7894 lines
268 KiB
C++
7894 lines
268 KiB
C++
#ifdef RTAUDIO_ENABLED
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/************************************************************************/
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/*! \class RtAudio
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\brief Realtime audio i/o C++ classes.
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RtAudio provides a common API (Application Programming Interface)
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for realtime audio input/output across Linux (native ALSA, Jack,
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and OSS), Macintosh OS X (CoreAudio and Jack), and Windows
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(DirectSound and ASIO) operating systems.
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RtAudio WWW site: http://www.music.mcgill.ca/~gary/rtaudio/
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RtAudio: realtime audio i/o C++ classes
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Copyright (c) 2001-2009 Gary P. Scavone
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Permission is hereby granted, free of charge, to any person
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obtaining a copy of this software and associated documentation files
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(the "Software"), to deal in the Software without restriction,
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including without limitation the rights to use, copy, modify, merge,
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publish, distribute, sublicense, and/or sell copies of the Software,
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and to permit persons to whom the Software is furnished to do so,
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subject to the following conditions:
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The above copyright notice and this permission notice shall be
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included in all copies or substantial portions of the Software.
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Any person wishing to distribute modifications to the Software is
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asked to send the modifications to the original developer so that
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they can be incorporated into the canonical version. This is,
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however, not a binding provision of this license.
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THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
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EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
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MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
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IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
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ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
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CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
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WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
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*/
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/************************************************************************/
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// RtAudio: Version 4.0.6
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#include "RtAudio.h"
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#include <iostream>
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#include <cstdlib>
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#include <cstring>
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#include <limits.h>
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// Static variable definitions.
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const unsigned int RtApi::MAX_SAMPLE_RATES = 14;
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const unsigned int RtApi::SAMPLE_RATES[] = {
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4000, 5512, 8000, 9600, 11025, 16000, 22050,
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32000, 44100, 48000, 88200, 96000, 176400, 192000
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};
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#if defined(__WINDOWS_DS__) || defined(__WINDOWS_ASIO__)
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#define MUTEX_INITIALIZE(A) InitializeCriticalSection(A)
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#define MUTEX_DESTROY(A) DeleteCriticalSection(A)
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#define MUTEX_LOCK(A) EnterCriticalSection(A)
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#define MUTEX_UNLOCK(A) LeaveCriticalSection(A)
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#elif defined(__LINUX_ALSA__) || defined(__UNIX_JACK__) || defined(__LINUX_OSS__) || defined(__MACOSX_CORE__)
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// pthread API
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#define MUTEX_INITIALIZE(A) pthread_mutex_init(A, NULL)
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#define MUTEX_DESTROY(A) pthread_mutex_destroy(A)
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#define MUTEX_LOCK(A) pthread_mutex_lock(A)
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#define MUTEX_UNLOCK(A) pthread_mutex_unlock(A)
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#else
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#define MUTEX_INITIALIZE(A) abs(*A) // dummy definitions
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#define MUTEX_DESTROY(A) abs(*A) // dummy definitions
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#endif
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// *************************************************** //
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//
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// RtAudio definitions.
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//
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// *************************************************** //
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void RtAudio :: getCompiledApi( std::vector<RtAudio::Api> &apis ) throw()
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{
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apis.clear();
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// The order here will control the order of RtAudio's API search in
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// the constructor.
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#if defined(__UNIX_JACK__)
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apis.push_back( UNIX_JACK );
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#endif
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#if defined(__LINUX_ALSA__)
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apis.push_back( LINUX_ALSA );
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#endif
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#if defined(__LINUX_OSS__)
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apis.push_back( LINUX_OSS );
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#endif
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#if defined(__WINDOWS_ASIO__)
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apis.push_back( WINDOWS_ASIO );
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#endif
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#if defined(__WINDOWS_DS__)
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apis.push_back( WINDOWS_DS );
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#endif
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#if defined(__MACOSX_CORE__)
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apis.push_back( MACOSX_CORE );
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#endif
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#if defined(__RTAUDIO_DUMMY__)
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apis.push_back( RTAUDIO_DUMMY );
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#endif
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}
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void RtAudio :: openRtApi( RtAudio::Api api )
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{
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#if defined(__UNIX_JACK__)
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if ( api == UNIX_JACK )
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rtapi_ = new RtApiJack();
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#endif
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#if defined(__LINUX_ALSA__)
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if ( api == LINUX_ALSA )
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rtapi_ = new RtApiAlsa();
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#endif
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#if defined(__LINUX_OSS__)
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if ( api == LINUX_OSS )
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rtapi_ = new RtApiOss();
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#endif
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#if defined(__WINDOWS_ASIO__)
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if ( api == WINDOWS_ASIO )
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rtapi_ = new RtApiAsio();
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#endif
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#if defined(__WINDOWS_DS__)
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if ( api == WINDOWS_DS )
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rtapi_ = new RtApiDs();
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#endif
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#if defined(__MACOSX_CORE__)
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if ( api == MACOSX_CORE )
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rtapi_ = new RtApiCore();
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#endif
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#if defined(__RTAUDIO_DUMMY__)
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if ( api == RTAUDIO_DUMMY )
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rtapi_ = new RtApiDummy();
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#endif
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}
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RtAudio :: RtAudio( RtAudio::Api api ) throw()
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{
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rtapi_ = 0;
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if ( api != UNSPECIFIED ) {
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// Attempt to open the specified API.
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openRtApi( api );
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if ( rtapi_ ) return;
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// No compiled support for specified API value. Issue a debug
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// warning and continue as if no API was specified.
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std::cerr << "\nRtAudio: no compiled support for specified API argument!\n" << std::endl;
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}
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// Iterate through the compiled APIs and return as soon as we find
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// one with at least one device or we reach the end of the list.
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std::vector< RtAudio::Api > apis;
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getCompiledApi( apis );
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for ( unsigned int i=0; i<apis.size(); i++ ) {
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openRtApi( apis[i] );
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if ( rtapi_->getDeviceCount() ) break;
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}
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if ( rtapi_ ) return;
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// It should not be possible to get here because the preprocessor
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// definition __RTAUDIO_DUMMY__ is automatically defined if no
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// API-specific definitions are passed to the compiler. But just in
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// case something weird happens, we'll print out an error message.
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std::cerr << "\nRtAudio: no compiled API support found ... critical error!!\n\n";
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}
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RtAudio :: ~RtAudio() throw()
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{
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delete rtapi_;
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}
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void RtAudio :: openStream( RtAudio::StreamParameters *outputParameters,
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RtAudio::StreamParameters *inputParameters,
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RtAudioFormat format, unsigned int sampleRate,
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unsigned int *bufferFrames,
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RtAudioCallback callback, void *userData,
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RtAudio::StreamOptions *options )
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{
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return rtapi_->openStream( outputParameters, inputParameters, format,
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sampleRate, bufferFrames, callback,
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userData, options );
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}
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// *************************************************** //
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//
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// Public RtApi definitions (see end of file for
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// private or protected utility functions).
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//
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// *************************************************** //
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RtApi :: RtApi()
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{
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stream_.state = STREAM_CLOSED;
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stream_.mode = UNINITIALIZED;
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stream_.apiHandle = 0;
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stream_.userBuffer[0] = 0;
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stream_.userBuffer[1] = 0;
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MUTEX_INITIALIZE( &stream_.mutex );
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showWarnings_ = true;
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}
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RtApi :: ~RtApi()
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{
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MUTEX_DESTROY( &stream_.mutex );
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}
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void RtApi :: openStream( RtAudio::StreamParameters *oParams,
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RtAudio::StreamParameters *iParams,
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RtAudioFormat format, unsigned int sampleRate,
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unsigned int *bufferFrames,
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RtAudioCallback callback, void *userData,
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RtAudio::StreamOptions *options )
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{
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if ( stream_.state != STREAM_CLOSED ) {
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errorText_ = "RtApi::openStream: a stream is already open!";
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error( RtError::INVALID_USE );
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}
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if ( oParams && oParams->nChannels < 1 ) {
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errorText_ = "RtApi::openStream: a non-NULL output StreamParameters structure cannot have an nChannels value less than one.";
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error( RtError::INVALID_USE );
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}
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if ( iParams && iParams->nChannels < 1 ) {
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errorText_ = "RtApi::openStream: a non-NULL input StreamParameters structure cannot have an nChannels value less than one.";
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error( RtError::INVALID_USE );
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}
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if ( oParams == NULL && iParams == NULL ) {
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errorText_ = "RtApi::openStream: input and output StreamParameters structures are both NULL!";
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error( RtError::INVALID_USE );
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}
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if ( formatBytes(format) == 0 ) {
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errorText_ = "RtApi::openStream: 'format' parameter value is undefined.";
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error( RtError::INVALID_USE );
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}
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unsigned int nDevices = getDeviceCount();
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unsigned int oChannels = 0;
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if ( oParams ) {
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oChannels = oParams->nChannels;
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if ( oParams->deviceId >= nDevices ) {
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errorText_ = "RtApi::openStream: output device parameter value is invalid.";
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error( RtError::INVALID_USE );
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}
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}
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unsigned int iChannels = 0;
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if ( iParams ) {
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iChannels = iParams->nChannels;
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if ( iParams->deviceId >= nDevices ) {
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errorText_ = "RtApi::openStream: input device parameter value is invalid.";
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error( RtError::INVALID_USE );
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}
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}
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clearStreamInfo();
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bool result;
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if ( oChannels > 0 ) {
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result = probeDeviceOpen( oParams->deviceId, OUTPUT, oChannels, oParams->firstChannel,
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sampleRate, format, bufferFrames, options );
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if ( result == false ) error( RtError::SYSTEM_ERROR );
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}
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if ( iChannels > 0 ) {
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result = probeDeviceOpen( iParams->deviceId, INPUT, iChannels, iParams->firstChannel,
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sampleRate, format, bufferFrames, options );
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if ( result == false ) {
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if ( oChannels > 0 ) closeStream();
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error( RtError::SYSTEM_ERROR );
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}
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}
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stream_.callbackInfo.callback = (void *) callback;
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stream_.callbackInfo.userData = userData;
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if ( options ) options->numberOfBuffers = stream_.nBuffers;
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stream_.state = STREAM_STOPPED;
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}
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unsigned int RtApi :: getDefaultInputDevice( void )
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{
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// Should be implemented in subclasses if possible.
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return 0;
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}
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unsigned int RtApi :: getDefaultOutputDevice( void )
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{
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// Should be implemented in subclasses if possible.
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return 0;
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}
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void RtApi :: closeStream( void )
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{
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// MUST be implemented in subclasses!
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return;
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}
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bool RtApi :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
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unsigned int firstChannel, unsigned int sampleRate,
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RtAudioFormat format, unsigned int *bufferSize,
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RtAudio::StreamOptions *options )
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{
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// MUST be implemented in subclasses!
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return FAILURE;
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}
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void RtApi :: tickStreamTime( void )
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{
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// Subclasses that do not provide their own implementation of
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// getStreamTime should call this function once per buffer I/O to
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// provide basic stream time support.
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stream_.streamTime += ( stream_.bufferSize * 1.0 / stream_.sampleRate );
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#if defined( HAVE_GETTIMEOFDAY )
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gettimeofday( &stream_.lastTickTimestamp, NULL );
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#endif
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}
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long RtApi :: getStreamLatency( void )
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{
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verifyStream();
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long totalLatency = 0;
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if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
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totalLatency = stream_.latency[0];
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if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
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totalLatency += stream_.latency[1];
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return totalLatency;
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}
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double RtApi :: getStreamTime( void )
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{
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verifyStream();
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#if defined( HAVE_GETTIMEOFDAY )
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// Return a very accurate estimate of the stream time by
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// adding in the elapsed time since the last tick.
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struct timeval then;
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struct timeval now;
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if ( stream_.state != STREAM_RUNNING || stream_.streamTime == 0.0 )
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return stream_.streamTime;
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gettimeofday( &now, NULL );
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then = stream_.lastTickTimestamp;
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return stream_.streamTime +
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((now.tv_sec + 0.000001 * now.tv_usec) -
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(then.tv_sec + 0.000001 * then.tv_usec));
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#else
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return stream_.streamTime;
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#endif
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}
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unsigned int RtApi :: getStreamSampleRate( void )
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{
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verifyStream();
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return stream_.sampleRate;
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}
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// *************************************************** //
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//
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// OS/API-specific methods.
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//
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// *************************************************** //
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#if defined(__MACOSX_CORE__)
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// The OS X CoreAudio API is designed to use a separate callback
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// procedure for each of its audio devices. A single RtAudio duplex
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// stream using two different devices is supported here, though it
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// cannot be guaranteed to always behave correctly because we cannot
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// synchronize these two callbacks.
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//
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// A property listener is installed for over/underrun information.
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// However, no functionality is currently provided to allow property
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// listeners to trigger user handlers because it is unclear what could
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// be done if a critical stream parameter (buffer size, sample rate,
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// device disconnect) notification arrived. The listeners entail
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// quite a bit of extra code and most likely, a user program wouldn't
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// be prepared for the result anyway. However, we do provide a flag
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// to the client callback function to inform of an over/underrun.
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//
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// The mechanism for querying and setting system parameters was
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// updated (and perhaps simplified) in OS-X version 10.4. However,
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// since 10.4 support is not necessarily available to all users, I've
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// decided not to update the respective code at this time. Perhaps
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// this will happen when Apple makes 10.4 free for everyone. :-)
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// A structure to hold various information related to the CoreAudio API
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// implementation.
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struct CoreHandle {
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AudioDeviceID id[2]; // device ids
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#if defined( MAC_OS_X_VERSION_10_5 ) && ( MAC_OS_X_VERSION_MIN_REQUIRED >= MAC_OS_X_VERSION_10_5 )
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AudioDeviceIOProcID procId[2];
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#endif
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UInt32 iStream[2]; // device stream index (or first if using multiple)
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UInt32 nStreams[2]; // number of streams to use
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bool xrun[2];
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char *deviceBuffer;
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pthread_cond_t condition;
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int drainCounter; // Tracks callback counts when draining
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bool internalDrain; // Indicates if stop is initiated from callback or not.
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CoreHandle()
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:deviceBuffer(0), drainCounter(0), internalDrain(false) { nStreams[0] = 1; nStreams[1] = 1; id[0] = 0; id[1] = 0; xrun[0] = false; xrun[1] = false; }
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};
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RtApiCore :: RtApiCore()
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{
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// Nothing to do here.
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}
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RtApiCore :: ~RtApiCore()
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{
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// The subclass destructor gets called before the base class
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// destructor, so close an existing stream before deallocating
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// apiDeviceId memory.
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if ( stream_.state != STREAM_CLOSED ) closeStream();
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}
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unsigned int RtApiCore :: getDeviceCount( void )
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{
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// Find out how many audio devices there are, if any.
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UInt32 dataSize;
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OSStatus result = AudioHardwareGetPropertyInfo( kAudioHardwarePropertyDevices, &dataSize, NULL );
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if ( result != noErr ) {
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errorText_ = "RtApiCore::getDeviceCount: OS-X error getting device info!";
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error( RtError::WARNING );
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return 0;
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}
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return dataSize / sizeof( AudioDeviceID );
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}
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unsigned int RtApiCore :: getDefaultInputDevice( void )
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{
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unsigned int nDevices = getDeviceCount();
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if ( nDevices <= 1 ) return 0;
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AudioDeviceID id;
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UInt32 dataSize = sizeof( AudioDeviceID );
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OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultInputDevice,
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&dataSize, &id );
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if ( result != noErr ) {
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errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device.";
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error( RtError::WARNING );
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return 0;
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}
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dataSize *= nDevices;
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AudioDeviceID deviceList[ nDevices ];
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result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
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if ( result != noErr ) {
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errorText_ = "RtApiCore::getDefaultInputDevice: OS-X system error getting device IDs.";
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error( RtError::WARNING );
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return 0;
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}
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for ( unsigned int i=0; i<nDevices; i++ )
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if ( id == deviceList[i] ) return i;
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errorText_ = "RtApiCore::getDefaultInputDevice: No default device found!";
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error( RtError::WARNING );
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return 0;
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}
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unsigned int RtApiCore :: getDefaultOutputDevice( void )
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{
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unsigned int nDevices = getDeviceCount();
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if ( nDevices <= 1 ) return 0;
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AudioDeviceID id;
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UInt32 dataSize = sizeof( AudioDeviceID );
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OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDefaultOutputDevice,
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&dataSize, &id );
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if ( result != noErr ) {
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errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device.";
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error( RtError::WARNING );
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return 0;
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}
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dataSize *= nDevices;
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AudioDeviceID deviceList[ nDevices ];
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result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
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if ( result != noErr ) {
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errorText_ = "RtApiCore::getDefaultOutputDevice: OS-X system error getting device IDs.";
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error( RtError::WARNING );
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return 0;
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}
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for ( unsigned int i=0; i<nDevices; i++ )
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if ( id == deviceList[i] ) return i;
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errorText_ = "RtApiCore::getDefaultOutputDevice: No default device found!";
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error( RtError::WARNING );
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return 0;
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}
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|
|
RtAudio::DeviceInfo RtApiCore :: getDeviceInfo( unsigned int device )
|
|
{
|
|
RtAudio::DeviceInfo info;
|
|
info.probed = false;
|
|
|
|
// Get device ID
|
|
unsigned int nDevices = getDeviceCount();
|
|
if ( nDevices == 0 ) {
|
|
errorText_ = "RtApiCore::getDeviceInfo: no devices found!";
|
|
error( RtError::INVALID_USE );
|
|
}
|
|
|
|
if ( device >= nDevices ) {
|
|
errorText_ = "RtApiCore::getDeviceInfo: device ID is invalid!";
|
|
error( RtError::INVALID_USE );
|
|
}
|
|
|
|
AudioDeviceID deviceList[ nDevices ];
|
|
UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
|
|
OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
|
|
if ( result != noErr ) {
|
|
errorText_ = "RtApiCore::getDeviceInfo: OS-X system error getting device IDs.";
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
AudioDeviceID id = deviceList[ device ];
|
|
|
|
// Get the device name.
|
|
info.name.erase();
|
|
char name[256];
|
|
dataSize = 256;
|
|
result = AudioDeviceGetProperty( id, 0, false,
|
|
kAudioDevicePropertyDeviceManufacturer,
|
|
&dataSize, name );
|
|
|
|
if ( result != noErr ) {
|
|
errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device manufacturer.";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
info.name.append( (const char *)name, strlen(name) );
|
|
info.name.append( ": " );
|
|
|
|
dataSize = 256;
|
|
result = AudioDeviceGetProperty( id, 0, false,
|
|
kAudioDevicePropertyDeviceName,
|
|
&dataSize, name );
|
|
if ( result != noErr ) {
|
|
errorStream_ << "RtApiCore::probeDeviceInfo: system error (" << getErrorCode( result ) << ") getting device name.";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
info.name.append( (const char *)name, strlen(name) );
|
|
|
|
// Get the output stream "configuration".
|
|
AudioBufferList *bufferList = nil;
|
|
result = AudioDeviceGetPropertyInfo( id, 0, false,
|
|
kAudioDevicePropertyStreamConfiguration,
|
|
&dataSize, NULL );
|
|
if (result != noErr || dataSize == 0) {
|
|
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration info for device (" << device << ").";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
// Allocate the AudioBufferList.
|
|
bufferList = (AudioBufferList *) malloc( dataSize );
|
|
if ( bufferList == NULL ) {
|
|
errorText_ = "RtApiCore::getDeviceInfo: memory error allocating output AudioBufferList.";
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
result = AudioDeviceGetProperty( id, 0, false,
|
|
kAudioDevicePropertyStreamConfiguration,
|
|
&dataSize, bufferList );
|
|
if ( result != noErr ) {
|
|
free( bufferList );
|
|
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting output stream configuration for device (" << device << ").";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
// Get output channel information.
|
|
unsigned int i, nStreams = bufferList->mNumberBuffers;
|
|
for ( i=0; i<nStreams; i++ )
|
|
info.outputChannels += bufferList->mBuffers[i].mNumberChannels;
|
|
free( bufferList );
|
|
|
|
// Get the input stream "configuration".
|
|
result = AudioDeviceGetPropertyInfo( id, 0, true,
|
|
kAudioDevicePropertyStreamConfiguration,
|
|
&dataSize, NULL );
|
|
if (result != noErr || dataSize == 0) {
|
|
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration info for device (" << device << ").";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
// Allocate the AudioBufferList.
|
|
bufferList = (AudioBufferList *) malloc( dataSize );
|
|
if ( bufferList == NULL ) {
|
|
errorText_ = "RtApiCore::getDeviceInfo: memory error allocating input AudioBufferList.";
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
result = AudioDeviceGetProperty( id, 0, true,
|
|
kAudioDevicePropertyStreamConfiguration,
|
|
&dataSize, bufferList );
|
|
if ( result != noErr ) {
|
|
free( bufferList );
|
|
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting input stream configuration for device (" << device << ").";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
// Get input channel information.
|
|
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.
