collection.";
+ goto Exit;
}
- err = snd_pcm_hw_params_get_channels_min(hw_params, &value);
- if (err < 0 ) {
- snd_pcm_close(handle);
- sprintf(message_, "RtApiAlsa: error getting min channels count on device (%s).", name);
- error(RtError::WARNING);
- return FAILURE;
+ hr = renderDevices->GetCount( &renderDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device count.";
+ goto Exit;
}
- device_channels = value;
- 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_, "RtApiAlsa: error setting channels (%d) on device (%s): %s.",
- device_channels, name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+ // validate device index
+ if ( device >= captureDeviceCount + renderDeviceCount ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Invalid device index.";
+ errorType = RtAudioError::INVALID_USE;
+ goto Exit;
}
- // Set the buffer number, which in ALSA is referred to as the "period".
- int dir;
- unsigned 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, &value, &dir);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message_, "RtApiAlsa: error getting min periods on device (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+ // determine whether index falls within capture or render devices
+ if ( device >= renderDeviceCount ) {
+ hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve capture device handle.";
+ goto Exit;
+ }
+ isCaptureDevice = true;
}
- if (value > periods) periods = value;
- err = snd_pcm_hw_params_get_periods_max(hw_params, &value, &dir);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message_, "RtApiAlsa: error getting max periods on device (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+ else {
+ hr = renderDevices->Item( device, &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve render device handle.";
+ goto Exit;
+ }
+ isCaptureDevice = false;
}
- if (value < periods) periods = value;
- err = snd_pcm_hw_params_set_periods(handle, hw_params, periods, 0);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message_, "RtApiAlsa: error setting periods (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+ // get default device name
+ if ( isCaptureDevice ) {
+ hr = deviceEnumerator_->GetDefaultAudioEndpoint( eCapture, eConsole, &defaultDevicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default capture device handle.";
+ goto Exit;
+ }
+ }
+ else {
+ hr = deviceEnumerator_->GetDefaultAudioEndpoint( eRender, eConsole, &defaultDevicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default render device handle.";
+ goto Exit;
+ }
}
- // Set the buffer (or period) size.
- snd_pcm_uframes_t period_size;
- err = snd_pcm_hw_params_get_period_size_min(hw_params, &period_size, &dir);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message_, "RtApiAlsa: error getting period size (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+ hr = defaultDevicePtr->OpenPropertyStore( STGM_READ, &defaultDevicePropStore );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open default device property store.";
+ goto Exit;
}
- if (*bufferSize < (int) period_size) *bufferSize = (int) period_size;
+ PropVariantInit( &defaultDeviceNameProp );
- err = snd_pcm_hw_params_set_period_size(handle, hw_params, *bufferSize, 0);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message_, "RtApiAlsa: error setting period size (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+ hr = defaultDevicePropStore->GetValue( PKEY_Device_FriendlyName, &defaultDeviceNameProp );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve default device property: PKEY_Device_FriendlyName.";
+ goto Exit;
}
- // 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_, "RtApiAlsa: error setting buffer size for duplex stream on device (%s).",
- name );
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ defaultDeviceName = convertCharPointerToStdString(defaultDeviceNameProp.pwszVal);
+
+ // name
+ hr = devicePtr->OpenPropertyStore( STGM_READ, &devicePropStore );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to open device property store.";
+ goto Exit;
}
- stream_.bufferSize = *bufferSize;
+ PropVariantInit( &deviceNameProp );
- // Install the hardware configuration
- err = snd_pcm_hw_params(handle, hw_params);
- if (err < 0) {
- snd_pcm_close(handle);
- sprintf(message_, "RtApiAlsa: error installing hardware configuration (%s): %s.",
- name, snd_strerror(err));
- error(RtError::WARNING);
- return FAILURE;
+ hr = devicePropStore->GetValue( PKEY_Device_FriendlyName, &deviceNameProp );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device property: PKEY_Device_FriendlyName.";
+ goto Exit;
}
-#if defined(__RTAUDIO_DEBUG__)
- fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
- snd_pcm_hw_params_dump(hw_params, out);
-#endif
+ info.name =convertCharPointerToStdString(deviceNameProp.pwszVal);
- // Allocate the stream handle if necessary and then save.
- snd_pcm_t **handles;
- if ( stream_.apiHandle == 0 ) {
- handles = (snd_pcm_t **) calloc(2, sizeof(snd_pcm_t *));
- if ( handle == NULL ) {
- sprintf(message_, "RtApiAlsa: error allocating handle memory (%s).",
- devices_[device].name.c_str());
- goto error;
- }
- stream_.apiHandle = (void *) handles;
- handles[0] = 0;
- handles[1] = 0;
+ // is default
+ if ( isCaptureDevice ) {
+ info.isDefaultInput = info.name == defaultDeviceName;
+ info.isDefaultOutput = false;
}
else {
- handles = (snd_pcm_t **) stream_.apiHandle;
+ info.isDefaultInput = false;
+ info.isDefaultOutput = info.name == defaultDeviceName;
}
- handles[mode] = handle;
- // 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;
+ // channel count
+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL, NULL, ( void** ) &audioClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device audio client.";
+ goto Exit;
+ }
- // Allocate necessary internal buffers
- if ( stream_.nUserChannels[0] != stream_.nUserChannels[1] ) {
+ hr = audioClient->GetMixFormat( &deviceFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::getDeviceInfo: Unable to retrieve device mix format.";
+ goto Exit;
+ }
- 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) {
- sprintf(message_, "RtApiAlsa: error allocating user buffer memory (%s).",
- devices_[device].name.c_str());
- goto error;
- }
+ if ( isCaptureDevice ) {
+ info.inputChannels = deviceFormat->nChannels;
+ info.outputChannels = 0;
+ info.duplexChannels = 0;
+ }
+ else {
+ info.inputChannels = 0;
+ info.outputChannels = deviceFormat->nChannels;
+ info.duplexChannels = 0;
}
- if ( stream_.doConvertBuffer[mode] ) {
+ // sample rates (WASAPI only supports the one native sample rate)
+ info.preferredSampleRate = deviceFormat->nSamplesPerSec;
- 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;
- }
- }
+ info.sampleRates.clear();
+ info.sampleRates.push_back( deviceFormat->nSamplesPerSec );
- if ( makeBuffer ) {
- buffer_bytes *= *bufferSize;
- if (stream_.deviceBuffer) free(stream_.deviceBuffer);
- stream_.deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream_.deviceBuffer == NULL) {
- sprintf(message_, "RtApiAlsa: error allocating device buffer memory (%s).",
- devices_[device].name.c_str());
- goto error;
- }
+ // native format
+ info.nativeFormats = 0;
+
+ if ( deviceFormat->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
+ ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
+ ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT ) )
+ {
+ if ( deviceFormat->wBitsPerSample == 32 ) {
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+ }
+ else if ( deviceFormat->wBitsPerSample == 64 ) {
+ info.nativeFormats |= RTAUDIO_FLOAT64;
+ }
+ }
+ else if ( deviceFormat->wFormatTag == WAVE_FORMAT_PCM ||
+ ( deviceFormat->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
+ ( ( WAVEFORMATEXTENSIBLE* ) deviceFormat )->SubFormat == KSDATAFORMAT_SUBTYPE_PCM ) )
+ {
+ if ( deviceFormat->wBitsPerSample == 8 ) {
+ info.nativeFormats |= RTAUDIO_SINT8;
+ }
+ else if ( deviceFormat->wBitsPerSample == 16 ) {
+ info.nativeFormats |= RTAUDIO_SINT16;
+ }
+ else if ( deviceFormat->wBitsPerSample == 24 ) {
+ info.nativeFormats |= RTAUDIO_SINT24;
+ }
+ else if ( deviceFormat->wBitsPerSample == 32 ) {
+ info.nativeFormats |= RTAUDIO_SINT32;
}
}
- 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;
+ // probed
+ info.probed = true;
- return SUCCESS;
+Exit:
+ // release all references
+ PropVariantClear( &deviceNameProp );
+ PropVariantClear( &defaultDeviceNameProp );
- error:
- if (handles) {
- if (handles[0])
- snd_pcm_close(handles[0]);
- if (handles[1])
- snd_pcm_close(handles[1]);
- free(handles);
- stream_.apiHandle = 0;
- }
+ SAFE_RELEASE( captureDevices );
+ SAFE_RELEASE( renderDevices );
+ SAFE_RELEASE( devicePtr );
+ SAFE_RELEASE( defaultDevicePtr );
+ SAFE_RELEASE( audioClient );
+ SAFE_RELEASE( devicePropStore );
+ SAFE_RELEASE( defaultDevicePropStore );
- if (stream_.userBuffer) {
- free(stream_.userBuffer);
- stream_.userBuffer = 0;
- }
+ CoTaskMemFree( deviceFormat );
+ CoTaskMemFree( closestMatchFormat );
- error(RtError::WARNING);
- return FAILURE;
+ if ( !errorText_.empty() )
+ error( errorType );
+ return info;
}
-void RtApiAlsa :: closeStream()
+//-----------------------------------------------------------------------------
+
+unsigned int RtApiWasapi::getDefaultOutputDevice( void )
{
- // We don't want an exception to be thrown here because this
- // function is called by our class destructor. So, do our own
- // stream check.
- if ( stream_.mode == UNINITIALIZED ) {
- sprintf(message_, "RtApiAlsa::closeStream(): no open stream to close!");
- error(RtError::WARNING);
- return;
+ for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
+ if ( getDeviceInfo( i ).isDefaultOutput ) {
+ return i;
+ }
}
- snd_pcm_t **handle = (snd_pcm_t **) stream_.apiHandle;
- if (stream_.state == STREAM_RUNNING) {
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX)
- snd_pcm_drop(handle[0]);
- if (stream_.mode == INPUT || stream_.mode == DUPLEX)
- snd_pcm_drop(handle[1]);
- stream_.state = STREAM_STOPPED;
- }
+ return 0;
+}
- if (stream_.callbackInfo.usingCallback) {
- stream_.callbackInfo.usingCallback = false;
- pthread_join(stream_.callbackInfo.thread, NULL);
- }
+//-----------------------------------------------------------------------------
- if (handle) {
- if (handle[0]) snd_pcm_close(handle[0]);
- if (handle[1]) snd_pcm_close(handle[1]);
- free(handle);
- handle = 0;
+unsigned int RtApiWasapi::getDefaultInputDevice( void )
+{
+ for ( unsigned int i = 0; i < getDeviceCount(); i++ ) {
+ if ( getDeviceInfo( i ).isDefaultInput ) {
+ return i;
+ }
}
- if (stream_.userBuffer) {
- free(stream_.userBuffer);
- stream_.userBuffer = 0;
- }
+ return 0;
+}
- if (stream_.deviceBuffer) {
- free(stream_.deviceBuffer);
- stream_.deviceBuffer = 0;
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::closeStream( void )
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiWasapi::closeStream: No open stream to close.";
+ error( RtAudioError::WARNING );
+ return;
}
- stream_.mode = UNINITIALIZED;
-}
+ if ( stream_.state != STREAM_STOPPED )
+ stopStream();
-void RtApiAlsa :: startStream()
-{
- // This method calls snd_pcm_prepare if the device isn't already in that state.
+ // clean up stream memory
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient )
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient )
- verifyStream();
- if (stream_.state == STREAM_RUNNING) return;
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->captureClient )
+ SAFE_RELEASE( ( ( WasapiHandle* ) stream_.apiHandle )->renderClient )
- MUTEX_LOCK(&stream_.mutex);
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent )
+ CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent );
- int err;
- snd_pcm_state_t state;
- snd_pcm_t **handle = (snd_pcm_t **) stream_.apiHandle;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
- state = snd_pcm_state(handle[0]);
- if (state != SND_PCM_STATE_PREPARED) {
- err = snd_pcm_prepare(handle[0]);
- if (err < 0) {
- sprintf(message_, "RtApiAlsa: error preparing pcm device (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), snd_strerror(err));
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
- }
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent )
+ CloseHandle( ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent );
+
+ delete ( WasapiHandle* ) stream_.apiHandle;
+ stream_.apiHandle = NULL;
+
+ for ( int i = 0; i < 2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
}
}
- if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
- state = snd_pcm_state(handle[1]);
- if (state != SND_PCM_STATE_PREPARED) {
- err = snd_pcm_prepare(handle[1]);
- if (err < 0) {
- sprintf(message_, "RtApiAlsa: error preparing pcm device (%s): %s.",
- devices_[stream_.device[1]].name.c_str(), snd_strerror(err));
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
}
- stream_.state = STREAM_RUNNING;
- MUTEX_UNLOCK(&stream_.mutex);
+ // update stream state
+ stream_.state = STREAM_CLOSED;
}
-void RtApiAlsa :: stopStream()
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::startStream( void )
{
verifyStream();
- if (stream_.state == STREAM_STOPPED) return;
-
- // Change the state before the lock to improve shutdown response
- // when using a callback.
- stream_.state = STREAM_STOPPED;
- MUTEX_LOCK(&stream_.mutex);
- int err;
- snd_pcm_t **handle = (snd_pcm_t **) stream_.apiHandle;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
- err = snd_pcm_drain(handle[0]);
- if (err < 0) {
- sprintf(message_, "RtApiAlsa: error draining pcm device (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), snd_strerror(err));
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
- }
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiWasapi::startStream: The stream is already running.";
+ error( RtAudioError::WARNING );
+ return;
}
- if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
- err = snd_pcm_drain(handle[1]);
- if (err < 0) {
- sprintf(message_, "RtApiAlsa: error draining pcm device (%s): %s.",
- devices_[stream_.device[1]].name.c_str(), snd_strerror(err));
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
+ // update stream state
+ stream_.state = STREAM_RUNNING;
+
+ // create WASAPI stream thread
+ stream_.callbackInfo.thread = ( ThreadHandle ) CreateThread( NULL, 0, runWasapiThread, this, CREATE_SUSPENDED, NULL );
- MUTEX_UNLOCK(&stream_.mutex);
+ if ( !stream_.callbackInfo.thread ) {
+ errorText_ = "RtApiWasapi::startStream: Unable to instantiate callback thread.";
+ error( RtAudioError::THREAD_ERROR );
+ }
+ else {
+ SetThreadPriority( ( void* ) stream_.callbackInfo.thread, stream_.callbackInfo.priority );
+ ResumeThread( ( void* ) stream_.callbackInfo.thread );
+ }
}
-void RtApiAlsa :: abortStream()
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::stopStream( void )
{
verifyStream();
- if (stream_.state == STREAM_STOPPED) return;
-
- // Change the state before the lock to improve shutdown response
- // when using a callback.
- stream_.state = STREAM_STOPPED;
- MUTEX_LOCK(&stream_.mutex);
-
- int err;
- snd_pcm_t **handle = (snd_pcm_t **) stream_.apiHandle;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
- err = snd_pcm_drop(handle[0]);
- if (err < 0) {
- sprintf(message_, "RtApiAlsa: error draining pcm device (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), snd_strerror(err));
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
- if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
- err = snd_pcm_drop(handle[1]);
- if (err < 0) {
- sprintf(message_, "RtApiAlsa: error draining pcm device (%s): %s.",
- devices_[stream_.device[1]].name.c_str(), snd_strerror(err));
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
- }
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiWasapi::stopStream: The stream is already stopped.";
+ error( RtAudioError::WARNING );
+ return;
}
- MUTEX_UNLOCK(&stream_.mutex);
-}
+ // inform stream thread by setting stream state to STREAM_STOPPING
+ stream_.state = STREAM_STOPPING;
-int RtApiAlsa :: streamWillBlock()
-{
- verifyStream();
- if (stream_.state == STREAM_STOPPED) return 0;
+ // wait until stream thread is stopped
+ while( stream_.state != STREAM_STOPPED ) {
+ Sleep( 1 );
+ }
- MUTEX_LOCK(&stream_.mutex);
+ // Wait for the last buffer to play before stopping.
+ Sleep( 1000 * stream_.bufferSize / stream_.sampleRate );
- int err = 0, frames = 0;
- snd_pcm_t **handle = (snd_pcm_t **) stream_.apiHandle;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
- err = snd_pcm_avail_update(handle[0]);
- if (err < 0) {
- sprintf(message_, "RtApiAlsa: error getting available frames for device (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), snd_strerror(err));
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
+ // stop capture client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::stopStream: Unable to stop capture stream.";
+ error( RtAudioError::DRIVER_ERROR );
+ return;
}
}
- frames = err;
-
- if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
- err = snd_pcm_avail_update(handle[1]);
- if (err < 0) {
- sprintf(message_, "RtApiAlsa: error getting available frames for device (%s): %s.",
- devices_[stream_.device[1]].name.c_str(), snd_strerror(err));
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
+ // stop render client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::stopStream: Unable to stop render stream.";
+ error( RtAudioError::DRIVER_ERROR );
+ return;
}
- if (frames > err) frames = err;
}
- frames = stream_.bufferSize - frames;
- if (frames < 0) frames = 0;
+ // close thread handle
+ if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
+ errorText_ = "RtApiWasapi::stopStream: Unable to close callback thread.";
+ error( RtAudioError::THREAD_ERROR );
+ return;
+ }
- MUTEX_UNLOCK(&stream_.mutex);
- return frames;
+ stream_.callbackInfo.thread = (ThreadHandle) NULL;
}
-void RtApiAlsa :: tickStream()
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::abortStream( void )
{
verifyStream();
- int stopStream = 0;
- if (stream_.state == STREAM_STOPPED) {
- if (stream_.callbackInfo.usingCallback) usleep(50000); // sleep 50 milliseconds
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiWasapi::abortStream: The stream is already stopped.";
+ error( RtAudioError::WARNING );
return;
}
- else if (stream_.callbackInfo.usingCallback) {
- RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
- stopStream = callback(stream_.userBuffer, stream_.bufferSize, stream_.callbackInfo.userData);
- }
-
- MUTEX_LOCK(&stream_.mutex);
- // The state might change while waiting on a mutex.
- if (stream_.state == STREAM_STOPPED)
- goto unlock;
+ // inform stream thread by setting stream state to STREAM_STOPPING
+ stream_.state = STREAM_STOPPING;
- int err;
- char *buffer;
- int channels;
- snd_pcm_t **handle;
- RtAudioFormat format;
- handle = (snd_pcm_t **) stream_.apiHandle;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
+ // wait until stream thread is stopped
+ while ( stream_.state != STREAM_STOPPED ) {
+ Sleep( 1 );
+ }
- // Setup parameters and do buffer conversion if necessary.
- if (stream_.doConvertBuffer[0]) {
- convertStreamBuffer(OUTPUT);
- buffer = stream_.deviceBuffer;
- channels = stream_.nDeviceChannels[0];
- format = stream_.deviceFormat[0];
- }
- else {
- buffer = stream_.userBuffer;
- channels = stream_.nUserChannels[0];
- format = stream_.userFormat;
+ // stop capture client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::abortStream: Unable to stop capture stream.";
+ error( RtAudioError::DRIVER_ERROR );
+ return;
}
+ }
- // Do byte swapping if necessary.
- if (stream_.doByteSwap[0])
- byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
-
- // Write samples to device in interleaved/non-interleaved format.
- if (stream_.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(handle[0], bufs, stream_.bufferSize);
+ // stop render client if applicable
+ if ( ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient ) {
+ HRESULT hr = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient->Stop();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::abortStream: Unable to stop render stream.";
+ error( RtAudioError::DRIVER_ERROR );
+ return;
}
- else
- err = snd_pcm_writei(handle[0], buffer, stream_.bufferSize);
+ }
- if (err < stream_.bufferSize) {
- // Either an error or underrun occured.
- if (err == -EPIPE) {
- snd_pcm_state_t state = snd_pcm_state(handle[0]);
- if (state == SND_PCM_STATE_XRUN) {
- sprintf(message_, "RtApiAlsa: underrun detected.");
- error(RtError::WARNING);
- err = snd_pcm_prepare(handle[0]);
- if (err < 0) {
- sprintf(message_, "RtApiAlsa: error preparing handle after underrun: %s.",
- snd_strerror(err));
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
- else {
- sprintf(message_, "RtApiAlsa: tickStream() error, current state is %s.",
- snd_pcm_state_name(state));
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
- }
- goto unlock;
- }
- else {
- sprintf(message_, "RtApiAlsa: audio write error for device (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), snd_strerror(err));
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
+ // close thread handle
+ if ( stream_.callbackInfo.thread && !CloseHandle( ( void* ) stream_.callbackInfo.thread ) ) {
+ errorText_ = "RtApiWasapi::abortStream: Unable to close callback thread.";
+ error( RtAudioError::THREAD_ERROR );
+ return;
}
- if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
+ stream_.callbackInfo.thread = (ThreadHandle) NULL;
+}
- // Setup parameters.
- if (stream_.doConvertBuffer[1]) {
- buffer = stream_.deviceBuffer;
- channels = stream_.nDeviceChannels[1];
- format = stream_.deviceFormat[1];
- }
- else {
- buffer = stream_.userBuffer;
- channels = stream_.nUserChannels[1];
- format = stream_.userFormat;
- }
+//-----------------------------------------------------------------------------
- // 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(handle[1], bufs, stream_.bufferSize);
- }
- else
- err = snd_pcm_readi(handle[1], buffer, stream_.bufferSize);
+bool RtApiWasapi::probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int* bufferSize,
+ RtAudio::StreamOptions* options )
+{
+ bool methodResult = FAILURE;
+ unsigned int captureDeviceCount = 0;
+ unsigned int renderDeviceCount = 0;
- if (err < stream_.bufferSize) {
- // Either an error or underrun occured.
- if (err == -EPIPE) {
- snd_pcm_state_t state = snd_pcm_state(handle[1]);
- if (state == SND_PCM_STATE_XRUN) {
- sprintf(message_, "RtApiAlsa: overrun detected.");
- error(RtError::WARNING);
- err = snd_pcm_prepare(handle[1]);
- if (err < 0) {
- sprintf(message_, "RtApiAlsa: error preparing handle after overrun: %s.",
- snd_strerror(err));
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
- else {
- sprintf(message_, "RtApiAlsa: tickStream() error, current state is %s.",
- snd_pcm_state_name(state));
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
- }
- goto unlock;
- }
- else {
- sprintf(message_, "RtApiAlsa: audio read error for device (%s): %s.",
- devices_[stream_.device[1]].name.c_str(), snd_strerror(err));
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
- }
- }
+ IMMDeviceCollection* captureDevices = NULL;
+ IMMDeviceCollection* renderDevices = NULL;
+ IMMDevice* devicePtr = NULL;
+ WAVEFORMATEX* deviceFormat = NULL;
+ unsigned int bufferBytes;
+ stream_.state = STREAM_STOPPED;
+ RtAudio::DeviceInfo deviceInfo;
- // Do byte swapping if necessary.
- if (stream_.doByteSwap[1])
- byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
+ // create API Handle if not already created
+ if ( !stream_.apiHandle )
+ stream_.apiHandle = ( void* ) new WasapiHandle();
- // Do buffer conversion if necessary.
- if (stream_.doConvertBuffer[1])
- convertStreamBuffer(INPUT);
+ // Count capture devices
+ errorText_.clear();
+ RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
+ HRESULT hr = deviceEnumerator_->EnumAudioEndpoints( eCapture, DEVICE_STATE_ACTIVE, &captureDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device collection.";
+ goto Exit;
}
- unlock:
- MUTEX_UNLOCK(&stream_.mutex);
-
- if (stream_.callbackInfo.usingCallback && stopStream)
- this->stopStream();
-}
+ hr = captureDevices->GetCount( &captureDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device count.";
+ goto Exit;
+ }
-void RtApiAlsa :: setStreamCallback(RtAudioCallback callback, void *userData)
-{
- verifyStream();
+ // Count render devices
+ hr = deviceEnumerator_->EnumAudioEndpoints( eRender, DEVICE_STATE_ACTIVE, &renderDevices );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device collection.";
+ goto Exit;
+ }
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
- if ( info->usingCallback ) {
- sprintf(message_, "RtApiAlsa: A callback is already set for this stream!");
- error(RtError::WARNING);
- return;
+ hr = renderDevices->GetCount( &renderDeviceCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device count.";
+ goto Exit;
}
- info->callback = (void *) callback;
- info->userData = userData;
- info->usingCallback = true;
- info->object = (void *) this;
+ // validate device index
+ if ( device >= captureDeviceCount + renderDeviceCount ) {
+ errorType = RtAudioError::INVALID_USE;
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Invalid device index.";
+ goto Exit;
+ }
- // 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);
- pthread_attr_setschedpolicy(&attr, SCHED_RR);
+ deviceInfo = getDeviceInfo( device );
- int err = pthread_create(&info->thread, &attr, alsaCallbackHandler, &stream_.callbackInfo);
- pthread_attr_destroy(&attr);
- if (err) {
- info->usingCallback = false;
- sprintf(message_, "RtApiAlsa: error starting callback thread!");
- error(RtError::THREAD_ERROR);
+ // validate sample rate
+ if ( sampleRate != deviceInfo.preferredSampleRate )
+ {
+ errorType = RtAudioError::INVALID_USE;
+ std::stringstream ss;
+ ss << "RtApiWasapi::probeDeviceOpen: " << sampleRate
+ << "Hz sample rate not supported. This device only supports "
+ << deviceInfo.preferredSampleRate << "Hz.";
+ errorText_ = ss.str();
+ goto Exit;
}
-}
-void RtApiAlsa :: cancelStreamCallback()
-{
- verifyStream();
+ // determine whether index falls within capture or render devices
+ if ( device >= renderDeviceCount ) {
+ if ( mode != INPUT ) {
+ errorType = RtAudioError::INVALID_USE;
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Capture device selected as output device.";
+ goto Exit;
+ }
- if (stream_.callbackInfo.usingCallback) {
+ // retrieve captureAudioClient from devicePtr
+ IAudioClient*& captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
- if (stream_.state == STREAM_RUNNING)
- stopStream();
+ hr = captureDevices->Item( device - renderDeviceCount, &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve capture device handle.";
+ goto Exit;
+ }
- MUTEX_LOCK(&stream_.mutex);
+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+ NULL, ( void** ) &captureAudioClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
+ goto Exit;
+ }
- stream_.callbackInfo.usingCallback = false;
- pthread_join(stream_.callbackInfo.thread, NULL);
- stream_.callbackInfo.thread = 0;
- stream_.callbackInfo.callback = NULL;
- stream_.callbackInfo.userData = NULL;
+ hr = captureAudioClient->GetMixFormat( &deviceFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
+ goto Exit;
+ }
- MUTEX_UNLOCK(&stream_.mutex);
+ stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+ captureAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
}
-}
+ else {
+ if ( mode != OUTPUT ) {
+ errorType = RtAudioError::INVALID_USE;
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Render device selected as input device.";
+ goto Exit;
+ }
-extern "C" void *alsaCallbackHandler(void *ptr)
-{
- CallbackInfo *info = (CallbackInfo *) ptr;
- RtApiAlsa *object = (RtApiAlsa *) info->object;
- bool *usingCallback = &info->usingCallback;
+ // retrieve renderAudioClient from devicePtr
+ IAudioClient*& renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
- while ( *usingCallback ) {
- try {
- object->tickStream();
+ hr = renderDevices->Item( device, &devicePtr );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve render device handle.";
+ goto Exit;
}
- catch (RtError &exception) {
- fprintf(stderr, "\nRtApiAlsa: callback thread error (%s) ... closing thread.\n\n",
- exception.getMessageString());
- break;
+
+ hr = devicePtr->Activate( __uuidof( IAudioClient ), CLSCTX_ALL,
+ NULL, ( void** ) &renderAudioClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device audio client.";
+ goto Exit;
+ }
+
+ hr = renderAudioClient->GetMixFormat( &deviceFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Unable to retrieve device mix format.";
+ goto Exit;
}
+
+ stream_.nDeviceChannels[mode] = deviceFormat->nChannels;
+ renderAudioClient->GetStreamLatency( ( long long* ) &stream_.latency[mode] );
}
- pthread_exit(NULL);
-}
+ // fill stream data
+ if ( ( stream_.mode == OUTPUT && mode == INPUT ) ||
+ ( stream_.mode == INPUT && mode == OUTPUT ) ) {
+ stream_.mode = DUPLEX;
+ }
+ else {
+ stream_.mode = mode;
+ }
-//******************** End of __LINUX_ALSA__ *********************//
-#endif
+ stream_.device[mode] = device;
+ stream_.doByteSwap[mode] = false;
+ stream_.sampleRate = sampleRate;
+ stream_.bufferSize = *bufferSize;
+ stream_.nBuffers = 1;
+ stream_.nUserChannels[mode] = channels;
+ stream_.channelOffset[mode] = firstChannel;
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = deviceInfo.nativeFormats;
-#if defined(__WINDOWS_ASIO__) // ASIO API on Windows
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
+ stream_.userInterleaved = false;
+ else
+ stream_.userInterleaved = true;
+ stream_.deviceInterleaved[mode] = true;
-// 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.