|
|
result = AudioDeviceGetPropertyInfo( id, 0, isInput,
|
|
kAudioDevicePropertyAvailableNominalSampleRates,
|
|
&dataSize, NULL );
|
|
|
|
if ( result != kAudioHardwareNoError || dataSize == 0 ) {
|
|
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rate info.";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
UInt32 nRanges = dataSize / sizeof( AudioValueRange );
|
|
AudioValueRange rangeList[ nRanges ];
|
|
result = AudioDeviceGetProperty( id, 0, isInput,
|
|
kAudioDevicePropertyAvailableNominalSampleRates,
|
|
&dataSize, &rangeList );
|
|
|
|
if ( result != kAudioHardwareNoError ) {
|
|
errorStream_ << "RtApiCore::getDeviceInfo: system error (" << getErrorCode( result ) << ") getting sample rates.";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
Float64 minimumRate = 100000000.0, maximumRate = 0.0;
|
|
for ( UInt32 i=0; i<nRanges; i++ ) {
|
|
if ( rangeList[i].mMinimum < minimumRate ) minimumRate = rangeList[i].mMinimum;
|
|
if ( rangeList[i].mMaximum > maximumRate ) maximumRate = rangeList[i].mMaximum;
|
|
}
|
|
|
|
info.sampleRates.clear();
|
|
for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
|
|
if ( SAMPLE_RATES[k] >= (unsigned int) minimumRate && SAMPLE_RATES[k] <= (unsigned int) maximumRate )
|
|
info.sampleRates.push_back( SAMPLE_RATES[k] );
|
|
}
|
|
|
|
if ( info.sampleRates.size() == 0 ) {
|
|
errorStream_ << "RtApiCore::probeDeviceInfo: No supported sample rates found for device (" << device << ").";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
// 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 ( getDefaultOutputDevice() == device )
|
|
info.isDefaultOutput = true;
|
|
if ( getDefaultInputDevice() == device )
|
|
info.isDefaultInput = true;
|
|
|
|
info.probed = true;
|
|
return info;
|
|
}
|
|
|
|
OSStatus callbackHandler( AudioDeviceID inDevice,
|
|
const AudioTimeStamp* inNow,
|
|
const AudioBufferList* inInputData,
|
|
const AudioTimeStamp* inInputTime,
|
|
AudioBufferList* outOutputData,
|
|
const AudioTimeStamp* inOutputTime,
|
|
void* infoPointer )
|
|
{
|
|
CallbackInfo *info = (CallbackInfo *) infoPointer;
|
|
|
|
RtApiCore *object = (RtApiCore *) info->object;
|
|
if ( object->callbackEvent( inDevice, inInputData, outOutputData ) == false )
|
|
return kAudioHardwareUnspecifiedError;
|
|
else
|
|
return kAudioHardwareNoError;
|
|
}
|
|
|
|
OSStatus deviceListener( AudioDeviceID inDevice,
|
|
UInt32 channel,
|
|
Boolean isInput,
|
|
AudioDevicePropertyID propertyID,
|
|
void* handlePointer )
|
|
{
|
|
CoreHandle *handle = (CoreHandle *) handlePointer;
|
|
if ( propertyID == kAudioDeviceProcessorOverload ) {
|
|
if ( isInput )
|
|
handle->xrun[1] = true;
|
|
else
|
|
handle->xrun[0] = true;
|
|
}
|
|
|
|
return kAudioHardwareNoError;
|
|
}
|
|
|
|
static bool hasProperty( AudioDeviceID id, UInt32 channel, bool isInput, AudioDevicePropertyID property )
|
|
{
|
|
OSStatus result = AudioDeviceGetPropertyInfo( id, channel, isInput, property, NULL, NULL );
|
|
return result == 0;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
AudioDeviceID deviceList[ nDevices ];
|
|
UInt32 dataSize = sizeof( AudioDeviceID ) * nDevices;
|
|
OSStatus result = AudioHardwareGetProperty( kAudioHardwarePropertyDevices, &dataSize, (void *) &deviceList );
|
|
if ( result != noErr ) {
|
|
errorText_ = "RtApiCore::probeDeviceOpen: OS-X system error getting device IDs.";
|
|
return FAILURE;
|
|
}
|
|
|
|
AudioDeviceID id = deviceList[ device ];
|
|
|
|
// Setup for stream mode.
|
|
bool isInput = false;
|
|
if ( mode == INPUT ) isInput = true;
|
|
|
|
// Set or disable "hog" mode.
|
|
dataSize = sizeof( UInt32 );
|
|
UInt32 doHog = 0;
|
|
if ( options && options->flags & RTAUDIO_HOG_DEVICE ) doHog = 1;
|
|
result = AudioHardwareSetProperty( kAudioHardwarePropertyHogModeIsAllowed, dataSize, &doHog );
|
|
if ( result != noErr ) {
|
|
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting 'hog' state!";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Get the stream "configuration".
|
|
AudioBufferList *bufferList;
|
|
result = AudioDeviceGetPropertyInfo( id, 0, isInput,
|
|
kAudioDevicePropertyStreamConfiguration,
|
|
&dataSize, NULL );
|
|
if (result != noErr || dataSize == 0) {
|
|
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration info for device (" << device << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Allocate the AudioBufferList.
|
|
bufferList = (AudioBufferList *) malloc( dataSize );
|
|
if ( bufferList == NULL ) {
|
|
errorText_ = "RtApiCore::probeDeviceOpen: memory error allocating AudioBufferList.";
|
|
return FAILURE;
|
|
}
|
|
|
|
result = AudioDeviceGetProperty( id, 0, isInput,
|
|
kAudioDevicePropertyStreamConfiguration,
|
|
&dataSize, bufferList );
|
|
if ( result != noErr ) {
|
|
free( bufferList );
|
|
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream configuration for device (" << device << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Search for 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;
|
|
}
|
|
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;
|
|
}
|
|
|
|
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++;
|
|
}
|
|
}
|
|
|
|
free( bufferList );
|
|
|
|
// Determine the buffer size.
|
|
AudioValueRange bufferRange;
|
|
dataSize = sizeof( AudioValueRange );
|
|
result = AudioDeviceGetProperty( id, 0, isInput,
|
|
kAudioDevicePropertyBufferFrameSizeRange,
|
|
&dataSize, &bufferRange );
|
|
if ( result != noErr ) {
|
|
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting buffer size range for device (" << device << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
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 );
|
|
result = AudioDeviceSetProperty( id, NULL, 0, isInput,
|
|
kAudioDevicePropertyBufferFrameSize,
|
|
dataSize, &theSize );
|
|
|
|
if ( result != noErr ) {
|
|
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting the buffer size for device (" << device << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// If attempting to setup a duplex stream, the bufferSize parameter
|
|
// MUST be the same in both directions!
|
|
*bufferSize = 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;
|
|
|
|
// Get the stream ID(s) so we can set the stream format. We'll have
|
|
// to do this for each stream.
|
|
AudioStreamID streamIDs[ nStreams ];
|
|
dataSize = nStreams * sizeof( AudioStreamID );
|
|
result = AudioDeviceGetProperty( id, 0, isInput,
|
|
kAudioDevicePropertyStreams,
|
|
&dataSize, &streamIDs );
|
|
if ( result != noErr ) {
|
|
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream ID(s) for device (" << device << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Now set the stream format. Also, check the physical format of the
|
|
// device and change that if necessary.
|
|
AudioStreamBasicDescription description;
|
|
dataSize = sizeof( AudioStreamBasicDescription );
|
|
|
|
bool updateFormat;
|
|
for ( UInt32 i=0; i<streamCount; i++ ) {
|
|
|
|
result = AudioStreamGetProperty( streamIDs[firstStream+i], 0,
|
|
kAudioStreamPropertyVirtualFormat,
|
|
&dataSize, &description );
|
|
|
|
if ( result != noErr ) {
|
|
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream format for device (" << device << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Set the sample rate and data format id. However, only make the
|
|
// change if the sample rate is not within 1.0 of the desired
|
|
// rate and the format is not linear pcm.
|
|
updateFormat = false;
|
|
if ( fabs( description.mSampleRate - (double)sampleRate ) > 1.0 ) {
|
|
description.mSampleRate = (double) sampleRate;
|
|
updateFormat = true;
|
|
}
|
|
|
|
if ( description.mFormatID != kAudioFormatLinearPCM ) {
|
|
description.mFormatID = kAudioFormatLinearPCM;
|
|
updateFormat = true;
|
|
}
|
|
|
|
if ( updateFormat ) {
|
|
result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0,
|
|
kAudioStreamPropertyVirtualFormat,
|
|
dataSize, &description );
|
|
if ( result != noErr ) {
|
|
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting sample rate or data format for device (" << device << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
}
|
|
|
|
// Now check the physical format.
|
|
result = AudioStreamGetProperty( streamIDs[firstStream+i], 0,
|
|
kAudioStreamPropertyPhysicalFormat,
|
|
&dataSize, &description );
|
|
if ( result != noErr ) {
|
|
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream physical format for device (" << device << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
if ( description.mFormatID != kAudioFormatLinearPCM || description.mBitsPerChannel < 24 ) {
|
|
description.mFormatID = kAudioFormatLinearPCM;
|
|
AudioStreamBasicDescription testDescription = description;
|
|
unsigned long formatFlags;
|
|
|
|
// We'll try higher bit rates first and then work our way down.
|
|
testDescription.mBitsPerChannel = 32;
|
|
formatFlags = description.mFormatFlags | kLinearPCMFormatFlagIsFloat & ~kLinearPCMFormatFlagIsSignedInteger;
|
|
testDescription.mFormatFlags = formatFlags;
|
|
result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
|
|
if ( result == noErr ) continue;
|
|
|
|
testDescription = description;
|
|
testDescription.mBitsPerChannel = 32;
|
|
formatFlags = (description.mFormatFlags | kLinearPCMFormatFlagIsSignedInteger) & ~kLinearPCMFormatFlagIsFloat;
|
|
testDescription.mFormatFlags = formatFlags;
|
|
result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
|
|
if ( result == noErr ) continue;
|
|
|
|
testDescription = description;
|
|
testDescription.mBitsPerChannel = 24;
|
|
testDescription.mFormatFlags = formatFlags;
|
|
result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
|
|
if ( result == noErr ) continue;
|
|
|
|
testDescription = description;
|
|
testDescription.mBitsPerChannel = 16;
|
|
testDescription.mFormatFlags = formatFlags;
|
|
result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
|
|
if ( result == noErr ) continue;
|
|
|
|
testDescription = description;
|
|
testDescription.mBitsPerChannel = 8;
|
|
testDescription.mFormatFlags = formatFlags;
|
|
result = AudioStreamSetProperty( streamIDs[firstStream+i], NULL, 0, kAudioStreamPropertyPhysicalFormat, dataSize, &testDescription );
|
|
if ( result != noErr ) {
|
|
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") setting physical data format for device (" << device << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
}
|
|
}
|
|
|
|
// Get the stream latency. There can be latency in both the device
|
|
// and the stream. First, attempt to get the device latency on the
|
|
// master channel or the first open channel. Errors that might
|
|
// occur here are not deemed critical.
|
|
|
|
// ***** CHECK THIS ***** //
|
|
UInt32 latency, channel = 0;
|
|
dataSize = sizeof( UInt32 );
|
|
AudioDevicePropertyID property = kAudioDevicePropertyLatency;
|
|
if ( hasProperty( id, channel, isInput, property ) == true ) {
|
|
result = AudioDeviceGetProperty( id, channel, isInput, property, &dataSize, &latency );
|
|
if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] = latency;
|
|
else {
|
|
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting device latency for device (" << device << ").";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
}
|
|
}
|
|
|
|
// Now try to get the stream latency. For multiple streams, I assume the
|
|
// latency is equal for each.
|
|
result = AudioStreamGetProperty( streamIDs[firstStream], 0, property, &dataSize, &latency );
|
|
if ( result == kAudioHardwareNoError ) stream_.latency[ mode ] += latency;
|
|
else {
|
|
errorStream_ << "RtApiCore::probeDeviceOpen: system error (" << getErrorCode( result ) << ") getting stream latency for device (" << device << ").";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
}
|
|
|
|
// 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] )
|
|
stream_.doConvertBuffer[mode] = true;
|
|
if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
|
|
stream_.doConvertBuffer[mode] = true;
|
|
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 our CoreHandle structure for the stream.
|
|
CoreHandle *handle = 0;
|
|
if ( stream_.apiHandle == 0 ) {
|
|
try {
|
|
handle = new CoreHandle;
|
|
}
|
|
catch ( std::bad_alloc& ) {
|
|
errorText_ = "RtApiCore::probeDeviceOpen: error allocating CoreHandle memory.";
|
|
goto error;
|
|
}
|
|
|
|
if ( 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;
|
|
|
|
// 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_ = "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 ) {
|
|
|
|
bool makeBuffer = true;
|
|
bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
|
|
if ( mode == INPUT ) {
|
|
if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
|
|
unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
|
|
if ( bufferBytes <= bytesOut ) makeBuffer = false;
|
|
}
|
|
}
|
|
|
|
if ( makeBuffer ) {
|
|
bufferBytes *= *bufferSize;
|
|
if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
|
|
stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
|
|
if ( stream_.deviceBuffer == NULL ) {
|
|
errorText_ = "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;
|
|
|
|
// Setup the buffer conversion information structure.
|
|
if ( stream_.doConvertBuffer[mode] ) {
|
|
if ( streamCount > 1 ) setConvertInfo( mode, 0 );
|
|
else setConvertInfo( mode, channelOffset );
|
|
}
|
|
|
|
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.
|
|
result = AudioDeviceAddPropertyListener( id, 0, isInput,
|
|
kAudioDeviceProcessorOverload,
|
|
deviceListener, (void *) handle );
|
|
|
|
return SUCCESS;
|
|
|
|
error:
|
|
if ( handle ) {
|
|
pthread_cond_destroy( &handle->condition );
|
|
delete handle;
|
|
stream_.apiHandle = 0;
|
|
}
|
|
|
|
for ( int i=0; i<2; i++ ) {
|
|
if ( stream_.userBuffer[i] ) {
|
|
free( stream_.userBuffer[i] );
|
|
stream_.userBuffer[i] = 0;
|
|
}
|
|
}
|
|
|
|
if ( stream_.deviceBuffer ) {
|
|
free( stream_.deviceBuffer );
|
|
stream_.deviceBuffer = 0;
|
|
}
|
|
|
|
return FAILURE;
|
|
}
|
|
|
|
void RtApiCore :: closeStream( void )
|
|
{
|
|
if ( stream_.state == STREAM_CLOSED ) {
|
|
errorText_ = "RtApiCore::closeStream(): no open stream to close!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
|
|
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
|
|
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_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
|
|
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
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
// Destroy pthread condition variable.
|
|
pthread_cond_destroy( &handle->condition );
|
|
delete handle;
|
|
stream_.apiHandle = 0;
|
|
|
|
stream_.mode = UNINITIALIZED;
|
|
stream_.state = STREAM_CLOSED;
|
|
}
|
|
|
|
void RtApiCore :: startStream( void )
|
|
{
|
|
verifyStream();
|
|
if ( stream_.state == STREAM_RUNNING ) {
|
|
errorText_ = "RtApiCore::startStream(): the stream is already running!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
|
|
OSStatus result = noErr;
|
|
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
|
|
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
|
|
|
|
result = AudioDeviceStart( handle->id[0], callbackHandler );
|
|
if ( result != noErr ) {
|
|
errorStream_ << "RtApiCore::startStream: system error (" << getErrorCode( result ) << ") starting callback procedure on device (" << stream_.device[0] << ").";
|
|
errorText_ = errorStream_.str();
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
if ( stream_.mode == INPUT ||
|
|
( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
|
|
|
|
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;
|
|
|
|
unlock:
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
|
|
if ( result == noErr ) return;
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
|
|
void RtApiCore :: stopStream( void )
|
|
{
|
|
verifyStream();
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
errorText_ = "RtApiCore::stopStream(): the stream is already stopped!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
return;
|
|
}
|
|
|
|
OSStatus result = noErr;
|
|
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
|
|
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
|
|
|
|
if ( handle->drainCounter == 0 ) {
|
|
handle->drainCounter = 1;
|
|
pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
|
|
}
|
|
|
|
result = AudioDeviceStop( handle->id[0], callbackHandler );
|
|
if ( result != noErr ) {
|
|
errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping callback procedure on device (" << stream_.device[0] << ").";
|
|
errorText_ = errorStream_.str();
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && stream_.device[0] != stream_.device[1] ) ) {
|
|
|
|
result = AudioDeviceStop( handle->id[1], callbackHandler );
|
|
if ( result != noErr ) {
|
|
errorStream_ << "RtApiCore::stopStream: system error (" << getErrorCode( result ) << ") stopping input callback procedure on device (" << stream_.device[1] << ").";
|
|
errorText_ = errorStream_.str();
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
stream_.state = STREAM_STOPPED;
|
|
|
|
unlock:
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
|
|
if ( result == noErr ) return;
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
|
|
void RtApiCore :: abortStream( void )
|
|
{
|
|
verifyStream();
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
errorText_ = "RtApiCore::abortStream(): the stream is already stopped!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
|
|
handle->drainCounter = 1;
|
|
|
|
stopStream();
|
|
}
|
|
|
|
bool RtApiCore :: callbackEvent( AudioDeviceID deviceId,
|
|
const AudioBufferList *inBufferList,
|
|
const AudioBufferList *outBufferList )
|
|
{
|
|
if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
|
|
if ( stream_.state == STREAM_CLOSED ) {
|
|
errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
|
|
error( RtError::WARNING );
|
|
return FAILURE;
|
|
}
|
|
|
|
CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
|
|
CoreHandle *handle = (CoreHandle *) stream_.apiHandle;
|
|
|
|
// Check if we were draining the stream and signal is finished.
|
|
if ( handle->drainCounter > 3 ) {
|
|
if ( handle->internalDrain == false )
|
|
pthread_cond_signal( &handle->condition );
|
|
else
|
|
stopStream();
|
|
return SUCCESS;
|
|
}
|
|
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
|
|
// The state might change while waiting on a mutex.
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
return SUCCESS;
|
|
}
|
|
|
|
AudioDeviceID outputDevice = handle->id[0];
|
|
|
|
// Invoke user callback to get fresh output data UNLESS we are
|
|
// draining stream or duplex mode AND the input/output devices are
|
|
// different AND this function is called for the input device.
|
|
if ( handle->drainCounter == 0 && ( stream_.mode != DUPLEX || deviceId == outputDevice ) ) {
|
|
RtAudioCallback callback = (RtAudioCallback) info->callback;
|
|
double streamTime = getStreamTime();
|
|
RtAudioStreamStatus status = 0;
|
|
if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
|
|
status |= RTAUDIO_OUTPUT_UNDERFLOW;
|
|
handle->xrun[0] = false;
|
|
}
|
|
if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
|
|
status |= RTAUDIO_INPUT_OVERFLOW;
|
|
handle->xrun[1] = false;
|
|
}
|
|
handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
|
|
stream_.bufferSize, streamTime, status, info->userData );
|
|
if ( handle->drainCounter == 2 ) {
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
abortStream();
|
|
return SUCCESS;
|
|
}
|
|
else if ( handle->drainCounter == 1 )
|
|
handle->internalDrain = true;
|
|
}
|
|
|
|
if ( stream_.mode == OUTPUT || ( stream_.mode == DUPLEX && deviceId == outputDevice ) ) {
|
|
|
|
if ( handle->drainCounter > 1 ) { // write zeros to the output stream
|
|
|
|
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 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;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
// Determine input buffer offsets and skips
|
|
if ( inInterleaved ) {
|
|
inJump = inChannels;
|
|
in += inChannels - channelsLeft;
|
|
}
|
|
else {
|
|
inJump = 1;
|
|
in += (inChannels - channelsLeft) * inOffset;
|
|
}
|
|
|
|
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 ( handle->drainCounter ) {
|
|
handle->drainCounter++;
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
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;
|
|
|
|
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 ( 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;
|
|
}
|
|
|
|
// Account for possible unread channels at end of the last stream
|
|
if ( streamChannels > channelsLeft ) {
|
|
inJump = streamChannels - channelsLeft;
|
|
streamChannels = channelsLeft;
|
|
}
|
|
|
|
// 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 );
|
|
|
|
RtApi::tickStreamTime();
|
|
return SUCCESS;
|
|
}
|
|
|
|
const char* RtApiCore :: getErrorCode( OSStatus code )
|
|
{
|
|
switch( code ) {
|
|
|
|
case kAudioHardwareNotRunningError:
|
|
return "kAudioHardwareNotRunningError";
|
|
|
|
case kAudioHardwareUnspecifiedError:
|
|
return "kAudioHardwareUnspecifiedError";
|
|
|
|
case kAudioHardwareUnknownPropertyError:
|
|
return "kAudioHardwareUnknownPropertyError";
|
|
|
|
case kAudioHardwareBadPropertySizeError:
|
|
return "kAudioHardwareBadPropertySizeError";
|
|
|
|
case kAudioHardwareIllegalOperationError:
|
|
return "kAudioHardwareIllegalOperationError";
|
|
|
|
case kAudioHardwareBadObjectError:
|
|
return "kAudioHardwareBadObjectError";
|
|
|
|
case kAudioHardwareBadDeviceError:
|
|
return "kAudioHardwareBadDeviceError";
|
|
|
|
case kAudioHardwareBadStreamError:
|
|
return "kAudioHardwareBadStreamError";
|
|
|
|
case kAudioHardwareUnsupportedOperationError:
|
|
return "kAudioHardwareUnsupportedOperationError";
|
|
|
|
case kAudioDeviceUnsupportedFormatError:
|
|
return "kAudioDeviceUnsupportedFormatError";
|
|
|
|
case kAudioDevicePermissionsError:
|
|
return "kAudioDevicePermissionsError";
|
|
|
|
default:
|
|
return "CoreAudio unknown error";
|
|
}
|
|
}
|
|
|
|
//******************** End of __MACOSX_CORE__ *********************//
|
|
#endif
|
|
|
|
#if defined(__UNIX_JACK__)
|
|
|
|
// JACK is a low-latency audio server, originally written for the
|
|
// GNU/Linux operating system and now also ported to OS-X. It can
|
|
// connect a number of different applications to an audio device, as
|
|
// well as allowing them to share audio between themselves.
|
|
//
|
|
// When using JACK with RtAudio, "devices" refer to JACK clients that
|
|
// have ports connected to the server. The JACK server is typically
|
|
// started in a terminal as follows:
|
|
//
|
|
// .jackd -d alsa -d hw:0
|
|
//
|
|
// or through an interface program such as qjackctl. Many of the
|
|
// parameters normally set for a stream are fixed by the JACK server
|
|
// and can be specified when the JACK server is started. In
|
|
// particular,
|
|
//
|
|
// .jackd -d alsa -d hw:0 -r 44100 -p 512 -n 4
|
|
//
|
|
// specifies a sample rate of 44100 Hz, a buffer size of 512 sample
|
|
// frames, and number of buffers = 4. Once the server is running, it
|
|
// is not possible to override these values. If the values are not
|
|
// specified in the command-line, the JACK server uses default values.
|
|
//
|
|
// The JACK server does not have to be running when an instance of
|
|
// RtApiJack is created, though the function getDeviceCount() will
|
|
// report 0 devices found until JACK has been started. When no
|
|
// devices are available (i.e., the JACK server is not running), a
|
|
// stream cannot be opened.
|
|
|
|
#include <jack/jack.h>
|
|
#include <unistd.h>
|
|
|
|
// A structure to hold various information related to the Jack API
|
|
// implementation.
|
|
struct JackHandle {
|
|
jack_client_t *client;
|
|
jack_port_t **ports[2];
|
|
std::string deviceName[2];
|
|
bool xrun[2];
|
|
pthread_cond_t condition;
|
|
int drainCounter; // Tracks callback counts when draining
|
|
bool internalDrain; // Indicates if stop is initiated from callback or not.
|
|
|
|
JackHandle()
|
|
:client(0), drainCounter(0), internalDrain(false) { ports[0] = 0; ports[1] = 0; xrun[0] = false; xrun[1] = false; }
|
|
};
|
|
|
|
void jackSilentError( const char * ) {};
|
|
|
|
RtApiJack :: RtApiJack()
|
|
{
|
|
// Nothing to do here.
|
|
#if !defined(__RTAUDIO_DEBUG__)
|
|
// Turn off Jack's internal error reporting.
|
|
jack_set_error_function( &jackSilentError );
|
|
#endif
|
|
}
|
|
|
|
RtApiJack :: ~RtApiJack()
|
|
{
|
|
if ( stream_.state != STREAM_CLOSED ) closeStream();
|
|
}
|
|
|
|
unsigned int RtApiJack :: getDeviceCount( void )
|
|
{
|
|
// See if we can become a jack client.
|
|
jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //JackNullOption;
|
|
jack_status_t *status = NULL;
|
|
jack_client_t *client = jack_client_open( "RtApiJackCount", options, status );
|
|
if ( client == 0 ) return 0;
|
|
|
|
const char **ports;
|
|
std::string port, previousPort;
|
|
unsigned int nChannels = 0, nDevices = 0;
|
|
ports = jack_get_ports( client, NULL, NULL, 0 );
|
|
if ( ports ) {
|
|
// Parse the port names up to the first colon (:).
|
|
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 );
|
|
}
|
|
|
|
jack_client_close( client );
|
|
return nDevices;
|
|
}
|
|
|
|
RtAudio::DeviceInfo RtApiJack :: getDeviceInfo( unsigned int device )
|
|
{
|
|
RtAudio::DeviceInfo info;
|
|
info.probed = false;
|
|
|
|
jack_options_t options = (jack_options_t) ( JackNoStartServer | JackUseExactName ); //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( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
const char **ports;
|
|
std::string port, previousPort;
|
|
unsigned int nPorts = 0, nDevices = 0;
|
|
ports = jack_get_ports( client, NULL, NULL, 0 );
|
|
if ( ports ) {
|
|
// Parse the port names up to the first colon (:).
|
|
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 );
|
|
}
|
|
|
|
if ( device >= nDevices ) {
|
|
errorText_ = "RtApiJack::getDeviceInfo: device ID is invalid!";
|
|
error( RtError::INVALID_USE );
|
|
}
|
|
|
|
// Get the current jack server sample rate.
|
|
info.sampleRates.clear();
|
|
info.sampleRates.push_back( jack_get_sample_rate( client ) );
|
|
|
|
// Count the available ports containing the client name as device
|
|
// channels. Jack "input ports" equal RtAudio output channels.
|
|
unsigned int nChannels = 0;
|
|
ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsInput );
|
|
if ( ports ) {
|
|
while ( ports[ nChannels ] ) nChannels++;
|
|
free( ports );
|
|
info.outputChannels = nChannels;
|
|
}
|
|
|
|
// Jack "output ports" equal RtAudio input channels.
|
|
nChannels = 0;
|
|
ports = jack_get_ports( client, info.name.c_str(), NULL, JackPortIsOutput );
|
|
if ( ports ) {
|
|
while ( ports[ nChannels ] ) nChannels++;
|
|
free( ports );
|
|
info.inputChannels = nChannels;
|
|
}
|
|
|
|
if ( info.outputChannels == 0 && info.inputChannels == 0 ) {
|
|
jack_client_close(client);
|
|
errorText_ = "RtApiJack::getDeviceInfo: error determining Jack input/output channels!";
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
// If device opens for both playback and capture, we determine the channels.
|
|
if ( info.outputChannels > 0 && info.inputChannels > 0 )
|
|
info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
|
|
|
|
// Jack always uses 32-bit floats.
|
|
info.nativeFormats = RTAUDIO_FLOAT32;
|
|
|
|
// 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;
|
|
|
|
jack_client_close(client);
|
|
info.probed = true;
|
|
return info;
|
|
}
|
|
|
|
int jackCallbackHandler( jack_nframes_t nframes, void *infoPointer )
|
|
{
|
|
CallbackInfo *info = (CallbackInfo *) infoPointer;
|
|
|
|
RtApiJack *object = (RtApiJack *) info->object;
|
|
if ( object->callbackEvent( (unsigned long) nframes ) == false ) return 1;
|
|
|
|
return 0;
|
|
}
|
|
|
|
void jackShutdown( void *infoPointer )
|
|
{
|
|
CallbackInfo *info = (CallbackInfo *) infoPointer;
|
|
RtApiJack *object = (RtApiJack *) info->object;
|
|
|
|
// Check current stream state. If stopped, then we'll assume this
|
|
// was called as a result of a call to RtApiJack::stopStream (the
|
|
// deactivation of a client handle causes this function to be called).
|
|
// If not, we'll assume the Jack server is shutting down or some
|
|
// other problem occurred and we should close the stream.
|
|
if ( object->isStreamRunning() == false ) return;
|
|
|
|
object->closeStream();
|
|
std::cerr << "\nRtApiJack: the Jack server is shutting down this client ... stream stopped and closed!!\n" << std::endl;
|
|
}
|
|
|
|
int jackXrun( void *infoPointer )
|
|
{
|
|
JackHandle *handle = (JackHandle *) infoPointer;
|
|
|
|
if ( handle->ports[0] ) handle->xrun[0] = true;
|
|
if ( handle->ports[1] ) handle->xrun[1] = true;
|
|
|
|
return 0;
|
|
}
|
|
|
|
bool RtApiJack :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
|
|
unsigned int firstChannel, unsigned int sampleRate,
|
|
RtAudioFormat format, unsigned int *bufferSize,
|
|
RtAudio::StreamOptions *options )
|
|
{
|
|
JackHandle *handle = (JackHandle *) stream_.apiHandle;
|
|
|
|
// 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 | JackUseExactName ); //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( RtError::WARNING );
|
|
return FAILURE;
|
|
}
|
|
}
|
|
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, NULL, 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;
|
|
}
|
|
}
|
|
} while ( ports[++nPorts] );
|
|
free( ports );
|
|
}
|
|
|
|
if ( device >= nDevices ) {
|
|
errorText_ = "RtApiJack::probeDeviceOpen: device ID is invalid!";
|
|
return FAILURE;
|
|
}
|
|
|
|
// Count the available ports containing the client name as device
|
|
// channels. Jack "input ports" equal RtAudio output channels.
|
|
unsigned int nChannels = 0;
|
|
unsigned long flag = JackPortIsInput;
|
|
if ( mode == INPUT ) flag = JackPortIsOutput;
|
|
ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
|
|
if ( ports ) {
|
|
while ( ports[ nChannels ] ) nChannels++;
|
|
free( ports );
|
|
}
|
|
|
|
// Compare the jack ports for specified client to the requested number of channels.
|
|
if ( nChannels < (channels + firstChannel) ) {
|
|
errorStream_ << "RtApiJack::probeDeviceOpen: requested number of channels (" << channels << ") + offset (" << firstChannel << ") not found for specified device (" << device << ":" << deviceName << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// 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;
|
|
|
|
// Get the latency of the JACK port.
|
|
ports = jack_get_ports( client, deviceName.c_str(), NULL, flag );
|
|
if ( ports[ firstChannel ] )
|
|
stream_.latency[mode] = jack_port_get_latency( jack_port_by_name( client, ports[ firstChannel ] ) );
|
|
free( ports );
|
|
|
|
// The jack server always uses 32-bit floating-point data.
|
|
stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
|
|
stream_.userFormat = format;
|
|
|
|
if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
|
|
else stream_.userInterleaved = true;
|
|
|
|
// Jack always uses non-interleaved buffers.
|
|
stream_.deviceInterleaved[mode] = false;
|
|
|
|
// Jack always provides host byte-ordered data.
|
|
stream_.doByteSwap[mode] = false;
|
|
|
|
// 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;
|
|
|
|
stream_.nDeviceChannels[mode] = channels;
|
|
stream_.nUserChannels[mode] = channels;
|
|
|
|
// Set flags for buffer conversion.
|
|
stream_.doConvertBuffer[mode] = false;
|
|
if ( stream_.userFormat != stream_.deviceFormat[mode] )
|
|
stream_.doConvertBuffer[mode] = true;
|
|
if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
|
|
stream_.nUserChannels[mode] > 1 )
|
|
stream_.doConvertBuffer[mode] = true;
|
|
|
|
// Allocate our JackHandle structure for the stream.
|
|
if ( handle == 0 ) {
|
|
try {
|
|
handle = new JackHandle;
|
|
}
|
|
catch ( std::bad_alloc& ) {
|
|
errorText_ = "RtApiJack::probeDeviceOpen: error allocating JackHandle memory.";
|
|
goto error;
|
|
}
|
|
|
|
if ( pthread_cond_init(&handle->condition, NULL) ) {
|
|
errorText_ = "RtApiJack::probeDeviceOpen: error initializing pthread condition variable.";
|
|
goto error;
|
|
}
|
|
stream_.apiHandle = (void *) handle;
|
|
handle->client = client;
|
|
}
|
|
handle->deviceName[mode] = deviceName;
|
|
|
|
// Allocate necessary internal buffers.
|
|
unsigned long bufferBytes;
|
|
bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
|
|
stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
|
|
if ( stream_.userBuffer[mode] == NULL ) {
|
|
errorText_ = "RtApiJack::probeDeviceOpen: error allocating user buffer memory.";
|
|
goto error;
|
|
}
|
|
|
|
if ( stream_.doConvertBuffer[mode] ) {
|
|
|
|
bool makeBuffer = true;
|
|
if ( mode == OUTPUT )
|
|
bufferBytes = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
|
|
else { // mode == INPUT
|
|
bufferBytes = stream_.nDeviceChannels[1] * formatBytes( stream_.deviceFormat[1] );
|
|
if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
|
|
unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes(stream_.deviceFormat[0]);
|
|
if ( bufferBytes < bytesOut ) makeBuffer = false;
|
|
}
|
|
}
|
|
|
|
if ( makeBuffer ) {
|
|
bufferBytes *= *bufferSize;
|
|
if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
|
|
stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
|
|
if ( stream_.deviceBuffer == NULL ) {
|
|
errorText_ = "RtApiJack::probeDeviceOpen: error allocating device buffer memory.";
|
|
goto error;
|
|
}
|
|
}
|
|
}
|
|
|
|
// 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 )
|
|
// We had already set up the stream for output.
|
|
stream_.mode = DUPLEX;
|
|
else {
|
|
stream_.mode = mode;
|
|
jack_set_process_callback( handle->client, jackCallbackHandler, (void *) &stream_.callbackInfo );
|
|
jack_set_xrun_callback( handle->client, jackXrun, (void *) &handle );
|
|
jack_on_shutdown( handle->client, jackShutdown, (void *) &stream_.callbackInfo );
|
|
}
|
|
|
|
// 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 );
|
|
}
|
|
}
|
|
|
|
// Setup the buffer conversion information structure. We don't use
|
|
// buffers to do channel offsets, so we override that parameter
|
|
// here.
|
|
if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
|
|
|
|
return SUCCESS;
|
|
|
|
error:
|
|
if ( 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;
|
|
}
|
|
|
|
for ( int i=0; i<2; i++ ) {
|
|
if ( stream_.userBuffer[i] ) {
|
|
free( stream_.userBuffer[i] );
|
|
stream_.userBuffer[i] = 0;
|
|
}
|
|
}
|
|
|
|
if ( stream_.deviceBuffer ) {
|
|
free( stream_.deviceBuffer );
|
|
stream_.deviceBuffer = 0;
|
|
}
|
|
|
|
return FAILURE;
|
|
}
|
|
|
|
void RtApiJack :: closeStream( void )
|
|
{
|
|
if ( stream_.state == STREAM_CLOSED ) {
|
|
errorText_ = "RtApiJack::closeStream(): no open stream to close!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
JackHandle *handle = (JackHandle *) stream_.apiHandle;
|
|
if ( handle ) {
|
|
|
|
if ( stream_.state == STREAM_RUNNING )
|
|
jack_deactivate( handle->client );
|
|
|
|
jack_client_close( handle->client );
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
for ( int i=0; i<2; i++ ) {
|
|
if ( stream_.userBuffer[i] ) {
|
|
free( stream_.userBuffer[i] );
|
|
stream_.userBuffer[i] = 0;
|
|
}
|
|
}
|
|
|
|
if ( stream_.deviceBuffer ) {
|
|
free( stream_.deviceBuffer );
|
|
stream_.deviceBuffer = 0;
|
|
}
|
|
|
|
stream_.mode = UNINITIALIZED;
|
|
stream_.state = STREAM_CLOSED;
|
|
}
|
|
|
|
void RtApiJack :: startStream( void )
|
|
{
|
|
verifyStream();
|
|
if ( stream_.state == STREAM_RUNNING ) {
|
|
errorText_ = "RtApiJack::startStream(): the stream is already running!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
MUTEX_LOCK(&stream_.mutex);
|
|
|
|
JackHandle *handle = (JackHandle *) stream_.apiHandle;
|
|
int result = jack_activate( handle->client );
|
|
if ( result ) {
|
|
errorText_ = "RtApiJack::startStream(): unable to activate JACK client!";
|
|
goto unlock;
|
|
}
|
|
|
|
const char **ports;
|
|
|
|
// Get the list of available ports.
|
|
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
|
|
result = 1;
|
|
ports = jack_get_ports( handle->client, handle->deviceName[0].c_str(), NULL, JackPortIsInput);
|
|
if ( ports == NULL) {
|
|
errorText_ = "RtApiJack::startStream(): error determining available JACK input ports!";
|
|
goto unlock;
|
|
}
|
|
|
|
// 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 ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
|
|
result = 1;
|
|
ports = jack_get_ports( handle->client, handle->deviceName[1].c_str(), NULL, JackPortIsOutput );
|
|
if ( ports == NULL) {
|
|
errorText_ = "RtApiJack::startStream(): error determining available JACK output ports!";
|
|
goto unlock;
|
|
}
|
|
|
|
// 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;
|
|
|
|
unlock:
|
|
MUTEX_UNLOCK(&stream_.mutex);
|
|
|
|
if ( result == 0 ) return;
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
|
|
void RtApiJack :: stopStream( void )
|
|
{
|
|
verifyStream();
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
errorText_ = "RtApiJack::stopStream(): the stream is already stopped!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
return;
|
|
}
|
|
|
|
JackHandle *handle = (JackHandle *) stream_.apiHandle;
|
|
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
|
|
|
|
if ( handle->drainCounter == 0 ) {
|
|
handle->drainCounter = 1;
|
|
pthread_cond_wait( &handle->condition, &stream_.mutex ); // block until signaled
|
|
}
|
|
}
|
|
|
|
jack_deactivate( handle->client );
|
|
stream_.state = STREAM_STOPPED;
|
|
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
}
|
|
|
|
void RtApiJack :: abortStream( void )
|
|
{
|
|
verifyStream();
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
errorText_ = "RtApiJack::abortStream(): the stream is already stopped!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
JackHandle *handle = (JackHandle *) stream_.apiHandle;
|
|
handle->drainCounter = 1;
|
|
|
|
stopStream();
|
|
}
|
|
|
|
// 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.
|
|
extern "C" void *jackStopStream( void *ptr )
|
|
{
|
|
CallbackInfo *info = (CallbackInfo *) ptr;
|
|
RtApiJack *object = (RtApiJack *) info->object;
|
|
|
|
object->stopStream();
|
|
|
|
pthread_exit( NULL );
|
|
}
|
|
|
|
bool RtApiJack :: callbackEvent( unsigned long nframes )
|
|
{
|
|
if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
|
|
if ( stream_.state == STREAM_CLOSED ) {
|
|
errorText_ = "RtApiCore::callbackEvent(): the stream is closed ... this shouldn't happen!";
|
|
error( RtError::WARNING );
|
|
return FAILURE;
|
|
}
|
|
if ( stream_.bufferSize != nframes ) {
|
|
errorText_ = "RtApiCore::callbackEvent(): the JACK buffer size has changed ... cannot process!";
|
|
error( RtError::WARNING );
|
|
return FAILURE;
|
|
}
|
|
|
|
CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
|
|
JackHandle *handle = (JackHandle *) stream_.apiHandle;
|
|
|
|
// Check if we were draining the stream and signal is finished.
|
|
if ( handle->drainCounter > 3 ) {
|
|
if ( handle->internalDrain == true ) {
|
|
ThreadHandle id;
|
|
pthread_create( &id, NULL, jackStopStream, info );
|
|
}
|
|
else
|
|
pthread_cond_signal( &handle->condition );
|
|
return SUCCESS;
|
|
}
|
|
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
|
|
// The state might change while waiting on a mutex.