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] ||
+ stream_.nUserChannels != stream_.nDeviceChannels )
+ stream_.doConvertBuffer[mode] = true;
+ else if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
-#include "asio/asiosys.h"
-#include "asio/asio.h"
-#include "asio/asiodrivers.h"
-#include <math.h>
+ if ( stream_.doConvertBuffer[mode] )
+ setConvertInfo( mode, 0 );
-AsioDrivers drivers;
-ASIOCallbacks asioCallbacks;
-ASIODriverInfo driverInfo;
-CallbackInfo *asioCallbackInfo;
+ // Allocate necessary internal buffers
+ bufferBytes = stream_.nUserChannels[mode] * stream_.bufferSize * formatBytes( stream_.userFormat );
-struct AsioHandle {
- bool stopStream;
- ASIOBufferInfo *bufferInfos;
- HANDLE condition;
+ stream_.userBuffer[mode] = ( char* ) calloc( bufferBytes, 1 );
+ if ( !stream_.userBuffer[mode] ) {
+ errorType = RtAudioError::MEMORY_ERROR;
+ errorText_ = "RtApiWasapi::probeDeviceOpen: Error allocating user buffer memory.";
+ goto Exit;
+ }
- AsioHandle()
- :stopStream(false), bufferInfos(0) {}
-};
+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME )
+ stream_.callbackInfo.priority = 15;
+ else
+ stream_.callbackInfo.priority = 0;
-RtApiAsio :: RtApiAsio()
+ ///! TODO: RTAUDIO_MINIMIZE_LATENCY // Provide stream buffers directly to callback
+ ///! TODO: RTAUDIO_HOG_DEVICE // Exclusive mode
+
+ methodResult = SUCCESS;
+
+Exit:
+ //clean up
+ SAFE_RELEASE( captureDevices );
+ SAFE_RELEASE( renderDevices );
+ SAFE_RELEASE( devicePtr );
+ CoTaskMemFree( deviceFormat );
+
+ // if method failed, close the stream
+ if ( methodResult == FAILURE )
+ closeStream();
+
+ if ( !errorText_.empty() )
+ error( errorType );
+ return methodResult;
+}
+
+//=============================================================================
+
+DWORD WINAPI RtApiWasapi::runWasapiThread( void* wasapiPtr )
{
- this->initialize();
+ if ( wasapiPtr )
+ ( ( RtApiWasapi* ) wasapiPtr )->wasapiThread();
- if (nDevices_ <= 0) {
- sprintf(message_, "RtApiAsio: no Windows ASIO audio drivers found!");
- error(RtError::NO_DEVICES_FOUND);
- }
+ return 0;
}
-RtApiAsio :: ~RtApiAsio()
+DWORD WINAPI RtApiWasapi::stopWasapiThread( void* wasapiPtr )
{
- if ( stream_.mode != UNINITIALIZED ) closeStream();
+ if ( wasapiPtr )
+ ( ( RtApiWasapi* ) wasapiPtr )->stopStream();
+
+ return 0;
}
-void RtApiAsio :: initialize(void)
+DWORD WINAPI RtApiWasapi::abortWasapiThread( void* wasapiPtr )
{
- nDevices_ = drivers.asioGetNumDev();
- if (nDevices_ <= 0) return;
+ if ( wasapiPtr )
+ ( ( RtApiWasapi* ) wasapiPtr )->abortStream();
+
+ return 0;
+}
+
+//-----------------------------------------------------------------------------
+
+void RtApiWasapi::wasapiThread()
+{
+ // as this is a new thread, we must CoInitialize it
+ CoInitialize( NULL );
+
+ HRESULT hr;
+
+ IAudioClient* captureAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureAudioClient;
+ IAudioClient* renderAudioClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderAudioClient;
+ IAudioCaptureClient* captureClient = ( ( WasapiHandle* ) stream_.apiHandle )->captureClient;
+ IAudioRenderClient* renderClient = ( ( WasapiHandle* ) stream_.apiHandle )->renderClient;
+ HANDLE captureEvent = ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent;
+ HANDLE renderEvent = ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent;
+
+ WAVEFORMATEX* captureFormat = NULL;
+ WAVEFORMATEX* renderFormat = NULL;
+ WasapiBuffer captureBuffer;
+ WasapiBuffer renderBuffer;
+
+ // declare local stream variables
+ RtAudioCallback callback = ( RtAudioCallback ) stream_.callbackInfo.callback;
+ BYTE* streamBuffer = NULL;
+ unsigned long captureFlags = 0;
+ unsigned int bufferFrameCount = 0;
+ unsigned int numFramesPadding = 0;
+ bool callbackPushed = false;
+ bool callbackPulled = false;
+ bool callbackStopped = false;
+ int callbackResult = 0;
+
+ unsigned int deviceBuffSize = 0;
+
+ errorText_.clear();
+ RtAudioError::Type errorType = RtAudioError::DRIVER_ERROR;
+
+ // Attempt to assign "Pro Audio" characteristic to thread
+ HMODULE AvrtDll = LoadLibrary( (LPCTSTR) "AVRT.dll" );
+ if ( AvrtDll ) {
+ DWORD taskIndex = 0;
+ TAvSetMmThreadCharacteristicsPtr AvSetMmThreadCharacteristicsPtr = ( TAvSetMmThreadCharacteristicsPtr ) GetProcAddress( AvrtDll, "AvSetMmThreadCharacteristicsW" );
+ AvSetMmThreadCharacteristicsPtr( L"Pro Audio", &taskIndex );
+ FreeLibrary( AvrtDll );
+ }
+
+ // start capture stream if applicable
+ if ( captureAudioClient ) {
+ hr = captureAudioClient->GetMixFormat( &captureFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
+ goto Exit;
+ }
+
+ // initialize capture stream according to desire buffer size
+ REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) stream_.bufferSize * 10000000 / captureFormat->nSamplesPerSec );
+
+ if ( !captureClient ) {
+ hr = captureAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ desiredBufferPeriod,
+ desiredBufferPeriod,
+ captureFormat,
+ NULL );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize capture audio client.";
+ goto Exit;
+ }
+
+ hr = captureAudioClient->GetService( __uuidof( IAudioCaptureClient ),
+ ( void** ) &captureClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture client handle.";
+ goto Exit;
+ }
+
+ // configure captureEvent to trigger on every available capture buffer
+ captureEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
+ if ( !captureEvent ) {
+ errorType = RtAudioError::SYSTEM_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to create capture event.";
+ goto Exit;
+ }
- // Create device structures and write device driver names to each.
- RtApiDevice device;
- char name[128];
- for (int i=0; i<nDevices_; i++) {
- if ( drivers.asioGetDriverName( i, name, 128 ) == 0 ) {
- device.name.erase();
- device.name.append( (const char *)name, strlen(name)+1);
- devices_.push_back(device);
+ hr = captureAudioClient->SetEventHandle( captureEvent );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to set capture event handle.";
+ goto Exit;
+ }
+
+ ( ( WasapiHandle* ) stream_.apiHandle )->captureClient = captureClient;
+ ( ( WasapiHandle* ) stream_.apiHandle )->captureEvent = captureEvent;
}
- else {
- sprintf(message_, "RtApiAsio: error getting driver name for device index %d!", i);
- error(RtError::WARNING);
+
+ unsigned int inBufferSize = 0;
+ hr = captureAudioClient->GetBufferSize( &inBufferSize );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to get capture buffer size.";
+ goto Exit;
}
- }
- nDevices_ = (int) devices_.size();
+ // scale outBufferSize according to stream->user sample rate ratio
+ unsigned int outBufferSize = ( unsigned int ) stream_.bufferSize * stream_.nDeviceChannels[INPUT];
+ inBufferSize *= stream_.nDeviceChannels[INPUT];
- drivers.removeCurrentDriver();
- driverInfo.asioVersion = 2;
- // See note in DirectSound implementation about GetDesktopWindow().
- driverInfo.sysRef = GetForegroundWindow();
-}
+ // set captureBuffer size
+ captureBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[INPUT] ) );
-void RtApiAsio :: probeDeviceInfo(RtApiDevice *info)
-{
- // Don't probe if a stream is already open.
- if ( stream_.mode != UNINITIALIZED ) {
- sprintf(message_, "RtApiAsio: unable to probe driver while a stream is open.");
- error(RtError::DEBUG_WARNING);
- return;
- }
+ // reset the capture stream
+ hr = captureAudioClient->Reset();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to reset capture stream.";
+ goto Exit;
+ }
- if ( !drivers.loadDriver( (char *)info->name.c_str() ) ) {
- sprintf(message_, "RtApiAsio: error loading driver (%s).", info->name.c_str());
- error(RtError::DEBUG_WARNING);
- return;
+ // start the capture stream
+ hr = captureAudioClient->Start();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to start capture stream.";
+ goto Exit;
+ }
}
- 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_, "RtApiAsio: error (%s) initializing driver (%s).", details, info->name.c_str());
- error(RtError::DEBUG_WARNING);
- return;
- }
+ // start render stream if applicable
+ if ( renderAudioClient ) {
+ hr = renderAudioClient->GetMixFormat( &renderFormat );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve device mix format.";
+ goto Exit;
+ }
- // Determine the device channel information.
- long inputChannels, outputChannels;
- result = ASIOGetChannels( &inputChannels, &outputChannels );
- if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- sprintf(message_, "RtApiAsio: error getting input/output channel count (%s).", info->name.c_str());
- error(RtError::DEBUG_WARNING);
- return;
- }
+ // initialize render stream according to desire buffer size
+ REFERENCE_TIME desiredBufferPeriod = ( REFERENCE_TIME ) ( ( float ) stream_.bufferSize * 10000000 / renderFormat->nSamplesPerSec );
- info->maxOutputChannels = outputChannels;
- if ( outputChannels > 0 ) info->minOutputChannels = 1;
+ if ( !renderClient ) {
+ hr = renderAudioClient->Initialize( AUDCLNT_SHAREMODE_SHARED,
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+ desiredBufferPeriod,
+ desiredBufferPeriod,
+ renderFormat,
+ NULL );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to initialize render audio client.";
+ goto Exit;
+ }
- info->maxInputChannels = inputChannels;
- if ( inputChannels > 0 ) info->minInputChannels = 1;
+ hr = renderAudioClient->GetService( __uuidof( IAudioRenderClient ),
+ ( void** ) &renderClient );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render client handle.";
+ goto Exit;
+ }
- // 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;
- }
+ // configure renderEvent to trigger on every available render buffer
+ renderEvent = CreateEvent( NULL, FALSE, FALSE, NULL );
+ if ( !renderEvent ) {
+ errorType = RtAudioError::SYSTEM_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to create render event.";
+ goto Exit;
+ }
- // 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] );
- }
+ hr = renderAudioClient->SetEventHandle( renderEvent );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to set render event handle.";
+ goto Exit;
+ }
- if (info->sampleRates.size() == 0) {
- drivers.removeCurrentDriver();
- sprintf( message_, "RtApiAsio: No supported sample rates found for driver (%s).", info->name.c_str() );
- error(RtError::DEBUG_WARNING);
- return;
- }
+ ( ( WasapiHandle* ) stream_.apiHandle )->renderClient = renderClient;
+ ( ( WasapiHandle* ) stream_.apiHandle )->renderEvent = renderEvent;
+ }
- // 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_, "RtApiAsio: error getting driver (%s) channel information.", info->name.c_str());
- error(RtError::DEBUG_WARNING);
- return;
- }
+ unsigned int outBufferSize = 0;
+ hr = renderAudioClient->GetBufferSize( &outBufferSize );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to get render buffer size.";
+ goto Exit;
+ }
- 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;
+ // scale inBufferSize according to user->stream sample rate ratio
+ unsigned int inBufferSize = ( unsigned int ) stream_.bufferSize * stream_.nDeviceChannels[OUTPUT];
+ outBufferSize *= stream_.nDeviceChannels[OUTPUT];
- // Check that we have at least one supported format.
- if (info->nativeFormats == 0) {
- drivers.removeCurrentDriver();
- sprintf(message_, "RtApiAsio: driver (%s) data format not supported by RtAudio.",
- info->name.c_str());
- error(RtError::DEBUG_WARNING);
- return;
- }
+ // set renderBuffer size
+ renderBuffer.setBufferSize( inBufferSize + outBufferSize, formatBytes( stream_.deviceFormat[OUTPUT] ) );
- info->probed = true;
- drivers.removeCurrentDriver();
-}
+ // reset the render stream
+ hr = renderAudioClient->Reset();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to reset render stream.";
+ goto Exit;
+ }
-void bufferSwitch(long index, ASIOBool processNow)
-{
- RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
- try {
- object->callbackEvent( index );
- }
- catch (RtError &exception) {
- fprintf(stderr, "\nRtApiAsio: callback handler error (%s)!\n\n", exception.getMessageString());
- return;
+ // start the render stream
+ hr = renderAudioClient->Start();
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to start render stream.";
+ goto Exit;
+ }
}
- return;
-}
-
-void sampleRateChanged(ASIOSampleRate sRate)
-{
- // The ASIO documentation says that this usually only happens during
- // external sync. Audio processing is not stopped by the driver,
- // actual sample rate might not have even changed, maybe only the
- // sample rate status of an AES/EBU or S/PDIF digital input at the
- // audio device.
-
- RtAudio *object = (RtAudio *) asioCallbackInfo->object;
- try {
- object->stopStream();
+ if ( stream_.mode == INPUT ) {
+ using namespace std; // for roundf
+ deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] );
}
- catch (RtError &exception) {
- fprintf(stderr, "\nRtApiAsio: sampleRateChanged() error (%s)!\n\n", exception.getMessageString());
- return;
+ else if ( stream_.mode == OUTPUT ) {
+ deviceBuffSize = stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] );
+ }
+ else if ( stream_.mode == DUPLEX ) {
+ deviceBuffSize = std::max( stream_.bufferSize * stream_.nDeviceChannels[INPUT] * formatBytes( stream_.deviceFormat[INPUT] ),
+ stream_.bufferSize * stream_.nDeviceChannels[OUTPUT] * formatBytes( stream_.deviceFormat[OUTPUT] ) );
}
- fprintf(stderr, "\nRtApiAsio: driver reports sample rate changed to %d ... stream stopped!!!", (int) sRate);
-}
+ stream_.deviceBuffer = ( char* ) malloc( deviceBuffSize );
+ if ( !stream_.deviceBuffer ) {
+ errorType = RtAudioError::MEMORY_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Error allocating device buffer memory.";
+ goto Exit;
+ }
+
+ // stream process loop
+ while ( stream_.state != STREAM_STOPPING ) {
+ if ( !callbackPulled ) {
+ // Callback Input
+ // ==============
+ // 1. Pull callback buffer from inputBuffer
+ // 2. If 1. was successful: Convert callback buffer to user format
+
+ if ( captureAudioClient ) {
+ // Pull callback buffer from inputBuffer
+ callbackPulled = captureBuffer.pullBuffer( stream_.deviceBuffer,
+ ( unsigned int ) stream_.bufferSize * stream_.nDeviceChannels[INPUT],
+ stream_.deviceFormat[INPUT] );
+
+ if ( callbackPulled ) {
+ if ( stream_.doConvertBuffer[INPUT] ) {
+ // Convert callback buffer to user format
+ convertBuffer( stream_.userBuffer[INPUT],
+ stream_.deviceBuffer,
+ stream_.convertInfo[INPUT] );
+ }
+ else {
+ // no further conversion, simple copy deviceBuffer to userBuffer
+ memcpy( stream_.userBuffer[INPUT],
+ stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nUserChannels[INPUT] * formatBytes( stream_.userFormat ) );
+ }
+ }
+ }
+ else {
+ // if there is no capture stream, set callbackPulled flag
+ callbackPulled = true;
+ }
+
+ // Execute Callback
+ // ================
+ // 1. Execute user callback method
+ // 2. Handle return value from callback
+
+ // if callback has not requested the stream to stop
+ if ( callbackPulled && !callbackStopped ) {
+ // Execute user callback method
+ callbackResult = callback( stream_.userBuffer[OUTPUT],
+ stream_.userBuffer[INPUT],
+ stream_.bufferSize,
+ getStreamTime(),
+ captureFlags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY ? RTAUDIO_INPUT_OVERFLOW : 0,
+ stream_.callbackInfo.userData );
+
+ // Handle return value from callback
+ if ( callbackResult == 1 ) {
+ // instantiate a thread to stop this thread
+ HANDLE threadHandle = CreateThread( NULL, 0, stopWasapiThread, this, 0, NULL );
+ if ( !threadHandle ) {
+ errorType = RtAudioError::THREAD_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream stop thread.";
+ goto Exit;
+ }
+ else if ( !CloseHandle( threadHandle ) ) {
+ errorType = RtAudioError::THREAD_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream stop thread handle.";
+ goto Exit;
+ }
+
+ callbackStopped = true;
+ }
+ else if ( callbackResult == 2 ) {
+ // instantiate a thread to stop this thread
+ HANDLE threadHandle = CreateThread( NULL, 0, abortWasapiThread, this, 0, NULL );
+ if ( !threadHandle ) {
+ errorType = RtAudioError::THREAD_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to instantiate stream abort thread.";
+ goto Exit;
+ }
+ else if ( !CloseHandle( threadHandle ) ) {
+ errorType = RtAudioError::THREAD_ERROR;
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to close stream abort thread handle.";
+ goto Exit;
+ }
+
+ callbackStopped = true;
+ }
+ }
+ }
+
+ // Callback Output
+ // ===============
+ // 1. Convert callback buffer to stream format
+ // 2. Push callback buffer into outputBuffer
+
+ if ( renderAudioClient && callbackPulled ) {
+ if ( stream_.doConvertBuffer[OUTPUT] ) {
+ // Convert callback buffer to stream format
+ convertBuffer( stream_.deviceBuffer,
+ stream_.userBuffer[OUTPUT],
+ stream_.convertInfo[OUTPUT] );
+
+ }
+
+ // Push callback buffer into outputBuffer
+ callbackPushed = renderBuffer.pushBuffer( stream_.deviceBuffer,
+ stream_.bufferSize * stream_.nDeviceChannels[OUTPUT],
+ stream_.deviceFormat[OUTPUT] );
+ }
+ else {
+ // if there is no render stream, set callbackPushed flag
+ callbackPushed = true;
+ }
+
+ // Stream Capture
+ // ==============
+ // 1. Get capture buffer from stream
+ // 2. Push capture buffer into inputBuffer
+ // 3. If 2. was successful: Release capture buffer
+
+ if ( captureAudioClient ) {
+ // if the callback input buffer was not pulled from captureBuffer, wait for next capture event
+ if ( !callbackPulled ) {
+ WaitForSingleObject( captureEvent, INFINITE );
+ }
+
+ // Get capture buffer from stream
+ hr = captureClient->GetBuffer( &streamBuffer,
+ &bufferFrameCount,
+ &captureFlags, NULL, NULL );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve capture buffer.";
+ goto Exit;
+ }
+
+ if ( bufferFrameCount != 0 ) {
+ // Push capture buffer into inputBuffer
+ if ( captureBuffer.pushBuffer( ( char* ) streamBuffer,
+ bufferFrameCount * stream_.nDeviceChannels[INPUT],
+ stream_.deviceFormat[INPUT] ) )
+ {
+ // Release capture buffer
+ hr = captureClient->ReleaseBuffer( bufferFrameCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
+ goto Exit;
+ }
+ }
+ else
+ {
+ // Inform WASAPI that capture was unsuccessful
+ hr = captureClient->ReleaseBuffer( 0 );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
+ goto Exit;
+ }
+ }
+ }
+ else
+ {
+ // Inform WASAPI that capture was unsuccessful
+ hr = captureClient->ReleaseBuffer( 0 );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release capture buffer.";
+ goto Exit;
+ }
+ }
+ }
+
+ // Stream Render
+ // =============
+ // 1. Get render buffer from stream
+ // 2. Pull next buffer from outputBuffer
+ // 3. If 2. was successful: Fill render buffer with next buffer
+ // Release render buffer
+
+ if ( renderAudioClient ) {
+ // if the callback output buffer was not pushed to renderBuffer, wait for next render event
+ if ( callbackPulled && !callbackPushed ) {
+ WaitForSingleObject( renderEvent, INFINITE );
+ }
+
+ // Get render buffer from stream
+ hr = renderAudioClient->GetBufferSize( &bufferFrameCount );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer size.";
+ goto Exit;
+ }
+
+ hr = renderAudioClient->GetCurrentPadding( &numFramesPadding );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer padding.";
+ goto Exit;
+ }
+
+ bufferFrameCount -= numFramesPadding;
+
+ if ( bufferFrameCount != 0 ) {
+ hr = renderClient->GetBuffer( bufferFrameCount, &streamBuffer );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to retrieve render buffer.";
+ goto Exit;
+ }
+
+ // Pull next buffer from outputBuffer
+ // Fill render buffer with next buffer
+ if ( renderBuffer.pullBuffer( ( char* ) streamBuffer,
+ bufferFrameCount * stream_.nDeviceChannels[OUTPUT],
+ stream_.deviceFormat[OUTPUT] ) )
+ {
+ // Release render buffer
+ hr = renderClient->ReleaseBuffer( bufferFrameCount, 0 );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
+ goto Exit;
+ }
+ }
+ else
+ {
+ // Inform WASAPI that render was unsuccessful
+ hr = renderClient->ReleaseBuffer( 0, 0 );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
+ goto Exit;
+ }
+ }
+ }
+ else
+ {
+ // Inform WASAPI that render was unsuccessful
+ hr = renderClient->ReleaseBuffer( 0, 0 );
+ if ( FAILED( hr ) ) {
+ errorText_ = "RtApiWasapi::wasapiThread: Unable to release render buffer.";
+ goto Exit;
+ }
+ }
+ }
+
+ // if the callback buffer was pushed renderBuffer reset callbackPulled flag
+ if ( callbackPushed ) {
+ callbackPulled = false;
+ // tick stream time
+ RtApi::tickStreamTime();
+ }
-long asioMessages(long selector, long value, void* message, double* opt)
-{
- long ret = 0;
- switch(selector) {
- case kAsioSelectorSupported:
- if(value == kAsioResetRequest
- || value == kAsioEngineVersion
- || value == kAsioResyncRequest
- || value == kAsioLatenciesChanged
- // The following three were added for ASIO 2.0, you don't
- // necessarily have to support them.
- || value == kAsioSupportsTimeInfo
- || value == kAsioSupportsTimeCode
- || value == kAsioSupportsInputMonitor)
- ret = 1L;
- break;
- case kAsioResetRequest:
- // Defer the task and perform the reset of the driver during the
- // next "safe" situation. You cannot reset the driver right now,
- // as this code is called from the driver. Reset the driver is
- // done by completely destruct is. I.e. ASIOStop(),
- // ASIODisposeBuffers(), Destruction Afterwards you initialize the
- // driver again.
- fprintf(stderr, "\nRtApiAsio: driver reset requested!!!");
- ret = 1L;
- break;
- case kAsioResyncRequest:
- // This informs the application that the driver encountered some
- // non-fatal data loss. It is used for synchronization purposes
- // of different media. Added mainly to work around the Win16Mutex
- // problems in Windows 95/98 with the Windows Multimedia system,
- // which could lose data because the Mutex was held too long by
- // another thread. However a driver can issue it in other
- // situations, too.
- fprintf(stderr, "\nRtApiAsio: driver resync requested!!!");
- ret = 1L;
- break;
- case kAsioLatenciesChanged:
- // This will inform the host application that the drivers were
- // latencies changed. Beware, it this does not mean that the
- // buffer sizes have changed! You might need to update internal
- // delay data.
- fprintf(stderr, "\nRtApiAsio: driver latency may have changed!!!");
- ret = 1L;
- break;
- case kAsioEngineVersion:
- // Return the supported ASIO version of the host application. If
- // a host application does not implement this selector, ASIO 1.0
- // is assumed by the driver.
- ret = 2L;
- break;
- case kAsioSupportsTimeInfo:
- // Informs the driver whether the
- // asioCallbacks.bufferSwitchTimeInfo() callback is supported.
- // For compatibility with ASIO 1.0 drivers the host application
- // should always support the "old" bufferSwitch method, too.
- ret = 0;
- break;
- case kAsioSupportsTimeCode:
- // Informs the driver wether application is interested in time
- // code info. If an application does not need to know about time
- // code, the driver has less work to do.
- ret = 0;
- break;
}
- return ret;
+
+Exit:
+ // clean up
+ CoTaskMemFree( captureFormat );
+ CoTaskMemFree( renderFormat );
+
+ CoUninitialize();
+
+ // update stream state
+ stream_.state = STREAM_STOPPED;
+
+ if ( errorText_.empty() )
+ return;
+ else
+ error( errorType );
+}
+
+//******************** End of __WINDOWS_WASAPI__ *********************//
+#endif
+
+
+#if defined(__WINDOWS_DS__) // Windows DirectSound API
+
+// Modified by Robin Davies, October 2005
+// - Improvements to DirectX pointer chasing.
+// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
+// - Auto-call CoInitialize for DSOUND and ASIO platforms.
+// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
+// Changed device query structure for RtAudio 4.0.7, January 2010
+
+#include <windows.h>
+#include <process.h>
+#include <mmsystem.h>
+#include <mmreg.h>
+#include <dsound.h>
+#include <assert.h>
+#include <algorithm>
+
+#if defined(__MINGW32__)
+ // missing from latest mingw winapi
+#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
+#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
+#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
+#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
+#endif
+
+#define MINIMUM_DEVICE_BUFFER_SIZE 32768
+
+#ifdef _MSC_VER // if Microsoft Visual C++
+#pragma comment( lib, "winmm.lib" ) // then, auto-link winmm.lib. Otherwise, it has to be added manually.
+#endif
+
+static inline DWORD dsPointerBetween( DWORD pointer, DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
+{
+ if ( pointer > bufferSize ) pointer -= bufferSize;
+ if ( laterPointer < earlierPointer ) laterPointer += bufferSize;
+ if ( pointer < earlierPointer ) pointer += bufferSize;
+ return pointer >= earlierPointer && pointer < laterPointer;
+}
+
+// A structure to hold various information related to the DirectSound
+// API implementation.
+struct DsHandle {
+ unsigned int drainCounter; // Tracks callback counts when draining
+ bool internalDrain; // Indicates if stop is initiated from callback or not.
+ void *id[2];
+ void *buffer[2];
+ bool xrun[2];
+ UINT bufferPointer[2];
+ DWORD dsBufferSize[2];
+ DWORD dsPointerLeadTime[2]; // the number of bytes ahead of the safe pointer to lead by.
+ HANDLE condition;
+
+ DsHandle()
+ :drainCounter(0), internalDrain(false) { id[0] = 0; id[1] = 0; buffer[0] = 0; buffer[1] = 0; xrun[0] = false; xrun[1] = false; bufferPointer[0] = 0; bufferPointer[1] = 0; }
+};
+
+// Declarations for utility functions, callbacks, and structures
+// specific to the DirectSound implementation.