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
return SUCCESS;
|
|
}
|
|
|
|
// Invoke user callback first, to get fresh output data.
|
|
if ( handle->drainCounter == 0 ) {
|
|
RtAudioCallback callback = (RtAudioCallback) info->callback;
|
|
double streamTime = getStreamTime();
|
|
RtAudioStreamStatus status = 0;
|
|
if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
|
|
status |= RTAUDIO_OUTPUT_UNDERFLOW;
|
|
handle->xrun[0] = false;
|
|
}
|
|
if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
|
|
status |= RTAUDIO_INPUT_OVERFLOW;
|
|
handle->xrun[1] = false;
|
|
}
|
|
handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
|
|
stream_.bufferSize, streamTime, status, info->userData );
|
|
if ( handle->drainCounter == 2 ) {
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
ThreadHandle id;
|
|
pthread_create( &id, NULL, jackStopStream, info );
|
|
return SUCCESS;
|
|
}
|
|
else if ( handle->drainCounter == 1 )
|
|
handle->internalDrain = true;
|
|
}
|
|
|
|
jack_default_audio_sample_t *jackbuffer;
|
|
unsigned long bufferBytes = nframes * sizeof( jack_default_audio_sample_t );
|
|
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
|
|
|
|
if ( handle->drainCounter > 0 ) { // write zeros to the output stream
|
|
|
|
for ( unsigned int i=0; i<stream_.nDeviceChannels[0]; i++ ) {
|
|
jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[0][i], (jack_nframes_t) nframes );
|
|
memset( jackbuffer, 0, bufferBytes );
|
|
}
|
|
|
|
}
|
|
else if ( stream_.doConvertBuffer[0] ) {
|
|
|
|
convertBuffer( stream_.deviceBuffer, stream_.userBuffer[0], stream_.convertInfo[0] );
|
|
|
|
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 { // 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 ( handle->drainCounter ) {
|
|
handle->drainCounter++;
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
|
|
|
|
if ( stream_.doConvertBuffer[1] ) {
|
|
for ( unsigned int i=0; i<stream_.nDeviceChannels[1]; i++ ) {
|
|
jackbuffer = (jack_default_audio_sample_t *) jack_port_get_buffer( handle->ports[1][i], (jack_nframes_t) nframes );
|
|
memcpy( &stream_.deviceBuffer[i*bufferBytes], jackbuffer, bufferBytes );
|
|
}
|
|
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 );
|
|
}
|
|
}
|
|
}
|
|
|
|
unlock:
|
|
MUTEX_UNLOCK(&stream_.mutex);
|
|
|
|
RtApi::tickStreamTime();
|
|
return SUCCESS;
|
|
}
|
|
//******************** End of __UNIX_JACK__ *********************//
|
|
#endif
|
|
|
|
#if defined(__WINDOWS_ASIO__) // ASIO API on Windows
|
|
|
|
// The ASIO API is designed around a callback scheme, so this
|
|
// implementation is similar to that used for OS-X CoreAudio 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.
|
|
|
|
#include "asiosys.h"
|
|
#include "asio.h"
|
|
#include "iasiothiscallresolver.h"
|
|
#include "asiodrivers.h"
|
|
#include <cmath>
|
|
|
|
AsioDrivers drivers;
|
|
ASIOCallbacks asioCallbacks;
|
|
ASIODriverInfo driverInfo;
|
|
CallbackInfo *asioCallbackInfo;
|
|
bool asioXRun;
|
|
|
|
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) {}
|
|
};
|
|
|
|
// Function declarations (definitions at end of section)
|
|
static const char* getAsioErrorString( ASIOError result );
|
|
void sampleRateChanged( ASIOSampleRate sRate );
|
|
long asioMessages( long selector, long value, void* message, double* opt );
|
|
|
|
RtApiAsio :: RtApiAsio()
|
|
{
|
|
// ASIO cannot run on a multi-threaded appartment. You can call
|
|
// CoInitialize beforehand, but it must be for appartment threading
|
|
// (in which case, CoInitilialize will return S_FALSE here).
|
|
coInitialized_ = false;
|
|
HRESULT hr = CoInitialize( NULL );
|
|
if ( FAILED(hr) ) {
|
|
errorText_ = "RtApiAsio::ASIO requires a single-threaded appartment. Call CoInitializeEx(0,COINIT_APARTMENTTHREADED)";
|
|
error( RtError::WARNING );
|
|
}
|
|
coInitialized_ = true;
|
|
|
|
drivers.removeCurrentDriver();
|
|
driverInfo.asioVersion = 2;
|
|
|
|
// See note in DirectSound implementation about GetDesktopWindow().
|
|
driverInfo.sysRef = GetForegroundWindow();
|
|
}
|
|
|
|
RtApiAsio :: ~RtApiAsio()
|
|
{
|
|
if ( stream_.state != STREAM_CLOSED ) closeStream();
|
|
if ( coInitialized_ ) CoUninitialize();
|
|
}
|
|
|
|
unsigned int RtApiAsio :: getDeviceCount( void )
|
|
{
|
|
return (unsigned int) drivers.asioGetNumDev();
|
|
}
|
|
|
|
RtAudio::DeviceInfo RtApiAsio :: getDeviceInfo( unsigned int device )
|
|
{
|
|
RtAudio::DeviceInfo info;
|
|
info.probed = false;
|
|
|
|
// Get device ID
|
|
unsigned int nDevices = getDeviceCount();
|
|
if ( nDevices == 0 ) {
|
|
errorText_ = "RtApiAsio::getDeviceInfo: no devices found!";
|
|
error( RtError::INVALID_USE );
|
|
}
|
|
|
|
if ( device >= nDevices ) {
|
|
errorText_ = "RtApiAsio::getDeviceInfo: device ID is invalid!";
|
|
error( RtError::INVALID_USE );
|
|
}
|
|
|
|
// If a stream is already open, we cannot probe other devices. Thus, use the saved results.
|
|
if ( stream_.state != STREAM_CLOSED ) {
|
|
if ( device >= devices_.size() ) {
|
|
errorText_ = "RtApiAsio::getDeviceInfo: device ID was not present before stream was opened.";
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
return devices_[ device ];
|
|
}
|
|
|
|
char driverName[32];
|
|
ASIOError result = drivers.asioGetDriverName( (int) device, driverName, 32 );
|
|
if ( result != ASE_OK ) {
|
|
errorStream_ << "RtApiAsio::getDeviceInfo: unable to get driver name (" << getAsioErrorString( result ) << ").";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
info.name = driverName;
|
|
|
|
if ( !drivers.loadDriver( driverName ) ) {
|
|
errorStream_ << "RtApiAsio::getDeviceInfo: unable to load driver (" << driverName << ").";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
result = ASIOInit( &driverInfo );
|
|
if ( result != ASE_OK ) {
|
|
errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
// Determine the device channel information.
|
|
long inputChannels, outputChannels;
|
|
result = ASIOGetChannels( &inputChannels, &outputChannels );
|
|
if ( result != ASE_OK ) {
|
|
drivers.removeCurrentDriver();
|
|
errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
info.outputChannels = outputChannels;
|
|
info.inputChannels = inputChannels;
|
|
if ( info.outputChannels > 0 && info.inputChannels > 0 )
|
|
info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
|
|
|
|
// 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] );
|
|
}
|
|
|
|
// Determine supported data types ... just check first channel and assume rest are the same.
|
|
ASIOChannelInfo channelInfo;
|
|
channelInfo.channel = 0;
|
|
channelInfo.isInput = true;
|
|
if ( info.inputChannels <= 0 ) channelInfo.isInput = false;
|
|
result = ASIOGetChannelInfo( &channelInfo );
|
|
if ( result != ASE_OK ) {
|
|
drivers.removeCurrentDriver();
|
|
errorStream_ << "RtApiAsio::getDeviceInfo: error (" << getAsioErrorString( result ) << ") getting driver channel info (" << driverName << ").";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
info.nativeFormats = 0;
|
|
if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB )
|
|
info.nativeFormats |= RTAUDIO_SINT16;
|
|
else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB )
|
|
info.nativeFormats |= RTAUDIO_SINT32;
|
|
else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB )
|
|
info.nativeFormats |= RTAUDIO_FLOAT32;
|
|
else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB )
|
|
info.nativeFormats |= RTAUDIO_FLOAT64;
|
|
|
|
if ( getDefaultOutputDevice() == device )
|
|
info.isDefaultOutput = true;
|
|
if ( getDefaultInputDevice() == device )
|
|
info.isDefaultInput = true;
|
|
|
|
info.probed = true;
|
|
drivers.removeCurrentDriver();
|
|
return info;
|
|
}
|
|
|
|
void bufferSwitch( long index, ASIOBool processNow )
|
|
{
|
|
RtApiAsio *object = (RtApiAsio *) asioCallbackInfo->object;
|
|
object->callbackEvent( index );
|
|
}
|
|
|
|
void RtApiAsio :: saveDeviceInfo( void )
|
|
{
|
|
devices_.clear();
|
|
|
|
unsigned int nDevices = getDeviceCount();
|
|
devices_.resize( nDevices );
|
|
for ( unsigned int i=0; i<nDevices; i++ )
|
|
devices_[i] = getDeviceInfo( i );
|
|
}
|
|
|
|
bool RtApiAsio :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
|
|
unsigned int firstChannel, unsigned int sampleRate,
|
|
RtAudioFormat format, unsigned int *bufferSize,
|
|
RtAudio::StreamOptions *options )
|
|
{
|
|
// For ASIO, a duplex stream MUST use the same driver.
|
|
if ( mode == INPUT && stream_.mode == OUTPUT && stream_.device[0] != device ) {
|
|
errorText_ = "RtApiAsio::probeDeviceOpen: an ASIO duplex stream must use the same device for input and output!";
|
|
return FAILURE;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
// The getDeviceInfo() function will not work when a stream is open
|
|
// because ASIO does not allow multiple devices to run at the same
|
|
// time. Thus, we'll probe the system before opening a stream and
|
|
// save the results for use by getDeviceInfo().
|
|
this->saveDeviceInfo();
|
|
|
|
// Only load the driver once for duplex stream.
|
|
if ( mode != INPUT || stream_.mode != OUTPUT ) {
|
|
if ( !drivers.loadDriver( driverName ) ) {
|
|
errorStream_ << "RtApiAsio::probeDeviceOpen: unable to load driver (" << driverName << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
result = ASIOInit( &driverInfo );
|
|
if ( result != ASE_OK ) {
|
|
errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") initializing driver (" << driverName << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
}
|
|
|
|
// Check the device channel count.
|
|
long inputChannels, outputChannels;
|
|
result = ASIOGetChannels( &inputChannels, &outputChannels );
|
|
if ( result != ASE_OK ) {
|
|
drivers.removeCurrentDriver();
|
|
errorStream_ << "RtApiAsio::probeDeviceOpen: error (" << getAsioErrorString( result ) << ") getting channel count (" << driverName << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
if ( ( mode == OUTPUT && (channels+firstChannel) > (unsigned int) outputChannels) ||
|
|
( mode == INPUT && (channels+firstChannel) > (unsigned int) inputChannels) ) {
|
|
drivers.removeCurrentDriver();
|
|
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested channel count (" << channels << ") + offset (" << firstChannel << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
stream_.nDeviceChannels[mode] = channels;
|
|
stream_.nUserChannels[mode] = channels;
|
|
stream_.channelOffset[mode] = firstChannel;
|
|
|
|
// Verify the sample rate is supported.
|
|
result = ASIOCanSampleRate( (ASIOSampleRate) sampleRate );
|
|
if ( result != ASE_OK ) {
|
|
drivers.removeCurrentDriver();
|
|
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") does not support requested sample rate (" << sampleRate << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Get the current sample rate
|
|
ASIOSampleRate currentRate;
|
|
result = ASIOGetSampleRate( ¤tRate );
|
|
if ( result != ASE_OK ) {
|
|
drivers.removeCurrentDriver();
|
|
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error getting sample rate.";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Set the sample rate only if necessary
|
|
if ( currentRate != sampleRate ) {
|
|
result = ASIOSetSampleRate( (ASIOSampleRate) sampleRate );
|
|
if ( result != ASE_OK ) {
|
|
drivers.removeCurrentDriver();
|
|
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error setting sample rate (" << sampleRate << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
}
|
|
|
|
// Determine the driver data type.
|
|
ASIOChannelInfo channelInfo;
|
|
channelInfo.channel = 0;
|
|
if ( mode == OUTPUT ) channelInfo.isInput = false;
|
|
else channelInfo.isInput = true;
|
|
result = ASIOGetChannelInfo( &channelInfo );
|
|
if ( result != ASE_OK ) {
|
|
drivers.removeCurrentDriver();
|
|
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting data format.";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Assuming WINDOWS host is always little-endian.
|
|
stream_.doByteSwap[mode] = false;
|
|
stream_.userFormat = format;
|
|
stream_.deviceFormat[mode] = 0;
|
|
if ( channelInfo.type == ASIOSTInt16MSB || channelInfo.type == ASIOSTInt16LSB ) {
|
|
stream_.deviceFormat[mode] = RTAUDIO_SINT16;
|
|
if ( channelInfo.type == ASIOSTInt16MSB ) stream_.doByteSwap[mode] = true;
|
|
}
|
|
else if ( channelInfo.type == ASIOSTInt32MSB || channelInfo.type == ASIOSTInt32LSB ) {
|
|
stream_.deviceFormat[mode] = RTAUDIO_SINT32;
|
|
if ( channelInfo.type == ASIOSTInt32MSB ) stream_.doByteSwap[mode] = true;
|
|
}
|
|
else if ( channelInfo.type == ASIOSTFloat32MSB || channelInfo.type == ASIOSTFloat32LSB ) {
|
|
stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
|
|
if ( channelInfo.type == ASIOSTFloat32MSB ) stream_.doByteSwap[mode] = true;
|
|
}
|
|
else if ( channelInfo.type == ASIOSTFloat64MSB || channelInfo.type == ASIOSTFloat64LSB ) {
|
|
stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
|
|
if ( channelInfo.type == ASIOSTFloat64MSB ) stream_.doByteSwap[mode] = true;
|
|
}
|
|
|
|
if ( stream_.deviceFormat[mode] == 0 ) {
|
|
drivers.removeCurrentDriver();
|
|
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") data format not supported by RtAudio.";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Set the buffer size. For a duplex stream, this will end up
|
|
// setting the buffer size based on the input constraints, which
|
|
// should be ok.
|
|
long minSize, maxSize, preferSize, granularity;
|
|
result = ASIOGetBufferSize( &minSize, &maxSize, &preferSize, &granularity );
|
|
if ( result != ASE_OK ) {
|
|
drivers.removeCurrentDriver();
|
|
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting buffer size.";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
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;
|
|
|
|
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;
|
|
}
|
|
|
|
long min_delta = std::abs( (long)*bufferSize - ((long)1 << log2_of_min_size) );
|
|
int min_delta_num = log2_of_min_size;
|
|
|
|
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 ( mode == INPUT && stream_.mode == OUTPUT && stream_.bufferSize != *bufferSize ) {
|
|
drivers.removeCurrentDriver();
|
|
errorText_ = "RtApiAsio::probeDeviceOpen: input/output buffersize discrepancy!";
|
|
return FAILURE;
|
|
}
|
|
|
|
stream_.bufferSize = *bufferSize;
|
|
stream_.nBuffers = 2;
|
|
|
|
if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) stream_.userInterleaved = false;
|
|
else stream_.userInterleaved = true;
|
|
|
|
// ASIO always uses non-interleaved buffers.
|
|
stream_.deviceInterleaved[mode] = false;
|
|
|
|
// Allocate, if necessary, our AsioHandle structure for the stream.
|
|
AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
|
|
if ( handle == 0 ) {
|
|
try {
|
|
handle = new AsioHandle;
|
|
}
|
|
catch ( std::bad_alloc& ) {
|
|
//if ( handle == NULL ) {
|
|
drivers.removeCurrentDriver();
|
|
errorText_ = "RtApiAsio::probeDeviceOpen: error allocating AsioHandle memory.";
|
|
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;
|
|
}
|
|
|
|
// Create the ASIO internal buffers. Since RtAudio sets up input
|
|
// and output separately, we'll have to dispose of previously
|
|
// created output buffers for a duplex stream.
|
|
long inputLatency, outputLatency;
|
|
if ( mode == INPUT && stream_.mode == OUTPUT ) {
|
|
ASIODisposeBuffers();
|
|
if ( handle->bufferInfos ) free( handle->bufferInfos );
|
|
}
|
|
|
|
// Allocate, initialize, and save the bufferInfos in our stream callbackInfo structure.
|
|
bool buffersAllocated = false;
|
|
unsigned int i, nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
|
|
handle->bufferInfos = (ASIOBufferInfo *) malloc( nChannels * sizeof(ASIOBufferInfo) );
|
|
if ( handle->bufferInfos == NULL ) {
|
|
errorStream_ << "RtApiAsio::probeDeviceOpen: error allocating bufferInfo memory for driver (" << driverName << ").";
|
|
errorText_ = errorStream_.str();
|
|
goto error;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
// Set up the ASIO callback structure and create the ASIO data buffers.
|
|
asioCallbacks.bufferSwitch = &bufferSwitch;
|
|
asioCallbacks.sampleRateDidChange = &sampleRateChanged;
|
|
asioCallbacks.asioMessage = &asioMessages;
|
|
asioCallbacks.bufferSwitchTimeInfo = NULL;
|
|
result = ASIOCreateBuffers( handle->bufferInfos, nChannels, stream_.bufferSize, &asioCallbacks );
|
|
if ( result != ASE_OK ) {
|
|
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") creating buffers.";
|
|
errorText_ = errorStream_.str();
|
|
goto error;
|
|
}
|
|
buffersAllocated = true;
|
|
|
|
// Set flags for buffer conversion.
|
|
stream_.doConvertBuffer[mode] = false;
|
|
if ( stream_.userFormat != stream_.deviceFormat[mode] )
|
|
stream_.doConvertBuffer[mode] = true;
|
|
if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
|
|
stream_.nUserChannels[mode] > 1 )
|
|
stream_.doConvertBuffer[mode] = true;
|
|
|
|
// Allocate necessary internal buffers
|
|
unsigned long bufferBytes;
|
|
bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
|
|
stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
|
|
if ( stream_.userBuffer[mode] == NULL ) {
|
|
errorText_ = "RtApiAsio::probeDeviceOpen: error allocating user buffer memory.";
|
|
goto error;
|
|
}
|
|
|
|
if ( stream_.doConvertBuffer[mode] ) {
|
|
|
|
bool makeBuffer = true;
|
|
bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
|
|
if ( mode == INPUT ) {
|
|
if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
|
|
unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
|
|
if ( bufferBytes <= bytesOut ) makeBuffer = false;
|
|
}
|
|
}
|
|
|
|
if ( makeBuffer ) {
|
|
bufferBytes *= *bufferSize;
|
|
if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
|
|
stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
|
|
if ( stream_.deviceBuffer == NULL ) {
|
|
errorText_ = "RtApiAsio::probeDeviceOpen: error allocating device buffer memory.";
|
|
goto error;
|
|
}
|
|
}
|
|
}
|
|
|
|
stream_.sampleRate = sampleRate;
|
|
stream_.device[mode] = device;
|
|
stream_.state = STREAM_STOPPED;
|
|
asioCallbackInfo = &stream_.callbackInfo;
|
|
stream_.callbackInfo.object = (void *) this;
|
|
if ( stream_.mode == OUTPUT && mode == INPUT )
|
|
// We had already set up an output stream.
|
|
stream_.mode = DUPLEX;
|
|
else
|
|
stream_.mode = mode;
|
|
|
|
// Determine device latencies
|
|
result = ASIOGetLatencies( &inputLatency, &outputLatency );
|
|
if ( result != ASE_OK ) {
|
|
errorStream_ << "RtApiAsio::probeDeviceOpen: driver (" << driverName << ") error (" << getAsioErrorString( result ) << ") getting latency.";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING); // warn but don't fail
|
|
}
|
|
else {
|
|
stream_.latency[0] = outputLatency;
|
|
stream_.latency[1] = inputLatency;
|
|
}
|
|
|
|
// Setup the buffer conversion information structure. We don't use
|
|
// buffers to do channel offsets, so we override that parameter
|
|
// here.
|
|
if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, 0 );
|
|
|
|
return SUCCESS;
|
|
|
|
error:
|
|
if ( buffersAllocated )
|
|
ASIODisposeBuffers();
|
|
drivers.removeCurrentDriver();
|
|
|
|
if ( handle ) {
|
|
CloseHandle( handle->condition );
|
|
if ( handle->bufferInfos )
|
|
free( handle->bufferInfos );
|
|
delete handle;
|
|
stream_.apiHandle = 0;
|
|
}
|
|
|
|
for ( int i=0; i<2; i++ ) {
|
|
if ( stream_.userBuffer[i] ) {
|
|
free( stream_.userBuffer[i] );
|
|
stream_.userBuffer[i] = 0;
|
|
}
|
|
}
|
|
|
|
if ( stream_.deviceBuffer ) {
|
|
free( stream_.deviceBuffer );
|
|
stream_.deviceBuffer = 0;
|
|
}
|
|
|
|
return FAILURE;
|
|
}
|
|
|
|
void RtApiAsio :: closeStream()
|
|
{
|
|
if ( stream_.state == STREAM_CLOSED ) {
|
|
errorText_ = "RtApiAsio::closeStream(): no open stream to close!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
if ( stream_.state == STREAM_RUNNING ) {
|
|
stream_.state = STREAM_STOPPED;
|
|
ASIOStop();
|
|
}
|
|
ASIODisposeBuffers();
|
|
drivers.removeCurrentDriver();
|
|
|
|
AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
|
|
if ( handle ) {
|
|
CloseHandle( handle->condition );
|
|
if ( handle->bufferInfos )
|
|
free( handle->bufferInfos );
|
|
delete handle;
|
|
stream_.apiHandle = 0;
|
|
}
|
|
|
|
for ( int i=0; i<2; i++ ) {
|
|
if ( stream_.userBuffer[i] ) {
|
|
free( stream_.userBuffer[i] );
|
|
stream_.userBuffer[i] = 0;
|
|
}
|
|
}
|
|
|
|
if ( stream_.deviceBuffer ) {
|
|
free( stream_.deviceBuffer );
|
|
stream_.deviceBuffer = 0;
|
|
}
|
|
|
|
stream_.mode = UNINITIALIZED;
|
|
stream_.state = STREAM_CLOSED;
|
|
}
|
|
|
|
void RtApiAsio :: startStream()
|
|
{
|
|
verifyStream();
|
|
if ( stream_.state == STREAM_RUNNING ) {
|
|
errorText_ = "RtApiAsio::startStream(): the stream is already running!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
|
|
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;
|
|
}
|
|
|
|
handle->drainCounter = 0;
|
|
handle->internalDrain = false;
|
|
stream_.state = STREAM_RUNNING;
|
|
asioXRun = false;
|
|
|
|
unlock:
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
|
|
if ( result == ASE_OK ) return;
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
|
|
void RtApiAsio :: stopStream()
|
|
{
|
|
verifyStream();
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
errorText_ = "RtApiAsio::stopStream(): the stream is already stopped!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
return;
|
|
}
|
|
|
|
AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
|
|
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
|
|
if ( handle->drainCounter == 0 ) {
|
|
handle->drainCounter = 1;
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled
|
|
ResetEvent( handle->condition );
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
}
|
|
}
|
|
|
|
ASIOError result = ASIOStop();
|
|
if ( result != ASE_OK ) {
|
|
errorStream_ << "RtApiAsio::stopStream: error (" << getAsioErrorString( result ) << ") stopping device.";
|
|
errorText_ = errorStream_.str();
|
|
}
|
|
|
|
stream_.state = STREAM_STOPPED;
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
|
|
if ( result == ASE_OK ) return;
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
|
|
void RtApiAsio :: abortStream()
|
|
{
|
|
verifyStream();
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
errorText_ = "RtApiAsio::abortStream(): the stream is already stopped!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
// 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 = 1;
|
|
stopStream();
|
|
}
|
|
|
|
bool RtApiAsio :: callbackEvent( long bufferIndex )
|
|
{
|
|
if ( stream_.state == STREAM_STOPPED ) return SUCCESS;
|
|
if ( stream_.state == STREAM_CLOSED ) {
|
|
errorText_ = "RtApiAsio::callbackEvent(): the stream is closed ... this shouldn't happen!";
|
|
error( RtError::WARNING );
|
|
return FAILURE;
|
|
}
|
|
|
|
CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
|
|
AsioHandle *handle = (AsioHandle *) stream_.apiHandle;
|
|
|
|
// Check if we were draining the stream and signal is finished.
|
|
if ( handle->drainCounter > 3 ) {
|
|
if ( handle->internalDrain == false )
|
|
SetEvent( handle->condition );
|
|
else
|
|
stopStream();
|
|
return SUCCESS;
|
|
}
|
|
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
|
|
// The state might change while waiting on a mutex.
|
|
if ( stream_.state == STREAM_STOPPED ) goto unlock;
|
|
|
|
// 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;
|
|
}
|
|
handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
|
|
stream_.bufferSize, streamTime, status, info->userData );
|
|
if ( handle->drainCounter == 2 ) {
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
abortStream();
|
|
return SUCCESS;
|
|
}
|
|
else if ( handle->drainCounter == 1 )
|
|
handle->internalDrain = true;
|
|
}
|
|
|
|
unsigned int nChannels, bufferBytes, i, j;
|
|
nChannels = stream_.nDeviceChannels[0] + stream_.nDeviceChannels[1];
|
|
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
|
|
|
|
bufferBytes = stream_.bufferSize * formatBytes( stream_.deviceFormat[0] );
|
|
|
|
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 );
|
|
}
|
|
|
|
}
|
|
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 );
|
|
}
|
|
|
|
}
|
|
else {
|
|
|
|
if ( stream_.doByteSwap[0] )
|
|
byteSwapBuffer( stream_.userBuffer[0],
|
|
stream_.bufferSize * stream_.nUserChannels[0],
|
|
stream_.userFormat );
|
|
|
|
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 );
|
|
}
|
|
|
|
}
|
|
|
|
if ( handle->drainCounter ) {
|
|
handle->drainCounter++;
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
|
|
|
|
bufferBytes = stream_.bufferSize * formatBytes(stream_.deviceFormat[1]);
|
|
|
|
if (stream_.doConvertBuffer[1]) {
|
|
|
|
// 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 ( stream_.doByteSwap[1] )
|
|
byteSwapBuffer( stream_.deviceBuffer,
|
|
stream_.bufferSize * stream_.nDeviceChannels[1],
|
|
stream_.deviceFormat[1] );
|
|
convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
|
|
|
|
}
|
|
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 );
|
|
}
|
|
}
|
|
|
|
if ( stream_.doByteSwap[1] )
|
|
byteSwapBuffer( stream_.userBuffer[1],
|
|
stream_.bufferSize * stream_.nUserChannels[1],
|
|
stream_.userFormat );
|
|
}
|
|
}
|
|
|
|
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();
|
|
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
|
|
RtApi::tickStreamTime();
|
|
return SUCCESS;
|
|
}
|
|
|
|
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 ( RtError &exception ) {
|
|
std::cerr << "\nRtApiAsio: sampleRateChanged() error (" << exception.getMessage() << ")!\n" << std::endl;
|
|
return;
|
|
}
|
|
|
|
std::cerr << "\nRtApiAsio: driver reports sample rate changed to " << sRate << " ... stream stopped!!!\n" << std::endl;
|
|
}
|
|
|
|
long asioMessages( long selector, long value, void* message, double* opt )
|
|
{
|
|
long 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.
|
|
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;
|
|
}
|
|
|
|
static const char* getAsioErrorString( ASIOError result )
|
|
{
|
|
struct Messages
|
|
{
|
|
ASIOError value;
|
|
const char*message;
|
|
};
|
|
|
|
static Messages m[] =
|
|
{
|
|
{ ASE_NotPresent, "Hardware input or output is not present or available." },
|
|
{ ASE_HWMalfunction, "Hardware is malfunctioning." },
|
|
{ ASE_InvalidParameter, "Invalid input parameter." },
|
|
{ ASE_InvalidMode, "Invalid mode." },
|
|
{ ASE_SPNotAdvancing, "Sample position not advancing." },
|
|
{ ASE_NoClock, "Sample clock or rate cannot be determined or is not present." },
|
|
{ ASE_NoMemory, "Not enough memory to complete the request." }
|
|
};
|
|
|
|
for ( unsigned int i = 0; i < sizeof(m)/sizeof(m[0]); ++i )
|
|
if ( m[i].value == result ) return m[i].message;
|
|
|
|
return "Unknown error.";
|
|
}
|
|
//******************** End of __WINDOWS_ASIO__ *********************//
|
|
#endif
|
|
|
|
|
|
#if defined(__WINDOWS_DS__) // Windows DirectSound API
|
|
|
|
// Modified by Robin Davies, October 2005
|
|
// - Improvements to DirectX pointer chasing.
|
|
// - Backdoor RtDsStatistics hook provides DirectX performance information.
|
|
// - Bug fix for non-power-of-two Asio granularity used by Edirol PCR-A30.
|
|
// - Auto-call CoInitialize for DSOUND and ASIO platforms.
|
|
// Various revisions for RtAudio 4.0 by Gary Scavone, April 2007
|
|
|
|
#include <dsound.h>
|
|
#include <assert.h>
|
|
|
|
#if defined(__MINGW32__)
|
|
// missing from latest mingw winapi
|
|
#define WAVE_FORMAT_96M08 0x00010000 /* 96 kHz, Mono, 8-bit */
|
|
#define WAVE_FORMAT_96S08 0x00020000 /* 96 kHz, Stereo, 8-bit */
|
|
#define WAVE_FORMAT_96M16 0x00040000 /* 96 kHz, Mono, 16-bit */
|
|
#define WAVE_FORMAT_96S16 0x00080000 /* 96 kHz, Stereo, 16-bit */
|
|
#endif
|
|
|
|
#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 dsPointerDifference( DWORD laterPointer, DWORD earlierPointer, DWORD bufferSize )
|
|
{
|
|
if ( laterPointer > earlierPointer )
|
|
return laterPointer - earlierPointer;
|
|
else
|
|
return laterPointer - earlierPointer + bufferSize;
|
|
}
|
|
|
|
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; }
|
|
};
|
|
|
|
/*
|
|
RtApiDs::RtDsStatistics RtApiDs::statistics;
|
|
|
|
// Provides a backdoor hook to monitor for DirectSound read overruns and write underruns.
|
|
RtApiDs::RtDsStatistics RtApiDs::getDsStatistics()
|
|
{
|
|
RtDsStatistics s = statistics;
|
|
|
|
// update the calculated fields.
|
|
if ( s.inputFrameSize != 0 )
|
|
s.latency += s.readDeviceSafeLeadBytes * 1.0 / s.inputFrameSize / s.sampleRate;
|
|
|
|
if ( s.outputFrameSize != 0 )
|
|
s.latency += (s.writeDeviceSafeLeadBytes + s.writeDeviceBufferLeadBytes) * 1.0 / s.outputFrameSize / s.sampleRate;
|
|
|
|
return s;
|
|
}
|
|
*/
|
|
|
|
// 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 char* getErrorString( int code );
|
|
|
|
extern "C" unsigned __stdcall callbackHandler( void *ptr );
|
|
|
|
struct EnumInfo {
|
|
bool isInput;
|
|
bool getDefault;
|
|
bool findIndex;
|
|
unsigned int counter;
|
|
unsigned int index;
|
|
LPGUID id;
|
|
std::string name;
|
|
|
|
EnumInfo()
|
|
: isInput(false), getDefault(false), findIndex(false), counter(0), index(0) {}
|
|
};
|
|
|
|
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 ( coInitialized_ ) CoUninitialize(); // balanced call.
|
|
if ( stream_.state != STREAM_CLOSED ) closeStream();
|
|
}
|
|
|
|
unsigned int RtApiDs :: getDefaultInputDevice( void )
|
|
{
|
|
// Count output devices.
|
|
EnumInfo info;
|
|
HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") counting output devices!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return 0;
|
|
}
|
|
|
|
// Now enumerate input devices until we find the id = NULL.
|
|
info.isInput = true;
|
|
info.getDefault = true;
|
|
result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::getDefaultInputDevice: error (" << getErrorString( result ) << ") enumerating input devices!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return 0;
|
|
}
|
|
|
|
if ( info.counter > 0 ) return info.counter - 1;
|
|
return 0;
|
|
}
|
|
|
|
unsigned int RtApiDs :: getDefaultOutputDevice( void )
|
|
{
|
|
// Enumerate output devices until we find the id = NULL.
|
|
EnumInfo info;
|
|
info.getDefault = true;
|
|
HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::getDefaultOutputDevice: error (" << getErrorString( result ) << ") enumerating output devices!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return 0;
|
|
}
|
|
|
|
if ( info.counter > 0 ) return info.counter - 1;
|
|
return 0;
|
|
}
|
|
|
|
unsigned int RtApiDs :: getDeviceCount( void )
|
|
{
|
|
// Count DirectSound devices.
|
|
EnumInfo info;
|
|
HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating output devices!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
}
|
|
|
|
// Count DirectSoundCapture devices.
|
|
info.isInput = true;
|
|
result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &info );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::getDeviceCount: error (" << getErrorString( result ) << ") enumerating input devices!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
}
|
|
|
|
return info.counter;
|
|
}
|
|
|
|
RtAudio::DeviceInfo RtApiDs :: getDeviceInfo( unsigned int device )
|
|
{
|
|
// Because DirectSound always enumerates input and output devices
|
|
// separately (and because we don't attempt to combine devices
|
|
// internally), none of our "devices" will ever be duplex.
|
|
|
|
RtAudio::DeviceInfo info;
|
|
info.probed = false;
|
|
|
|
// Enumerate through devices to find the id (if it exists). Note
|
|
// that we have to do the output enumeration first, even if this is
|
|
// an input device, in order for the device counter to be correct.
|
|
EnumInfo dsinfo;
|
|
dsinfo.findIndex = true;
|
|
dsinfo.index = device;
|
|
HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating output devices!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
}
|
|
|
|
if ( dsinfo.name.empty() ) goto probeInput;
|
|
|
|
LPDIRECTSOUND output;
|
|
DSCAPS outCaps;
|
|
result = DirectSoundCreate( dsinfo.id, &output, NULL );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
outCaps.dwSize = sizeof( outCaps );
|
|
result = output->GetCaps( &outCaps );
|
|
if ( FAILED( result ) ) {
|
|
output->Release();
|
|
errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") getting capabilities!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
// Get 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] );
|
|
}
|
|
|
|
// 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;
|
|
|
|
// Copy name and return.
|
|
info.name = dsinfo.name;
|
|
|
|
info.probed = true;
|
|
return info;
|
|
|
|
probeInput:
|
|
|
|
dsinfo.isInput = true;
|
|
result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") enumerating input devices!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
}
|
|
|
|
if ( dsinfo.name.empty() ) return info;
|
|
|
|
LPDIRECTSOUNDCAPTURE input;
|
|
result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::getDeviceInfo: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::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 (" << dsinfo.name << ")!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
// Get input channel information.
|
|
info.inputChannels = inCaps.dwChannels;
|
|
|
|
// Get sample rate and format information.
|
|
if ( inCaps.dwChannels == 2 ) {
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.nativeFormats |= RTAUDIO_SINT16;
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.nativeFormats |= RTAUDIO_SINT16;
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.nativeFormats |= RTAUDIO_SINT16;
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.nativeFormats |= RTAUDIO_SINT16;
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.nativeFormats |= RTAUDIO_SINT8;
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.nativeFormats |= RTAUDIO_SINT8;
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.nativeFormats |= RTAUDIO_SINT8;
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.nativeFormats |= RTAUDIO_SINT8;
|
|
|
|
if ( info.nativeFormats & RTAUDIO_SINT16 ) {
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_1S16 ) info.sampleRates.push_back( 11025 );
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_2S16 ) info.sampleRates.push_back( 22050 );
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_4S16 ) info.sampleRates.push_back( 44100 );
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_96S16 ) info.sampleRates.push_back( 96000 );
|
|
}
|
|
else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_1S08 ) info.sampleRates.push_back( 11025 );
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_2S08 ) info.sampleRates.push_back( 22050 );
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_4S08 ) info.sampleRates.push_back( 44100 );
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_96S08 ) info.sampleRates.push_back( 44100 );
|
|
}
|
|
}
|
|
else if ( inCaps.dwChannels == 1 ) {
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.nativeFormats |= RTAUDIO_SINT16;
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.nativeFormats |= RTAUDIO_SINT16;
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.nativeFormats |= RTAUDIO_SINT16;
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.nativeFormats |= RTAUDIO_SINT16;
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.nativeFormats |= RTAUDIO_SINT8;
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.nativeFormats |= RTAUDIO_SINT8;
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.nativeFormats |= RTAUDIO_SINT8;
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.nativeFormats |= RTAUDIO_SINT8;
|
|
|
|
if ( info.nativeFormats & RTAUDIO_SINT16 ) {
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_1M16 ) info.sampleRates.push_back( 11025 );
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_2M16 ) info.sampleRates.push_back( 22050 );
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_4M16 ) info.sampleRates.push_back( 44100 );
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_96M16 ) info.sampleRates.push_back( 96000 );
|
|
}
|
|
else if ( info.nativeFormats & RTAUDIO_SINT8 ) {
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_1M08 ) info.sampleRates.push_back( 11025 );
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_2M08 ) info.sampleRates.push_back( 22050 );
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_4M08 ) info.sampleRates.push_back( 44100 );
|
|
if ( inCaps.dwFormats & WAVE_FORMAT_96M08 ) info.sampleRates.push_back( 96000 );
|
|
}
|
|
}
|
|
else info.inputChannels = 0; // technically, this would be an error
|
|
|
|
input->Release();
|
|
|
|
if ( info.inputChannels == 0 ) return info;
|
|
|
|
if ( getDefaultInputDevice() == device )
|
|
info.isDefaultInput = true;
|
|
|
|
// Copy name and return.
|
|
info.name = dsinfo.name;
|
|
info.probed = true;
|
|
return info;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
// Enumerate through devices to find the id (if it exists). Note
|
|
// that we have to do the output enumeration first, even if this is
|
|
// an input device, in order for the device counter to be correct.
|
|
EnumInfo dsinfo;
|
|
dsinfo.findIndex = true;
|
|
dsinfo.index = device;
|
|
HRESULT result = DirectSoundEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating output devices!";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
if ( mode == OUTPUT ) {
|
|
if ( dsinfo.name.empty() ) {
|
|
errorStream_ << "RtApiDs::probeDeviceOpen: device (" << device << ") does not support output!";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
}
|
|
else { // mode == INPUT
|
|
dsinfo.isInput = true;
|
|
HRESULT result = DirectSoundCaptureEnumerate( (LPDSENUMCALLBACK) deviceQueryCallback, &dsinfo );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") enumerating input devices!";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
if ( dsinfo.name.empty() ) {
|
|
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. However, for console
|
|
// applications, no sound was produced when using GetDesktopWindow().
|
|
HWND hWnd = GetForegroundWindow();
|
|
|
|
// Check the numberOfBuffers parameter and limit the lowest value to
|
|
// two. This is a judgement call and a value of two is probably too
|
|
// low for capture, but it should work for playback.
|
|
int nBuffers = 0;
|
|
if ( options ) nBuffers = options->numberOfBuffers;
|
|
if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) nBuffers = 2;
|
|
if ( nBuffers < 2 ) nBuffers = 3;
|
|
|
|
// 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, 32k, but we will
|
|
// grow it to make allowances for very large software buffer sizes.
|
|
DWORD dsBufferSize = 0;
|
|
DWORD dsPointerLeadTime = 0;
|
|
long bufferBytes = MINIMUM_DEVICE_BUFFER_SIZE; // sound cards will always *knock wood* support this
|
|
|
|
void *ohandle = 0, *bhandle = 0;
|
|
if ( mode == OUTPUT ) {
|
|
|
|
LPDIRECTSOUND output;
|
|
result = DirectSoundCreate( dsinfo.id, &output, NULL );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening output device (" << dsinfo.name << ")!";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
DSCAPS outCaps;
|
|
outCaps.dwSize = sizeof( outCaps );
|
|
result = output->GetCaps( &outCaps );
|
|
if ( FAILED( result ) ) {
|
|
output->Release();
|
|
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting capabilities (" << dsinfo.name << ")!";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Check channel information.
|
|
if ( channels + firstChannel == 2 && !( outCaps.dwFlags & DSCAPS_PRIMARYSTEREO ) ) {
|
|
errorStream_ << "RtApiDs::getDeviceInfo: the output device (" << dsinfo.name << ") does not support stereo playback.";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Check format information. Use 16-bit format unless not
|
|
// supported or user requests 8-bit.
|
|
if ( outCaps.dwFlags & DSCAPS_PRIMARY16BIT &&
|
|
!( format == RTAUDIO_SINT8 && outCaps.dwFlags & DSCAPS_PRIMARY8BIT ) ) {
|
|
waveFormat.wBitsPerSample = 16;
|
|
stream_.deviceFormat[mode] = RTAUDIO_SINT16;
|
|
}
|
|
else {
|
|
waveFormat.wBitsPerSample = 8;
|
|
stream_.deviceFormat[mode] = RTAUDIO_SINT8;
|
|
}
|
|
stream_.userFormat = format;
|
|
|
|
// Update wave format structure and buffer information.
|
|
waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
|
|
waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
|
|
dsPointerLeadTime = nBuffers * (*bufferSize) * (waveFormat.wBitsPerSample / 8) * channels;
|
|
|
|
// If the user wants an even bigger buffer, increase the device buffer size accordingly.
|
|
while ( dsPointerLeadTime * 2U > (DWORD) bufferBytes )
|
|
bufferBytes *= 2;
|
|
|
|
// Set cooperative level to DSSCL_EXCLUSIVE ... sound stops when window focus changes.
|
|
// result = output->SetCooperativeLevel( hWnd, DSSCL_EXCLUSIVE );
|
|
// Set cooperative level to DSSCL_PRIORITY ... sound remains when window focus changes.
|
|
result = output->SetCooperativeLevel( hWnd, DSSCL_PRIORITY );
|
|
if ( FAILED( result ) ) {
|
|
output->Release();
|
|
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") setting cooperative level (" << dsinfo.name << ")!";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Even though we will write to the secondary buffer, we need to
|
|
// access the primary buffer to set the correct output format
|
|
// (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 (" << dsinfo.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 (" << dsinfo.name << ")!";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Setup the secondary DS buffer description.
|
|
dsBufferSize = (DWORD) bufferBytes;
|
|
ZeroMemory( &bufferDescription, sizeof( DSBUFFERDESC ) );
|
|
bufferDescription.dwSize = sizeof( DSBUFFERDESC );
|
|
bufferDescription.dwFlags = ( DSBCAPS_STICKYFOCUS |
|
|
DSBCAPS_GLOBALFOCUS |
|
|
DSBCAPS_GETCURRENTPOSITION2 |
|
|
DSBCAPS_LOCHARDWARE ); // Force hardware mixing
|
|
bufferDescription.dwBufferBytes = bufferBytes;
|
|
bufferDescription.lpwfxFormat = &waveFormat;
|
|
|
|
// Try to create the secondary DS buffer. If that doesn't work,
|
|
// try to use software mixing. Otherwise, there's a problem.
|
|
result = 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 (" << dsinfo.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 (" << dsinfo.name << ")!";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
bufferBytes = dsbcaps.dwBufferBytes;
|
|
|
|
// Lock the DS buffer
|
|
LPVOID audioPtr;
|
|
DWORD dataLen;
|
|
result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 );
|
|
if ( FAILED( result ) ) {
|
|
output->Release();
|
|
buffer->Release();
|
|
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking buffer (" << dsinfo.name << ")!";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Zero the DS buffer
|
|
ZeroMemory( audioPtr, dataLen );
|
|
|
|
// 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 (" << dsinfo.name << ")!";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
dsBufferSize = bufferBytes;
|
|
ohandle = (void *) output;
|
|
bhandle = (void *) buffer;
|
|
}
|
|
|
|
if ( mode == INPUT ) {
|
|
|
|
LPDIRECTSOUNDCAPTURE input;
|
|
result = DirectSoundCaptureCreate( dsinfo.id, &input, NULL );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") opening input device (" << dsinfo.name << ")!";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
DSCCAPS inCaps;
|
|
inCaps.dwSize = sizeof( inCaps );
|
|
result = input->GetCaps( &inCaps );
|
|
if ( FAILED( result ) ) {
|
|
input->Release();
|
|
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") getting input capabilities (" << dsinfo.name << ")!";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// 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 > (DWORD) bufferBytes )
|
|
bufferBytes *= 2;
|
|
|
|
// Setup the secondary DS buffer description.