+static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
+ LPCTSTR description,
+ LPCTSTR module,
+ LPVOID lpContext );
+
+static const char* getErrorString( int code );
+
+static unsigned __stdcall callbackHandler( void *ptr );
+
+struct DsDevice {
+ LPGUID id[2];
+ bool validId[2];
+ bool found;
+ std::string name;
+
+ DsDevice()
+ : found(false) { validId[0] = false; validId[1] = false; }
+};
+
+struct DsProbeData {
+ bool isInput;
+ std::vector<struct DsDevice>* dsDevices;
+};
+
+RtApiDs :: RtApiDs()
+{
+ // Dsound will run both-threaded. If CoInitialize fails, then just
+ // accept whatever the mainline chose for a threading model.
+ coInitialized_ = false;
+ HRESULT hr = CoInitialize( NULL );
+ if ( !FAILED( hr ) ) coInitialized_ = true;
+}
+
+RtApiDs :: ~RtApiDs()
+{
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
+ if ( coInitialized_ ) CoUninitialize(); // balanced call.
+}
+
+// The DirectSound default output is always the first device.
+unsigned int RtApiDs :: getDefaultOutputDevice( void )
+{
+ return 0;
+}
+
+// The DirectSound default input is always the first input device,
+// which is the first capture device enumerated.
+unsigned int RtApiDs :: getDefaultInputDevice( void )
+{
+ return 0;
+}
+
+unsigned int RtApiDs :: getDeviceCount( void )
+{
+ // Set query flag for previously found devices to false, so that we
+ // can check for any devices that have disappeared.
+ for ( unsigned int i=0; i<dsDevices.size(); i++ )
+ dsDevices[i].found = false;
+
+ // Query DirectSound devices.
+ struct DsProbeData probeInfo;
+ probeInfo.isInput = false;
+ probeInfo.dsDevices = &dsDevices;
+ HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ }
+
+ // Query DirectSoundCapture devices.
+ probeInfo.isInput = true;
+ result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &probeInfo );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ }
+
+ // Clean out any devices that may have disappeared (code update submitted by Eli Zehngut).
+ for ( unsigned int i=0; i<dsDevices.size(); ) {
+ if ( dsDevices[i].found == false ) dsDevices.erase( dsDevices.begin() + i );
+ else i++;
+ }
+
+ return static_cast<unsigned int>(dsDevices.size());
+}
+
+RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = false;
+
+ if ( dsDevices.size() == 0 ) {
+ // Force a query of all devices
+ getDeviceCount();
+ if ( dsDevices.size() == 0 ) {
+ errorText_ = "RtApiDs::getDeviceInfo: no devices found!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+ }
+
+ if ( device >= dsDevices.size() ) {
+ errorText_ = "RtApiDs::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
+ }
+
+ HRESULT result;
+ if ( dsDevices[ device ].validId[0] == false ) goto probeInput;
+
+ LPDIRECTSOUND output;
+ DSCAPS outCaps;
+ result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto probeInput;
+ }
+
+ outCaps.dwSize = sizeof( outCaps );
+ result = output->GetCaps( &outCaps );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto probeInput;
+ }
+
+ // Get output channel information.
+ info.outputChannels = ( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
+
+ // Get sample rate information.
+ info.sampleRates.clear();
+ for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
+ if ( SAMPLE_RATES[k] >= (unsigned int) outCaps.dwMinSecondarySampleRate &&
+ SAMPLE_RATES[k] <= (unsigned int) outCaps.dwMaxSecondarySampleRate ) {
+ info.sampleRates.push_back( SAMPLE_RATES[k] );
+
+ if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
+ info.preferredSampleRate = SAMPLE_RATES[k];
+ }
+ }
+
+ // Get format information.
+ if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) info.nativeFormats |= RTAUDIO_SINT8;
+
+ output->Release();
+
+ if ( getDefaultOutputDevice() == device )
+ info.isDefaultOutput = true;
+
+ if ( dsDevices[ device ].validId[1] == false ) {
+ info.name = dsDevices[ device ].name;
+ info.probed = true;
+ return info;
+ }
+
+ probeInput:
+
+ LPDIRECTSOUNDCAPTURE input;
+ result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ DSCCAPS inCaps;
+ inCaps.dwSize = sizeof( inCaps );
+ result = input->GetCaps( &inCaps );
+ if ( FAILED( result ) ) {
+ input->Release();
+ errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting object capabilities (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Get input channel information.
+ info.inputChannels = inCaps.dwChannels;
+
+ // Get sample rate and format information.
+ std::vector<unsigned int> rates;
+ if ( inCaps.dwChannels >= 2 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
+
+ if ( info.nativeFormats & RTAUDIO_SINT16 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) rates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) rates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) rates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) rates.push_back( 96000 );
+ }
+ else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) rates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) rates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) rates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) rates.push_back( 96000 );
+ }
+ }
+ else if ( inCaps.dwChannels == 1 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
+
+ if ( info.nativeFormats & RTAUDIO_SINT16 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) rates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) rates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) rates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) rates.push_back( 96000 );
+ }
+ else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
+ if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) rates.push_back( 11025 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) rates.push_back( 22050 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) rates.push_back( 44100 );
+ if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) rates.push_back( 96000 );
+ }
+ }
+ else info.inputChannels = 0; // technically, this would be an error
+
+ input->Release();
+
+ if ( info.inputChannels == 0 ) return info;
+
+ // Copy the supported rates to the info structure but avoid duplication.
+ bool found;
+ for ( unsigned int i=0; i<rates.size(); i++ ) {
+ found = false;
+ for ( unsigned int j=0; j<info.sampleRates.size(); j++ ) {
+ if ( rates[i] == info.sampleRates[j] ) {
+ found = true;
+ break;
+ }
+ }
+ if ( found == false ) info.sampleRates.push_back( rates[i] );
+ }
+ std::sort( info.sampleRates.begin(), info.sampleRates.end() );
+
+ // If device opens for both playback and capture, we determine the channels.
+ if ( info.outputChannels > 0 && info.inputChannels > 0 )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+
+ if ( device == 0 ) info.isDefaultInput = true;
+
+ // Copy name and return.
+ info.name = dsDevices[ device ].name;
+ info.probed = true;
+ return info;
}
-bool RtApiAsio :: probeDeviceOpen(int device, StreamMode mode, int channels,
- int sampleRate, RtAudioFormat format,
- int *bufferSize, int numberOfBuffers)
-{
- // For ASIO, a duplex stream MUST use the same driver.
- if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) {
- sprintf(message_, "RtApiAsio: duplex stream must use the same device for input and output.");
- error(RtError::WARNING);
- return FAILURE;
+bool RtApiDs :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ if ( channels + firstChannel > 2 ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: DirectSound does not support more than 2 channels per device.";
+ return FAILURE;
+ }
+
+ size_t nDevices = dsDevices.size();
+ if ( nDevices == 0 ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiDs::probeDeviceOpen: no devices found!";
+ return FAILURE;
+ }
+
+ if ( device >= nDevices ) {
+ // This should not happen because a check is made before this function is called.
+ errorText_ = "RtApiDs::probeDeviceOpen: device ID is invalid!";
+ return FAILURE;
+ }
+
+ if ( mode == OUTPUT ) {
+ if ( dsDevices[ device ].validId[0] == false ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+ else { // mode == INPUT
+ if ( dsDevices[ device ].validId[1] == false ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support input!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // According to a note in PortAudio, using GetDesktopWindow()
+ // instead of GetForegroundWindow() is supposed to avoid problems
+ // that occur when the application's window is not the foreground
+ // window. Also, if the application window closes before the
+ // DirectSound buffer, DirectSound can crash. In the past, I had
+ // problems when using GetDesktopWindow() but it seems fine now
+ // (January 2010). I'll leave it commented here.
+ // HWND hWnd = GetForegroundWindow();
+ HWND hWnd = GetDesktopWindow();
+
+ // Check the numberOfBuffers parameter and limit the lowest value to
+ // two. This is a judgement call and a value of two is probably too
+ // low for capture, but it should work for playback.
+ int nBuffers = 0;
+ if ( options ) nBuffers = options->numberOfBuffers;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
+ if ( nBuffers < 2 ) nBuffers = 3;
+
+ // Check the lower range of the user-specified buffer size and set
+ // (arbitrarily) to a lower bound of 32.
+ if ( *bufferSize < 32 ) *bufferSize = 32;
+
+ // Create the wave format structure. The data format setting will
+ // be determined later.
+ WAVEFORMATEX waveFormat;
+ ZeroMemory( &waveFormat, sizeof(WAVEFORMATEX) );
+ waveFormat.wFormatTag = WAVE_FORMAT_PCM;
+ waveFormat.nChannels = channels + firstChannel;
+ waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
+
+ // Determine the device buffer size. By default, we'll use the value
+ // defined above (32K), but we will grow it to make allowances for
+ // very large software buffer sizes.
+ DWORD dsBufferSize = MINIMUM_DEVICE_BUFFER_SIZE;
+ DWORD dsPointerLeadTime = 0;
+
+ void *ohandle = 0, *bhandle = 0;
+ HRESULT result;
+ if ( mode == OUTPUT ) {
+
+ LPDIRECTSOUND output;
+ result = DirectSoundCreate( dsDevices[ device ].id[0], &output, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ DSCAPS outCaps;
+ outCaps.dwSize = sizeof( outCaps );
+ result = output->GetCaps( &outCaps );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Check channel information.
+ if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
+ errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsDevices[ device ].name << ") does not support stereo playback.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Check format information. Use 16-bit format unless not
+ // supported or user requests 8-bit.
+ if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
+ !( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
+ waveFormat.wBitsPerSample = 16;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ else {
+ waveFormat.wBitsPerSample = 8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ stream_.userFormat = format;
+
+ // Update wave format structure and buffer information.
+ waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+ waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
+
+ // If the user wants an even bigger buffer, increase the device buffer size accordingly.
+ while ( dsPointerLeadTime * 2U > dsBufferSize )
+ dsBufferSize *= 2;
+
+ // Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
+ // result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
+ // Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
+ result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Even though we will write to the secondary buffer, we need to
+ // access the primary buffer to set the correct output format
+ // (since the default is 8-bit, 22 kHz!). Setup the DS primary
+ // buffer description.
+ DSBUFFERDESC bufferDescription;
+ ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
+ bufferDescription.dwSize = sizeof( DSBUFFERDESC );
+ bufferDescription.dwFlags = DSBCAPS_PRIMARYBUFFER;
+
+ // Obtain the primary buffer
+ LPDIRECTSOUNDBUFFER buffer;
+ result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") accessing primary buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Set the primary DS buffer sound format.
+ result = buffer->SetFormat( &waveFormat );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting primary buffer format (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Setup the secondary DS buffer description.
+ ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
+ bufferDescription.dwSize = sizeof( DSBUFFERDESC );
+ bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
+ DSBCAPS_GLOBALFOCUS |
+ DSBCAPS_GETCURRENTPOSITION2 |
+ DSBCAPS_LOCHARDWARE ); // Force hardware mixing
+ bufferDescription.dwBufferBytes = dsBufferSize;
+ bufferDescription.lpwfxFormat = &waveFormat;
+
+ // Try to create the secondary DS buffer. If that doesn't work,
+ // try to use software mixing. Otherwise, there's a problem.
+ result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
+ DSBCAPS_GLOBALFOCUS |
+ DSBCAPS_GETCURRENTPOSITION2 |
+ DSBCAPS_LOCSOFTWARE ); // Force software mixing
+ result = output->CreateSoundBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ output->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating secondary buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+
+ // Get the buffer size ... might be different from what we specified.
+ DSBCAPS dsbcaps;
+ dsbcaps.dwSize = sizeof( DSBCAPS );
+ result = buffer->GetCaps( &dsbcaps );
+ if ( FAILED( result ) ) {
+ output->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ dsBufferSize = dsbcaps.dwBufferBytes;
+
+ // Lock the DS buffer
+ LPVOID audioPtr;
+ DWORD dataLen;
+ result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ output->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Zero the DS buffer
+ ZeroMemory( audioPtr, dataLen );
+
+ // Unlock the DS buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ output->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ ohandle = (void *) output;
+ bhandle = (void *) buffer;
+ }
+
+ if ( mode == INPUT ) {
+
+ LPDIRECTSOUNDCAPTURE input;
+ result = DirectSoundCaptureCreate( dsDevices[ device ].id[1], &input, NULL );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ DSCCAPS inCaps;
+ inCaps.dwSize = sizeof( inCaps );
+ result = input->GetCaps( &inCaps );
+ if ( FAILED( result ) ) {
+ input->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Check channel information.
+ if ( inCaps.dwChannels < channels + firstChannel ) {
+ errorText_ = "RtApiDs::getDeviceInfo: the input device does not support requested input channels.";
+ return FAILURE;
+ }
+
+ // Check format information. Use 16-bit format unless user
+ // requests 8-bit.
+ DWORD deviceFormats;
+ if ( channels + firstChannel == 2 ) {
+ deviceFormats = WAVE_FORMAT_1S08 | WAVE_FORMAT_2S08 | WAVE_FORMAT_4S08 | WAVE_FORMAT_96S08;
+ if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
+ waveFormat.wBitsPerSample = 8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ else { // assume 16-bit is supported
+ waveFormat.wBitsPerSample = 16;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ }
+ else { // channel == 1
+ deviceFormats = WAVE_FORMAT_1M08 | WAVE_FORMAT_2M08 | WAVE_FORMAT_4M08 | WAVE_FORMAT_96M08;
+ if ( format == RTAUDIO_SINT8 && inCaps.dwFormats & deviceFormats ) {
+ waveFormat.wBitsPerSample = 8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ else { // assume 16-bit is supported
+ waveFormat.wBitsPerSample = 16;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ }
+ stream_.userFormat = format;
+
+ // Update wave format structure and buffer information.
+ waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+ waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
+
+ // If the user wants an even bigger buffer, increase the device buffer size accordingly.
+ while ( dsPointerLeadTime * 2U > dsBufferSize )
+ dsBufferSize *= 2;
+
+ // Setup the secondary DS buffer description.
+ DSCBUFFERDESC bufferDescription;
+ ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
+ bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
+ bufferDescription.dwFlags = 0;
+ bufferDescription.dwReserved = 0;
+ bufferDescription.dwBufferBytes = dsBufferSize;
+ bufferDescription.lpwfxFormat = &waveFormat;
+
+ // Create the capture buffer.
+ LPDIRECTSOUNDCAPTUREBUFFER buffer;
+ result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
+ if ( FAILED( result ) ) {
+ input->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Get the buffer size ... might be different from what we specified.
+ DSCBCAPS dscbcaps;
+ dscbcaps.dwSize = sizeof( DSCBCAPS );
+ result = buffer->GetCaps( &dscbcaps );
+ if ( FAILED( result ) ) {
+ input->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting buffer settings (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ dsBufferSize = dscbcaps.dwBufferBytes;
+
+ // NOTE: We could have a problem here if this is a duplex stream
+ // and the play and capture hardware buffer sizes are different
+ // (I'm actually not sure if that is a problem or not).
+ // Currently, we are not verifying that.
+
+ // Lock the capture buffer
+ LPVOID audioPtr;
+ DWORD dataLen;
+ result = buffer->Lock( 0, dsBufferSize, &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ input->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Zero the buffer
+ ZeroMemory( audioPtr, dataLen );
+
+ // Unlock the buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ input->Release();
+ buffer->Release();
+ errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") unlocking input buffer (" << dsDevices[ device ].name << ")!";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ ohandle = (void *) input;
+ bhandle = (void *) buffer;
+ }
+
+ // Set various stream parameters
+ DsHandle *handle = 0;
+ stream_.nDeviceChannels[mode] = channels + firstChannel;
+ stream_.nUserChannels[mode] = channels;
+ stream_.bufferSize = *bufferSize;
+ stream_.channelOffset[mode] = firstChannel;
+ stream_.deviceInterleaved[mode] = true;
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
+
+ // Set flag for buffer conversion
+ stream_.doConvertBuffer[mode] = false;
+ if (stream_.nUserChannels[mode] != stream_.nDeviceChannels[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if (stream_.userFormat != stream_.deviceFormat[mode])
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
+ stream_.nUserChannels[mode] > 1 )
+ stream_.doConvertBuffer[mode] = true;
+
+ // Allocate necessary internal buffers
+ long bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
}
- // Only load the driver once for duplex stream.
- ASIOError result;
- if ( mode != INPUT || stream_.mode != OUTPUT ) {
- if ( !drivers.loadDriver( (char *)devices_[device].name.c_str() ) ) {
- sprintf(message_, "RtApiAsio: error loading driver (%s).", devices_[device].name.c_str());
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ if ( stream_.doConvertBuffer[mode] ) {
+
+ bool makeBuffer = true;
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
+ if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
+ unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
+ if ( bufferBytes <= (long) bytesOut ) makeBuffer = false;
+ }
}
- 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_, "RtApiAsio: error (%s) initializing driver (%s).", details, devices_[device].name.c_str());
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
}
}
- // Check the device channel count.
- long inputChannels, outputChannels;
- result = ASIOGetChannels( &inputChannels, &outputChannels );
- if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- sprintf(message_, "RtApiAsio: error getting input/output channel count (%s).",
- devices_[device].name.c_str());
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
+ // 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;
+ }
- if ( ( mode == OUTPUT && channels > outputChannels) ||
- ( mode == INPUT && channels > inputChannels) ) {
- drivers.removeCurrentDriver();
- sprintf(message_, "RtApiAsio: driver (%s) does not support requested channel count (%d).",
- devices_[device].name.c_str(), channels);
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ // 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_.nDeviceChannels[mode] = channels;
- stream_.nUserChannels[mode] = channels;
+ else
+ handle = (DsHandle *) stream_.apiHandle;
+ handle->id[mode] = ohandle;
+ handle->buffer[mode] = bhandle;
+ handle->dsBufferSize[mode] = dsBufferSize;
+ handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
- // Verify the sample rate is supported.
- result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
- if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- sprintf(message_, "RtApiAsio: driver (%s) does not support requested sample rate (%d).",
- devices_[device].name.c_str(), sampleRate);
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
+ stream_.device[mode] = device;
+ stream_.state = STREAM_STOPPED;
+ if ( stream_.mode == OUTPUT && mode == INPUT )
+ // We had already set up an output stream.
+ stream_.mode = DUPLEX;
+ else
+ stream_.mode = mode;
+ stream_.nBuffers = nBuffers;
+ stream_.sampleRate = sampleRate;
- // Set the sample rate.
- result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
- if ( result != ASE_OK ) {
- drivers.removeCurrentDriver();
- sprintf(message_, "RtApiAsio: driver (%s) error setting sample rate (%d).",
- devices_[device].name.c_str(), sampleRate);
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
+
+ // Setup the callback thread.
+ if ( stream_.callbackInfo.isRunning == false ) {
+ unsigned threadId;
+ stream_.callbackInfo.isRunning = true;
+ stream_.callbackInfo.object = (void *) this;
+ stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
+ &stream_.callbackInfo, 0, &threadId );
+ if ( stream_.callbackInfo.thread == 0 ) {
+ errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
+ goto error;
+ }
+
+ // Boost DS thread priority
+ SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
}
+ return SUCCESS;
- // 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_, "RtApiAsio: driver (%s) error getting data format.",
- devices_[device].name.c_str());
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ error:
+ if ( handle ) {
+ if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
+ LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ if ( buffer ) buffer->Release();
+ object->Release();
+ }
+ if ( handle->buffer[1] ) {
+ LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ if ( buffer ) buffer->Release();
+ object->Release();
+ }
+ CloseHandle( handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
}
- // 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;
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
}
- else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
- stream_.deviceFormat[mode] = RTAUDIO_SINT32;
- if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
}
- else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
- if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
+
+ stream_.state = STREAM_CLOSED;
+ return FAILURE;
+}
+
+void RtApiDs :: closeStream()
+{
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiDs::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
+ return;
}
- else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
- stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
- if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
+
+ // Stop the callback thread.
+ stream_.callbackInfo.isRunning = false;
+ WaitForSingleObject( (HANDLE) stream_.callbackInfo.thread, INFINITE );
+ CloseHandle( (HANDLE) stream_.callbackInfo.thread );
+
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ if ( handle ) {
+ if ( handle->buffer[0] ) { // the object pointer can be NULL and valid
+ LPDIRECTSOUND object = (LPDIRECTSOUND) handle->id[0];
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ if ( buffer ) {
+ buffer->Stop();
+ buffer->Release();
+ }
+ object->Release();
+ }
+ if ( handle->buffer[1] ) {
+ LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handle->id[1];
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ if ( buffer ) {
+ buffer->Stop();
+ buffer->Release();
+ }
+ object->Release();
+ }
+ CloseHandle( handle->condition );
+ delete handle;
+ stream_.apiHandle = 0;
}
- if ( stream_.deviceFormat[mode] == 0 ) {
- drivers.removeCurrentDriver();
- sprintf(message_, "RtApiAsio: driver (%s) data format not supported by RtAudio.",
- devices_[device].name.c_str());
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
}
- // 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_, "RtApiAsio: driver (%s) error getting buffer size.",
- devices_[device].name.c_str());
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
}
- 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( (double) *bufferSize ) / log10( 2.0 );
- *bufferSize = (int) pow( 2.0, floor(power+0.5) );
- if ( *bufferSize < minSize ) *bufferSize = minSize;
- else if ( *bufferSize > maxSize ) *bufferSize = maxSize;
- else *bufferSize = preferSize;
+ stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
+}
+
+void RtApiDs :: startStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiDs::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
}
- if ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize )
- std::cerr << "Possible input/output buffersize discrepancy!" << std::endl;
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
- stream_.bufferSize = *bufferSize;
- stream_.nBuffers = 2;
+ // Increase scheduler frequency on lesser windows (a side-effect of
+ // increasing timer accuracy). On greater windows (Win2K or later),
+ // this is already in effect.
+ timeBeginPeriod( 1 );
- // ASIO always uses deinterleaved channels.
- stream_.deInterleave[mode] = true;
+ buffersRolling = false;
+ duplexPrerollBytes = 0;
- // Allocate, if necessary, our AsioHandle structure for the stream.
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
- if ( handle == 0 ) {
- handle = (AsioHandle *) calloc(1, sizeof(AsioHandle));
- if ( handle == NULL ) {
- drivers.removeCurrentDriver();
- sprintf(message_, "RtApiAsio: error allocating AsioHandle memory (%s).",
- devices_[device].name.c_str());
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
- handle->bufferInfos = 0;
- // Create a manual-reset event.
- handle->condition = CreateEvent(NULL, // no security
- TRUE, // manual-reset
- FALSE, // non-signaled initially
- NULL); // unnamed
- stream_.apiHandle = (void *) handle;
+ 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] );
}
- // Create the ASIO internal buffers. Since RtAudio sets up input
- // and output separately, we'll have to dispose of previously
- // created output buffers for a duplex stream.
- if ( mode == INPUT && stream_.mode == OUTPUT ) {
- ASIODisposeBuffers();
- if ( handle->bufferInfos ) free( handle->bufferInfos );
- }
+ HRESULT result = 0;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
- // Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
- int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
- handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
- if (handle->bufferInfos == NULL) {
- sprintf(message_, "RtApiAsio: error allocating bufferInfo memory (%s).",
- devices_[device].name.c_str());
- goto error;
- }
- ASIOBufferInfo *infos;
- infos = handle->bufferInfos;
- for ( i=0; i<stream_.nDeviceChannels[0]; i++, infos++ ) {
- infos->isInput = ASIOFalse;
- infos->channelNum = i;
- infos->buffers[0] = infos->buffers[1] = 0;
- }
- for ( i=0; i<stream_.nDeviceChannels[1]; i++, infos++ ) {
- infos->isInput = ASIOTrue;
- infos->channelNum = i;
- infos->buffers[0] = infos->buffers[1] = 0;
+ 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;
+ }
}
- // 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 ) {
- sprintf(message_, "RtApiAsio: driver (%s) error creating buffers.",
- devices_[device].name.c_str());
- goto error;
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ result = buffer->Start( DSCBSTART_LOOPING );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
}
- // 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;
+ handle->drainCounter = 0;
+ handle->internalDrain = false;
+ ResetEvent( handle->condition );
+ stream_.state = STREAM_RUNNING;
- // Allocate necessary internal buffers
- if ( stream_.nUserChannels[0] != stream_.nUserChannels[1] ) {
+ unlock:
+ if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
+}
- 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) {
- sprintf(message_, "RtApiAsio: error allocating user buffer memory (%s).",
- devices_[device].name.c_str());
- goto error;
- }
+void RtApiDs :: stopStream()
+{
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
}
- if ( stream_.doConvertBuffer[mode] ) {
+ HRESULT result = 0;
+ LPVOID audioPtr;
+ DWORD dataLen;
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( handle->drainCounter == 0 ) {
+ handle->drainCounter = 2;
+ WaitForSingleObject( handle->condition, INFINITE ); // block until signaled
+ }
- 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;
- }
+ stream_.state = STREAM_STOPPED;
+
+ MUTEX_LOCK( &stream_.mutex );
+
+ // Stop the buffer and clear memory
+ LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ result = buffer->Stop();
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
- if ( makeBuffer ) {
- buffer_bytes *= *bufferSize;
- if (stream_.deviceBuffer) free(stream_.deviceBuffer);
- stream_.deviceBuffer = (char *) calloc(buffer_bytes, 1);
- if (stream_.deviceBuffer == NULL) {
- sprintf(message_, "RtApiAsio: error allocating device buffer memory (%s).",
- devices_[device].name.c_str());
- goto 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::stopStream: error (" << getErrorString( result ) << ") locking output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // Zero the DS buffer
+ ZeroMemory( audioPtr, dataLen );
+
+ // Unlock the DS buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking output buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
+
+ // If we start playing again, we must begin at beginning of buffer.
+ handle->bufferPointer[0] = 0;
}
- 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;
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ audioPtr = NULL;
+ dataLen = 0;
+
+ stream_.state = STREAM_STOPPED;
+
+ if ( stream_.mode != DUPLEX )
+ MUTEX_LOCK( &stream_.mutex );
+
+ result = buffer->Stop();
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
+
+ // Lock the buffer and clear it so that if we start to play again,
+ // we won't have old data playing.
+ result = buffer->Lock( 0, handle->dsBufferSize[1], &audioPtr, &dataLen, NULL, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") locking input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
- return SUCCESS;
+ // Zero the DS buffer
+ ZeroMemory( audioPtr, dataLen );
- error:
- ASIODisposeBuffers();
- drivers.removeCurrentDriver();
+ // Unlock the DS buffer
+ result = buffer->Unlock( audioPtr, dataLen, NULL, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") unlocking input buffer!";
+ errorText_ = errorStream_.str();
+ goto unlock;
+ }
- if ( handle ) {
- CloseHandle( handle->condition );
- if ( handle->bufferInfos )
- free( handle->bufferInfos );
- free( handle );
- stream_.apiHandle = 0;
+ // If we start recording again, we must begin at beginning of buffer.
+ handle->bufferPointer[1] = 0;
}
- if (stream_.userBuffer) {
- free(stream_.userBuffer);
- stream_.userBuffer = 0;
- }
+ unlock:
+ timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
+ MUTEX_UNLOCK( &stream_.mutex );
- error(RtError::WARNING);
- return FAILURE;
+ if ( FAILED( result ) ) error( RtAudioError::SYSTEM_ERROR );
}
-void RtApiAsio :: closeStream()
+void RtApiDs :: abortStream()
{
- // 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 ( stream_.mode == UNINITIALIZED ) {
- sprintf(message_, "RtApiAsio::closeStream(): no open stream to close!");
- error(RtError::WARNING);
+ verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
return;
}
- if (stream_.state == STREAM_RUNNING)
- ASIOStop();
-
- ASIODisposeBuffers();
- drivers.removeCurrentDriver();
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
+ handle->drainCounter = 2;
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
- if ( handle ) {
- CloseHandle( handle->condition );
- if ( handle->bufferInfos )
- free( handle->bufferInfos );
- free( handle );
- stream_.apiHandle = 0;
- }
+ stopStream();
+}
- if (stream_.userBuffer) {
- free(stream_.userBuffer);
- stream_.userBuffer = 0;
+void RtApiDs :: callbackEvent()
+{
+ if ( stream_.state == STREAM_STOPPED || stream_.state == STREAM_STOPPING ) {
+ Sleep( 50 ); // sleep 50 milliseconds
+ return;
}
- if (stream_.deviceBuffer) {
- free(stream_.deviceBuffer);
- stream_.deviceBuffer = 0;
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return;
}
- stream_.mode = UNINITIALIZED;
-}
+ CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
+ DsHandle *handle = (DsHandle *) stream_.apiHandle;
-void RtApiAsio :: setStreamCallback(RtAudioCallback callback, void *userData)
-{
- verifyStream();
+ // Check if we were draining the stream and signal is finished.