|
|
dsBufferSize = bufferBytes;
|
|
DSCBUFFERDESC bufferDescription;
|
|
ZeroMemory( &bufferDescription, sizeof( DSCBUFFERDESC ) );
|
|
bufferDescription.dwSize = sizeof( DSCBUFFERDESC );
|
|
bufferDescription.dwFlags = 0;
|
|
bufferDescription.dwReserved = 0;
|
|
bufferDescription.dwBufferBytes = bufferBytes;
|
|
bufferDescription.lpwfxFormat = &waveFormat;
|
|
|
|
// Create the capture buffer.
|
|
LPDIRECTSOUNDCAPTUREBUFFER buffer;
|
|
result = input->CreateCaptureBuffer( &bufferDescription, &buffer, NULL );
|
|
if ( FAILED( result ) ) {
|
|
input->Release();
|
|
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") creating input buffer (" << dsinfo.name << ")!";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// 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 (" << dsinfo.name << ")!";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
bufferBytes = dscbcaps.dwBufferBytes;
|
|
|
|
// Lock the capture buffer
|
|
LPVOID audioPtr;
|
|
DWORD dataLen;
|
|
result = buffer->Lock( 0, bufferBytes, &audioPtr, &dataLen, NULL, NULL, 0 );
|
|
if ( FAILED( result ) ) {
|
|
input->Release();
|
|
buffer->Release();
|
|
errorStream_ << "RtApiDs::probeDeviceOpen: error (" << getErrorString( result ) << ") locking input buffer (" << dsinfo.name << ")!";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Zero the buffer
|
|
ZeroMemory( audioPtr, dataLen );
|
|
|
|
// 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 (" << dsinfo.name << ")!";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
dsBufferSize = bufferBytes;
|
|
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
|
|
bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
|
|
stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
|
|
if ( stream_.userBuffer[mode] == NULL ) {
|
|
errorText_ = "RtApiDs::probeDeviceOpen: error allocating user buffer memory.";
|
|
goto error;
|
|
}
|
|
|
|
if ( stream_.doConvertBuffer[mode] ) {
|
|
|
|
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;
|
|
}
|
|
}
|
|
|
|
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;
|
|
}
|
|
}
|
|
}
|
|
|
|
// Allocate our DsHandle structures for the stream.
|
|
if ( stream_.apiHandle == 0 ) {
|
|
try {
|
|
handle = new DsHandle;
|
|
}
|
|
catch ( std::bad_alloc& ) {
|
|
errorText_ = "RtApiDs::probeDeviceOpen: error allocating AsioHandle memory.";
|
|
goto error;
|
|
}
|
|
|
|
// Create a manual-reset event.
|
|
handle->condition = CreateEvent( NULL, // no security
|
|
TRUE, // manual-reset
|
|
FALSE, // non-signaled initially
|
|
NULL ); // unnamed
|
|
stream_.apiHandle = (void *) handle;
|
|
}
|
|
else
|
|
handle = (DsHandle *) stream_.apiHandle;
|
|
handle->id[mode] = ohandle;
|
|
handle->buffer[mode] = bhandle;
|
|
handle->dsBufferSize[mode] = dsBufferSize;
|
|
handle->dsPointerLeadTime[mode] = dsPointerLeadTime;
|
|
|
|
stream_.device[mode] = device;
|
|
stream_.state = STREAM_STOPPED;
|
|
if ( stream_.mode == OUTPUT && mode == INPUT )
|
|
// We had already set up an output stream.
|
|
stream_.mode = DUPLEX;
|
|
else
|
|
stream_.mode = mode;
|
|
stream_.nBuffers = nBuffers;
|
|
stream_.sampleRate = sampleRate;
|
|
|
|
// Setup the buffer conversion information structure.
|
|
if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
|
|
|
|
// Setup the callback thread.
|
|
unsigned threadId;
|
|
stream_.callbackInfo.object = (void *) this;
|
|
stream_.callbackInfo.isRunning = true;
|
|
stream_.callbackInfo.thread = _beginthreadex( NULL, 0, &callbackHandler,
|
|
&stream_.callbackInfo, 0, &threadId );
|
|
if ( stream_.callbackInfo.thread == 0 ) {
|
|
errorText_ = "RtApiDs::probeDeviceOpen: error creating callback thread!";
|
|
goto error;
|
|
}
|
|
|
|
// Boost DS thread priority
|
|
SetThreadPriority( (HANDLE) stream_.callbackInfo.thread, THREAD_PRIORITY_HIGHEST );
|
|
return SUCCESS;
|
|
|
|
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;
|
|
}
|
|
|
|
for ( int i=0; i<2; i++ ) {
|
|
if ( stream_.userBuffer[i] ) {
|
|
free( stream_.userBuffer[i] );
|
|
stream_.userBuffer[i] = 0;
|
|
}
|
|
}
|
|
|
|
if ( stream_.deviceBuffer ) {
|
|
free( stream_.deviceBuffer );
|
|
stream_.deviceBuffer = 0;
|
|
}
|
|
|
|
return FAILURE;
|
|
}
|
|
|
|
void RtApiDs :: closeStream()
|
|
{
|
|
if ( stream_.state == STREAM_CLOSED ) {
|
|
errorText_ = "RtApiDs::closeStream(): no open stream to close!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
// 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;
|
|
}
|
|
|
|
for ( int i=0; i<2; i++ ) {
|
|
if ( stream_.userBuffer[i] ) {
|
|
free( stream_.userBuffer[i] );
|
|
stream_.userBuffer[i] = 0;
|
|
}
|
|
}
|
|
|
|
if ( stream_.deviceBuffer ) {
|
|
free( stream_.deviceBuffer );
|
|
stream_.deviceBuffer = 0;
|
|
}
|
|
|
|
stream_.mode = UNINITIALIZED;
|
|
stream_.state = STREAM_CLOSED;
|
|
}
|
|
|
|
void RtApiDs :: startStream()
|
|
{
|
|
verifyStream();
|
|
if ( stream_.state == STREAM_RUNNING ) {
|
|
errorText_ = "RtApiDs::startStream(): the stream is already running!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
// Increase scheduler frequency on lesser windows (a side-effect of
|
|
// increasing timer accuracy). On greater windows (Win2K or later),
|
|
// this is already in effect.
|
|
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
|
|
DsHandle *handle = (DsHandle *) stream_.apiHandle;
|
|
|
|
timeBeginPeriod( 1 );
|
|
|
|
/*
|
|
memset( &statistics, 0, sizeof( statistics ) );
|
|
statistics.sampleRate = stream_.sampleRate;
|
|
statistics.writeDeviceBufferLeadBytes = handle->dsPointerLeadTime[0];
|
|
*/
|
|
|
|
buffersRolling = false;
|
|
duplexPrerollBytes = 0;
|
|
|
|
if ( stream_.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] );
|
|
}
|
|
|
|
HRESULT result = 0;
|
|
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
|
|
//statistics.outputFrameSize = formatBytes( stream_.deviceFormat[0] ) * stream_.nDeviceChannels[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;
|
|
}
|
|
}
|
|
|
|
if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
|
|
//statistics.inputFrameSize = formatBytes( stream_.deviceFormat[1]) * stream_.nDeviceChannels[1];
|
|
|
|
LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
|
|
result = buffer->Start( DSCBSTART_LOOPING );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::startStream: error (" << getErrorString( result ) << ") starting input buffer!";
|
|
errorText_ = errorStream_.str();
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
handle->drainCounter = 0;
|
|
handle->internalDrain = false;
|
|
stream_.state = STREAM_RUNNING;
|
|
|
|
unlock:
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
|
|
if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
|
|
}
|
|
|
|
void RtApiDs :: stopStream()
|
|
{
|
|
verifyStream();
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
errorText_ = "RtApiDs::stopStream(): the stream is already stopped!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
return;
|
|
}
|
|
|
|
HRESULT result = 0;
|
|
LPVOID audioPtr;
|
|
DWORD dataLen;
|
|
DsHandle *handle = (DsHandle *) stream_.apiHandle;
|
|
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
|
|
if ( handle->drainCounter == 0 ) {
|
|
handle->drainCounter = 1;
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
WaitForMultipleObjects( 1, &handle->condition, FALSE, INFINITE ); // block until signaled
|
|
ResetEvent( handle->condition );
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
}
|
|
|
|
// Stop the buffer and clear memory
|
|
LPDIRECTSOUNDBUFFER buffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
|
|
result = buffer->Stop();
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::stopStream: error (" << getErrorString( result ) << ") stopping output 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[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;
|
|
}
|
|
|
|
if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
|
|
LPDIRECTSOUNDCAPTUREBUFFER buffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
|
|
audioPtr = NULL;
|
|
dataLen = 0;
|
|
|
|
result = buffer->Stop();
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::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;
|
|
}
|
|
|
|
// 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 input buffer!";
|
|
errorText_ = errorStream_.str();
|
|
goto unlock;
|
|
}
|
|
|
|
// If we start recording again, we must begin at beginning of buffer.
|
|
handle->bufferPointer[1] = 0;
|
|
}
|
|
|
|
unlock:
|
|
timeEndPeriod( 1 ); // revert to normal scheduler frequency on lesser windows.
|
|
stream_.state = STREAM_STOPPED;
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
|
|
if ( FAILED( result ) ) error( RtError::SYSTEM_ERROR );
|
|
}
|
|
|
|
void RtApiDs :: abortStream()
|
|
{
|
|
verifyStream();
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
errorText_ = "RtApiDs::abortStream(): the stream is already stopped!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
DsHandle *handle = (DsHandle *) stream_.apiHandle;
|
|
handle->drainCounter = 1;
|
|
|
|
stopStream();
|
|
}
|
|
|
|
void RtApiDs :: callbackEvent()
|
|
{
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
Sleep(50); // sleep 50 milliseconds
|
|
return;
|
|
}
|
|
|
|
if ( stream_.state == STREAM_CLOSED ) {
|
|
errorText_ = "RtApiDs::callbackEvent(): the stream is closed ... this shouldn't happen!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
CallbackInfo *info = (CallbackInfo *) &stream_.callbackInfo;
|
|
DsHandle *handle = (DsHandle *) stream_.apiHandle;
|
|
|
|
// Check if we were draining the stream and signal is finished.
|
|
if ( handle->drainCounter > stream_.nBuffers + 2 ) {
|
|
if ( handle->internalDrain == false )
|
|
SetEvent( handle->condition );
|
|
else
|
|
stopStream();
|
|
return;
|
|
}
|
|
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
|
|
// The state might change while waiting on a mutex.
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
return;
|
|
}
|
|
|
|
// Invoke user callback to get fresh output data UNLESS we are
|
|
// draining stream.
|
|
if ( handle->drainCounter == 0 ) {
|
|
RtAudioCallback callback = (RtAudioCallback) info->callback;
|
|
double streamTime = getStreamTime();
|
|
RtAudioStreamStatus status = 0;
|
|
if ( stream_.mode != INPUT && handle->xrun[0] == true ) {
|
|
status |= RTAUDIO_OUTPUT_UNDERFLOW;
|
|
handle->xrun[0] = false;
|
|
}
|
|
if ( stream_.mode != OUTPUT && handle->xrun[1] == true ) {
|
|
status |= RTAUDIO_INPUT_OVERFLOW;
|
|
handle->xrun[1] = false;
|
|
}
|
|
handle->drainCounter = callback( stream_.userBuffer[0], stream_.userBuffer[1],
|
|
stream_.bufferSize, streamTime, status, info->userData );
|
|
if ( handle->drainCounter == 2 ) {
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
abortStream();
|
|
return;
|
|
}
|
|
else if ( handle->drainCounter == 1 )
|
|
handle->internalDrain = true;
|
|
}
|
|
|
|
HRESULT result;
|
|
DWORD currentWritePos, safeWritePos;
|
|
DWORD currentReadPos, safeReadPos;
|
|
DWORD leadPos;
|
|
UINT nextWritePos;
|
|
|
|
#ifdef GENERATE_DEBUG_LOG
|
|
DWORD writeTime, readTime;
|
|
#endif
|
|
|
|
LPVOID buffer1 = NULL;
|
|
LPVOID buffer2 = NULL;
|
|
DWORD bufferSize1 = 0;
|
|
DWORD bufferSize2 = 0;
|
|
|
|
char *buffer;
|
|
long bufferBytes;
|
|
|
|
if ( stream_.mode == DUPLEX && !buffersRolling ) {
|
|
//assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
|
|
|
|
// It takes a while for the devices to get rolling. As a result,
|
|
// there's no guarantee that the capture and write device pointers
|
|
// will move in lockstep. Wait here for both devices to start
|
|
// rolling, and then set our buffer pointers accordingly.
|
|
// e.g. Crystal Drivers: the capture buffer starts up 5700 to 9600
|
|
// bytes later than the write buffer.
|
|
|
|
// Stub: a serious risk of having a pre-emptive scheduling round
|
|
// take place between the two GetCurrentPosition calls... but I'm
|
|
// really not sure how to solve the problem. Temporarily boost to
|
|
// Realtime priority, maybe; but I'm not sure what priority the
|
|
// DirectSound service threads run at. We *should* be roughly
|
|
// within a ms or so of correct.
|
|
|
|
LPDIRECTSOUNDBUFFER dsWriteBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
|
|
LPDIRECTSOUNDCAPTUREBUFFER dsCaptureBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
|
|
|
|
DWORD initialWritePos, initialSafeWritePos;
|
|
DWORD initialReadPos, initialSafeReadPos;
|
|
|
|
result = dsWriteBuffer->GetCurrentPosition( &initialWritePos, &initialSafeWritePos );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
result = dsCaptureBuffer->GetCurrentPosition( &initialReadPos, &initialSafeReadPos );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
while ( true ) {
|
|
result = dsWriteBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
result = dsCaptureBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
if ( safeWritePos != initialSafeWritePos && safeReadPos != initialSafeReadPos ) break;
|
|
Sleep( 1 );
|
|
}
|
|
|
|
//assert( handle->dsBufferSize[0] == handle->dsBufferSize[1] );
|
|
|
|
buffersRolling = true;
|
|
handle->bufferPointer[0] = ( safeWritePos + handle->dsPointerLeadTime[0] );
|
|
handle->bufferPointer[1] = safeReadPos;
|
|
}
|
|
|
|
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
|
|
|
|
LPDIRECTSOUNDBUFFER dsBuffer = (LPDIRECTSOUNDBUFFER) handle->buffer[0];
|
|
|
|
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 );
|
|
}
|
|
|
|
// 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 );
|
|
}
|
|
|
|
// No byte swapping necessary in DirectSound implementation.
|
|
|
|
// 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 );
|
|
|
|
DWORD dsBufferSize = handle->dsBufferSize[0];
|
|
nextWritePos = handle->bufferPointer[0];
|
|
|
|
DWORD endWrite;
|
|
while ( true ) {
|
|
// Find out where the read and "safe write" pointers are.
|
|
result = dsBuffer->GetCurrentPosition( ¤tWritePos, &safeWritePos );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current write position!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
|
|
leadPos = safeWritePos + handle->dsPointerLeadTime[0];
|
|
if ( leadPos > dsBufferSize ) leadPos -= dsBufferSize;
|
|
if ( leadPos < nextWritePos ) leadPos += dsBufferSize; // unwrap offset
|
|
endWrite = nextWritePos + bufferBytes;
|
|
|
|
// Check whether the entire write region is behind the play pointer.
|
|
if ( leadPos >= endWrite ) break;
|
|
|
|
// If we are here, then we must wait until the play pointer gets
|
|
// beyond the write region. The approach here is to use the
|
|
// Sleep() function to suspend operation until safePos catches
|
|
// up. Calculate number of milliseconds to wait as:
|
|
// time = distance * (milliseconds/second) * fudgefactor /
|
|
// ((bytes/sample) * (samples/second))
|
|
// A "fudgefactor" less than 1 is used because it was found
|
|
// that sleeping too long was MUCH worse than sleeping for
|
|
// several shorter periods.
|
|
double millis = ( endWrite - leadPos ) * 900.0;
|
|
millis /= ( formatBytes( stream_.deviceFormat[0]) * stream_.nDeviceChannels[0] * stream_.sampleRate);
|
|
if ( millis < 1.0 ) millis = 1.0;
|
|
if ( millis > 50.0 ) {
|
|
static int nOverruns = 0;
|
|
++nOverruns;
|
|
}
|
|
Sleep( (DWORD) millis );
|
|
}
|
|
|
|
//if ( statistics.writeDeviceSafeLeadBytes < dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] ) ) {
|
|
// statistics.writeDeviceSafeLeadBytes = dsPointerDifference( safeWritePos, currentWritePos, handle->dsBufferSize[0] );
|
|
//}
|
|
|
|
if ( dsPointerBetween( nextWritePos, safeWritePos, currentWritePos, dsBufferSize )
|
|
|| dsPointerBetween( endWrite, safeWritePos, currentWritePos, dsBufferSize ) ) {
|
|
// We've strayed into the forbidden zone ... resync the read pointer.
|
|
//++statistics.numberOfWriteUnderruns;
|
|
handle->xrun[0] = true;
|
|
nextWritePos = safeWritePos + handle->dsPointerLeadTime[0] - bufferBytes + dsBufferSize;
|
|
while ( nextWritePos >= dsBufferSize ) nextWritePos -= dsBufferSize;
|
|
handle->bufferPointer[0] = nextWritePos;
|
|
endWrite = nextWritePos + bufferBytes;
|
|
}
|
|
|
|
// Lock free space in the buffer
|
|
result = dsBuffer->Lock( nextWritePos, bufferBytes, &buffer1,
|
|
&bufferSize1, &buffer2, &bufferSize2, 0 );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking buffer during playback!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
|
|
// Copy our buffer into the DS buffer
|
|
CopyMemory( buffer1, buffer, bufferSize1 );
|
|
if ( buffer2 != NULL ) CopyMemory( buffer2, buffer+bufferSize1, bufferSize2 );
|
|
|
|
// Update our buffer offset and unlock sound buffer
|
|
dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking buffer during playback!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
nextWritePos = ( nextWritePos + bufferSize1 + bufferSize2 ) % dsBufferSize;
|
|
handle->bufferPointer[0] = nextWritePos;
|
|
|
|
if ( handle->drainCounter ) {
|
|
handle->drainCounter++;
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
|
|
|
|
// 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 );
|
|
}
|
|
|
|
LPDIRECTSOUNDCAPTUREBUFFER dsBuffer = (LPDIRECTSOUNDCAPTUREBUFFER) handle->buffer[1];
|
|
long nextReadPos = handle->bufferPointer[1];
|
|
DWORD dsBufferSize = handle->dsBufferSize[1];
|
|
|
|
// Find out where the write and "safe read" pointers are.
|
|
result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
|
|
if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset
|
|
DWORD endRead = nextReadPos + bufferBytes;
|
|
|
|
// Handling depends on whether we are INPUT or DUPLEX.
|
|
// If we're in INPUT mode then waiting is a good thing. If we're in DUPLEX mode,
|
|
// then a wait here will drag the write pointers into the forbidden zone.
|
|
//
|
|
// In DUPLEX mode, rather than wait, we will back off the read pointer until
|
|
// it's in a safe position. This causes dropouts, but it seems to be the only
|
|
// practical way to sync up the read and write pointers reliably, given the
|
|
// the very complex relationship between phase and increment of the read and write
|
|
// pointers.
|
|
//
|
|
// In order to minimize audible dropouts in DUPLEX mode, we will
|
|
// provide a pre-roll period of 0.5 seconds in which we return
|
|
// zeros from the read buffer while the pointers sync up.
|
|
|
|
if ( stream_.mode == DUPLEX ) {
|
|
if ( safeReadPos < endRead ) {
|
|
if ( duplexPrerollBytes <= 0 ) {
|
|
// Pre-roll time over. Be more agressive.
|
|
int adjustment = endRead-safeReadPos;
|
|
|
|
handle->xrun[1] = true;
|
|
//++statistics.numberOfReadOverruns;
|
|
// Two cases:
|
|
// - large adjustments: we've probably run out of CPU cycles, so just resync exactly,
|
|
// and perform fine adjustments later.
|
|
// - small adjustments: back off by twice as much.
|
|
if ( adjustment >= 2*bufferBytes )
|
|
nextReadPos = safeReadPos-2*bufferBytes;
|
|
else
|
|
nextReadPos = safeReadPos-bufferBytes-adjustment;
|
|
|
|
//statistics.readDeviceSafeLeadBytes = currentReadPos-nextReadPos;
|
|
//if ( statistics.readDeviceSafeLeadBytes < 0) statistics.readDeviceSafeLeadBytes += dsBufferSize;
|
|
if ( nextReadPos < 0 ) nextReadPos += dsBufferSize;
|
|
|
|
}
|
|
else {
|
|
// In pre=roll time. Just do it.
|
|
nextReadPos = safeReadPos-bufferBytes;
|
|
while ( nextReadPos < 0 ) nextReadPos += dsBufferSize;
|
|
}
|
|
endRead = nextReadPos + bufferBytes;
|
|
}
|
|
}
|
|
else { // mode == INPUT
|
|
while ( safeReadPos < endRead ) {
|
|
// See comments for playback.