+ if ( handle->drainCounter > stream_.nBuffers + 2 ) {
- if ( stream_.callbackInfo.usingCallback ) {
- sprintf(message_, "RtApiAsio: A callback is already set for this stream!");
- error(RtError::WARNING);
+ stream_.state = STREAM_STOPPING;
+ if ( handle->internalDrain == false )
+ SetEvent( handle->condition );
+ else
+ stopStream();
return;
}
- stream_.callbackInfo.callback = (void *) callback;
- stream_.callbackInfo.userData = userData;
- stream_.callbackInfo.usingCallback = true;
-}
+ // Invoke user callback to get fresh output data UNLESS we are
+ // draining stream.
+ if ( handle->drainCounter == 0 ) {
+ RtAudioCallback callback = (RtAudioCallback) info->callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
+ }
+ int cbReturnValue = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, info->userData );
+ if ( cbReturnValue == 2 ) {
+ stream_.state = STREAM_STOPPING;
+ handle->drainCounter = 2;
+ abortStream();
+ return;
+ }
+ else if ( cbReturnValue == 1 ) {
+ handle->drainCounter = 1;
+ handle->internalDrain = true;
+ }
+ }
-void RtApiAsio :: cancelStreamCallback()
-{
- verifyStream();
+ HRESULT result;
+ DWORD currentWritePointer, safeWritePointer;
+ DWORD currentReadPointer, safeReadPointer;
+ UINT nextWritePointer;
- if (stream_.callbackInfo.usingCallback) {
+ LPVOID buffer1 = NULL;
+ LPVOID buffer2 = NULL;
+ DWORD bufferSize1 = 0;
+ DWORD bufferSize2 = 0;
- if (stream_.state == STREAM_RUNNING)
- stopStream();
+ char *buffer;
+ long bufferBytes;
- MUTEX_LOCK(&stream_.mutex);
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
- stream_.callbackInfo.usingCallback = false;
- stream_.callbackInfo.userData = NULL;
- stream_.state = STREAM_STOPPED;
- stream_.callbackInfo.callback = NULL;
+ if ( buffersRolling == false ) {
+ if ( stream_.mode == DUPLEX ) {
+ //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
+
+ // It takes a while for the devices to get rolling. As a result,
+ // there's no guarantee that the capture and write device pointers
+ // will move in lockstep. Wait here for both devices to start
+ // rolling, and then set our buffer pointers accordingly.
+ // e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
+ // bytes later than the write buffer.
+
+ // Stub: a serious risk of having a pre-emptive scheduling round
+ // take place between the two GetCurrentPosition calls... but I'm
+ // really not sure how to solve the problem. Temporarily boost to
+ // Realtime priority, maybe; but I'm not sure what priority the
+ // DirectSound service threads run at. We *should* be roughly
+ // within a ms or so of correct.
+
+ LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+
+ DWORD startSafeWritePointer, startSafeReadPointer;
+
+ result = dsWriteBuffer->GetCurrentPosition( NULL, &startSafeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ result = dsCaptureBuffer->GetCurrentPosition( NULL, &startSafeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ while ( true ) {
+ result = dsWriteBuffer->GetCurrentPosition( NULL, &safeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ result = dsCaptureBuffer->GetCurrentPosition( NULL, &safeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ if ( safeWritePointer != startSafeWritePointer && safeReadPointer != startSafeReadPointer ) break;
+ Sleep( 1 );
+ }
- MUTEX_UNLOCK(&stream_.mutex);
- }
-}
+ //assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
-void RtApiAsio :: startStream()
-{
- verifyStream();
- if (stream_.state == STREAM_RUNNING) return;
+ handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
+ if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
+ handle->bufferPointer[1] = safeReadPointer;
+ }
+ else if ( stream_.mode == OUTPUT ) {
- MUTEX_LOCK(&stream_.mutex);
+ // Set the proper nextWritePosition after initial startup.
+ LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
+ result = dsWriteBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ handle->bufferPointer[0] = safeWritePointer + handle->dsPointerLeadTime[0];
+ if ( handle->bufferPointer[0] >= handle->dsBufferSize[0] ) handle->bufferPointer[0] -= handle->dsBufferSize[0];
+ }
- ASIOError result = ASIOStart();
- if ( result != ASE_OK ) {
- sprintf(message_, "RtApiAsio: error starting device (%s).",
- devices_[stream_.device[0]].name.c_str());
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
+ buffersRolling = true;
}
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
- handle->stopStream = false;
- stream_.state = STREAM_RUNNING;
- MUTEX_UNLOCK(&stream_.mutex);
-}
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
-void RtApiAsio :: stopStream()
-{
- verifyStream();
- if (stream_.state == STREAM_STOPPED) return;
+ 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 );
+ }
- // Change the state before the lock to improve shutdown response
- // when using a callback.
- stream_.state = STREAM_STOPPED;
- MUTEX_LOCK(&stream_.mutex);
+ // 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 );
+ }
- ASIOError result = ASIOStop();
- if ( result != ASE_OK ) {
- sprintf(message_, "RtApiAsio: error stopping device (%s).",
- devices_[stream_.device[0]].name.c_str());
- MUTEX_UNLOCK(&stream_.mutex);
- error(RtError::DRIVER_ERROR);
- }
+ // No byte swapping necessary in DirectSound implementation.
- MUTEX_UNLOCK(&stream_.mutex);
-}
+ // 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 );
-void RtApiAsio :: abortStream()
-{
- stopStream();
-}
+ DWORD dsBufferSize = handle->dsBufferSize[0];
+ nextWritePointer = handle->bufferPointer[0];
-void RtApiAsio :: tickStream()
-{
- verifyStream();
+ DWORD endWrite, leadPointer;
+ while ( true ) {
+ // Find out where the read and "safe write" pointers are.
+ result = dsBuffer->GetCurrentPosition( ¤tWritePointer, &safeWritePointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
- if (stream_.state == STREAM_STOPPED)
- return;
+ // We will copy our output buffer into the region between
+ // safeWritePointer and leadPointer. If leadPointer is not
+ // beyond the next endWrite position, wait until it is.
+ leadPointer = safeWritePointer + handle->dsPointerLeadTime[0];
+ //std::cout << "safeWritePointer = " << safeWritePointer << ", leadPointer = " << leadPointer << ", nextWritePointer = " << nextWritePointer << std::endl;
+ if ( leadPointer > dsBufferSize ) leadPointer -= dsBufferSize;
+ if ( leadPointer < nextWritePointer ) leadPointer += dsBufferSize; // unwrap offset
+ endWrite = nextWritePointer + bufferBytes;
- if (stream_.callbackInfo.usingCallback) {
- sprintf(message_, "RtApiAsio: tickStream() should not be used when a callback function is set!");
- error(RtError::WARNING);
- return;
- }
+ // Check whether the entire write region is behind the play pointer.
+ if ( leadPointer >= endWrite ) break;
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
+ // If we are here, then we must wait until the leadPointer advances
+ // beyond the end of our next write region. We use the
+ // Sleep() function to suspend operation until that happens.
+ double millis = ( endWrite - leadPointer ) * 1000.0;
+ millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
+ if ( millis < 1.0 ) millis = 1.0;
+ Sleep( (DWORD) millis );
+ }
- MUTEX_LOCK(&stream_.mutex);
+ if ( dsPointerBetween( nextWritePointer, safeWritePointer, currentWritePointer, dsBufferSize )
+ || dsPointerBetween( endWrite, safeWritePointer, currentWritePointer, dsBufferSize ) ) {
+ // We've strayed into the forbidden zone ... resync the read pointer.
+ handle->xrun[0] = true;
+ nextWritePointer = safeWritePointer + handle->dsPointerLeadTime[0] - bufferBytes;
+ if ( nextWritePointer >= dsBufferSize ) nextWritePointer -= dsBufferSize;
+ handle->bufferPointer[0] = nextWritePointer;
+ endWrite = nextWritePointer + bufferBytes;
+ }
- // Release the stream_mutex here and wait for the event
- // to become signaled by the callback process.
- MUTEX_UNLOCK(&stream_.mutex);
- WaitForMultipleObjects(1, &handle->condition, FALSE, INFINITE);
- ResetEvent( handle->condition );
-}
+ // Lock free space in the buffer
+ result = dsBuffer->Lock( nextWritePointer, bufferBytes, &buffer1,
+ &bufferSize1, &buffer2, &bufferSize2, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
-void RtApiAsio :: callbackEvent(long bufferIndex)
-{
- verifyStream();
+ // Copy our buffer into the DS buffer
+ CopyMemory( buffer1, buffer, bufferSize1 );
+ if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
- if (stream_.state == STREAM_STOPPED) return;
+ // Update our buffer offset and unlock sound buffer
+ dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ nextWritePointer = ( nextWritePointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
+ handle->bufferPointer[0] = nextWritePointer;
+ }
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
- AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
- if ( info->usingCallback && handle->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();
- return;
+ // Don't bother draining input
+ if ( handle->drainCounter ) {
+ handle->drainCounter++;
+ goto unlock;
}
- MUTEX_LOCK(&stream_.mutex);
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
- // Invoke user callback first, to get fresh output data.
- if ( info->usingCallback ) {
- RtAudioCallback callback = (RtAudioCallback) info->callback;
- if ( callback(stream_.userBuffer, stream_.bufferSize, info->userData) )
- handle->stopStream = true;
- }
+ // 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 );
+ }
- int bufferBytes, j;
- int nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
- if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
+ long nextReadPointer = handle->bufferPointer[1];
+ DWORD dsBufferSize = handle->dsBufferSize[1];
+
+ // Find out where the write and "safe read" pointers are.
+ result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
- bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[0]);
- if (stream_.doConvertBuffer[0]) {
+ if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
+ DWORD endRead = nextReadPointer + bufferBytes;
+
+ // Handling depends on whether we are INPUT or DUPLEX.
+ // If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
+ // then a wait here will drag the write pointers into the forbidden zone.
+ //
+ // In DUPLEX mode, rather than wait, we will back off the read pointer until
+ // it's in a safe position. This causes dropouts, but it seems to be the only
+ // practical way to sync up the read and write pointers reliably, given the
+ // the very complex relationship between phase and increment of the read and write
+ // pointers.
+ //
+ // In order to minimize audible dropouts in DUPLEX mode, we will
+ // provide a pre-roll period of 0.5 seconds in which we return
+ // zeros from the read buffer while the pointers sync up.
+
+ if ( stream_.mode == DUPLEX ) {
+ if ( safeReadPointer < endRead ) {
+ if ( duplexPrerollBytes <= 0 ) {
+ // Pre-roll time over. Be more agressive.
+ int adjustment = endRead-safeReadPointer;
+
+ handle->xrun[1] = true;
+ // Two cases:
+ // - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
+ // and perform fine adjustments later.
+ // - small adjustments: back off by twice as much.
+ if ( adjustment >= 2*bufferBytes )
+ nextReadPointer = safeReadPointer-2*bufferBytes;
+ else
+ nextReadPointer = safeReadPointer-bufferBytes-adjustment;
- convertStreamBuffer(OUTPUT);
- if ( stream_.doByteSwap[0] )
- byteSwapBuffer(stream_.deviceBuffer,
- stream_.bufferSize * stream_.nDeviceChannels[0],
- stream_.deviceFormat[0]);
+ if ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
- // Always de-interleave ASIO output data.
- j = 0;
- for ( int i=0; i<nChannels; i++ ) {
- if ( handle->bufferInfos[i].isInput != ASIOTrue )
- memcpy(handle->bufferInfos[i].buffers[bufferIndex],
- &stream_.deviceBuffer[j++*bufferBytes], bufferBytes );
+ }
+ else {
+ // In pre=roll time. Just do it.
+ nextReadPointer = safeReadPointer - bufferBytes;
+ while ( nextReadPointer < 0 ) nextReadPointer += dsBufferSize;
+ }
+ endRead = nextReadPointer + bufferBytes;
}
}
- else { // single channel only
-
- if (stream_.doByteSwap[0])
- byteSwapBuffer(stream_.userBuffer,
- stream_.bufferSize * stream_.nUserChannels[0],
- stream_.userFormat);
+ else { // mode == INPUT
+ while ( safeReadPointer < endRead && stream_.callbackInfo.isRunning ) {
+ // See comments for playback.
+ double millis = (endRead - safeReadPointer) * 1000.0;
+ millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
+ if ( millis < 1.0 ) millis = 1.0;
+ Sleep( (DWORD) millis );
- for ( int i=0; i<nChannels; i++ ) {
- if ( handle->bufferInfos[i].isInput != ASIOTrue ) {
- memcpy(handle->bufferInfos[i].buffers[bufferIndex], stream_.userBuffer, bufferBytes );
- break;
+ // Wake up and find out where we are now.
+ result = dsBuffer->GetCurrentPosition( ¤tReadPointer, &safeReadPointer );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
}
+
+ if ( safeReadPointer < (DWORD)nextReadPointer ) safeReadPointer += dsBufferSize; // unwrap offset
}
}
+
+ // Lock free space in the buffer
+ result = dsBuffer->Lock( nextReadPointer, bufferBytes, &buffer1,
+ &bufferSize1, &buffer2, &bufferSize2, 0 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+
+ if ( duplexPrerollBytes <= 0 ) {
+ // Copy our buffer into the DS buffer
+ CopyMemory( buffer, buffer1, bufferSize1 );
+ if ( buffer2 != NULL ) CopyMemory( buffer+bufferSize1, buffer2, bufferSize2 );
+ }
+ else {
+ memset( buffer, 0, bufferSize1 );
+ if ( buffer2 != NULL ) memset( buffer + bufferSize1, 0, bufferSize2 );
+ duplexPrerollBytes -= bufferSize1 + bufferSize2;
+ }
+
+ // Update our buffer offset and unlock sound buffer
+ nextReadPointer = ( nextReadPointer + bufferSize1 + bufferSize2 ) % dsBufferSize;
+ dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
+ if ( FAILED( result ) ) {
+ errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ handle->bufferPointer[1] = nextReadPointer;
+
+ // No byte swapping necessary in DirectSound implementation.
+
+ // If necessary, convert 8-bit data from unsigned to signed.
+ if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
+ for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
+
+ // Do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[1] )
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
}
- if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
+ RtApi::tickStreamTime();
+}
- bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
- if (stream_.doConvertBuffer[1]) {
+// Definitions for utility functions and callbacks
+// specific to the DirectSound implementation.
- // Always interleave ASIO input data.
- j = 0;
- for ( int i=0; i<nChannels; i++ ) {
- if ( handle->bufferInfos[i].isInput == ASIOTrue )
- memcpy(&stream_.deviceBuffer[j++*bufferBytes],
- handle->bufferInfos[i].buffers[bufferIndex],
- bufferBytes );
- }
+static unsigned __stdcall callbackHandler( void *ptr )
+{
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiDs *object = (RtApiDs *) info->object;
+ bool* isRunning = &info->isRunning;
- if ( stream_.doByteSwap[1] )
- byteSwapBuffer(stream_.deviceBuffer,
- stream_.bufferSize * stream_.nDeviceChannels[1],
- stream_.deviceFormat[1]);
- convertStreamBuffer(INPUT);
+ while ( *isRunning == true ) {
+ object->callbackEvent();
+ }
+
+ _endthreadex( 0 );
+ return 0;
+}
+
+static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
+ LPCTSTR description,
+ LPCTSTR /*module*/,
+ LPVOID lpContext )
+{
+ struct DsProbeData& probeInfo = *(struct DsProbeData*) lpContext;
+ std::vector<struct DsDevice>& dsDevices = *probeInfo.dsDevices;
+
+ HRESULT hr;
+ bool validDevice = false;
+ if ( probeInfo.isInput == true ) {
+ DSCCAPS caps;
+ LPDIRECTSOUNDCAPTURE object;
+ hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
+ if ( hr != DS_OK ) return TRUE;
+
+ caps.dwSize = sizeof(caps);
+ hr = object->GetCaps( &caps );
+ if ( hr == DS_OK ) {
+ if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
+ validDevice = true;
}
- else { // single channel only
- for ( int i=0; i<nChannels; i++ ) {
- if ( handle->bufferInfos[i].isInput == ASIOTrue ) {
- memcpy(stream_.userBuffer,
- handle->bufferInfos[i].buffers[bufferIndex],
- bufferBytes );
- break;
+ object->Release();
+ }
+ else {
+ DSCAPS caps;
+ LPDIRECTSOUND object;
+ hr = DirectSoundCreate( lpguid, &object, NULL );
+ if ( hr != DS_OK ) return TRUE;
+
+ caps.dwSize = sizeof(caps);
+ hr = object->GetCaps( &caps );
+ if ( hr == DS_OK ) {
+ if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
+ validDevice = true;
+ }
+ object->Release();
+ }
+
+ // If good device, then save its name and guid.
+ std::string name = convertCharPointerToStdString( description );
+ //if ( name == "Primary Sound Driver" || name == "Primary Sound Capture Driver" )
+ if ( lpguid == NULL )
+ name = "Default Device";
+ if ( validDevice ) {
+ for ( unsigned int i=0; i<dsDevices.size(); i++ ) {
+ if ( dsDevices[i].name == name ) {
+ dsDevices[i].found = true;
+ if ( probeInfo.isInput ) {
+ dsDevices[i].id[1] = lpguid;
+ dsDevices[i].validId[1] = true;
+ }
+ else {
+ dsDevices[i].id[0] = lpguid;
+ dsDevices[i].validId[0] = true;
}
+ return TRUE;
}
+ }
- if (stream_.doByteSwap[1])
- byteSwapBuffer(stream_.userBuffer,
- stream_.bufferSize * stream_.nUserChannels[1],
- stream_.userFormat);
+ DsDevice device;
+ device.name = name;
+ device.found = true;
+ if ( probeInfo.isInput ) {
+ device.id[1] = lpguid;
+ device.validId[1] = true;
+ }
+ else {
+ device.id[0] = lpguid;
+ device.validId[0] = true;
}
+ dsDevices.push_back( device );
}
- if ( !info->usingCallback )
- SetEvent( handle->condition );
-
- MUTEX_UNLOCK(&stream_.mutex);
+ return TRUE;
}
-//******************** End of __WINDOWS_ASIO__ *********************//
-#endif
+static const char* getErrorString( int code )
+{
+ switch ( code ) {
-#if defined(__WINDOWS_DS__) // Windows DirectSound API
+ case DSERR_ALLOCATED:
+ return "Already allocated";
-#include <dsound.h>
+ case DSERR_CONTROLUNAVAIL:
+ return "Control unavailable";
-// A structure to hold various information related to the DirectSound
-// API implementation.
-struct DsHandle {
- void *object;
- void *buffer;
- UINT bufferPointer;
-};
+ case DSERR_INVALIDPARAM:
+ return "Invalid parameter";
-// Declarations for utility functions, callbacks, and structures
-// specific to the DirectSound implementation.
-static bool CALLBACK deviceCountCallback(LPGUID lpguid,
- LPCSTR lpcstrDescription,
- LPCSTR lpcstrModule,
- LPVOID lpContext);
+ 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";
-static bool CALLBACK deviceInfoCallback(LPGUID lpguid,
- LPCSTR lpcstrDescription,
- LPCSTR lpcstrModule,
- LPVOID lpContext);
+ case DSERR_UNINITIALIZED:
+ return "Uninitialized";
-static bool CALLBACK defaultDeviceCallback(LPGUID lpguid,
- LPCSTR lpcstrDescription,
- LPCSTR lpcstrModule,
- LPVOID lpContext);
+ default:
+ return "DirectSound unknown error";
+ }
+}
+//******************** End of __WINDOWS_DS__ *********************//
+#endif
-static bool CALLBACK deviceIdCallback(LPGUID lpguid,
- LPCSTR lpcstrDescription,
- LPCSTR lpcstrModule,
- LPVOID lpContext);
-static char* getErrorString(int code);
+#if defined(__LINUX_ALSA__)
-extern "C" unsigned __stdcall callbackHandler(void *ptr);
+#include <alsa/asoundlib.h>
+#include <unistd.h>
-struct enum_info {
- char name[64];
- LPGUID id;
- bool isInput;
- bool isValid;
+ // A structure to hold various information related to the ALSA API
+ // implementation.
+struct AlsaHandle {
+ snd_pcm_t *handles[2];
+ bool synchronized;
+ bool xrun[2];
+ pthread_cond_t runnable_cv;
+ bool runnable;
+
+ AlsaHandle()
+ :synchronized(false), runnable(false) { xrun[0] = false; xrun[1] = false; }
};
-RtApiDs :: RtApiDs()
-{
- this->initialize();
+static void *alsaCallbackHandler( void * ptr );
- if (nDevices_ <= 0) {
- sprintf(message_, "RtApiDs: no Windows DirectSound audio devices found!");
- error(RtError::NO_DEVICES_FOUND);
- }
+RtApiAlsa :: RtApiAlsa()
+{
+ // Nothing to do here.
}
-RtApiDs :: ~RtApiDs()
+RtApiAlsa :: ~RtApiAlsa()
{
- if ( stream_.mode != UNINITIALIZED ) closeStream();
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
}
-int RtApiDs :: getDefaultInputDevice(void)
+unsigned int RtApiAlsa :: getDeviceCount( void )
{
- enum_info info;
- info.name[0] = '\0';
+ unsigned nDevices = 0;
+ int result, subdevice, card;
+ char name[64];
+ snd_ctl_t *handle;
- // Enumerate through devices to find the default output.
- HRESULT result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)defaultDeviceCallback, &info);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Error performing default input device enumeration: %s.",
- getErrorString(result));
- error(RtError::WARNING);
- return 0;
+ // Count cards and devices
+ card = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &handle, name, 0 );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto nextcard;
+ }
+ subdevice = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( handle, &subdevice );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ break;
+ }
+ if ( subdevice < 0 )
+ break;
+ nDevices++;
+ }
+ nextcard:
+ snd_ctl_close( handle );
+ snd_card_next( &card );
}
- for ( int i=0; i<nDevices_; i++ ) {
- if ( strncmp( info.name, devices_[i].name.c_str(), 64 ) == 0 ) return i;
+ result = snd_ctl_open( &handle, "default", 0 );
+ if (result == 0) {
+ nDevices++;
+ snd_ctl_close( handle );
}
-
- return 0;
+ return nDevices;
}
-int RtApiDs :: getDefaultOutputDevice(void)
+RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
{
- 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_, "RtApiDs: Error performing default output device enumeration: %s.",
- getErrorString(result));
- error(RtError::WARNING);
- return 0;
- }
+ RtAudio::DeviceInfo info;
+ info.probed = false;
- for ( int i=0; i<nDevices_; i++ )
- if ( strncmp( info.name, devices_[i].name.c_str(), 64 ) == 0 ) return i;
-
- return 0;
-}
-
-void RtApiDs :: initialize(void)
-{
- int i, ins = 0, outs = 0, count = 0;
- HRESULT result;
- nDevices_ = 0;
+ unsigned nDevices = 0;
+ int result, subdevice, card;
+ char name[64];
+ snd_ctl_t *chandle;
- // Count DirectSound devices.
- result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceCountCallback, &outs);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to enumerate through sound playback devices: %s.",
- getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ // Count cards and devices
+ card = -1;
+ subdevice = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto nextcard;
+ }
+ subdevice = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( chandle, &subdevice );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ break;
+ }
+ if ( subdevice < 0 ) break;
+ if ( nDevices == device ) {
+ sprintf( name, "hw:%d,%d", card, subdevice );
+ goto foundDevice;
+ }
+ nDevices++;
+ }
+ nextcard:
+ snd_ctl_close( chandle );
+ snd_card_next( &card );
}
- // Count DirectSoundCapture devices.
- result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceCountCallback, &ins);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to enumerate through sound capture devices: %s.",
- getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
+ if ( result == 0 ) {
+ if ( nDevices == device ) {
+ strcpy( name, "default" );
+ goto foundDevice;
+ }
+ nDevices++;
}
- count = ins + outs;
- if (count == 0) return;
-
- 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;
+ if ( nDevices == 0 ) {
+ errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
+ error( RtAudioError::INVALID_USE );
+ return info;
}
- // Get playback device info and check capabilities.
- result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceInfoCallback, &info[0]);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to enumerate through sound playback devices: %s.",
- getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ if ( device >= nDevices ) {
+ errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
}
- // Get capture device info and check capabilities.
- result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceInfoCallback, &info[0]);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to enumerate through sound capture devices: %s.",
- getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
+ foundDevice:
- // Create device structures for valid devices and write device names
- // to each. Devices are considered invalid if they cannot be
- // opened, they report < 1 supported channels, or they report no
- // supported data (capture only).
- RtApiDevice device;
- int index = 0;
- for (i=0; i<count; i++) {
- if ( info[i].isValid ) {
- device.name.erase();
- device.name.append( (const char *)info[i].name, strlen(info[i].name)+1);
- devices_.push_back(device);
+ // If a stream is already open, we cannot probe the stream devices.
+ // Thus, use the saved results.
+ if ( stream_.state != STREAM_CLOSED &&
+ ( stream_.device[0] == device || stream_.device[1] == device ) ) {
+ snd_ctl_close( chandle );
+ if ( device >= devices_.size() ) {
+ errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
+ error( RtAudioError::WARNING );
+ return info;
}
+ return devices_[ device ];
}
- nDevices_ = devices_.size();
- return;
-}
+ 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 );
-void RtApiDs :: probeDeviceInfo(RtApiDevice *info)
-{
- enum_info dsinfo;
- strncpy( dsinfo.name, info->name.c_str(), 64 );
- dsinfo.isValid = false;
+ // First try for playback unless default device (which has subdev -1)
+ stream = SND_PCM_STREAM_PLAYBACK;
+ snd_pcm_info_set_stream( pcminfo, stream );
+ if ( subdevice != -1 ) {
+ snd_pcm_info_set_device( pcminfo, subdevice );
+ snd_pcm_info_set_subdevice( pcminfo, 0 );
- // Enumerate through input devices to find the id (if it exists).
- HRESULT result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Error performing input device id enumeration: %s.",
- getErrorString(result));
- error(RtError::WARNING);
- return;
+ result = snd_ctl_pcm_info( chandle, pcminfo );
+ if ( result < 0 ) {
+ // Device probably doesn't support playback.
+ goto captureProbe;
+ }
}
- // Do capture probe first.
- if ( dsinfo.isValid == false )
- goto playback_probe;
-
- LPDIRECTSOUNDCAPTURE input;
- result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL );
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Could not create capture object (%s): %s.",
- info->name.c_str(), getErrorString(result));
- error(RtError::WARNING);
- goto playback_probe;
+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto captureProbe;
}
- DSCCAPS in_caps;
- in_caps.dwSize = sizeof(in_caps);
- result = input->GetCaps( &in_caps );
- if ( FAILED(result) ) {
- input->Release();
- sprintf(message_, "RtApiDs: Could not get capture capabilities (%s): %s.",
- info->name.c_str(), getErrorString(result));
- error(RtError::WARNING);
- goto playback_probe;
+ // The device is open ... fill the parameter structure.
+ result = snd_pcm_hw_params_any( phandle, params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto captureProbe;
}
- // Get input channel information.
- info->minInputChannels = 1;
- info->maxInputChannels = in_caps.dwChannels;
+ // Get output channel information.
+ unsigned int value;
+ result = snd_pcm_hw_params_get_channels_max( params, &value );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ goto captureProbe;
+ }
+ info.outputChannels = value;
+ snd_pcm_close( phandle );
- // Get sample rate and format information.
- info->sampleRates.clear();
- 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.push_back( 11025 );
- if( in_caps.dwFormats & WAVE_FORMAT_2S16 ) info->sampleRates.push_back( 22050 );
- if( in_caps.dwFormats & WAVE_FORMAT_4S16 ) info->sampleRates.push_back( 44100 );
- }
- else if ( info->nativeFormats & RTAUDIO_SINT8 ) {
- if( in_caps.dwFormats & WAVE_FORMAT_1S08 ) info->sampleRates.push_back( 11025 );
- if( in_caps.dwFormats & WAVE_FORMAT_2S08 ) info->sampleRates.push_back( 22050 );
- if( in_caps.dwFormats & WAVE_FORMAT_4S08 ) info->sampleRates.push_back( 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.push_back( 11025 );
- if( in_caps.dwFormats & WAVE_FORMAT_2M16 ) info->sampleRates.push_back( 22050 );
- if( in_caps.dwFormats & WAVE_FORMAT_4M16 ) info->sampleRates.push_back( 44100 );
- }
- else if ( info->nativeFormats & RTAUDIO_SINT8 ) {
- if( in_caps.dwFormats & WAVE_FORMAT_1M08 ) info->sampleRates.push_back( 11025 );
- if( in_caps.dwFormats & WAVE_FORMAT_2M08 ) info->sampleRates.push_back( 22050 );
- if( in_caps.dwFormats & WAVE_FORMAT_4M08 ) info->sampleRates.push_back( 44100 );
- }
- }
- else info->minInputChannels = 0; // technically, this would be an error
+ captureProbe:
+ stream = SND_PCM_STREAM_CAPTURE;
+ snd_pcm_info_set_stream( pcminfo, stream );
- input->Release();
+ // Now try for capture unless default device (with subdev = -1)
+ if ( subdevice != -1 ) {
+ result = snd_ctl_pcm_info( chandle, pcminfo );
+ snd_ctl_close( chandle );
+ if ( result < 0 ) {
+ // Device probably doesn't support capture.
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
+ }
+ else
+ snd_ctl_close( chandle );
- playback_probe:
+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
- dsinfo.isValid = false;
+ // The device is open ... fill the parameter structure.
+ result = snd_pcm_hw_params_any( phandle, params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
+ }
- // Enumerate through output devices to find the id (if it exists).
- result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Error performing output device id enumeration: %s.",
- getErrorString(result));
- error(RtError::WARNING);
- return;
+ result = snd_pcm_hw_params_get_channels_max( params, &value );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ if ( info.outputChannels == 0 ) return info;
+ goto probeParameters;
}
+ info.inputChannels = value;
+ snd_pcm_close( phandle );
- // Now do playback probe.
- if ( dsinfo.isValid == false )
- goto check_parameters;
+ // 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;
- LPDIRECTSOUND output;
- DSCAPS out_caps;
- result = DirectSoundCreate( dsinfo.id, &output, NULL );
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Could not create playback object (%s): %s.",
- info->name.c_str(), getErrorString(result));
- error(RtError::WARNING);
- goto check_parameters;
- }
+ // 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;
- out_caps.dwSize = sizeof(out_caps);
- result = output->GetCaps( &out_caps );
- if ( FAILED(result) ) {
- output->Release();
- sprintf(message_, "RtApiDs: Could not get playback capabilities (%s): %s.",
- info->name.c_str(), getErrorString(result));
- error(RtError::WARNING);
- goto check_parameters;
- }
+ 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.
- // Get output channel information.
- info->minOutputChannels = 1;
- info->maxOutputChannels = ( out_caps.dwFlags & DSCAPS_PRIMARYSTEREO ) ? 2 : 1;
+ if ( info.outputChannels >= info.inputChannels )
+ stream = SND_PCM_STREAM_PLAYBACK;
+ else
+ stream = SND_PCM_STREAM_CAPTURE;
+ snd_pcm_info_set_stream( pcminfo, stream );
- // Get sample rate information. Use capture device rate information
- // if it exists.
- if ( info->sampleRates.size() == 0 ) {
- info->sampleRates.push_back( (int) out_caps.dwMinSecondarySampleRate );
- info->sampleRates.push_back( (int) out_caps.dwMaxSecondarySampleRate );
- }
- else {
- // Check input rates against output rate range.
- for ( int i=info->sampleRates.size()-1; i>=0; i-- ) {
- if ( (unsigned int) info->sampleRates[i] > out_caps.dwMaxSecondarySampleRate )
- info->sampleRates.erase( info->sampleRates.begin() + i );
- }
- while ( info->sampleRates.size() > 0 &&
- ((unsigned int) info->sampleRates[0] < out_caps.dwMinSecondarySampleRate) ) {
- info->sampleRates.erase( info->sampleRates.begin() );
- }
+ result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
}
- // Get format information.
- if ( out_caps.dwFlags & DSCAPS_PRIMARY16BIT ) info->nativeFormats |= RTAUDIO_SINT16;
- if ( out_caps.dwFlags & DSCAPS_PRIMARY8BIT ) info->nativeFormats |= RTAUDIO_SINT8;
+ // The device is open ... fill the parameter structure.
+ result = snd_pcm_hw_params_any( phandle, params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
- output->Release();
+ // 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] );
- check_parameters:
- if ( info->maxInputChannels == 0 && info->maxOutputChannels == 0 ) {
- sprintf(message_, "RtApiDs: no reported input or output channels for device (%s).",
- info->name.c_str());
- error(RtError::DEBUG_WARNING);
- return;
+ if ( !info.preferredSampleRate || ( SAMPLE_RATES[i] <= 48000 && SAMPLE_RATES[i] > info.preferredSampleRate ) )
+ info.preferredSampleRate = SAMPLE_RATES[i];
+ }
}
- if ( info->sampleRates.size() == 0 || info->nativeFormats == 0 ) {
- sprintf(message_, "RtApiDs: no reported sample rates or data formats for device (%s).",
- info->name.c_str());
- error(RtError::DEBUG_WARNING);
- return;
+ if ( info.sampleRates.size() == 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
}
- // 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;
+ // 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;
- if ( info->maxDuplexChannels > 0 ) info->hasDuplexSupport = true;
- else info->hasDuplexSupport = false;
+ // Check that we have at least one supported format
+ if ( info.nativeFormats == 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
- info->probed = true;
+ // Get the device name
+ char *cardname;
+ result = snd_card_get_name( card, &cardname );
+ if ( result >= 0 ) {
+ sprintf( name, "hw:%s,%d", cardname, subdevice );
+ free( cardname );
+ }
+ info.name = name;
- return;
+ // That's all ... close the device and return
+ snd_pcm_close( phandle );
+ info.probed = true;
+ return info;
}
-bool RtApiDs :: probeDeviceOpen( int device, StreamMode mode, int channels,
- int sampleRate, RtAudioFormat format,
- int *bufferSize, int numberOfBuffers)
+void RtApiAlsa :: saveDeviceInfo( void )
{
- 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;
+ devices_.clear();
- // 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;
+ unsigned int nDevices = getDeviceCount();
+ devices_.resize( nDevices );
+ for ( unsigned int i=0; i<nDevices; i++ )
+ devices_[i] = getDeviceInfo( i );
+}
- // Define the wave format structure (16-bit PCM, srate, channels)
- WAVEFORMATEX waveFormat;
- ZeroMemory(&waveFormat, sizeof(WAVEFORMATEX));
- waveFormat.wFormatTag = WAVE_FORMAT_PCM;
- waveFormat.nChannels = channels;
- waveFormat.nSamplesPerSec = (unsigned long) sampleRate;
+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 )
- // 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_, "RtApiDs: no reported data formats for device (%s).",
- devices_[device].name.c_str());
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
+{
+#if defined(__RTAUDIO_DEBUG__)
+ snd_output_t *out;
+ snd_output_stdio_attach(&out, stderr, 0);
+#endif
- waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
+ // I'm not using the "plug" interface ... too much inconsistent behavior.
- enum_info dsinfo;
- void *ohandle = 0, *bhandle = 0;
- strncpy( dsinfo.name, devices_[device].name.c_str(), 64 );
- dsinfo.isValid = false;
- if ( mode == OUTPUT ) {
+ unsigned nDevices = 0;
+ int result, subdevice, card;
+ char name[64];
+ snd_ctl_t *chandle;
- if ( devices_[device].maxOutputChannels < channels ) {
- sprintf(message_, "RtApiDs: requested channels (%d) > than supported (%d) by device (%s).",
- channels, devices_[device].maxOutputChannels, devices_[device].name.c_str());
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ if ( options && options->flags & RTAUDIO_ALSA_USE_DEFAULT )
+ snprintf(name, sizeof(name), "%s", "default");
+ else {
+ // Count cards and devices
+ card = -1;
+ snd_card_next( &card );
+ while ( card >= 0 ) {
+ sprintf( name, "hw:%d", card );
+ result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ subdevice = -1;
+ while( 1 ) {
+ result = snd_ctl_pcm_next_device( chandle, &subdevice );
+ if ( result < 0 ) break;
+ if ( subdevice < 0 ) break;
+ if ( nDevices == device ) {
+ sprintf( name, "hw:%d,%d", card, subdevice );
+ snd_ctl_close( chandle );
+ goto foundDevice;
+ }
+ nDevices++;
+ }
+ snd_ctl_close( chandle );
+ snd_card_next( &card );
}
- // Enumerate through output devices to find the id (if it exists).
- result = DirectSoundEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Error performing output device id enumeration: %s.",
- getErrorString(result));
- error(RtError::DEBUG_WARNING);
- return FAILURE;
+ result = snd_ctl_open( &chandle, "default", SND_CTL_NONBLOCK );
+ if ( result == 0 ) {
+ if ( nDevices == device ) {
+ strcpy( name, "default" );
+ goto foundDevice;
+ }
+ nDevices++;
}
- if ( dsinfo.isValid == false ) {
- sprintf(message_, "RtApiDs: output device (%s) id not found!", devices_[device].name.c_str());
- error(RtError::DEBUG_WARNING);
+ 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;
}
- LPGUID id = dsinfo.id;
- LPDIRECTSOUND object;
- LPDIRECTSOUNDBUFFER buffer;
- DSBUFFERDESC bufferDescription;
-
- result = DirectSoundCreate( id, &object, NULL );
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Could not create playback object (%s): %s.",
- devices_[device].name.c_str(), getErrorString(result));
- error(RtError::DEBUG_WARNING);
+ 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;
}
+ }
- // Set cooperative level to DSSCL_EXCLUSIVE
- result = object->SetCooperativeLevel(hWnd, DSSCL_EXCLUSIVE);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message_, "RtApiDs: Unable to set cooperative level (%s): %s.",
- devices_[device].name.c_str(), getErrorString(result));
- error(RtError::WARNING);
- return FAILURE;
- }
+ foundDevice:
- // Even though we will write to the secondary buffer, we need to
- // access the primary buffer to set the correct output format
- // (since the default is 8-bit, 22 kHz!). Setup the DS primary
- // buffer description.
- 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_, "RtApiDs: Unable to access primary buffer (%s): %s.",
- devices_[device].name.c_str(), getErrorString(result));
- error(RtError::WARNING);
- return FAILURE;
- }
+ // 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();
- // Set the primary DS buffer sound format.
- result = buffer->SetFormat(&waveFormat);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message_, "RtApiDs: Unable to set primary buffer format (%s): %s.",
- devices_[device].name.c_str(), getErrorString(result));
- error(RtError::WARNING);
- return FAILURE;
- }
+ snd_pcm_stream_t stream;
+ if ( mode == OUTPUT )
+ stream = SND_PCM_STREAM_PLAYBACK;
+ else
+ stream = SND_PCM_STREAM_CAPTURE;
- // Setup the secondary DS buffer description.
- buffer_size = channels * *bufferSize * nBuffers * waveFormat.wBitsPerSample / 8;
- ZeroMemory(&bufferDescription, sizeof(DSBUFFERDESC));
- bufferDescription.dwSize = sizeof(DSBUFFERDESC);
- bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
- DSBCAPS_GETCURRENTPOSITION2 |
- DSBCAPS_LOCHARDWARE ); // Force hardware mixing
- bufferDescription.dwBufferBytes = buffer_size;
- bufferDescription.lpwfxFormat = &waveFormat;
+ 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;
+ }
- // 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) ) {
- bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
- DSBCAPS_GETCURRENTPOSITION2 |
- DSBCAPS_LOCSOFTWARE ); // Force software mixing
- result = object->CreateSoundBuffer(&bufferDescription, &buffer, NULL);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message_, "RtApiDs: Unable to create secondary DS buffer (%s): %s.",
- devices_[device].name.c_str(), getErrorString(result));
- error(RtError::WARNING);
- 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;
+ }
- // Get the buffer size ... might be different from what we specified.
- DSBCAPS dsbcaps;
- dsbcaps.dwSize = sizeof(DSBCAPS);
- buffer->GetCaps(&dsbcaps);
- buffer_size = dsbcaps.dwBufferBytes;
+#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
- // Lock the DS buffer
- result = buffer->Lock(0, buffer_size, &audioPtr, &dataLen, NULL, NULL, 0);
- if ( FAILED(result) ) {
- object->Release();
- buffer->Release();
- sprintf(message_, "RtApiDs: Unable to lock buffer (%s): %s.",
- devices_[device].name.c_str(), getErrorString(result));
- error(RtError::WARNING);
- return FAILURE;
+ // 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;
+ }
- // Zero the DS buffer
- ZeroMemory(audioPtr, dataLen);
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
- // Unlock the DS buffer
- result = buffer->Unlock(audioPtr, dataLen, NULL, 0);
- if ( FAILED(result) ) {
- object->Release();
- buffer->Release();
- sprintf(message_, "RtApiDs: Unable to unlock buffer(%s): %s.",
- devices_[device].name.c_str(), getErrorString(result));
- error(RtError::WARNING);
- 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;
+ }
- ohandle = (void *) object;
- bhandle = (void *) buffer;
- stream_.nDeviceChannels[0] = channels;
+ // 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;
}
- if ( mode == INPUT ) {
+ deviceFormat = SND_PCM_FORMAT_S24;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ goto setFormat;
+ }
- if ( devices_[device].maxInputChannels < channels )
- return FAILURE;
+ deviceFormat = SND_PCM_FORMAT_S16;
+ if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ goto setFormat;
+ }
- // Enumerate through input devices to find the id (if it exists).
- result = DirectSoundCaptureEnumerate((LPDSENUMCALLBACK)deviceIdCallback, &dsinfo);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Error performing input device id enumeration: %s.",
- getErrorString(result));
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
+ 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 ( dsinfo.isValid == false ) {
- sprintf(message_, "RtAudioDS: input device (%s) id not found!", devices_[device].name.c_str());
- error(RtError::DEBUG_WARNING);
- return FAILURE;
- }
+ // If we get here, no supported format was found.
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
- LPGUID id = dsinfo.id;
- LPDIRECTSOUNDCAPTURE object;
- LPDIRECTSOUNDCAPTUREBUFFER buffer;
- DSCBUFFERDESC bufferDescription;
+ 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;
+ }
- result = DirectSoundCaptureCreate( id, &object, NULL );
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Could not create capture object (%s): %s.",
- devices_[device].name.c_str(), getErrorString(result));
- error(RtError::WARNING);
+ // 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;
}
+ }
- // 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;
+ // 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;
+ }
- // Create the capture buffer.
- result = object->CreateCaptureBuffer(&bufferDescription, &buffer, NULL);
- if ( FAILED(result) ) {
- object->Release();
- sprintf(message_, "RtApiDs: Unable to create capture buffer (%s): %s.",
- devices_[device].name.c_str(), getErrorString(result));
- error(RtError::WARNING);
- 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;
+ }
- // Lock the capture buffer
- result = buffer->Lock(0, buffer_size, &audioPtr, &dataLen, NULL, NULL, 0);
- if ( FAILED(result) ) {
- object->Release();
- buffer->Release();
- sprintf(message_, "RtApiDs: Unable to lock capture buffer (%s): %s.",
- devices_[device].name.c_str(), getErrorString(result));
- error(RtError::WARNING);
- 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;
- // Zero the buffer
- ZeroMemory(audioPtr, dataLen);
+ // 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;
+ }
- // Unlock the buffer
- result = buffer->Unlock(audioPtr, dataLen, NULL, 0);
- if ( FAILED(result) ) {
- object->Release();
- buffer->Release();
- sprintf(message_, "RtApiDs: Unable to unlock capture buffer (%s): %s.",
- devices_[device].name.c_str(), getErrorString(result));
- error(RtError::WARNING);
- return FAILURE;
- }
+ // Set the buffer (or period) size.
+ int dir = 0;
+ snd_pcm_uframes_t periodSize = *bufferSize;
+ result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ *bufferSize = periodSize;
- ohandle = (void *) object;
- bhandle = (void *) buffer;
- stream_.nDeviceChannels[1] = channels;
+ // Set the buffer number, which in ALSA is referred to as the "period".
+ unsigned int periods = 0;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
+ if ( options && options->numberOfBuffers > 0 ) periods = options->numberOfBuffers;
+ if ( periods < 2 ) periods = 4; // a fairly safe default value
+ result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // If attempting to setup a duplex stream, the bufferSize parameter
+ // MUST be the same in both directions!
+ if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- stream_.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;
+ // Install the hardware configuration
+ result = snd_pcm_hw_params( phandle, hw_params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+#if defined(__RTAUDIO_DEBUG__)
+ fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
+ snd_pcm_hw_params_dump( hw_params, out );
+#endif
+
+ // Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
+ snd_pcm_sw_params_t *sw_params = NULL;
+ snd_pcm_sw_params_alloca( &sw_params );
+ snd_pcm_sw_params_current( phandle, sw_params );
+ snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
+ snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, ULONG_MAX );
+ snd_pcm_sw_params_set_silence_threshold( phandle, sw_params, 0 );
+
+ // The following two settings were suggested by Theo Veenker
+ //snd_pcm_sw_params_set_avail_min( phandle, sw_params, *bufferSize );
+ //snd_pcm_sw_params_set_xfer_align( phandle, sw_params, 1 );
+
+ // here are two options for a fix
+ //snd_pcm_sw_params_set_silence_size( phandle, sw_params, ULONG_MAX );
+ snd_pcm_uframes_t val;
+ snd_pcm_sw_params_get_boundary( sw_params, &val );
+ snd_pcm_sw_params_set_silence_size( phandle, sw_params, val );
+
+ result = snd_pcm_sw_params( phandle, sw_params );
+ if ( result < 0 ) {
+ snd_pcm_close( phandle );
+ errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing software configuration on device (" << name << "), " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+#if defined(__RTAUDIO_DEBUG__)
+ fprintf(stderr, "\nRtApiAlsa: dump software params after installation:\n\n");
+ snd_pcm_sw_params_dump( sw_params, out );
+#endif
+
// Set flags for buffer conversion
stream_.doConvertBuffer[mode] = false;
- if (stream_.userFormat != stream_.deviceFormat[mode])
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
stream_.doConvertBuffer[mode] = true;
- if (stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode])
+ 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 necessary internal buffers
- if ( stream_.nUserChannels[0] != stream_.nUserChannels[1] ) {
+ // 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;
+ }
- 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) {
- sprintf(message_, "RtApiDs: error allocating user buffer memory (%s).",
- devices_[device].name.c_str());
+ if ( pthread_cond_init( &apiInfo->runnable_cv, NULL ) ) {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: error initializing pthread condition variable.";
goto error;
}
+
+ stream_.apiHandle = (void *) apiInfo;
+ apiInfo->handles[0] = 0;
+ apiInfo->handles[1] = 0;
+ }
+ else {
+ apiInfo = (AlsaHandle *) stream_.apiHandle;
+ }
+ apiInfo->handles[mode] = phandle;
+ phandle = 0;
+
+ // Allocate necessary internal buffers.
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
}
if ( stream_.doConvertBuffer[mode] ) {
- 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]);
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
- long bytes_out = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
- if ( buffer_bytes < bytes_out ) makeBuffer = false;
+ 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) {
- sprintf(message_, "RtApiDs: error allocating device buffer memory (%s).",
- devices_[device].name.c_str());
+ 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;
}
}
}
- // Allocate our DsHandle structures for the stream.
- DsHandle *handles;
- if ( stream_.apiHandle == 0 ) {
- handles = (DsHandle *) calloc(2, sizeof(DsHandle));
- if ( handles == NULL ) {
- sprintf(message_, "RtApiDs: Error allocating DsHandle memory (%s).",
- devices_[device].name.c_str());
- goto error;
- }
- handles[0].object = 0;
- handles[1].object = 0;
- stream_.apiHandle = (void *) handles;
- }
- else
- handles = (DsHandle *) stream_.apiHandle;
- handles[mode].object = ohandle;
- handles[mode].buffer = bhandle;
-
+ stream_.sampleRate = sampleRate;
+ stream_.nBuffers = periods;
stream_.device[mode] = device;
stream_.state = STREAM_STOPPED;
- if ( stream_.mode == OUTPUT && mode == INPUT )
+
+ // 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
+ // Link the streams if possible.
+ apiInfo->synchronized = false;
+ if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
+ apiInfo->synchronized = true;
+ else {
+ errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
+ error( RtAudioError::WARNING );
+ }
+ }
+ else {
stream_.mode = mode;
- stream_.nBuffers = nBuffers;
- stream_.sampleRate = sampleRate;
+
+ // Setup callback thread.
+ stream_.callbackInfo.object = (void *) this;
+
+ // Set the thread attributes for joinable and realtime scheduling
+ // priority (optional). The higher priority will only take affect
+ // if the program is run as root or suid. Note, under Linux
+ // processes with CAP_SYS_NICE privilege, a user can change
+ // scheduling policy and priority (thus need not be root). See
+ // POSIX "capabilities".
+ pthread_attr_t attr;
+ pthread_attr_init( &attr );
+ pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
+#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+ stream_.callbackInfo.doRealtime = true;
+ struct sched_param param;
+ int priority = options->priority;
+ int min = sched_get_priority_min( SCHED_RR );
+ int max = sched_get_priority_max( SCHED_RR );
+ if ( priority < min ) priority = min;
+ else if ( priority > max ) priority = max;
+ param.sched_priority = priority;
+
+ // Set the policy BEFORE the priority. Otherwise it fails.
+ pthread_attr_setschedpolicy(&attr, SCHED_RR);
+ pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
+ // This is definitely required. Otherwise it fails.
+ pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
+ pthread_attr_setschedparam(&attr, ¶m);
+ }
+ else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#endif
+
+ stream_.callbackInfo.isRunning = true;
+ result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
+ pthread_attr_destroy( &attr );
+ if ( result ) {
+ // Failed. Try instead with default attributes.
+ result = pthread_create( &stream_.callbackInfo.thread, NULL, alsaCallbackHandler, &stream_.callbackInfo );
+ if ( result ) {
+ stream_.callbackInfo.isRunning = false;
+ errorText_ = "RtApiAlsa::error creating callback thread!";
+ goto error;
+ }
+ }
+ }
return SUCCESS;
error:
- if (handles) {
- if (handles[0].object) {
- LPDIRECTSOUND object = (LPDIRECTSOUND) handles[0].object;
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handles[0].buffer;
- if (buffer) buffer->Release();
- object->Release();
- }
- if (handles[1].object) {
- LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handles[1].object;
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handles[1].buffer;
- if (buffer) buffer->Release();
- object->Release();
- }
- free(handles);
+ if ( apiInfo ) {
+ pthread_cond_destroy( &apiInfo->runnable_cv );
+ if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
+ if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
+ delete apiInfo;
stream_.apiHandle = 0;
}
- if (stream_.userBuffer) {
- free(stream_.userBuffer);
- stream_.userBuffer = 0;
+ if ( phandle) snd_pcm_close( phandle );
+
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
+
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
}
- error(RtError::WARNING);
+ stream_.state = STREAM_CLOSED;
return FAILURE;
}
-void RtApiDs :: setStreamCallback(RtAudioCallback callback, void *userData)
+void RtApiAlsa :: closeStream()
{
- verifyStream();
-
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
- if ( info->usingCallback ) {
- sprintf(message_, "RtApiDs: A callback is already set for this stream!");
- error(RtError::WARNING);
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
return;
}
- info->callback = (void *) callback;
- info->userData = userData;
- info->usingCallback = true;
- info->object = (void *) this;
-
- unsigned thread_id;
- info->thread = _beginthreadex(NULL, 0, &callbackHandler,
- &stream_.callbackInfo, 0, &thread_id);
- if (info->thread == 0) {
- info->usingCallback = false;
- sprintf(message_, "RtApiDs: error starting callback thread!");
- error(RtError::THREAD_ERROR);
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ stream_.callbackInfo.isRunning = false;
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_STOPPED ) {
+ apiInfo->runnable = true;
+ pthread_cond_signal( &apiInfo->runnable_cv );
}
+ MUTEX_UNLOCK( &stream_.mutex );
+ pthread_join( stream_.callbackInfo.thread, NULL );
- // When spawning multiple threads in quick succession, it appears to be
- // necessary to wait a bit for each to initialize ... another windoism!
- Sleep(1);
-}
-
-void RtApiDs :: cancelStreamCallback()
-{
- verifyStream();
-
- if (stream_.callbackInfo.usingCallback) {
-
- if (stream_.state == STREAM_RUNNING)
- stopStream();
+ 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] );
+ }
- MUTEX_LOCK(&stream_.mutex);
+ if ( apiInfo ) {
+ pthread_cond_destroy( &apiInfo->runnable_cv );
+ if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
+ if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
+ delete apiInfo;
+ stream_.apiHandle = 0;
+ }
- 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 RtApiDs :: closeStream()
+void RtApiAlsa :: 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 ( stream_.mode == UNINITIALIZED ) {
- sprintf(message_, "RtApiDs::closeStream(): no open stream to close!");
- error(RtError::WARNING);
+ // This method calls snd_pcm_prepare if the device isn't already in that state.
+
+ verifyStream();
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
return;
}
- if (stream_.callbackInfo.usingCallback) {
- stream_.callbackInfo.usingCallback = false;
- WaitForSingleObject( (HANDLE)stream_.callbackInfo.thread, INFINITE );
- CloseHandle( (HANDLE)stream_.callbackInfo.thread );
- }
+ MUTEX_LOCK( &stream_.mutex );
- DsHandle *handles = (DsHandle *) stream_.apiHandle;
- if (handles) {
- if (handles[0].object) {
- LPDIRECTSOUND object = (LPDIRECTSOUND) handles[0].object;
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handles[0].buffer;
- if (buffer) {
- buffer->Stop();
- buffer->Release();
+ 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;
}
- object->Release();
}
+ }
- if (handles[1].object) {
- LPDIRECTSOUNDCAPTURE object = (LPDIRECTSOUNDCAPTURE) handles[1].object;
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handles[1].buffer;
- if (buffer) {
- buffer->Stop();
- buffer->Release();
+ if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
+ result = snd_pcm_drop(handle[1]); // fix to remove stale data received since device has been open
+ state = snd_pcm_state( handle[1] );
+ if ( state != SND_PCM_STATE_PREPARED ) {
+ result = snd_pcm_prepare( handle[1] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
- object->Release();
}
- free(handles);
- stream_.apiHandle = 0;
- }
-
- if (stream_.userBuffer) {
- free(stream_.userBuffer);
- stream_.userBuffer = 0;
}
- if (stream_.deviceBuffer) {
- free(stream_.deviceBuffer);
- stream_.deviceBuffer = 0;
- }
+ stream_.state = STREAM_RUNNING;
- stream_.mode = UNINITIALIZED;
+ unlock:
+ apiInfo->runnable = true;
+ pthread_cond_signal( &apiInfo->runnable_cv );
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result >= 0 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
}
-void RtApiDs :: startStream()
+void RtApiAlsa :: stopStream()
{
verifyStream();
- if (stream_.state == STREAM_RUNNING) return;
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
- MUTEX_LOCK(&stream_.mutex);
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK( &stream_.mutex );
- HRESULT result;
- DsHandle *handles = (DsHandle *) stream_.apiHandle;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handles[0].buffer;
- result = buffer->Play(0, 0, DSBPLAY_LOOPING );
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to start buffer (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), 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 (stream_.mode == INPUT || stream_.mode == DUPLEX) {
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handles[1].buffer;
- result = buffer->Start(DSCBSTART_LOOPING );
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to start capture buffer (%s): %s.",
- devices_[stream_.device[1]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ 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;
}
}
- stream_.state = STREAM_RUNNING;
- MUTEX_UNLOCK(&stream_.mutex);
+ unlock:
+ apiInfo->runnable = false; // fixes high CPU usage when stopped
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ if ( result >= 0 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
}
-void RtApiDs :: stopStream()
+void RtApiAlsa :: abortStream()
{
verifyStream();
- if (stream_.state == STREAM_STOPPED) return;
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
- // Change the state before the lock to improve shutdown response
- // when using a callback.
stream_.state = STREAM_STOPPED;
- MUTEX_LOCK(&stream_.mutex);
-
- // 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.