|
|
double millis = (endRead - safeReadPos) * 900.0;
|
|
millis /= ( formatBytes(stream_.deviceFormat[1]) * stream_.nDeviceChannels[1] * stream_.sampleRate);
|
|
if ( millis < 1.0 ) millis = 1.0;
|
|
Sleep( (DWORD) millis );
|
|
|
|
// Wake up, find out where we are now
|
|
result = dsBuffer->GetCurrentPosition( ¤tReadPos, &safeReadPos );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") getting current read position!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
|
|
if ( safeReadPos < (DWORD)nextReadPos ) safeReadPos += dsBufferSize; // unwrap offset
|
|
}
|
|
}
|
|
|
|
//if (statistics.readDeviceSafeLeadBytes < dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize ) )
|
|
// statistics.readDeviceSafeLeadBytes = dsPointerDifference( currentReadPos, nextReadPos, dsBufferSize );
|
|
|
|
// Lock free space in the buffer
|
|
result = dsBuffer->Lock( nextReadPos, bufferBytes, &buffer1,
|
|
&bufferSize1, &buffer2, &bufferSize2, 0 );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") locking capture buffer!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
|
|
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
|
|
nextReadPos = ( nextReadPos + bufferSize1 + bufferSize2 ) % dsBufferSize;
|
|
dsBuffer->Unlock( buffer1, bufferSize1, buffer2, bufferSize2 );
|
|
if ( FAILED( result ) ) {
|
|
errorStream_ << "RtApiDs::callbackEvent: error (" << getErrorString( result ) << ") unlocking capture buffer!";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
handle->bufferPointer[1] = nextReadPos;
|
|
|
|
// No byte swapping necessary in DirectSound implementation.
|
|
|
|
// If necessary, convert 8-bit data from unsigned to signed.
|
|
if ( stream_.deviceFormat[1] == RTAUDIO_SINT8 )
|
|
for ( int j=0; j<bufferBytes; j++ ) buffer[j] = (signed char) ( buffer[j] - 128 );
|
|
|
|
// Do buffer conversion if necessary.
|
|
if ( stream_.doConvertBuffer[1] )
|
|
convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
|
|
}
|
|
#ifdef GENERATE_DEBUG_LOG
|
|
if ( currentDebugLogEntry < debugLog.size() )
|
|
{
|
|
TTickRecord &r = debugLog[currentDebugLogEntry++];
|
|
r.currentReadPointer = currentReadPos;
|
|
r.safeReadPointer = safeReadPos;
|
|
r.currentWritePointer = currentWritePos;
|
|
r.safeWritePointer = safeWritePos;
|
|
r.readTime = readTime;
|
|
r.writeTime = writeTime;
|
|
r.nextReadPointer = handles[1].bufferPointer;
|
|
r.nextWritePointer = handles[0].bufferPointer;
|
|
}
|
|
#endif
|
|
|
|
unlock:
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
|
|
RtApi::tickStreamTime();
|
|
}
|
|
|
|
// Definitions for utility functions and callbacks
|
|
// specific to the DirectSound implementation.
|
|
|
|
extern "C" unsigned __stdcall callbackHandler( void *ptr )
|
|
{
|
|
CallbackInfo *info = (CallbackInfo *) ptr;
|
|
RtApiDs *object = (RtApiDs *) info->object;
|
|
bool* isRunning = &info->isRunning;
|
|
|
|
while ( *isRunning == true ) {
|
|
object->callbackEvent();
|
|
}
|
|
|
|
_endthreadex( 0 );
|
|
return 0;
|
|
}
|
|
|
|
#include "tchar.h"
|
|
|
|
std::string convertTChar( LPCTSTR name )
|
|
{
|
|
std::string s;
|
|
|
|
#if defined( UNICODE ) || defined( _UNICODE )
|
|
// Yes, this conversion doesn't make sense for two-byte characters
|
|
// but RtAudio is currently written to return an std::string of
|
|
// one-byte chars for the device name.
|
|
for ( unsigned int i=0; i<wcslen( name ); i++ )
|
|
s.push_back( name[i] );
|
|
#else
|
|
s.append( std::string( name ) );
|
|
#endif
|
|
|
|
return s;
|
|
}
|
|
|
|
static BOOL CALLBACK deviceQueryCallback( LPGUID lpguid,
|
|
LPCTSTR description,
|
|
LPCTSTR module,
|
|
LPVOID lpContext )
|
|
{
|
|
EnumInfo *info = (EnumInfo *) lpContext;
|
|
|
|
HRESULT hr;
|
|
if ( info->isInput == true ) {
|
|
DSCCAPS caps;
|
|
LPDIRECTSOUNDCAPTURE object;
|
|
|
|
hr = DirectSoundCaptureCreate( lpguid, &object, NULL );
|
|
if ( hr != DS_OK ) return TRUE;
|
|
|
|
caps.dwSize = sizeof(caps);
|
|
hr = object->GetCaps( &caps );
|
|
if ( hr == DS_OK ) {
|
|
if ( caps.dwChannels > 0 && caps.dwFormats > 0 )
|
|
info->counter++;
|
|
}
|
|
object->Release();
|
|
}
|
|
else {
|
|
DSCAPS caps;
|
|
LPDIRECTSOUND object;
|
|
hr = DirectSoundCreate( lpguid, &object, NULL );
|
|
if ( hr != DS_OK ) return TRUE;
|
|
|
|
caps.dwSize = sizeof(caps);
|
|
hr = object->GetCaps( &caps );
|
|
if ( hr == DS_OK ) {
|
|
if ( caps.dwFlags & DSCAPS_PRIMARYMONO || caps.dwFlags & DSCAPS_PRIMARYSTEREO )
|
|
info->counter++;
|
|
}
|
|
object->Release();
|
|
}
|
|
|
|
if ( info->getDefault && lpguid == NULL ) return FALSE;
|
|
|
|
if ( info->findIndex && info->counter > info->index ) {
|
|
info->id = lpguid;
|
|
info->name = convertTChar( description );
|
|
return FALSE;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static char* getErrorString( int code )
|
|
{
|
|
switch ( code ) {
|
|
|
|
case DSERR_ALLOCATED:
|
|
return "Already allocated";
|
|
|
|
case DSERR_CONTROLUNAVAIL:
|
|
return "Control unavailable";
|
|
|
|
case DSERR_INVALIDPARAM:
|
|
return "Invalid parameter";
|
|
|
|
case DSERR_INVALIDCALL:
|
|
return "Invalid call";
|
|
|
|
case DSERR_GENERIC:
|
|
return "Generic error";
|
|
|
|
case DSERR_PRIOLEVELNEEDED:
|
|
return "Priority level needed";
|
|
|
|
case DSERR_OUTOFMEMORY:
|
|
return "Out of memory";
|
|
|
|
case DSERR_BADFORMAT:
|
|
return "The sample rate or the channel format is not supported";
|
|
|
|
case DSERR_UNSUPPORTED:
|
|
return "Not supported";
|
|
|
|
case DSERR_NODRIVER:
|
|
return "No driver";
|
|
|
|
case DSERR_ALREADYINITIALIZED:
|
|
return "Already initialized";
|
|
|
|
case DSERR_NOAGGREGATION:
|
|
return "No aggregation";
|
|
|
|
case DSERR_BUFFERLOST:
|
|
return "Buffer lost";
|
|
|
|
case DSERR_OTHERAPPHASPRIO:
|
|
return "Another application already has priority";
|
|
|
|
case DSERR_UNINITIALIZED:
|
|
return "Uninitialized";
|
|
|
|
default:
|
|
return "DirectSound unknown error";
|
|
}
|
|
}
|
|
//******************** End of __WINDOWS_DS__ *********************//
|
|
#endif
|
|
|
|
|
|
#if defined(__LINUX_ALSA__)
|
|
|
|
#include <alsa/asoundlib.h>
|
|
#include <unistd.h>
|
|
|
|
// A structure to hold various information related to the ALSA API
|
|
// implementation.
|
|
struct AlsaHandle {
|
|
snd_pcm_t *handles[2];
|
|
bool synchronized;
|
|
bool xrun[2];
|
|
pthread_cond_t runnable;
|
|
|
|
AlsaHandle()
|
|
:synchronized(false) { xrun[0] = false; xrun[1] = false; }
|
|
};
|
|
|
|
extern "C" void *alsaCallbackHandler( void * ptr );
|
|
|
|
RtApiAlsa :: RtApiAlsa()
|
|
{
|
|
// Nothing to do here.
|
|
}
|
|
|
|
RtApiAlsa :: ~RtApiAlsa()
|
|
{
|
|
if ( stream_.state != STREAM_CLOSED ) closeStream();
|
|
}
|
|
|
|
unsigned int RtApiAlsa :: getDeviceCount( void )
|
|
{
|
|
unsigned nDevices = 0;
|
|
int result, subdevice, card;
|
|
char name[64];
|
|
snd_ctl_t *handle;
|
|
|
|
// Count cards and devices
|
|
card = -1;
|
|
snd_card_next( &card );
|
|
while ( card >= 0 ) {
|
|
sprintf( name, "hw:%d", card );
|
|
result = snd_ctl_open( &handle, name, 0 );
|
|
if ( result < 0 ) {
|
|
errorStream_ << "RtApiAlsa::getDeviceCount: control open, card = " << card << ", " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
goto nextcard;
|
|
}
|
|
subdevice = -1;
|
|
while( 1 ) {
|
|
result = snd_ctl_pcm_next_device( handle, &subdevice );
|
|
if ( result < 0 ) {
|
|
errorStream_ << "RtApiAlsa::getDeviceCount: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
break;
|
|
}
|
|
if ( subdevice < 0 )
|
|
break;
|
|
nDevices++;
|
|
}
|
|
nextcard:
|
|
snd_ctl_close( handle );
|
|
snd_card_next( &card );
|
|
}
|
|
|
|
return nDevices;
|
|
}
|
|
|
|
RtAudio::DeviceInfo RtApiAlsa :: getDeviceInfo( unsigned int device )
|
|
{
|
|
RtAudio::DeviceInfo info;
|
|
info.probed = false;
|
|
|
|
unsigned nDevices = 0;
|
|
int result, subdevice, card;
|
|
char name[64];
|
|
snd_ctl_t *chandle;
|
|
|
|
// Count cards and devices
|
|
card = -1;
|
|
snd_card_next( &card );
|
|
while ( card >= 0 ) {
|
|
sprintf( name, "hw:%d", card );
|
|
result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
|
|
if ( result < 0 ) {
|
|
errorStream_ << "RtApiAlsa::getDeviceInfo: control open, card = " << card << ", " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
goto nextcard;
|
|
}
|
|
subdevice = -1;
|
|
while( 1 ) {
|
|
result = snd_ctl_pcm_next_device( chandle, &subdevice );
|
|
if ( result < 0 ) {
|
|
errorStream_ << "RtApiAlsa::getDeviceInfo: control next device, card = " << card << ", " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
break;
|
|
}
|
|
if ( subdevice < 0 ) break;
|
|
if ( nDevices == device ) {
|
|
sprintf( name, "hw:%d,%d", card, subdevice );
|
|
goto foundDevice;
|
|
}
|
|
nDevices++;
|
|
}
|
|
nextcard:
|
|
snd_ctl_close( chandle );
|
|
snd_card_next( &card );
|
|
}
|
|
|
|
if ( nDevices == 0 ) {
|
|
errorText_ = "RtApiAlsa::getDeviceInfo: no devices found!";
|
|
error( RtError::INVALID_USE );
|
|
}
|
|
|
|
if ( device >= nDevices ) {
|
|
errorText_ = "RtApiAlsa::getDeviceInfo: device ID is invalid!";
|
|
error( RtError::INVALID_USE );
|
|
}
|
|
|
|
foundDevice:
|
|
|
|
// If a stream is already open, we cannot probe the stream devices.
|
|
// Thus, use the saved results.
|
|
if ( stream_.state != STREAM_CLOSED &&
|
|
( stream_.device[0] == device || stream_.device[1] == device ) ) {
|
|
if ( device >= devices_.size() ) {
|
|
errorText_ = "RtApiAlsa::getDeviceInfo: device ID was not present before stream was opened.";
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
return devices_[ device ];
|
|
}
|
|
|
|
int openMode = SND_PCM_ASYNC;
|
|
snd_pcm_stream_t stream;
|
|
snd_pcm_info_t *pcminfo;
|
|
snd_pcm_info_alloca( &pcminfo );
|
|
snd_pcm_t *phandle;
|
|
snd_pcm_hw_params_t *params;
|
|
snd_pcm_hw_params_alloca( ¶ms );
|
|
|
|
// First try for playback
|
|
stream = SND_PCM_STREAM_PLAYBACK;
|
|
snd_pcm_info_set_device( pcminfo, subdevice );
|
|
snd_pcm_info_set_subdevice( pcminfo, 0 );
|
|
snd_pcm_info_set_stream( pcminfo, stream );
|
|
|
|
result = snd_ctl_pcm_info( chandle, pcminfo );
|
|
if ( result < 0 ) {
|
|
// Device probably doesn't support playback.
|
|
goto captureProbe;
|
|
}
|
|
|
|
result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK );
|
|
if ( result < 0 ) {
|
|
errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
goto captureProbe;
|
|
}
|
|
|
|
// The device is open ... fill the parameter structure.
|
|
result = snd_pcm_hw_params_any( phandle, params );
|
|
if ( result < 0 ) {
|
|
snd_pcm_close( phandle );
|
|
errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
goto captureProbe;
|
|
}
|
|
|
|
// Get output channel information.
|
|
unsigned int value;
|
|
result = snd_pcm_hw_params_get_channels_max( params, &value );
|
|
if ( result < 0 ) {
|
|
snd_pcm_close( phandle );
|
|
errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") output channels, " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
goto captureProbe;
|
|
}
|
|
info.outputChannels = value;
|
|
snd_pcm_close( phandle );
|
|
|
|
captureProbe:
|
|
// Now try for capture
|
|
stream = SND_PCM_STREAM_CAPTURE;
|
|
snd_pcm_info_set_stream( pcminfo, stream );
|
|
|
|
result = snd_ctl_pcm_info( chandle, pcminfo );
|
|
snd_ctl_close( chandle );
|
|
if ( result < 0 ) {
|
|
// Device probably doesn't support capture.
|
|
if ( info.outputChannels == 0 ) return info;
|
|
goto probeParameters;
|
|
}
|
|
|
|
result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
|
|
if ( result < 0 ) {
|
|
errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
if ( info.outputChannels == 0 ) return info;
|
|
goto probeParameters;
|
|
}
|
|
|
|
// The device is open ... fill the parameter structure.
|
|
result = snd_pcm_hw_params_any( phandle, params );
|
|
if ( result < 0 ) {
|
|
snd_pcm_close( phandle );
|
|
errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
if ( info.outputChannels == 0 ) return info;
|
|
goto probeParameters;
|
|
}
|
|
|
|
result = snd_pcm_hw_params_get_channels_max( params, &value );
|
|
if ( result < 0 ) {
|
|
snd_pcm_close( phandle );
|
|
errorStream_ << "RtApiAlsa::getDeviceInfo: error getting device (" << name << ") input channels, " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
if ( info.outputChannels == 0 ) return info;
|
|
goto probeParameters;
|
|
}
|
|
info.inputChannels = value;
|
|
snd_pcm_close( phandle );
|
|
|
|
// If device opens for both playback and capture, we determine the channels.
|
|
if ( info.outputChannels > 0 && info.inputChannels > 0 )
|
|
info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
|
|
|
|
// ALSA doesn't provide default devices so we'll use the first available one.
|
|
if ( device == 0 && info.outputChannels > 0 )
|
|
info.isDefaultOutput = true;
|
|
if ( device == 0 && info.inputChannels > 0 )
|
|
info.isDefaultInput = true;
|
|
|
|
probeParameters:
|
|
// At this point, we just need to figure out the supported data
|
|
// formats and sample rates. We'll proceed by opening the device in
|
|
// the direction with the maximum number of channels, or playback if
|
|
// they are equal. This might limit our sample rate options, but so
|
|
// be it.
|
|
|
|
if ( info.outputChannels >= info.inputChannels )
|
|
stream = SND_PCM_STREAM_PLAYBACK;
|
|
else
|
|
stream = SND_PCM_STREAM_CAPTURE;
|
|
snd_pcm_info_set_stream( pcminfo, stream );
|
|
|
|
result = snd_pcm_open( &phandle, name, stream, openMode | SND_PCM_NONBLOCK);
|
|
if ( result < 0 ) {
|
|
errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_open error for device (" << name << "), " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
// The device is open ... fill the parameter structure.
|
|
result = snd_pcm_hw_params_any( phandle, params );
|
|
if ( result < 0 ) {
|
|
snd_pcm_close( phandle );
|
|
errorStream_ << "RtApiAlsa::getDeviceInfo: snd_pcm_hw_params error for device (" << name << "), " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
// Test our discrete set of sample rate values.
|
|
info.sampleRates.clear();
|
|
for ( unsigned int i=0; i<MAX_SAMPLE_RATES; i++ ) {
|
|
if ( snd_pcm_hw_params_test_rate( phandle, params, SAMPLE_RATES[i], 0 ) == 0 )
|
|
info.sampleRates.push_back( SAMPLE_RATES[i] );
|
|
}
|
|
if ( info.sampleRates.size() == 0 ) {
|
|
snd_pcm_close( phandle );
|
|
errorStream_ << "RtApiAlsa::getDeviceInfo: no supported sample rates found for device (" << name << ").";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
// Probe the supported data formats ... we don't care about endian-ness just yet
|
|
snd_pcm_format_t format;
|
|
info.nativeFormats = 0;
|
|
format = SND_PCM_FORMAT_S8;
|
|
if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
|
|
info.nativeFormats |= RTAUDIO_SINT8;
|
|
format = SND_PCM_FORMAT_S16;
|
|
if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
|
|
info.nativeFormats |= RTAUDIO_SINT16;
|
|
format = SND_PCM_FORMAT_S24;
|
|
if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
|
|
info.nativeFormats |= RTAUDIO_SINT24;
|
|
format = SND_PCM_FORMAT_S32;
|
|
if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
|
|
info.nativeFormats |= RTAUDIO_SINT32;
|
|
format = SND_PCM_FORMAT_FLOAT;
|
|
if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
|
|
info.nativeFormats |= RTAUDIO_FLOAT32;
|
|
format = SND_PCM_FORMAT_FLOAT64;
|
|
if ( snd_pcm_hw_params_test_format( phandle, params, format ) == 0 )
|
|
info.nativeFormats |= RTAUDIO_FLOAT64;
|
|
|
|
// Check that we have at least one supported format
|
|
if ( info.nativeFormats == 0 ) {
|
|
errorStream_ << "RtApiAlsa::getDeviceInfo: pcm device (" << name << ") data format not supported by RtAudio.";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
// Get the device name
|
|
char *cardname;
|
|
result = snd_card_get_name( card, &cardname );
|
|
if ( result >= 0 )
|
|
sprintf( name, "hw:%s,%d", cardname, subdevice );
|
|
info.name = name;
|
|
|
|
// That's all ... close the device and return
|
|
snd_pcm_close( phandle );
|
|
info.probed = true;
|
|
return info;
|
|
}
|
|
|
|
void RtApiAlsa :: saveDeviceInfo( void )
|
|
{
|
|
devices_.clear();
|
|
|
|
unsigned int nDevices = getDeviceCount();
|
|
devices_.resize( nDevices );
|
|
for ( unsigned int i=0; i<nDevices; i++ )
|
|
devices_[i] = getDeviceInfo( i );
|
|
}
|
|
|
|
bool RtApiAlsa :: probeDeviceOpen( unsigned int device, StreamMode mode, unsigned int channels,
|
|
unsigned int firstChannel, unsigned int sampleRate,
|
|
RtAudioFormat format, unsigned int *bufferSize,
|
|
RtAudio::StreamOptions *options )
|
|
|
|
{
|
|
#if defined(__RTAUDIO_DEBUG__)
|
|
snd_output_t *out;
|
|
snd_output_stdio_attach(&out, stderr, 0);
|
|
#endif
|
|
|
|
// I'm not using the "plug" interface ... too much inconsistent behavior.
|
|
|
|
unsigned nDevices = 0;
|
|
int result, subdevice, card;
|
|
char name[64];
|
|
snd_ctl_t *chandle;
|
|
|
|
// Count cards and devices
|
|
card = -1;
|
|
snd_card_next( &card );
|
|
while ( card >= 0 ) {
|
|
sprintf( name, "hw:%d", card );
|
|
result = snd_ctl_open( &chandle, name, SND_CTL_NONBLOCK );
|
|
if ( result < 0 ) {
|
|
errorStream_ << "RtApiAlsa::probeDeviceOpen: control open, card = " << card << ", " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
subdevice = -1;
|
|
while( 1 ) {
|
|
result = snd_ctl_pcm_next_device( chandle, &subdevice );
|
|
if ( result < 0 ) break;
|
|
if ( subdevice < 0 ) break;
|
|
if ( nDevices == device ) {
|
|
sprintf( name, "hw:%d,%d", card, subdevice );
|
|
snd_ctl_close( chandle );
|
|
goto foundDevice;
|
|
}
|
|
nDevices++;
|
|
}
|
|
snd_ctl_close( chandle );
|
|
snd_card_next( &card );
|
|
}
|
|
|
|
if ( nDevices == 0 ) {
|
|
// This should not happen because a check is made before this function is called.