- HRESULT result;
- DWORD dsBufferSize;
- LPVOID buffer1 = NULL;
- LPVOID buffer2 = NULL;
- DWORD bufferSize1 = 0;
- DWORD bufferSize2 = 0;
- DsHandle *handles = (DsHandle *) stream_.apiHandle;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
-
- DWORD currentPos, safePos;
- long buffer_bytes = stream_.bufferSize * stream_.nDeviceChannels[0];
- buffer_bytes *= formatBytes(stream_.deviceFormat[0]);
-
- LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handles[0].buffer;
- UINT nextWritePos = handles[0].bufferPointer;
- dsBufferSize = buffer_bytes * stream_.nBuffers;
-
- // Write zeroes for nBuffer counts.
- for (int i=0; i<stream_.nBuffers; i++) {
-
- // Find out where the read and "safe write" pointers are.
- result = dsBuffer->GetCurrentPosition(¤tPos, &safePos);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to get current position (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- 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 ) {
- double 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_, "RtApiDs: Unable to get current position (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
- if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset
- }
-
- // Lock free space in the buffer
- result = dsBuffer->Lock (nextWritePos, buffer_bytes, &buffer1,
- &bufferSize1, &buffer2, &bufferSize2, 0);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to lock buffer during playback (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- // Zero the free space
- ZeroMemory(buffer1, bufferSize1);
- if (buffer2 != NULL) ZeroMemory(buffer2, bufferSize2);
+ MUTEX_LOCK( &stream_.mutex );
- // Update our buffer offset and unlock sound buffer
- dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to unlock buffer during playback (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
- nextWritePos = (nextWritePos + bufferSize1 + bufferSize2) % dsBufferSize;
- handles[0].bufferPointer = nextWritePos;
+ int result = 0;
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ result = snd_pcm_drop( handle[0] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::abortStream: error aborting output pcm device, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
-
- // If we play again, start at the beginning of the buffer.
- handles[0].bufferPointer = 0;
}
- if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handles[1].buffer;
- buffer1 = NULL;
- bufferSize1 = 0;
-
- result = buffer->Stop();
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to stop capture buffer (%s): %s",
- devices_[stream_.device[1]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ 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;
}
+ }
- dsBufferSize = stream_.bufferSize * stream_.nDeviceChannels[1];
- dsBufferSize *= formatBytes(stream_.deviceFormat[1]) * stream_.nBuffers;
+ unlock:
+ apiInfo->runnable = false; // fixes high CPU usage when stopped
+ MUTEX_UNLOCK( &stream_.mutex );
- // 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_, "RtApiDs: Unable to lock capture buffer (%s): %s.",
- devices_[stream_.device[1]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ if ( result >= 0 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
+}
+
+void RtApiAlsa :: callbackEvent()
+{
+ AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_LOCK( &stream_.mutex );
+ while ( !apiInfo->runnable )
+ pthread_cond_wait( &apiInfo->runnable_cv, &stream_.mutex );
+
+ if ( stream_.state != STREAM_RUNNING ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
}
+ MUTEX_UNLOCK( &stream_.mutex );
+ }
- // Zero the DS buffer
- ZeroMemory(buffer1, bufferSize1);
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return;
+ }
- // Unlock the DS buffer
- result = buffer->Unlock(buffer1, bufferSize1, NULL, 0);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to unlock capture buffer (%s): %s.",
- devices_[stream_.device[1]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
+ int doStopStream = 0;
+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ if ( stream_.mode != INPUT && apiInfo->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ apiInfo->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && apiInfo->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ apiInfo->xrun[1] = false;
+ }
+ doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
- // If we start recording again, we must begin at beginning of buffer.
- handles[1].bufferPointer = 0;
+ if ( doStopStream == 2 ) {
+ abortStream();
+ return;
}
- MUTEX_UNLOCK(&stream_.mutex);
-}
+ MUTEX_LOCK( &stream_.mutex );
-void RtApiDs :: abortStream()
-{
- verifyStream();
- if (stream_.state == STREAM_STOPPED) return;
+ // The state might change while waiting on a mutex.
+ 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 result;
+ char *buffer;
+ int channels;
+ snd_pcm_t **handle;
+ snd_pcm_sframes_t frames;
+ RtAudioFormat format;
+ handle = (snd_pcm_t **) apiInfo->handles;
- HRESULT result;
- long dsBufferSize;
- LPVOID audioPtr;
- DWORD dataLen;
- DsHandle *handles = (DsHandle *) stream_.apiHandle;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
- LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handles[0].buffer;
- result = buffer->Stop();
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to stop buffer (%s): %s",
- devices_[stream_.device[0]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
- dsBufferSize = stream_.bufferSize * stream_.nDeviceChannels[0];
- dsBufferSize *= formatBytes(stream_.deviceFormat[0]) * stream_.nBuffers;
+ // Setup parameters.
+ if ( stream_.doConvertBuffer[1] ) {
+ buffer = stream_.deviceBuffer;
+ channels = stream_.nDeviceChannels[1];
+ format = stream_.deviceFormat[1];
+ }
+ else {
+ buffer = stream_.userBuffer[1];
+ channels = stream_.nUserChannels[1];
+ format = stream_.userFormat;
+ }
- // 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_, "RtApiDs: Unable to lock buffer (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), 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 ( result < (int) stream_.bufferSize ) {
+ // Either an error or overrun occured.
+ if ( result == -EPIPE ) {
+ snd_pcm_state_t state = snd_pcm_state( handle[1] );
+ if ( state == SND_PCM_STATE_XRUN ) {
+ apiInfo->xrun[1] = true;
+ result = snd_pcm_prepare( handle[1] );
+ if ( result < 0 ) {
+ errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after overrun, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ }
+ else {
+ errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ }
+ else {
+ errorStream_ << "RtApiAlsa::callbackEvent: audio read error, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ error( RtAudioError::WARNING );
+ goto tryOutput;
}
- // Zero the DS buffer
- ZeroMemory(audioPtr, dataLen);
+ // Do byte swapping if necessary.
+ if ( stream_.doByteSwap[1] )
+ byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
- // Unlock the DS buffer
- result = buffer->Unlock(audioPtr, dataLen, NULL, 0);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to unlock buffer (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
+ // Do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[1] )
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
- // If we start playing again, we must begin at beginning of buffer.
- handles[0].bufferPointer = 0;
+ // 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) {
- LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handles[1].buffer;
- audioPtr = NULL;
- dataLen = 0;
+ tryOutput:
- result = buffer->Stop();
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to stop capture buffer (%s): %s",
- devices_[stream_.device[1]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+
+ // Setup parameters and do buffer conversion if necessary.
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ convertBuffer( buffer, stream_.userBuffer[0], stream_.convertInfo[0] );
+ channels = stream_.nDeviceChannels[0];
+ format = stream_.deviceFormat[0];
+ }
+ else {
+ buffer = stream_.userBuffer[0];
+ channels = stream_.nUserChannels[0];
+ format = stream_.userFormat;
}
- dsBufferSize = stream_.bufferSize * stream_.nDeviceChannels[1];
- dsBufferSize *= formatBytes(stream_.deviceFormat[1]) * stream_.nBuffers;
+ // Do byte swapping if necessary.
+ if ( stream_.doByteSwap[0] )
+ byteSwapBuffer(buffer, stream_.bufferSize * channels, format);
- // 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_, "RtApiDs: Unable to lock capture buffer (%s): %s.",
- devices_[stream_.device[1]].name.c_str(), 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 );
}
- // Zero the DS buffer
- ZeroMemory(audioPtr, dataLen);
-
- // Unlock the DS buffer
- result = buffer->Unlock(audioPtr, dataLen, NULL, 0);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to unlock capture buffer (%s): %s.",
- devices_[stream_.device[1]].name.c_str(), 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
+ errorText_ = "RtApiAlsa::callbackEvent: audio write error, underrun.";
+ }
+ else {
+ errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ }
+ else {
+ errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
+ errorText_ = errorStream_.str();
+ }
+ error( RtAudioError::WARNING );
+ goto unlock;
}
- // If we start recording again, we must begin at beginning of buffer.
- handles[1].bufferPointer = 0;
+ // 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 );
+
+ RtApi::tickStreamTime();
+ if ( doStopStream == 1 ) this->stopStream();
}
-int RtApiDs :: streamWillBlock()
+static void *alsaCallbackHandler( void *ptr )
{
- verifyStream();
- if (stream_.state == STREAM_STOPPED) return 0;
-
- MUTEX_LOCK(&stream_.mutex);
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiAlsa *object = (RtApiAlsa *) info->object;
+ bool *isRunning = &info->isRunning;
- int channels;
- int frames = 0;
- HRESULT result;
- DWORD currentPos, safePos;
- channels = 1;
- DsHandle *handles = (DsHandle *) stream_.apiHandle;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
+#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
+ if ( info->doRealtime ) {
+ std::cerr << "RtAudio alsa: " <<
+ (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
+ "running realtime scheduling" << std::endl;
+ }
+#endif
- LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handles[0].buffer;
- UINT nextWritePos = handles[0].bufferPointer;
- channels = stream_.nDeviceChannels[0];
- DWORD dsBufferSize = stream_.bufferSize * channels;
- dsBufferSize *= formatBytes(stream_.deviceFormat[0]) * stream_.nBuffers;
+ while ( *isRunning == true ) {
+ pthread_testcancel();
+ object->callbackEvent();
+ }
- // Find out where the read and "safe write" pointers are.
- result = dsBuffer->GetCurrentPosition(¤tPos, &safePos);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to get current position (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
+ pthread_exit( NULL );
+}
- if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset
- frames = currentPos - nextWritePos;
- frames /= channels * formatBytes(stream_.deviceFormat[0]);
- }
+//******************** End of __LINUX_ALSA__ *********************//
+#endif
- if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
+#if defined(__LINUX_PULSE__)
- LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handles[1].buffer;
- UINT nextReadPos = handles[1].bufferPointer;
- channels = stream_.nDeviceChannels[1];
- DWORD dsBufferSize = stream_.bufferSize * channels;
- dsBufferSize *= formatBytes(stream_.deviceFormat[1]) * stream_.nBuffers;
+// Code written by Peter Meerwald, pmeerw@pmeerw.net
+// and Tristan Matthews.
- // Find out where the write and "safe read" pointers are.
- result = dsBuffer->GetCurrentPosition(¤tPos, &safePos);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to get current capture position (%s): %s.",
- devices_[stream_.device[1]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
+#include <pulse/error.h>
+#include <pulse/simple.h>
+#include <cstdio>
- if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset
+static const unsigned int SUPPORTED_SAMPLERATES[] = { 8000, 16000, 22050, 32000,
+ 44100, 48000, 96000, 0};
- 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]);
- }
- }
+struct rtaudio_pa_format_mapping_t {
+ RtAudioFormat rtaudio_format;
+ pa_sample_format_t pa_format;
+};
- frames = stream_.bufferSize - frames;
- if (frames < 0) frames = 0;
+static const rtaudio_pa_format_mapping_t supported_sampleformats[] = {
+ {RTAUDIO_SINT16, PA_SAMPLE_S16LE},
+ {RTAUDIO_SINT32, PA_SAMPLE_S32LE},
+ {RTAUDIO_FLOAT32, PA_SAMPLE_FLOAT32LE},
+ {0, PA_SAMPLE_INVALID}};
+
+struct PulseAudioHandle {
+ pa_simple *s_play;
+ pa_simple *s_rec;
+ pthread_t thread;
+ pthread_cond_t runnable_cv;
+ bool runnable;
+ PulseAudioHandle() : s_play(0), s_rec(0), runnable(false) { }
+};
- MUTEX_UNLOCK(&stream_.mutex);
- return frames;
+RtApiPulse::~RtApiPulse()
+{
+ if ( stream_.state != STREAM_CLOSED )
+ closeStream();
}
-void RtApiDs :: tickStream()
+unsigned int RtApiPulse::getDeviceCount( void )
{
- verifyStream();
+ return 1;
+}
- int stopStream = 0;
- if (stream_.state == STREAM_STOPPED) {
- if (stream_.callbackInfo.usingCallback) Sleep(50); // sleep 50 milliseconds
- return;
- }
- else if (stream_.callbackInfo.usingCallback) {
- RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
- stopStream = callback(stream_.userBuffer, stream_.bufferSize, stream_.callbackInfo.userData);
- }
+RtAudio::DeviceInfo RtApiPulse::getDeviceInfo( unsigned int /*device*/ )
+{
+ RtAudio::DeviceInfo info;
+ info.probed = true;
+ info.name = "PulseAudio";
+ info.outputChannels = 2;
+ info.inputChannels = 2;
+ info.duplexChannels = 2;
+ info.isDefaultOutput = true;
+ info.isDefaultInput = true;
- MUTEX_LOCK(&stream_.mutex);
+ for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr )
+ info.sampleRates.push_back( *sr );
- // The state might change while waiting on a mutex.
- if (stream_.state == STREAM_STOPPED) {
- MUTEX_UNLOCK(&stream_.mutex);
- return;
- }
+ info.preferredSampleRate = 48000;
+ info.nativeFormats = RTAUDIO_SINT16 | RTAUDIO_SINT32 | RTAUDIO_FLOAT32;
- HRESULT result;
- DWORD currentPos, safePos;
- LPVOID buffer1 = NULL;
- LPVOID buffer2 = NULL;
- DWORD bufferSize1 = 0;
- DWORD bufferSize2 = 0;
- char *buffer;
- long buffer_bytes;
- DsHandle *handles = (DsHandle *) stream_.apiHandle;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
+ return info;
+}
- // Setup parameters and do buffer conversion if necessary.
- if (stream_.doConvertBuffer[0]) {
- convertStreamBuffer(OUTPUT);
- buffer = stream_.deviceBuffer;
- buffer_bytes = stream_.bufferSize * stream_.nDeviceChannels[0];
- buffer_bytes *= formatBytes(stream_.deviceFormat[0]);
- }
- else {
- buffer = stream_.userBuffer;
- buffer_bytes = stream_.bufferSize * stream_.nUserChannels[0];
- buffer_bytes *= formatBytes(stream_.userFormat);
- }
+static void *pulseaudio_callback( void * user )
+{
+ CallbackInfo *cbi = static_cast<CallbackInfo *>( user );
+ RtApiPulse *context = static_cast<RtApiPulse *>( cbi->object );
+ volatile bool *isRunning = &cbi->isRunning;
+
+#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
+ if (cbi->doRealtime) {
+ std::cerr << "RtAudio pulse: " <<
+ (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
+ "running realtime scheduling" << std::endl;
+ }
+#endif
+
+ while ( *isRunning ) {
+ pthread_testcancel();
+ context->callbackEvent();
+ }
- // No byte swapping necessary in DirectSound implementation.
+ pthread_exit( NULL );
+}
- LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handles[0].buffer;
- UINT nextWritePos = handles[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_, "RtApiDs: Unable to get current position (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- 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.
- double millis = (endWrite - currentPos) * 900.0;
- millis /= ( formatBytes(stream_.deviceFormat[0]) * stream_.sampleRate);
- if ( millis < 1.0 ) millis = 1.0;
- Sleep( (DWORD) millis );
+void RtApiPulse::closeStream( void )
+{
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
- // Wake up, find out where we are now
- result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos );
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to get current position (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
- if ( currentPos < nextWritePos ) currentPos += dsBufferSize; // unwrap offset
+ stream_.callbackInfo.isRunning = false;
+ if ( pah ) {
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_STOPPED ) {
+ pah->runnable = true;
+ pthread_cond_signal( &pah->runnable_cv );
}
+ MUTEX_UNLOCK( &stream_.mutex );
- // Lock free space in the buffer
- result = dsBuffer->Lock (nextWritePos, buffer_bytes, &buffer1,
- &bufferSize1, &buffer2, &bufferSize2, 0);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to lock buffer during playback (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ pthread_join( pah->thread, 0 );
+ if ( pah->s_play ) {
+ pa_simple_flush( pah->s_play, NULL );
+ pa_simple_free( pah->s_play );
}
+ if ( pah->s_rec )
+ pa_simple_free( pah->s_rec );
- // Copy our buffer into the DS buffer
- CopyMemory(buffer1, buffer, bufferSize1);
- if (buffer2 != NULL) CopyMemory(buffer2, buffer+bufferSize1, bufferSize2);
+ pthread_cond_destroy( &pah->runnable_cv );
+ delete pah;
+ stream_.apiHandle = 0;
+ }
- // Update our buffer offset and unlock sound buffer
- dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to unlock buffer during playback (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
- nextWritePos = (nextWritePos + bufferSize1 + bufferSize2) % dsBufferSize;
- handles[0].bufferPointer = nextWritePos;
+ if ( stream_.userBuffer[0] ) {
+ free( stream_.userBuffer[0] );
+ stream_.userBuffer[0] = 0;
+ }
+ if ( stream_.userBuffer[1] ) {
+ free( stream_.userBuffer[1] );
+ stream_.userBuffer[1] = 0;
}
- if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
+ stream_.state = STREAM_CLOSED;
+ stream_.mode = UNINITIALIZED;
+}
- // Setup parameters.
- if (stream_.doConvertBuffer[1]) {
- buffer = stream_.deviceBuffer;
- buffer_bytes = stream_.bufferSize * stream_.nDeviceChannels[1];
- buffer_bytes *= formatBytes(stream_.deviceFormat[1]);
- }
- else {
- buffer = stream_.userBuffer;
- buffer_bytes = stream_.bufferSize * stream_.nUserChannels[1];
- buffer_bytes *= formatBytes(stream_.userFormat);
- }
+void RtApiPulse::callbackEvent( void )
+{
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
- LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handles[1].buffer;
- UINT nextReadPos = handles[1].bufferPointer;
- DWORD dsBufferSize = buffer_bytes * stream_.nBuffers;
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_LOCK( &stream_.mutex );
+ while ( !pah->runnable )
+ pthread_cond_wait( &pah->runnable_cv, &stream_.mutex );
- // Find out where the write and "safe read" pointers are.
- result = dsBuffer->GetCurrentPosition(¤tPos, &safePos);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to get current capture position (%s): %s.",
- devices_[stream_.device[1]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ if ( stream_.state != STREAM_RUNNING ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
}
+ MUTEX_UNLOCK( &stream_.mutex );
+ }
- if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset
- DWORD endRead = nextReadPos + buffer_bytes;
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiPulse::callbackEvent(): the stream is closed ... "
+ "this shouldn't happen!";
+ error( RtAudioError::WARNING );
+ return;
+ }
- // Check whether the entire write region is behind the play pointer.
- while ( safePos < endRead ) {
- // See comments for playback.
- double millis = (endRead - safePos) * 900.0;
- millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.sampleRate);
- if ( millis < 1.0 ) millis = 1.0;
- Sleep( (DWORD) millis );
+ RtAudioCallback callback = (RtAudioCallback) stream_.callbackInfo.callback;
+ double streamTime = getStreamTime();
+ RtAudioStreamStatus status = 0;
+ int doStopStream = callback( stream_.userBuffer[OUTPUT], stream_.userBuffer[INPUT],
+ stream_.bufferSize, streamTime, status,
+ stream_.callbackInfo.userData );
- // Wake up, find out where we are now
- result = dsBuffer->GetCurrentPosition( ¤tPos, &safePos );
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to get current capture position (%s): %s.",
- devices_[stream_.device[1]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
-
- if ( safePos < nextReadPos ) safePos += dsBufferSize; // unwrap offset
- }
+ if ( doStopStream == 2 ) {
+ abortStream();
+ return;
+ }
- // Lock free space in the buffer
- result = dsBuffer->Lock (nextReadPos, buffer_bytes, &buffer1,
- &bufferSize1, &buffer2, &bufferSize2, 0);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to lock buffer during capture (%s): %s.",
- devices_[stream_.device[1]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
- }
+ MUTEX_LOCK( &stream_.mutex );
+ void *pulse_in = stream_.doConvertBuffer[INPUT] ? stream_.deviceBuffer : stream_.userBuffer[INPUT];
+ void *pulse_out = stream_.doConvertBuffer[OUTPUT] ? stream_.deviceBuffer : stream_.userBuffer[OUTPUT];
- // Copy our buffer into the DS buffer
- CopyMemory(buffer, buffer1, bufferSize1);
- if (buffer2 != NULL) CopyMemory(buffer+bufferSize1, buffer2, bufferSize2);
+ if ( stream_.state != STREAM_RUNNING )
+ goto unlock;
- // Update our buffer offset and unlock sound buffer
- nextReadPos = (nextReadPos + bufferSize1 + bufferSize2) % dsBufferSize;
- dsBuffer->Unlock (buffer1, bufferSize1, buffer2, bufferSize2);
- if ( FAILED(result) ) {
- sprintf(message_, "RtApiDs: Unable to unlock buffer during capture (%s): %s.",
- devices_[stream_.device[1]].name.c_str(), getErrorString(result));
- error(RtError::DRIVER_ERROR);
+ int pa_error;
+ size_t bytes;
+ if (stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
+ if ( stream_.doConvertBuffer[OUTPUT] ) {
+ convertBuffer( stream_.deviceBuffer,
+ stream_.userBuffer[OUTPUT],
+ stream_.convertInfo[OUTPUT] );
+ bytes = stream_.nDeviceChannels[OUTPUT] * stream_.bufferSize *
+ formatBytes( stream_.deviceFormat[OUTPUT] );
+ } else
+ bytes = stream_.nUserChannels[OUTPUT] * stream_.bufferSize *
+ formatBytes( stream_.userFormat );
+
+ if ( pa_simple_write( pah->s_play, pulse_out, bytes, &pa_error ) < 0 ) {
+ errorStream_ << "RtApiPulse::callbackEvent: audio write error, " <<
+ pa_strerror( pa_error ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ }
+ }
+
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX) {
+ if ( stream_.doConvertBuffer[INPUT] )
+ bytes = stream_.nDeviceChannels[INPUT] * stream_.bufferSize *
+ formatBytes( stream_.deviceFormat[INPUT] );
+ else
+ bytes = stream_.nUserChannels[INPUT] * stream_.bufferSize *
+ formatBytes( stream_.userFormat );
+
+ if ( pa_simple_read( pah->s_rec, pulse_in, bytes, &pa_error ) < 0 ) {
+ errorStream_ << "RtApiPulse::callbackEvent: audio read error, " <<
+ pa_strerror( pa_error ) << ".";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ }
+ if ( stream_.doConvertBuffer[INPUT] ) {
+ convertBuffer( stream_.userBuffer[INPUT],
+ stream_.deviceBuffer,
+ stream_.convertInfo[INPUT] );
}
- handles[1].bufferPointer = nextReadPos;
-
- // No byte swapping necessary in DirectSound implementation.
-
- // Do buffer conversion if necessary.
- if (stream_.doConvertBuffer[1])
- convertStreamBuffer(INPUT);
}
- MUTEX_UNLOCK(&stream_.mutex);
+ unlock:
+ MUTEX_UNLOCK( &stream_.mutex );
+ RtApi::tickStreamTime();
- if (stream_.callbackInfo.usingCallback && stopStream)
- this->stopStream();
+ if ( doStopStream == 1 )
+ stopStream();
}
-// Definitions for utility functions and callbacks
-// specific to the DirectSound implementation.
-
-extern "C" unsigned __stdcall callbackHandler(void *ptr)
+void RtApiPulse::startStream( void )
{
- CallbackInfo *info = (CallbackInfo *) ptr;
- RtApiDs *object = (RtApiDs *) info->object;
- bool *usingCallback = &info->usingCallback;
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
- while ( *usingCallback ) {
- try {
- object->tickStream();
- }
- catch (RtError &exception) {
- fprintf(stderr, "\nRtApiDs: callback thread error (%s) ... closing thread.\n\n",
- exception.getMessageString());
- break;
- }
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiPulse::startStream(): the stream is not open!";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiPulse::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
}
- _endthreadex( 0 );
- return 0;
-}
+ MUTEX_LOCK( &stream_.mutex );
-static bool CALLBACK deviceCountCallback(LPGUID lpguid,
- LPCSTR lpcstrDescription,
- LPCSTR lpcstrModule,
- LPVOID lpContext)
-{
- int *pointer = ((int *) lpContext);
- (*pointer)++;
+ stream_.state = STREAM_RUNNING;
- return true;
+ pah->runnable = true;
+ pthread_cond_signal( &pah->runnable_cv );
+ MUTEX_UNLOCK( &stream_.mutex );
}
-static bool CALLBACK deviceInfoCallback(LPGUID lpguid,
- LPCSTR lpcstrDescription,
- LPCSTR lpcstrModule,
- LPVOID lpContext)
+void RtApiPulse::stopStream( void )
{
- enum_info *info = ((enum_info *) lpContext);
- while (strlen(info->name) > 0) info++;
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
- strncpy(info->name, lpcstrDescription, 64);
- info->id = lpguid;
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiPulse::stopStream(): the stream is not open!";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiPulse::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
- HRESULT hr;
- info->isValid = false;
- if (info->isInput == true) {
- DSCCAPS caps;
- LPDIRECTSOUNDCAPTURE object;
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK( &stream_.mutex );
+
+ if ( pah && pah->s_play ) {
+ int pa_error;
+ if ( pa_simple_drain( pah->s_play, &pa_error ) < 0 ) {
+ errorStream_ << "RtApiPulse::stopStream: error draining output device, " <<
+ pa_strerror( pa_error ) << ".";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
+ }
+ }
- hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
- if( hr != DS_OK ) return true;
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
+}
- caps.dwSize = sizeof(caps);
- hr = object->GetCaps( &caps );
- if( hr == DS_OK ) {
- if (caps.dwChannels > 0 && caps.dwFormats > 0)
- info->isValid = true;
- }
- object->Release();
+void RtApiPulse::abortStream( void )
+{
+ PulseAudioHandle *pah = static_cast<PulseAudioHandle*>( stream_.apiHandle );
+
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiPulse::abortStream(): the stream is not open!";
+ error( RtAudioError::INVALID_USE );
+ return;
+ }
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiPulse::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
}
- 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;
+ stream_.state = STREAM_STOPPED;
+ MUTEX_LOCK( &stream_.mutex );
+
+ if ( pah && pah->s_play ) {
+ int pa_error;
+ if ( pa_simple_flush( pah->s_play, &pa_error ) < 0 ) {
+ errorStream_ << "RtApiPulse::abortStream: error flushing output device, " <<
+ pa_strerror( pa_error ) << ".";
+ errorText_ = errorStream_.str();
+ MUTEX_UNLOCK( &stream_.mutex );
+ error( RtAudioError::SYSTEM_ERROR );
+ return;
}
- object->Release();
}
- return true;
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
}
-static bool CALLBACK defaultDeviceCallback(LPGUID lpguid,
- LPCSTR lpcstrDescription,
- LPCSTR lpcstrModule,
- LPVOID lpContext)
+bool RtApiPulse::probeDeviceOpen( unsigned int device, StreamMode mode,
+ unsigned int channels, unsigned int firstChannel,
+ unsigned int sampleRate, RtAudioFormat format,
+ unsigned int *bufferSize, RtAudio::StreamOptions *options )
{
- enum_info *info = ((enum_info *) lpContext);
+ PulseAudioHandle *pah = 0;
+ unsigned long bufferBytes = 0;
+ pa_sample_spec ss;
- if ( lpguid == NULL ) {
- strncpy(info->name, lpcstrDescription, 64);
+ if ( device != 0 ) return false;
+ if ( mode != INPUT && mode != OUTPUT ) return false;
+ if ( channels != 1 && channels != 2 ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: unsupported number of channels.";
return false;
}
+ ss.channels = channels;
- return true;
-}
-
-static bool CALLBACK deviceIdCallback(LPGUID lpguid,
- LPCSTR lpcstrDescription,
- LPCSTR lpcstrModule,
- LPVOID lpContext)
-{
- enum_info *info = ((enum_info *) lpContext);
+ if ( firstChannel != 0 ) return false;
- if ( strncmp( info->name, lpcstrDescription, 64 ) == 0 ) {
- info->id = lpguid;
- info->isValid = true;
+ bool sr_found = false;
+ for ( const unsigned int *sr = SUPPORTED_SAMPLERATES; *sr; ++sr ) {
+ if ( sampleRate == *sr ) {
+ sr_found = true;
+ stream_.sampleRate = sampleRate;
+ ss.rate = sampleRate;
+ break;
+ }
+ }
+ if ( !sr_found ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: unsupported sample rate.";
return false;
}
- return true;
-}
+ bool sf_found = 0;
+ for ( const rtaudio_pa_format_mapping_t *sf = supported_sampleformats;
+ sf->rtaudio_format && sf->pa_format != PA_SAMPLE_INVALID; ++sf ) {
+ if ( format == sf->rtaudio_format ) {
+ sf_found = true;
+ stream_.userFormat = sf->rtaudio_format;
+ stream_.deviceFormat[mode] = stream_.userFormat;
+ ss.format = sf->pa_format;
+ break;
+ }
+ }
+ if ( !sf_found ) { // Use internal data format conversion.
+ stream_.userFormat = format;
+ stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
+ ss.format = PA_SAMPLE_FLOAT32LE;
+ }
+
+ // Set other stream parameters.
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
+ else stream_.userInterleaved = true;
+ stream_.deviceInterleaved[mode] = true;
+ stream_.nBuffers = 1;
+ stream_.doByteSwap[mode] = false;
+ stream_.nUserChannels[mode] = channels;
+ stream_.nDeviceChannels[mode] = channels + firstChannel;
+ stream_.channelOffset[mode] = 0;
+ std::string streamName = "RtAudio";
+
+ // Set flags for buffer conversion.
+ stream_.doConvertBuffer[mode] = false;
+ if ( stream_.userFormat != stream_.deviceFormat[mode] )
+ stream_.doConvertBuffer[mode] = true;
+ if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
+ stream_.doConvertBuffer[mode] = true;
-static char* getErrorString(int code)
-{
- switch (code) {
+ // Allocate necessary internal buffers.
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
+ }
+ stream_.bufferSize = *bufferSize;
- case DSERR_ALLOCATED:
- return "Direct Sound already allocated";
+ if ( stream_.doConvertBuffer[mode] ) {
- case DSERR_CONTROLUNAVAIL:
- return "Direct Sound control unavailable";
+ 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;
+ }
+ }
- case DSERR_INVALIDPARAM:
- return "Direct Sound invalid parameter";
+ if ( makeBuffer ) {
+ bufferBytes *= *bufferSize;
+ if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
+ stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.deviceBuffer == NULL ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error allocating device buffer memory.";
+ goto error;
+ }
+ }
+ }
- case DSERR_INVALIDCALL:
- return "Direct Sound invalid call";
+ stream_.device[mode] = device;
- case DSERR_GENERIC:
- return "Direct Sound generic error";
+ // Setup the buffer conversion information structure.
+ if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
- case DSERR_PRIOLEVELNEEDED:
- return "Direct Sound Priority level needed";
+ if ( !stream_.apiHandle ) {
+ PulseAudioHandle *pah = new PulseAudioHandle;
+ if ( !pah ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error allocating memory for handle.";
+ goto error;
+ }
- case DSERR_OUTOFMEMORY:
- return "Direct Sound out of memory";
+ stream_.apiHandle = pah;
+ if ( pthread_cond_init( &pah->runnable_cv, NULL ) != 0 ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error creating condition variable.";
+ goto error;
+ }
+ }
+ pah = static_cast<PulseAudioHandle *>( stream_.apiHandle );
- case DSERR_BADFORMAT:
- return "Direct Sound bad format";
+ int error;
+ if ( options && !options->streamName.empty() ) streamName = options->streamName;
+ switch ( mode ) {
+ case INPUT:
+ pa_buffer_attr buffer_attr;
+ buffer_attr.fragsize = bufferBytes;
+ buffer_attr.maxlength = -1;
- case DSERR_UNSUPPORTED:
- return "Direct Sound unsupported error";
+ pah->s_rec = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_RECORD, NULL, "Record", &ss, NULL, &buffer_attr, &error );
+ if ( !pah->s_rec ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error connecting input to PulseAudio server.";
+ goto error;
+ }
+ break;
+ case OUTPUT:
+ pah->s_play = pa_simple_new( NULL, streamName.c_str(), PA_STREAM_PLAYBACK, NULL, "Playback", &ss, NULL, NULL, &error );
+ if ( !pah->s_play ) {
+ errorText_ = "RtApiPulse::probeDeviceOpen: error connecting output to PulseAudio server.";
+ goto error;
+ }
+ break;
+ default:
+ goto error;
+ }
- case DSERR_NODRIVER:
- return "Direct Sound no driver error";
+ if ( stream_.mode == UNINITIALIZED )
+ stream_.mode = mode;
+ else if ( stream_.mode == mode )
+ goto error;
+ else
+ stream_.mode = DUPLEX;
- case DSERR_ALREADYINITIALIZED:
- return "Direct Sound already initialized";
+ if ( !stream_.callbackInfo.isRunning ) {
+ stream_.callbackInfo.object = this;
+
+ stream_.state = STREAM_STOPPED;
+ // Set the thread attributes for joinable and realtime scheduling
+ // priority (optional). The higher priority will only take affect
+ // if the program is run as root or suid. Note, under Linux
+ // processes with CAP_SYS_NICE privilege, a user can change
+ // scheduling policy and priority (thus need not be root). See
+ // POSIX "capabilities".
+ pthread_attr_t attr;
+ pthread_attr_init( &attr );
+ pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
+#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+ stream_.callbackInfo.doRealtime = true;
+ struct sched_param param;
+ int priority = options->priority;
+ int min = sched_get_priority_min( SCHED_RR );
+ int max = sched_get_priority_max( SCHED_RR );
+ if ( priority < min ) priority = min;
+ else if ( priority > max ) priority = max;
+ param.sched_priority = priority;
+
+ // Set the policy BEFORE the priority. Otherwise it fails.
+ pthread_attr_setschedpolicy(&attr, SCHED_RR);
+ pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
+ // This is definitely required. Otherwise it fails.
+ pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
+ pthread_attr_setschedparam(&attr, ¶m);
+ }
+ else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#endif
- case DSERR_NOAGGREGATION:
- return "Direct Sound no aggregation";
+ stream_.callbackInfo.isRunning = true;
+ int result = pthread_create( &pah->thread, &attr, pulseaudio_callback, (void *)&stream_.callbackInfo);
+ pthread_attr_destroy(&attr);
+ if(result != 0) {
+ // Failed. Try instead with default attributes.
+ result = pthread_create( &pah->thread, NULL, pulseaudio_callback, (void *)&stream_.callbackInfo);
+ if(result != 0) {
+ stream_.callbackInfo.isRunning = false;
+ errorText_ = "RtApiPulse::probeDeviceOpen: error creating thread.";
+ goto error;
+ }
+ }
+ }
- case DSERR_BUFFERLOST:
- return "Direct Sound buffer lost";
+ return SUCCESS;
+
+ error:
+ if ( pah && stream_.callbackInfo.isRunning ) {
+ pthread_cond_destroy( &pah->runnable_cv );
+ delete pah;
+ stream_.apiHandle = 0;
+ }
- case DSERR_OTHERAPPHASPRIO:
- return "Direct Sound other app has priority";
+ for ( int i=0; i<2; i++ ) {
+ if ( stream_.userBuffer[i] ) {
+ free( stream_.userBuffer[i] );
+ stream_.userBuffer[i] = 0;
+ }
+ }
- case DSERR_UNINITIALIZED:
- return "Direct Sound uninitialized";
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
+ stream_.deviceBuffer = 0;
+ }
- default:
- return "Direct Sound unknown error";
- }
+ stream_.state = STREAM_CLOSED;
+ return FAILURE;
}
-//******************** End of __WINDOWS_DS__ *********************//
+//******************** End of __LINUX_PULSE__ *********************//
#endif
-#if defined(__IRIX_AL__) // SGI's AL API for IRIX
+#if defined(__LINUX_OSS__)
-#include <dmedia/audio.h>
#include <unistd.h>
+#include <sys/ioctl.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <sys/soundcard.h>
#include <errno.h>
+#include <math.h>
-extern "C" void *callbackHandler(void * ptr);
+static void *ossCallbackHandler(void * ptr);
-RtApiAl :: RtApiAl()
+// A structure to hold various information related to the OSS API
+// implementation.
+struct OssHandle {
+ int id[2]; // device ids
+ bool xrun[2];
+ bool triggered;
+ pthread_cond_t runnable;
+
+ OssHandle()
+ :triggered(false) { id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
+};
+
+RtApiOss :: RtApiOss()
{
- this->initialize();
+ // Nothing to do here.
+}
- if (nDevices_ <= 0) {
- sprintf(message_, "RtApiAl: no Irix AL audio devices found!");
- error(RtError::NO_DEVICES_FOUND);
- }
+RtApiOss :: ~RtApiOss()
+{
+ if ( stream_.state != STREAM_CLOSED ) closeStream();
}
-RtApiAl :: ~RtApiAl()
+unsigned int RtApiOss :: getDeviceCount( void )
{
- // The subclass destructor gets called before the base class
- // destructor, so close any existing streams before deallocating
- // apiDeviceId memory.
- if ( stream_.mode != UNINITIALIZED ) closeStream();
+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+ if ( mixerfd == -1 ) {
+ errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
+ error( RtAudioError::WARNING );
+ return 0;
+ }
- // Free our allocated apiDeviceId memory.
- long *id;
- for ( unsigned int i=0; i<devices_.size(); i++ ) {
- id = (long *) devices_[i].apiDeviceId;
- if (id) free(id);
+ oss_sysinfo sysinfo;
+ if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
+ error( RtAudioError::WARNING );
+ return 0;
}
+
+ close( mixerfd );
+ return sysinfo.numaudios;
}
-void RtApiAl :: initialize(void)
+RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
{
- // Count cards and devices
- nDevices_ = 0;
+ RtAudio::DeviceInfo info;
+ info.probed = false;
- // Determine the total number of input and output devices.
- nDevices_ = alQueryValues(AL_SYSTEM, AL_DEVICES, 0, 0, 0, 0);
- if (nDevices_ < 0) {
- sprintf(message_, "RtApiAl: error counting devices: %s.",
- alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+ if ( mixerfd == -1 ) {
+ errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
+ error( RtAudioError::WARNING );
+ return info;
}
- if (nDevices_ <= 0) return;
-
- ALvalue *vls = (ALvalue *) new ALvalue[nDevices_];
+ oss_sysinfo sysinfo;
+ int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
+ if ( result == -1 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
+ error( RtAudioError::WARNING );
+ return info;
+ }
- // Create our list of devices and write their ascii identifiers and resource ids.
- char name[64];
- int outs, ins, i;
- ALpv pvs[1];
- pvs[0].param = AL_NAME;
- pvs[0].value.ptr = name;
- pvs[0].sizeIn = 64;
- RtApiDevice device;
- long *id;
-
- outs = alQueryValues(AL_SYSTEM, AL_DEFAULT_OUTPUT, vls, nDevices_, 0, 0);
- if (outs < 0) {
- delete [] vls;
- sprintf(message_, "RtApiAl: error getting output devices: %s.",
- alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
- }
-
- for (i=0; i<outs; i++) {
- if (alGetParams(vls[i].i, pvs, 1) < 0) {
- delete [] vls;
- sprintf(message_, "RtApiAl: error querying output devices: %s.",
- alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
- }
- device.name.erase();
- device.name.append( (const char *)name, strlen(name)+1);
- devices_.push_back(device);
- id = (long *) calloc(2, sizeof(long));
- id[0] = vls[i].i;
- devices_[i].apiDeviceId = (void *) id;
- }
-
- ins = alQueryValues(AL_SYSTEM, AL_DEFAULT_INPUT, &vls[outs], nDevices_-outs, 0, 0);
- if (ins < 0) {
- delete [] vls;
- sprintf(message_, "RtApiAl: error getting input devices: %s.",
- alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
- }
-
- for (i=outs; i<ins+outs; i++) {
- if (alGetParams(vls[i].i, pvs, 1) < 0) {
- delete [] vls;
- sprintf(message_, "RtApiAl: error querying input devices: %s.",
- alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
- }
- device.name.erase();
- device.name.append( (const char *)name, strlen(name)+1);
- devices_.push_back(device);
- id = (long *) calloc(2, sizeof(long));
- id[1] = vls[i].i;
- devices_[i].apiDeviceId = (void *) id;
- }
-
- delete [] vls;
-}
-
-int RtApiAl :: getDefaultInputDevice(void)
-{
- ALvalue value;
- long *id;
- int result = alQueryValues(AL_SYSTEM, AL_DEFAULT_INPUT, &value, 1, 0, 0);
- if (result < 0) {
- sprintf(message_, "RtApiAl: error getting default input device id: %s.",
- alGetErrorString(oserror()));
- error(RtError::WARNING);
+ unsigned nDevices = sysinfo.numaudios;
+ if ( nDevices == 0 ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
+ error( RtAudioError::INVALID_USE );
+ return info;
}
- else {
- for ( unsigned int i=0; i<devices_.size(); i++ ) {
- id = (long *) devices_[i].apiDeviceId;
- if ( id[1] == value.i ) return i;
- }
+
+ if ( device >= nDevices ) {
+ close( mixerfd );
+ errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
+ error( RtAudioError::INVALID_USE );
+ return info;
}
- return 0;
-}
+ oss_audioinfo ainfo;
+ ainfo.dev = device;
+ result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
+ close( mixerfd );
+ if ( result == -1 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
+ }
+
+ // Probe channels
+ if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
+ if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
+ if ( ainfo.caps & PCM_CAP_DUPLEX ) {
+ if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
+ info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
+ }
+
+ // Probe data formats ... do for input
+ unsigned long mask = ainfo.iformats;
+ if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
+ info.nativeFormats |= RTAUDIO_SINT16;
+ if ( mask & AFMT_S8 )
+ info.nativeFormats |= RTAUDIO_SINT8;
+ if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
+ info.nativeFormats |= RTAUDIO_SINT32;
+#ifdef AFMT_FLOAT
+ if ( mask & AFMT_FLOAT )
+ info.nativeFormats |= RTAUDIO_FLOAT32;
+#endif
+ if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
+ info.nativeFormats |= RTAUDIO_SINT24;
-int RtApiAl :: getDefaultOutputDevice(void)
-{
- ALvalue value;
- long *id;
- int result = alQueryValues(AL_SYSTEM, AL_DEFAULT_OUTPUT, &value, 1, 0, 0);
- if (result < 0) {
- sprintf(message_, "RtApiAl: error getting default output device id: %s.",
- alGetErrorString(oserror()));
- error(RtError::WARNING);
- }
- else {
- for ( unsigned int i=0; i<devices_.size(); i++ ) {
- id = (long *) devices_[i].apiDeviceId;
- if ( id[0] == value.i ) return i;
- }
+ // Check that we have at least one supported format
+ if ( info.nativeFormats == 0 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ return info;
}
- return 0;
-}
+ // 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] );
-void RtApiAl :: probeDeviceInfo(RtApiDevice *info)
-{
- int result;
- long resource;
- ALvalue value;
- ALparamInfo pinfo;
-
- // Get output resource ID if it exists.
- long *id = (long *) info->apiDeviceId;
- resource = id[0];
- if (resource > 0) {
-
- // Probe output device parameters.
- result = alQueryValues(resource, AL_CHANNELS, &value, 1, 0, 0);
- if (result < 0) {
- sprintf(message_, "RtApiAl: error getting device (%s) channels: %s.",
- info->name.c_str(), alGetErrorString(oserror()));
- error(RtError::WARNING);
- }
- else {
- info->maxOutputChannels = value.i;
- info->minOutputChannels = 1;
- }
+ if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
+ info.preferredSampleRate = SAMPLE_RATES[k];
- result = alGetParamInfo(resource, AL_RATE, &pinfo);
- if (result < 0) {
- sprintf(message_, "RtApiAl: error getting device (%s) rates: %s.",
- info->name.c_str(), alGetErrorString(oserror()));
- error(RtError::WARNING);
- }
- else {
- info->sampleRates.clear();
- for (unsigned int k=0; k<MAX_SAMPLE_RATES; k++) {
- if ( SAMPLE_RATES[k] >= pinfo.min.i && SAMPLE_RATES[k] <= pinfo.max.i )
- info->sampleRates.push_back( SAMPLE_RATES[k] );
+ break;
+ }
}
}
-
- // The AL library supports all our formats, except 24-bit and 32-bit ints.
- info->nativeFormats = (RtAudioFormat) 51;
}
+ 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] );
- // Now get input resource ID if it exists.
- resource = id[1];
- if (resource > 0) {
-
- // Probe input device parameters.
- result = alQueryValues(resource, AL_CHANNELS, &value, 1, 0, 0);
- if (result < 0) {
- sprintf(message_, "RtApiAl: error getting device (%s) channels: %s.",
- info->name.c_str(), alGetErrorString(oserror()));
- error(RtError::WARNING);
- }
- else {
- info->maxInputChannels = value.i;
- info->minInputChannels = 1;
- }
-
- result = alGetParamInfo(resource, AL_RATE, &pinfo);
- if (result < 0) {
- sprintf(message_, "RtApiAl: error getting device (%s) rates: %s.",
- info->name.c_str(), alGetErrorString(oserror()));
- error(RtError::WARNING);
- }
- 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->sampleRates.clear();
- for (unsigned int k=0; k<MAX_SAMPLE_RATES; k++) {
- if ( SAMPLE_RATES[k] >= pinfo.min.i && SAMPLE_RATES[k] <= pinfo.max.i )
- info->sampleRates.push_back( SAMPLE_RATES[k] );
+ if ( !info.preferredSampleRate || ( SAMPLE_RATES[k] <= 48000 && SAMPLE_RATES[k] > info.preferredSampleRate ) )
+ info.preferredSampleRate = SAMPLE_RATES[k];
}
}
+ }
- // The AL library supports all our formats, except 24-bit and 32-bit ints.
- info->nativeFormats = (RtAudioFormat) 51;
+ if ( info.sampleRates.size() == 0 ) {
+ errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ error( RtAudioError::WARNING );
+ }
+ else {
+ info.probed = true;
+ info.name = ainfo.name;
}
- if ( info->maxInputChannels == 0 && info->maxOutputChannels == 0 )
- return;
- if ( info->sampleRates.size() == 0 )
- return;
+ return info;
+}
- // 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;
+bool RtApiOss :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
+ unsigned int firstChannel, unsigned int sampleRate,
+ RtAudioFormat format, unsigned int *bufferSize,
+ RtAudio::StreamOptions *options )
+{
+ int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
+ if ( mixerfd == -1 ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error opening '/dev/mixer'.";
+ return FAILURE;
+ }
- info->probed = true;
+ 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;
+ }
- return;
-}
+ 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;
+ }
-bool RtApiAl :: probeDeviceOpen(int device, StreamMode mode, int channels,
- int sampleRate, RtAudioFormat format,
- int *bufferSize, int numberOfBuffers)
-{
- int result, nBuffers;
- long resource;
- ALconfig al_config;
- ALport port;
- ALpv pvs[2];
- long *id = (long *) devices_[device].apiDeviceId;
+ 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;
+ }
- // Get a new ALconfig structure.
- al_config = alNewConfig();
- if ( !al_config ) {
- sprintf(message_,"RtApiAl: can't get AL config: %s.",
- alGetErrorString(oserror()));
- error(RtError::WARNING);
+ 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;
}
- // Set the channels.
- result = alSetChannels(al_config, channels);
- if ( result < 0 ) {
- alFreeConfig(al_config);
- sprintf(message_,"RtApiAl: can't set %d channels in AL config: %s.",
- channels, alGetErrorString(oserror()));
- error(RtError::WARNING);
+ // 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;
}
- // 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 ) {
- alFreeConfig(al_config);
- sprintf(message_,"RtApiAl: can't set buffer size (%ld) in AL config: %s.",
- buffer_size, alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
+ int flags = 0;
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( mode == OUTPUT )
+ flags |= O_WRONLY;
+ else { // mode == INPUT
+ if (stream_.mode == OUTPUT && stream_.device[0] == device) {
+ // We just set the same device for playback ... close and reopen for duplex (OSS only).
+ close( handle->id[0] );
+ handle->id[0] = 0;
+ if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ // Check that the number previously set channels is the same.
+ if ( stream_.nUserChannels[0] != channels ) {
+ errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ flags |= O_RDWR;
}
- *bufferSize = buffer_size / nBuffers;
+ else
+ flags |= O_RDONLY;
}
- // 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;
+ // Set exclusive access if specified.
+ if ( options && options->flags & RTAUDIO_HOG_DEVICE ) flags |= O_EXCL;
+
+ // Try to open the device.
+ int fd;
+ fd = open( ainfo.devnode, flags, 0 );
+ if ( fd == -1 ) {
+ if ( errno == EBUSY )
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") is busy.";
+ else
+ errorStream_ << "RtApiOss::probeDeviceOpen: error opening device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- 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;
+
+ // For duplex operation, specifically set this mode (this doesn't seem to work).
+ /*
+ if ( flags | O_RDWR ) {
+ result = ioctl( fd, SNDCTL_DSP_SETDUPLEX, NULL );
+ if ( result == -1) {
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting duplex mode for device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ }
+ */
+
+ // Check the device channel support.
+ stream_.nUserChannels[mode] = channels;
+ if ( ainfo.max_channels < (int)(channels + firstChannel) ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: the device (" << ainfo.name << ") does not support requested channel parameters.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- else if (format == RTAUDIO_FLOAT32)
- result = alSetSampFmt(al_config, AL_SAMPFMT_FLOAT);
- else if (format == RTAUDIO_FLOAT64)
- result = alSetSampFmt(al_config, AL_SAMPFMT_DOUBLE);
- if ( result == -1 ) {
- alFreeConfig(al_config);
- sprintf(message_,"RtApiAl: error setting sample format in AL config: %s.",
- alGetErrorString(oserror()));
- error(RtError::WARNING);
+ // 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;
- if (mode == OUTPUT) {
+ // 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;
+ }
- // Set our device.
- if (device == 0)
- resource = AL_DEFAULT_OUTPUT;
- else
- resource = id[0];
- result = alSetDevice(al_config, resource);
- if ( result == -1 ) {
- alFreeConfig(al_config);
- sprintf(message_,"RtApiAl: error setting device (%s) in AL config: %s.",
- devices_[device].name.c_str(), alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
+ // Determine how to set the device format.
+ stream_.userFormat = format;
+ int deviceFormat = -1;
+ stream_.doByteSwap[mode] = false;
+ if ( format == RTAUDIO_SINT8 ) {
+ if ( mask & AFMT_S8 ) {
+ deviceFormat = AFMT_S8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
}
-
- // Open the port.
- port = alOpenPort("RtApiAl Output Port", "w", al_config);
- if( !port ) {
- alFreeConfig(al_config);
- sprintf(message_,"RtApiAl: error opening output port: %s.",
- alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
+ }
+ else if ( format == RTAUDIO_SINT16 ) {
+ if ( mask & AFMT_S16_NE ) {
+ deviceFormat = AFMT_S16_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
}
-
- // 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);
- alFreeConfig(al_config);
- sprintf(message_,"RtApiAl: error setting sample rate (%d) for device (%s): %s.",
- sampleRate, devices_[device].name.c_str(), alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
+ else if ( mask & AFMT_S16_OE ) {
+ deviceFormat = AFMT_S16_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ stream_.doByteSwap[mode] = true;
+ }
+ }
+ else if ( format == RTAUDIO_SINT24 ) {
+ if ( mask & AFMT_S24_NE ) {
+ deviceFormat = AFMT_S24_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ }
+ else if ( mask & AFMT_S24_OE ) {
+ deviceFormat = AFMT_S24_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ stream_.doByteSwap[mode] = true;
+ }
+ }
+ else if ( format == RTAUDIO_SINT32 ) {
+ if ( mask & AFMT_S32_NE ) {
+ deviceFormat = AFMT_S32_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ }
+ else if ( mask & AFMT_S32_OE ) {
+ deviceFormat = AFMT_S32_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ stream_.doByteSwap[mode] = true;
}
}
- else { // mode == INPUT
- // Set our device.
- if (device == 0)
- resource = AL_DEFAULT_INPUT;
- else
- resource = id[1];
- result = alSetDevice(al_config, resource);
- if ( result == -1 ) {
- alFreeConfig(al_config);
- sprintf(message_,"RtApiAl: error setting device (%s) in AL config: %s.",
- devices_[device].name.c_str(), alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
+ if ( deviceFormat == -1 ) {
+ // The user requested format is not natively supported by the device.
+ if ( mask & AFMT_S16_NE ) {
+ deviceFormat = AFMT_S16_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ }
+ else if ( mask & AFMT_S32_NE ) {
+ deviceFormat = AFMT_S32_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ }
+ else if ( mask & AFMT_S24_NE ) {
+ deviceFormat = AFMT_S24_NE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
}
+ else if ( mask & AFMT_S16_OE ) {
+ deviceFormat = AFMT_S16_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT16;
+ stream_.doByteSwap[mode] = true;
+ }
+ else if ( mask & AFMT_S32_OE ) {
+ deviceFormat = AFMT_S32_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT32;
+ stream_.doByteSwap[mode] = true;
+ }
+ else if ( mask & AFMT_S24_OE ) {
+ deviceFormat = AFMT_S24_OE;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT24;
+ stream_.doByteSwap[mode] = true;
+ }
+ else if ( mask & AFMT_S8) {
+ deviceFormat = AFMT_S8;
+ stream_.deviceFormat[mode] = RTAUDIO_SINT8;
+ }
+ }
+
+ if ( stream_.deviceFormat[mode] == 0 ) {
+ // This really shouldn't happen ...
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") data format not supported by RtAudio.";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Set the data format.
+ int temp = deviceFormat;
+ result = ioctl( fd, SNDCTL_DSP_SETFMT, &deviceFormat );
+ if ( result == -1 || deviceFormat != temp ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting data format on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+
+ // Attempt to set the buffer size. According to OSS, the minimum
+ // number of buffers is two. The supposed minimum buffer size is 16
+ // bytes, so that will be our lower bound. The argument to this
+ // call is in the form 0xMMMMSSSS (hex), where the buffer size (in
+ // bytes) is given as 2^SSSS and the number of buffers as 2^MMMM.
+ // We'll check the actual value used near the end of the setup
+ // procedure.
+ int ossBufferBytes = *bufferSize * formatBytes( stream_.deviceFormat[mode] ) * deviceChannels;
+ if ( ossBufferBytes < 16 ) ossBufferBytes = 16;
+ int buffers = 0;
+ if ( options ) buffers = options->numberOfBuffers;
+ if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) buffers = 2;
+ if ( buffers < 2 ) buffers = 3;
+ temp = ((int) buffers << 16) + (int)( log10( (double)ossBufferBytes ) / log10( 2.0 ) );
+ result = ioctl( fd, SNDCTL_DSP_SETFRAGMENT, &temp );
+ if ( result == -1 ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting buffer size on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.nBuffers = buffers;
- // Open the port.
- port = alOpenPort("RtApiAl Input Port", "r", al_config);
- if( !port ) {
- alFreeConfig(al_config);
- sprintf(message_,"RtApiAl: error opening input port: %s.",
- alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
- }
+ // Save buffer size (in sample frames).
+ *bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
+ stream_.bufferSize = *bufferSize;
- // 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);
- alFreeConfig(al_config);
- sprintf(message_,"RtApiAl: error setting sample rate (%d) for device (%s): %s.",
- sampleRate, devices_[device].name.c_str(), alGetErrorString(oserror()));
- error(RtError::WARNING);
- return FAILURE;
- }
+ // Set the sample rate.
+ int srate = sampleRate;
+ result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
+ if ( result == -1 ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
}
- alFreeConfig(al_config);
-
- stream_.nUserChannels[mode] = channels;
- stream_.nDeviceChannels[mode] = channels;
+ // Verify the sample rate setup worked.
+ if ( abs( srate - (int)sampleRate ) > 100 ) {
+ close( fd );
+ errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
+ errorText_ = errorStream_.str();
+ return FAILURE;
+ }
+ stream_.sampleRate = sampleRate;
- // Save stream handle.
- ALport *handle = (ALport *) stream_.apiHandle;
- if ( handle == 0 ) {
- handle = (ALport *) calloc(2, sizeof(ALport));
- if ( handle == NULL ) {
- sprintf(message_, "RtApiAl: Irix Al error allocating handle memory (%s).",
- devices_[device].name.c_str());
- goto error;
- }
- stream_.apiHandle = (void *) handle;
- handle[0] = 0;
- handle[1] = 0;
+ 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;
}
- handle[mode] = port;
+
+ // Set interleaving parameters.
+ stream_.userInterleaved = true;
+ stream_.deviceInterleaved[mode] = true;
+ if ( options && options->flags & RTAUDIO_NONINTERLEAVED )
+ stream_.userInterleaved = false;
// Set flags for buffer conversion
stream_.doConvertBuffer[mode] = false;
- if (stream_.userFormat != stream_.deviceFormat[mode])
+ 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 necessary internal buffers
- if ( stream_.nUserChannels[0] != stream_.nUserChannels[1] ) {
+ // Allocate the stream handles if necessary and then save.
+ if ( stream_.apiHandle == 0 ) {
+ try {
+ handle = new OssHandle;
+ }
+ catch ( std::bad_alloc& ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
+ goto error;
+ }
- 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) {
- sprintf(message_, "RtApiAl: error allocating user buffer memory (%s).",
- devices_[device].name.c_str());
+ if ( pthread_cond_init( &handle->runnable, NULL ) ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error initializing pthread condition variable.";
goto error;
}
+
+ stream_.apiHandle = (void *) handle;
+ }
+ else {
+ handle = (OssHandle *) stream_.apiHandle;
+ }
+ handle->id[mode] = fd;
+
+ // Allocate necessary internal buffers.
+ unsigned long bufferBytes;
+ bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
+ stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
+ if ( stream_.userBuffer[mode] == NULL ) {
+ errorText_ = "RtApiOss::probeDeviceOpen: error allocating user buffer memory.";
+ goto error;
}
if ( stream_.doConvertBuffer[mode] ) {
- 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]);
+ bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
+ if ( mode == INPUT ) {
if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
- long bytes_out = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
- if ( buffer_bytes < bytes_out ) makeBuffer = false;
+ 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) {
- sprintf(message_, "RtApiAl: error allocating device buffer memory (%s).",
- devices_[device].name.c_str());
+ 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 )
+
+ // 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
+ if ( stream_.device[0] == device ) handle->id[0] = fd;
+ }
+ else {
stream_.mode = mode;
- stream_.nBuffers = nBuffers;
- stream_.bufferSize = *bufferSize;
- stream_.sampleRate = sampleRate;
+
+ // 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 (e.g. NetBSD 1.6.x with GNU Pthread)
+ if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
+ stream_.callbackInfo.doRealtime = true;
+ struct sched_param param;
+ int priority = options->priority;
+ int min = sched_get_priority_min( SCHED_RR );
+ int max = sched_get_priority_max( SCHED_RR );
+ if ( priority < min ) priority = min;
+ else if ( priority > max ) priority = max;
+ param.sched_priority = priority;
+
+ // Set the policy BEFORE the priority. Otherwise it fails.
+ pthread_attr_setschedpolicy(&attr, SCHED_RR);
+ pthread_attr_setscope (&attr, PTHREAD_SCOPE_SYSTEM);
+ // This is definitely required. Otherwise it fails.
+ pthread_attr_setinheritsched(&attr, PTHREAD_EXPLICIT_SCHED);
+ pthread_attr_setschedparam(&attr, ¶m);
+ }
+ else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#else
+ pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
+#endif
+
+ stream_.callbackInfo.isRunning = true;
+ result = pthread_create( &stream_.callbackInfo.thread, &attr, ossCallbackHandler, &stream_.callbackInfo );
+ pthread_attr_destroy( &attr );
+ if ( result ) {
+ // Failed. Try instead with default attributes.
+ result = pthread_create( &stream_.callbackInfo.thread, NULL, ossCallbackHandler, &stream_.callbackInfo );
+ if ( result ) {
+ stream_.callbackInfo.isRunning = false;
+ errorText_ = "RtApiOss::error creating callback thread!";
+ goto error;
+ }
+ }
+ }
return SUCCESS;
error:
- if (handle) {
- if (handle[0])
- alClosePort(handle[0]);
- if (handle[1])
- alClosePort(handle[1]);
- free(handle);
+ if ( handle ) {
+ pthread_cond_destroy( &handle->runnable );
+ if ( handle->id[0] ) close( handle->id[0] );
+ if ( handle->id[1] ) close( handle->id[1] );
+ delete handle;
stream_.apiHandle = 0;
}
- if (stream_.userBuffer) {
- free(stream_.userBuffer);
- stream_.userBuffer = 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;
}
- error(RtError::WARNING);
+ stream_.state = STREAM_CLOSED;
return FAILURE;
}
-void RtApiAl :: closeStream()
+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 ( stream_.mode == UNINITIALIZED ) {
- sprintf(message_, "RtApiAl::closeStream(): no open stream to close!");
- error(RtError::WARNING);
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApiOss::closeStream(): no open stream to close!";
+ error( RtAudioError::WARNING );
return;
}
- ALport *handle = (ALport *) stream_.apiHandle;
- if (stream_.state == STREAM_RUNNING) {
- int buffer_size = stream_.bufferSize * stream_.nBuffers;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX)
- alDiscardFrames(handle[0], buffer_size);
- if (stream_.mode == INPUT || stream_.mode == DUPLEX)
- alDiscardFrames(handle[1], buffer_size);
- stream_.state = STREAM_STOPPED;
- }
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ stream_.callbackInfo.isRunning = false;
+ MUTEX_LOCK( &stream_.mutex );
+ if ( stream_.state == STREAM_STOPPED )
+ pthread_cond_signal( &handle->runnable );
+ MUTEX_UNLOCK( &stream_.mutex );
+ pthread_join( stream_.callbackInfo.thread, NULL );
- if (stream_.callbackInfo.usingCallback) {
- stream_.callbackInfo.usingCallback = false;
- pthread_join(stream_.callbackInfo.thread, NULL);
+ if ( stream_.state == STREAM_RUNNING ) {
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
+ ioctl( handle->id[0], SNDCTL_DSP_HALT, 0 );
+ else
+ ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
+ stream_.state = STREAM_STOPPED;
}
- if (handle) {
- if (handle[0]) alClosePort(handle[0]);
- if (handle[1]) alClosePort(handle[1]);
- free(handle);
+ if ( handle ) {
+ pthread_cond_destroy( &handle->runnable );
+ if ( handle->id[0] ) close( handle->id[0] );
+ if ( handle->id[1] ) close( handle->id[1] );
+ delete handle;
stream_.apiHandle = 0;
}
- if (stream_.userBuffer) {
- free(stream_.userBuffer);
- stream_.userBuffer = 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);
+ if ( stream_.deviceBuffer ) {
+ free( stream_.deviceBuffer );
stream_.deviceBuffer = 0;
}
stream_.mode = UNINITIALIZED;
+ stream_.state = STREAM_CLOSED;
}
-void RtApiAl :: startStream()
+void RtApiOss :: startStream()
{
verifyStream();
- if (stream_.state == STREAM_RUNNING) return;
+ if ( stream_.state == STREAM_RUNNING ) {
+ errorText_ = "RtApiOss::startStream(): the stream is already running!";
+ error( RtAudioError::WARNING );
+ return;
+ }
- MUTEX_LOCK(&stream_.mutex);
+ MUTEX_LOCK( &stream_.mutex );
- // The AL port is ready as soon as it is opened.
stream_.state = STREAM_RUNNING;
- MUTEX_UNLOCK(&stream_.mutex);
+ // No need to do anything else here ... OSS automatically starts
+ // when fed samples.
+
+ MUTEX_UNLOCK( &stream_.mutex );
+
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ pthread_cond_signal( &handle->runnable );
}
-void RtApiAl :: stopStream()
+void RtApiOss :: stopStream()
{
verifyStream();
- if (stream_.state == STREAM_STOPPED) return;
-
- // Change the state before the lock to improve shutdown response
- // when using a callback.
- stream_.state = STREAM_STOPPED;
- MUTEX_LOCK(&stream_.mutex);
-
- int result, buffer_size = stream_.bufferSize * stream_.nBuffers;
- ALport *handle = (ALport *) stream_.apiHandle;
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX)
- alZeroFrames(handle[0], buffer_size);
+ MUTEX_LOCK( &stream_.mutex );
- if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
- result = alDiscardFrames(handle[1], buffer_size);
- if (result == -1) {
- sprintf(message_, "RtApiAl: error draining stream device (%s): %s.",
- devices_[stream_.device[1]].name.c_str(), alGetErrorString(oserror()));
- error(RtError::DRIVER_ERROR);
- }
+ // The state might change while waiting on a mutex.
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
}
- MUTEX_UNLOCK(&stream_.mutex);
-}
+ int result = 0;
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
-void RtApiAl :: abortStream()
-{
- verifyStream();
- if (stream_.state == STREAM_STOPPED) return;
+ // Flush the output with zeros a few times.
+ char *buffer;
+ int samples;
+ RtAudioFormat format;
- // Change the state before the lock to improve shutdown response
- // when using a callback.
- stream_.state = STREAM_STOPPED;
- MUTEX_LOCK(&stream_.mutex);
+ if ( stream_.doConvertBuffer[0] ) {
+ buffer = stream_.deviceBuffer;
+ samples = stream_.bufferSize * stream_.nDeviceChannels[0];
+ format = stream_.deviceFormat[0];
+ }
+ else {
+ buffer = stream_.userBuffer[0];
+ samples = stream_.bufferSize * stream_.nUserChannels[0];
+ format = stream_.userFormat;
+ }
+
+ memset( buffer, 0, samples * formatBytes(format) );
+ for ( unsigned int i=0; i<stream_.nBuffers+1; i++ ) {
+ result = write( handle->id[0], buffer, samples * formatBytes(format) );
+ if ( result == -1 ) {
+ errorText_ = "RtApiOss::stopStream: audio write error.";
+ error( RtAudioError::WARNING );
+ }
+ }
- ALport *handle = (ALport *) stream_.apiHandle;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
+ 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;
+ }
- int buffer_size = stream_.bufferSize * stream_.nBuffers;
- int result = alDiscardFrames(handle[0], buffer_size);
- if (result == -1) {
- sprintf(message_, "RtApiAl: error aborting stream device (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), 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::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
+ errorText_ = errorStream_.str();
+ goto unlock;
}
}
- // There is no clear action to take on the input stream, since the
- // port will continue to run in any event.
+ unlock:
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
- MUTEX_UNLOCK(&stream_.mutex);
+ if ( result != -1 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
}
-int RtApiAl :: streamWillBlock()
+void RtApiOss :: abortStream()
{
verifyStream();
+ if ( stream_.state == STREAM_STOPPED ) {
+ errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
+ error( RtAudioError::WARNING );
+ return;
+ }
- if (stream_.state == STREAM_STOPPED) return 0;
+ 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;
+ }
- int frames = 0;
- int err = 0;
- ALport *handle = (ALport *) stream_.apiHandle;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
- err = alGetFillable(handle[0]);
- if (err < 0) {
- sprintf(message_, "RtApiAl: error getting available frames for stream (%s): %s.",
- devices_[stream_.device[0]].name.c_str(), 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(handle[1]);
- if (err < 0) {
- sprintf(message_, "RtApiAl: error getting available frames for stream (%s): %s.",
- devices_[stream_.device[1]].name.c_str(), 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:
+ stream_.state = STREAM_STOPPED;
+ MUTEX_UNLOCK( &stream_.mutex );
- MUTEX_UNLOCK(&stream_.mutex);
- return frames;
+ if ( result != -1 ) return;
+ error( RtAudioError::SYSTEM_ERROR );
}
-void RtApiAl :: tickStream()
+void RtApiOss :: callbackEvent()
{
- verifyStream();
+ OssHandle *handle = (OssHandle *) stream_.apiHandle;
+ if ( stream_.state == STREAM_STOPPED ) {
+ MUTEX_LOCK( &stream_.mutex );
+ pthread_cond_wait( &handle->runnable, &stream_.mutex );
+ if ( stream_.state != STREAM_RUNNING ) {
+ MUTEX_UNLOCK( &stream_.mutex );
+ return;
+ }
+ MUTEX_UNLOCK( &stream_.mutex );
+ }
- 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( RtAudioError::WARNING );
return;
}
- else if (stream_.callbackInfo.usingCallback) {
- RtAudioCallback callback = (RtAudioCallback) 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;
+ if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
+ status |= RTAUDIO_OUTPUT_UNDERFLOW;
+ handle->xrun[0] = false;
+ }
+ if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
+ status |= RTAUDIO_INPUT_OVERFLOW;
+ handle->xrun[1] = false;
+ }
+ doStopStream = callback( stream_.userBuffer[0], stream_.userBuffer[1],
+ stream_.bufferSize, streamTime, status, stream_.callbackInfo.userData );
+ if ( doStopStream == 2 ) {
+ this->abortStream();
+ return;
}
- MUTEX_LOCK(&stream_.mutex);
+ 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;
+ int samples;
RtAudioFormat format;
- ALport *handle = (ALport *) stream_.apiHandle;
- if (stream_.mode == OUTPUT || stream_.mode == DUPLEX) {
+
+ if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
// Setup parameters and do buffer conversion if necessary.
- if (stream_.doConvertBuffer[0]) {
- convertStreamBuffer(OUTPUT);
+ if ( stream_.doConvertBuffer[0] ) {
buffer = stream_.deviceBuffer;
- channels = stream_.nDeviceChannels[0];
+ 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];
+ 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(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( RtAudioError::WARNING );
+ // Continue on to input section.
+ }
}
- if (stream_.mode == INPUT || stream_.mode == DUPLEX) {
+ if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
// Setup parameters.
- if (stream_.doConvertBuffer[1]) {
+ if ( stream_.doConvertBuffer[1] ) {
buffer = stream_.deviceBuffer;
- channels = stream_.nDeviceChannels[1];
+ samples = stream_.bufferSize * stream_.nDeviceChannels[1];
format = stream_.deviceFormat[1];
}
else {
- buffer = stream_.userBuffer;
- channels = stream_.nUserChannels[1];
+ buffer = stream_.userBuffer[1];
+ samples = stream_.bufferSize * stream_.nUserChannels[1];
format = stream_.userFormat;
}
- // Read interleaved samples from device.
- alReadFrames(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( RtAudioError::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(INPUT);
+ if ( stream_.doConvertBuffer[1] )
+ convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
}
unlock:
- MUTEX_UNLOCK(&stream_.mutex);
-
- if (stream_.callbackInfo.usingCallback && stopStream)
- this->stopStream();
-}
-
-void RtApiAl :: setStreamCallback(RtAudioCallback callback, void *userData)
-{
- verifyStream();
-
- CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
- if ( info->usingCallback ) {
- sprintf(message_, "RtApiAl: 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;
+ MUTEX_UNLOCK( &stream_.mutex );
- // 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);
- pthread_attr_setschedpolicy(&attr, SCHED_RR);
-
- int err = pthread_create(&info->thread, &attr, callbackHandler, &stream_.callbackInfo);
- pthread_attr_destroy(&attr);
- if (err) {
- info->usingCallback = false;
- sprintf(message_, "RtApiAl: error starting callback thread!");
- error(RtError::THREAD_ERROR);
- }
+ RtApi::tickStreamTime();
+ if ( doStopStream == 1 ) this->stopStream();
}
-void RtApiAl :: cancelStreamCallback()
+static void *ossCallbackHandler( void *ptr )
{
- verifyStream();
-
- if (stream_.callbackInfo.usingCallback) {
-
- if (stream_.state == STREAM_RUNNING)
- stopStream();
-
- MUTEX_LOCK(&stream_.mutex);
-
- stream_.callbackInfo.usingCallback = false;
- pthread_join(stream_.callbackInfo.thread, NULL);
- stream_.callbackInfo.thread = 0;
- stream_.callbackInfo.callback = NULL;
- stream_.callbackInfo.userData = NULL;
+ CallbackInfo *info = (CallbackInfo *) ptr;
+ RtApiOss *object = (RtApiOss *) info->object;
+ bool *isRunning = &info->isRunning;
- MUTEX_UNLOCK(&stream_.mutex);
+#ifdef SCHED_RR // Undefined with some OSes (e.g. NetBSD 1.6.x with GNU Pthread)
+ if (info->doRealtime) {
+ std::cerr << "RtAudio oss: " <<
+ (sched_getscheduler(0) == SCHED_RR ? "" : "_NOT_ ") <<
+ "running realtime scheduling" << std::endl;
}
-}
-
-extern "C" void *callbackHandler(void *ptr)
-{
- CallbackInfo *info = (CallbackInfo *) ptr;
- RtApiAl *object = (RtApiAl *) info->object;
- bool *usingCallback = &info->usingCallback;
+#endif
- while ( *usingCallback ) {
- try {
- object->tickStream();
- }
- catch (RtError &exception) {
- fprintf(stderr, "\nRtApiAl: callback thread error (%s) ... closing thread.\n\n",
- exception.getMessageString());
- break;
- }
+ while ( *isRunning == true ) {
+ pthread_testcancel();
+ object->callbackEvent();
}
- return 0;
+ pthread_exit( NULL );
}
-//******************** End of __IRIX_AL__ *********************//
+//******************** End of __LINUX_OSS__ *********************//
#endif
// *************************************************** //
// This method can be modified to control the behavior of error
-// message reporting and throwing.
-void RtApi :: error(RtError::Type type)
+// message printing.
+void RtApi :: error( RtAudioError::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 {
-#if defined(__RTAUDIO_DEBUG__)
- fprintf(stderr, "\n%s\n\n", message_);
-#endif
- throw RtError(std::string(message_), type);
+ errorStream_.str(""); // clear the ostringstream
+
+ RtAudioErrorCallback errorCallback = (RtAudioErrorCallback) stream_.callbackInfo.errorCallback;
+ if ( errorCallback ) {
+ // abortStream() can generate new error messages. Ignore them. Just keep original one.
+
+ if ( firstErrorOccurred_ )
+ return;
+
+ firstErrorOccurred_ = true;
+ const std::string errorMessage = errorText_;
+
+ if ( type != RtAudioError::WARNING && stream_.state != STREAM_STOPPED) {
+ stream_.callbackInfo.isRunning = false; // exit from the thread
+ abortStream();
+ }
+
+ errorCallback( type, errorMessage );
+ firstErrorOccurred_ = false;
+ return;
}
+
+ if ( type == RtAudioError::WARNING && showWarnings_ == true )
+ std::cerr << '\n' << errorText_ << "\n\n";
+ else if ( type != RtAudioError::WARNING )
+ throw( RtAudioError( errorText_, type ) );
}
void RtApi :: verifyStream()
{
- if ( stream_.mode == UNINITIALIZED ) {
- sprintf(message_, "RtAudio: a stream was not previously opened!");
- error(RtError::INVALID_STREAM);
+ if ( stream_.state == STREAM_CLOSED ) {
+ errorText_ = "RtApi:: a stream is not open!";
+ error( RtAudioError::INVALID_USE );
}
}
-void RtApi :: clearDeviceInfo(RtApiDevice *info)
-{
- // 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->sampleRates.clear();
- info->nativeFormats = 0;
-}
-
void RtApi :: clearStreamInfo()
{
stream_.mode = UNINITIALIZED;
- stream_.state = STREAM_STOPPED;
+ stream_.state = STREAM_CLOSED;
stream_.sampleRate = 0;
stream_.bufferSize = 0;
stream_.nBuffers = 0;
stream_.userFormat = 0;
+ stream_.userInterleaved = true;
+ stream_.streamTime = 0.0;
+ stream_.apiHandle = 0;
+ stream_.deviceBuffer = 0;
+ stream_.callbackInfo.callback = 0;
+ stream_.callbackInfo.userData = 0;
+ stream_.callbackInfo.isRunning = false;
+ stream_.callbackInfo.errorCallback = 0;
for ( int i=0; i<2; i++ ) {
- stream_.device[i] = 0;
+ stream_.device[i] = 11111;
stream_.doConvertBuffer[i] = false;
- stream_.deInterleave[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 RtApi :: formatBytes(RtAudioFormat 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_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_SINT24 )
+ return 3;
+ else if ( format == RTAUDIO_SINT8 )
return 1;
- sprintf(message_,"RtApi: undefined format in formatBytes().");
- error(RtError::WARNING);
+ errorText_ = "RtApi::formatBytes: undefined format.";
+ error( RtAudioError::WARNING );
return 0;
}
-void RtApi :: convertStreamBuffer( StreamMode 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;
- RtAudioFormat 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 lower three bytes of a 32-bit integer.
+
+ // Clear our device buffer when in/out duplex device channels are different
+ if ( outBuffer == stream_.deviceBuffer && stream_.mode == DUPLEX &&
+ ( stream_.nDeviceChannels[0] < stream_.nDeviceChannels[1] ) )
+ memset( outBuffer, 0, stream_.bufferSize * info.outJump * formatBytes( info.outFormat ) );
- if (format_out == RTAUDIO_FLOAT64) {
+ int j;
+ if (info.outFormat == RTAUDIO_FLOAT64) {
Float64 scale;
- Float64 *out = (Float64 *)output;
-
- if (format_in == RTAUDIO_SINT8) {
- signed char *in = (signed char *)input;
- 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;
+ Float64 *out = (Float64 *)outBuffer;
+
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ scale = 1.0 / 127.5;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT16) {
- Int16 *in = (Int16 *)input;
- 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;
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ scale = 1.0 / 32767.5;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
}
- in += 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) {
+ Int24 *in = (Int24 *)inBuffer;
+ scale = 1.0 / 8388607.5;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) (in[info.inOffset[j]].asInt());
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT32) {
- 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]];
- out[offset_out[j]] *= scale;
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ scale = 1.0 / 2147483647.5;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float64) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
}
- in += 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) {
+ else if (info.outFormat == RTAUDIO_FLOAT32) {
Float32 scale;
- Float32 *out = (Float32 *)output;
-
- if (format_in == RTAUDIO_SINT8) {
- signed char *in = (signed char *)input;
- 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;
+ Float32 *out = (Float32 *)outBuffer;
+
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ scale = (Float32) ( 1.0 / 127.5 );
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT16) {
- Int16 *in = (Int16 *)input;
- 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;
+ else if (info.inFormat == RTAUDIO_SINT16) {
+ Int16 *in = (Int16 *)inBuffer;
+ scale = (Float32) ( 1.0 / 32767.5 );
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
}
- in += 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) {
+ Int24 *in = (Int24 *)inBuffer;
+ scale = (Float32) ( 1.0 / 8388607.5 );
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) (in[info.inOffset[j]].asInt());
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
- else if (format_in == RTAUDIO_SINT32) {
- 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]];
- out[offset_out[j]] *= scale;
+ else if (info.inFormat == RTAUDIO_SINT32) {
+ Int32 *in = (Int32 *)inBuffer;
+ scale = (Float32) ( 1.0 / 2147483647.5 );
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Float32) in[info.inOffset[j]];
+ out[info.outOffset[j]] += 0.5;
+ out[info.outOffset[j]] *= scale;
}
- in += 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) {
+ Int24 *in = (Int24 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) in[info.inOffset[j]].asInt();
+ out[info.outOffset[j]] <<= 8;
}
- in += 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]] * 2147483647.5 - 0.5);
}
- 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]] * 2147483647.5 - 0.5);
}
- 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) {
+ Int24 *out = (Int24 *)outBuffer;
+ if (info.inFormat == RTAUDIO_SINT8) {
+ signed char *in = (signed char *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int32) (in[info.inOffset[j]] << 16);
+ //out[info.outOffset[j]] <<= 16;
}
- in += 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]] << 8);
+ //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]];
+ Int24 *in = (Int24 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = in[info.inOffset[j]];
}
- in += 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]] >> 8);
+ //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]] * 8388607.5 - 0.5);
}
- 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]] * 8388607.5 - 0.5);
}
- 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) {
+ Int24 *in = (Int24 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (Int16) (in[info.inOffset[j]].asInt() >> 8);
}
- in += 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]] * 32767.5 - 0.5);
}
- 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]] * 32767.5 - 0.5);
}
- 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) {
+ Int24 *in = (Int24 *)inBuffer;
+ for (unsigned int i=0; i<stream_.bufferSize; i++) {
+ for (j=0; j<info.channels; j++) {
+ out[info.outOffset[j]] = (signed char) (in[info.inOffset[j]].asInt() >> 16);
}
- in += 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]] * 127.5 - 0.5);
}
- 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]] * 127.5 - 0.5);
}
- in += jump_in;
- out += jump_out;
+ in += info.inJump;
+ out += info.outJump;
}
}
}
}
-void RtApi :: byteSwapBuffer( char *buffer, int samples, RtAudioFormat format )
+//static inline uint16_t bswap_16(uint16_t x) { return (x>>8) | (x<<8); }
+//static inline uint32_t bswap_32(uint32_t x) { return (bswap_16(x&0xffff)<<16) | (bswap_16(x>>16)); }
+//static inline uint64_t bswap_64(uint64_t x) { return (((unsigned long long)bswap_32(x&0xffffffffull))<<32) | (bswap_32(x>>32)); }
+
+void RtApi :: byteSwapBuffer( char *buffer, unsigned int samples, RtAudioFormat format )
{
- register char val;
- register char *ptr;
+ char val;
+ 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_SINT32 ||
+ format == RTAUDIO_FLOAT32 ) {
+ for ( unsigned int i=0; i<samples; i++ ) {
// Swap 1st and 4th bytes.
val = *(ptr);
*(ptr) = *(ptr+3);
*(ptr) = *(ptr+1);
*(ptr+1) = val;
- // Increment 4 bytes.
- ptr += 4;
+ // Increment 3 more bytes.
+ ptr += 3;
+ }
+ }
+ else if ( format == RTAUDIO_SINT24 ) {
+ for ( unsigned int i=0; i<samples; i++ ) {
+ // Swap 1st and 3rd bytes.
+ val = *(ptr);
+ *(ptr) = *(ptr+2);
+ *(ptr+2) = val;
+
+ // Increment 2 more bytes.
+ ptr += 2;
}
}
- else if (format == RTAUDIO_FLOAT64) {
- for (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);
*(ptr) = *(ptr+1);
*(ptr+1) = val;
- // Increment 8 bytes.
- ptr += 8;
+ // Increment 5 more bytes.
+ ptr += 5;
}
}
}
+
+ // Indentation settings for Vim and Emacs
+ //
+ // Local Variables:
+ // c-basic-offset: 2
+ // indent-tabs-mode: nil
+ // End:
+ //
+ // vim: et sts=2 sw=2
+
projects / rtaudio-cdist.git / blobdiff