|
|
errorText_ = "RtApiAlsa::probeDeviceOpen: no devices found!";
|
|
return FAILURE;
|
|
}
|
|
|
|
if ( device >= nDevices ) {
|
|
// This should not happen because a check is made before this function is called.
|
|
errorText_ = "RtApiAlsa::probeDeviceOpen: device ID is invalid!";
|
|
return FAILURE;
|
|
}
|
|
|
|
foundDevice:
|
|
|
|
// The getDeviceInfo() function will not work for a device that is
|
|
// already open. Thus, we'll probe the system before opening a
|
|
// stream and save the results for use by getDeviceInfo().
|
|
if ( mode == OUTPUT || ( mode == INPUT && stream_.mode != OUTPUT ) ) // only do once
|
|
this->saveDeviceInfo();
|
|
|
|
snd_pcm_stream_t stream;
|
|
if ( mode == OUTPUT )
|
|
stream = SND_PCM_STREAM_PLAYBACK;
|
|
else
|
|
stream = SND_PCM_STREAM_CAPTURE;
|
|
|
|
snd_pcm_t *phandle;
|
|
int openMode = SND_PCM_ASYNC;
|
|
result = snd_pcm_open( &phandle, name, stream, openMode );
|
|
if ( result < 0 ) {
|
|
if ( mode == OUTPUT )
|
|
errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for output.";
|
|
else
|
|
errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device (" << name << ") won't open for input.";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Fill the parameter structure.
|
|
snd_pcm_hw_params_t *hw_params;
|
|
snd_pcm_hw_params_alloca( &hw_params );
|
|
result = snd_pcm_hw_params_any( phandle, hw_params );
|
|
if ( result < 0 ) {
|
|
snd_pcm_close( phandle );
|
|
errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") parameters, " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
#if defined(__RTAUDIO_DEBUG__)
|
|
fprintf( stderr, "\nRtApiAlsa: dump hardware params just after device open:\n\n" );
|
|
snd_pcm_hw_params_dump( hw_params, out );
|
|
#endif
|
|
|
|
// Set access ... check user preference.
|
|
if ( options && options->flags & RTAUDIO_NONINTERLEAVED ) {
|
|
stream_.userInterleaved = false;
|
|
result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
|
|
if ( result < 0 ) {
|
|
result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
|
|
stream_.deviceInterleaved[mode] = true;
|
|
}
|
|
else
|
|
stream_.deviceInterleaved[mode] = false;
|
|
}
|
|
else {
|
|
stream_.userInterleaved = true;
|
|
result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
|
|
if ( result < 0 ) {
|
|
result = snd_pcm_hw_params_set_access( phandle, hw_params, SND_PCM_ACCESS_RW_NONINTERLEAVED );
|
|
stream_.deviceInterleaved[mode] = false;
|
|
}
|
|
else
|
|
stream_.deviceInterleaved[mode] = true;
|
|
}
|
|
|
|
if ( result < 0 ) {
|
|
snd_pcm_close( phandle );
|
|
errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") access, " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Determine how to set the device format.
|
|
stream_.userFormat = format;
|
|
snd_pcm_format_t deviceFormat = SND_PCM_FORMAT_UNKNOWN;
|
|
|
|
if ( format == RTAUDIO_SINT8 )
|
|
deviceFormat = SND_PCM_FORMAT_S8;
|
|
else if ( format == RTAUDIO_SINT16 )
|
|
deviceFormat = SND_PCM_FORMAT_S16;
|
|
else if ( format == RTAUDIO_SINT24 )
|
|
deviceFormat = SND_PCM_FORMAT_S24;
|
|
else if ( format == RTAUDIO_SINT32 )
|
|
deviceFormat = SND_PCM_FORMAT_S32;
|
|
else if ( format == RTAUDIO_FLOAT32 )
|
|
deviceFormat = SND_PCM_FORMAT_FLOAT;
|
|
else if ( format == RTAUDIO_FLOAT64 )
|
|
deviceFormat = SND_PCM_FORMAT_FLOAT64;
|
|
|
|
if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat) == 0) {
|
|
stream_.deviceFormat[mode] = format;
|
|
goto setFormat;
|
|
}
|
|
|
|
// The user requested format is not natively supported by the device.
|
|
deviceFormat = SND_PCM_FORMAT_FLOAT64;
|
|
if ( snd_pcm_hw_params_test_format( phandle, hw_params, deviceFormat ) == 0 ) {
|
|
stream_.deviceFormat[mode] = RTAUDIO_FLOAT64;
|
|
goto setFormat;
|
|
}
|
|
|
|
deviceFormat = SND_PCM_FORMAT_FLOAT;
|
|
if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
|
|
stream_.deviceFormat[mode] = RTAUDIO_FLOAT32;
|
|
goto setFormat;
|
|
}
|
|
|
|
deviceFormat = SND_PCM_FORMAT_S32;
|
|
if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
|
|
stream_.deviceFormat[mode] = RTAUDIO_SINT32;
|
|
goto setFormat;
|
|
}
|
|
|
|
deviceFormat = SND_PCM_FORMAT_S24;
|
|
if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
|
|
stream_.deviceFormat[mode] = RTAUDIO_SINT24;
|
|
goto setFormat;
|
|
}
|
|
|
|
deviceFormat = SND_PCM_FORMAT_S16;
|
|
if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
|
|
stream_.deviceFormat[mode] = RTAUDIO_SINT16;
|
|
goto setFormat;
|
|
}
|
|
|
|
deviceFormat = SND_PCM_FORMAT_S8;
|
|
if ( snd_pcm_hw_params_test_format(phandle, hw_params, deviceFormat ) == 0 ) {
|
|
stream_.deviceFormat[mode] = RTAUDIO_SINT8;
|
|
goto setFormat;
|
|
}
|
|
|
|
// If we get here, no supported format was found.
|
|
errorStream_ << "RtApiAlsa::probeDeviceOpen: pcm device " << device << " data format not supported by RtAudio.";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
|
|
setFormat:
|
|
result = snd_pcm_hw_params_set_format( phandle, hw_params, deviceFormat );
|
|
if ( result < 0 ) {
|
|
snd_pcm_close( phandle );
|
|
errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting pcm device (" << name << ") data format, " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Determine whether byte-swaping is necessary.
|
|
stream_.doByteSwap[mode] = false;
|
|
if ( deviceFormat != SND_PCM_FORMAT_S8 ) {
|
|
result = snd_pcm_format_cpu_endian( deviceFormat );
|
|
if ( result == 0 )
|
|
stream_.doByteSwap[mode] = true;
|
|
else if (result < 0) {
|
|
snd_pcm_close( phandle );
|
|
errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting pcm device (" << name << ") endian-ness, " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
}
|
|
|
|
// Set the sample rate.
|
|
result = snd_pcm_hw_params_set_rate_near( phandle, hw_params, (unsigned int*) &sampleRate, 0 );
|
|
if ( result < 0 ) {
|
|
snd_pcm_close( phandle );
|
|
errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting sample rate on device (" << name << "), " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Determine the number of channels for this device. We support a possible
|
|
// minimum device channel number > than the value requested by the user.
|
|
stream_.nUserChannels[mode] = channels;
|
|
unsigned int value;
|
|
result = snd_pcm_hw_params_get_channels_max( hw_params, &value );
|
|
unsigned int deviceChannels = value;
|
|
if ( result < 0 || deviceChannels < channels + firstChannel ) {
|
|
snd_pcm_close( phandle );
|
|
errorStream_ << "RtApiAlsa::probeDeviceOpen: requested channel parameters not supported by device (" << name << "), " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
result = snd_pcm_hw_params_get_channels_min( hw_params, &value );
|
|
if ( result < 0 ) {
|
|
snd_pcm_close( phandle );
|
|
errorStream_ << "RtApiAlsa::probeDeviceOpen: error getting minimum channels for device (" << name << "), " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
deviceChannels = value;
|
|
if ( deviceChannels < channels + firstChannel ) deviceChannels = channels + firstChannel;
|
|
stream_.nDeviceChannels[mode] = deviceChannels;
|
|
|
|
// Set the device channels.
|
|
result = snd_pcm_hw_params_set_channels( phandle, hw_params, deviceChannels );
|
|
if ( result < 0 ) {
|
|
snd_pcm_close( phandle );
|
|
errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting channels for device (" << name << "), " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Set the buffer number, which in ALSA is referred to as the "period".
|
|
int totalSize, dir = 0;
|
|
unsigned int periods = 0;
|
|
if ( options ) periods = options->numberOfBuffers;
|
|
totalSize = *bufferSize * periods;
|
|
|
|
// Set the buffer (or period) size.
|
|
snd_pcm_uframes_t periodSize = *bufferSize;
|
|
result = snd_pcm_hw_params_set_period_size_near( phandle, hw_params, &periodSize, &dir );
|
|
if ( result < 0 ) {
|
|
snd_pcm_close( phandle );
|
|
errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting period size for device (" << name << "), " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
*bufferSize = periodSize;
|
|
|
|
if ( options && options->flags & RTAUDIO_MINIMIZE_LATENCY ) periods = 2;
|
|
else periods = totalSize / *bufferSize;
|
|
// Even though the hardware might allow 1 buffer, it won't work reliably.
|
|
if ( periods < 2 ) periods = 2;
|
|
result = snd_pcm_hw_params_set_periods_near( phandle, hw_params, &periods, &dir );
|
|
if ( result < 0 ) {
|
|
snd_pcm_close( phandle );
|
|
errorStream_ << "RtApiAlsa::probeDeviceOpen: error setting periods for device (" << name << "), " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// If attempting to setup a duplex stream, the bufferSize parameter
|
|
// MUST be the same in both directions!
|
|
if ( stream_.mode == OUTPUT && mode == INPUT && *bufferSize != stream_.bufferSize ) {
|
|
errorStream_ << "RtApiAlsa::probeDeviceOpen: system error setting buffer size for duplex stream on device (" << name << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
stream_.bufferSize = *bufferSize;
|
|
|
|
// Install the hardware configuration
|
|
result = snd_pcm_hw_params( phandle, hw_params );
|
|
if ( result < 0 ) {
|
|
snd_pcm_close( phandle );
|
|
errorStream_ << "RtApiAlsa::probeDeviceOpen: error installing hardware configuration on device (" << name << "), " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
#if defined(__RTAUDIO_DEBUG__)
|
|
fprintf(stderr, "\nRtApiAlsa: dump hardware params after installation:\n\n");
|
|
snd_pcm_hw_params_dump( hw_params, out );
|
|
#endif
|
|
|
|
// Set the software configuration to fill buffers with zeros and prevent device stopping on xruns.
|
|
snd_pcm_sw_params_t *sw_params = NULL;
|
|
snd_pcm_sw_params_alloca( &sw_params );
|
|
snd_pcm_sw_params_current( phandle, sw_params );
|
|
snd_pcm_sw_params_set_start_threshold( phandle, sw_params, *bufferSize );
|
|
snd_pcm_sw_params_set_stop_threshold( phandle, sw_params, 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] )
|
|
stream_.doConvertBuffer[mode] = true;
|
|
if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
|
|
stream_.doConvertBuffer[mode] = true;
|
|
if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
|
|
stream_.nUserChannels[mode] > 1 )
|
|
stream_.doConvertBuffer[mode] = true;
|
|
|
|
// Allocate the ApiHandle if necessary and then save.
|
|
AlsaHandle *apiInfo = 0;
|
|
if ( stream_.apiHandle == 0 ) {
|
|
try {
|
|
apiInfo = (AlsaHandle *) new AlsaHandle;
|
|
}
|
|
catch ( std::bad_alloc& ) {
|
|
errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating AlsaHandle memory.";
|
|
goto error;
|
|
}
|
|
|
|
if ( pthread_cond_init( &apiInfo->runnable, 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;
|
|
|
|
// Allocate necessary internal buffers.
|
|
unsigned long bufferBytes;
|
|
bufferBytes = stream_.nUserChannels[mode] * *bufferSize * formatBytes( stream_.userFormat );
|
|
stream_.userBuffer[mode] = (char *) calloc( bufferBytes, 1 );
|
|
if ( stream_.userBuffer[mode] == NULL ) {
|
|
errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating user buffer memory.";
|
|
goto error;
|
|
}
|
|
|
|
if ( stream_.doConvertBuffer[mode] ) {
|
|
|
|
bool makeBuffer = true;
|
|
bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
|
|
if ( mode == INPUT ) {
|
|
if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
|
|
unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
|
|
if ( bufferBytes <= bytesOut ) makeBuffer = false;
|
|
}
|
|
}
|
|
|
|
if ( makeBuffer ) {
|
|
bufferBytes *= *bufferSize;
|
|
if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
|
|
stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
|
|
if ( stream_.deviceBuffer == NULL ) {
|
|
errorText_ = "RtApiAlsa::probeDeviceOpen: error allocating device buffer memory.";
|
|
goto error;
|
|
}
|
|
}
|
|
}
|
|
|
|
stream_.sampleRate = sampleRate;
|
|
stream_.nBuffers = periods;
|
|
stream_.device[mode] = device;
|
|
stream_.state = STREAM_STOPPED;
|
|
|
|
// Setup the buffer conversion information structure.
|
|
if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
|
|
|
|
// Setup thread if necessary.
|
|
if ( stream_.mode == OUTPUT && mode == INPUT ) {
|
|
// We had already set up an output stream.
|
|
stream_.mode = DUPLEX;
|
|
// Link the streams if possible.
|
|
apiInfo->synchronized = false;
|
|
if ( snd_pcm_link( apiInfo->handles[0], apiInfo->handles[1] ) == 0 )
|
|
apiInfo->synchronized = true;
|
|
else {
|
|
errorText_ = "RtApiAlsa::probeDeviceOpen: unable to synchronize input and output devices.";
|
|
error( RtError::WARNING );
|
|
}
|
|
}
|
|
else {
|
|
stream_.mode = mode;
|
|
|
|
// Setup callback thread.
|
|
stream_.callbackInfo.object = (void *) this;
|
|
|
|
// Set the thread attributes for joinable and realtime scheduling
|
|
// priority (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 (eg: NetBSD 1.6.x with GNU Pthread)
|
|
if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
|
|
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;
|
|
pthread_attr_setschedparam( &attr, ¶m );
|
|
pthread_attr_setschedpolicy( &attr, SCHED_RR );
|
|
}
|
|
else
|
|
pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
|
|
#else
|
|
pthread_attr_setschedpolicy( &attr, SCHED_OTHER );
|
|
#endif
|
|
|
|
stream_.callbackInfo.isRunning = true;
|
|
result = pthread_create( &stream_.callbackInfo.thread, &attr, alsaCallbackHandler, &stream_.callbackInfo );
|
|
pthread_attr_destroy( &attr );
|
|
if ( result ) {
|
|
stream_.callbackInfo.isRunning = false;
|
|
errorText_ = "RtApiAlsa::error creating callback thread!";
|
|
goto error;
|
|
}
|
|
}
|
|
|
|
return SUCCESS;
|
|
|
|
error:
|
|
if ( apiInfo ) {
|
|
pthread_cond_destroy( &apiInfo->runnable );
|
|
if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
|
|
if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
|
|
delete apiInfo;
|
|
stream_.apiHandle = 0;
|
|
}
|
|
|
|
for ( int i=0; i<2; i++ ) {
|
|
if ( stream_.userBuffer[i] ) {
|
|
free( stream_.userBuffer[i] );
|
|
stream_.userBuffer[i] = 0;
|
|
}
|
|
}
|
|
|
|
if ( stream_.deviceBuffer ) {
|
|
free( stream_.deviceBuffer );
|
|
stream_.deviceBuffer = 0;
|
|
}
|
|
|
|
return FAILURE;
|
|
}
|
|
|
|
void RtApiAlsa :: closeStream()
|
|
{
|
|
if ( stream_.state == STREAM_CLOSED ) {
|
|
errorText_ = "RtApiAlsa::closeStream(): no open stream to close!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
|
|
stream_.callbackInfo.isRunning = false;
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
if ( stream_.state == STREAM_STOPPED )
|
|
pthread_cond_signal( &apiInfo->runnable );
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
pthread_join( stream_.callbackInfo.thread, NULL );
|
|
|
|
if ( stream_.state == STREAM_RUNNING ) {
|
|
stream_.state = STREAM_STOPPED;
|
|
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX )
|
|
snd_pcm_drop( apiInfo->handles[0] );
|
|
if ( stream_.mode == INPUT || stream_.mode == DUPLEX )
|
|
snd_pcm_drop( apiInfo->handles[1] );
|
|
}
|
|
|
|
if ( apiInfo ) {
|
|
pthread_cond_destroy( &apiInfo->runnable );
|
|
if ( apiInfo->handles[0] ) snd_pcm_close( apiInfo->handles[0] );
|
|
if ( apiInfo->handles[1] ) snd_pcm_close( apiInfo->handles[1] );
|
|
delete apiInfo;
|
|
stream_.apiHandle = 0;
|
|
}
|
|
|
|
for ( int i=0; i<2; i++ ) {
|
|
if ( stream_.userBuffer[i] ) {
|
|
free( stream_.userBuffer[i] );
|
|
stream_.userBuffer[i] = 0;
|
|
}
|
|
}
|
|
|
|
if ( stream_.deviceBuffer ) {
|
|
free( stream_.deviceBuffer );
|
|
stream_.deviceBuffer = 0;
|
|
}
|
|
|
|
stream_.mode = UNINITIALIZED;
|
|
stream_.state = STREAM_CLOSED;
|
|
}
|
|
|
|
void RtApiAlsa :: startStream()
|
|
{
|
|
// This method calls snd_pcm_prepare if the device isn't already in that state.
|
|
|
|
verifyStream();
|
|
if ( stream_.state == STREAM_RUNNING ) {
|
|
errorText_ = "RtApiAlsa::startStream(): the stream is already running!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
|
|
int result = 0;
|
|
snd_pcm_state_t state;
|
|
AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
|
|
snd_pcm_t **handle = (snd_pcm_t **) apiInfo->handles;
|
|
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
|
|
state = snd_pcm_state( handle[0] );
|
|
if ( state != SND_PCM_STATE_PREPARED ) {
|
|
result = snd_pcm_prepare( handle[0] );
|
|
if ( result < 0 ) {
|
|
errorStream_ << "RtApiAlsa::startStream: error preparing output pcm device, " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
goto unlock;
|
|
}
|
|
}
|
|
}
|
|
|
|
if ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
|
|
state = snd_pcm_state( handle[1] );
|
|
if ( state != SND_PCM_STATE_PREPARED ) {
|
|
result = snd_pcm_prepare( handle[1] );
|
|
if ( result < 0 ) {
|
|
errorStream_ << "RtApiAlsa::startStream: error preparing input pcm device, " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
goto unlock;
|
|
}
|
|
}
|
|
}
|
|
|
|
stream_.state = STREAM_RUNNING;
|
|
|
|
unlock:
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
|
|
pthread_cond_signal( &apiInfo->runnable );
|
|
|
|
if ( result >= 0 ) return;
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
|
|
void RtApiAlsa :: stopStream()
|
|
{
|
|
verifyStream();
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
errorText_ = "RtApiAlsa::stopStream(): the stream is already stopped!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
return;
|
|
}
|
|
|
|
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 ) && !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;
|
|
}
|
|
}
|
|
|
|
unlock:
|
|
stream_.state = STREAM_STOPPED;
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
|
|
if ( result >= 0 ) return;
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
|
|
void RtApiAlsa :: abortStream()
|
|
{
|
|
verifyStream();
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
errorText_ = "RtApiAlsa::abortStream(): the stream is already stopped!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
return;
|
|
}
|
|
|
|
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 ( ( stream_.mode == INPUT || stream_.mode == DUPLEX ) && !apiInfo->synchronized ) {
|
|
result = snd_pcm_drop( handle[1] );
|
|
if ( result < 0 ) {
|
|
errorStream_ << "RtApiAlsa::abortStream: error aborting input pcm device, " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
unlock:
|
|
stream_.state = STREAM_STOPPED;
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
|
|
if ( result >= 0 ) return;
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
|
|
void RtApiAlsa :: callbackEvent()
|
|
{
|
|
AlsaHandle *apiInfo = (AlsaHandle *) stream_.apiHandle;
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
pthread_cond_wait( &apiInfo->runnable, &stream_.mutex );
|
|
if ( stream_.state != STREAM_RUNNING ) {
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
return;
|
|
}
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
}
|
|
|
|
if ( stream_.state == STREAM_CLOSED ) {
|
|
errorText_ = "RtApiAlsa::callbackEvent(): the stream is closed ... this shouldn't happen!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
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 ( doStopStream == 2 ) {
|
|
abortStream();
|
|
return;
|
|
}
|
|
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
|
|
// The state might change while waiting on a mutex.
|
|
if ( stream_.state == STREAM_STOPPED ) goto unlock;
|
|
|
|
int result;
|
|
char *buffer;
|
|
int channels;
|
|
snd_pcm_t **handle;
|
|
snd_pcm_sframes_t frames;
|
|
RtAudioFormat format;
|
|
handle = (snd_pcm_t **) apiInfo->handles;
|
|
|
|
if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
|
|
|
|
// Setup parameters.
|
|
if ( stream_.doConvertBuffer[1] ) {
|
|
buffer = stream_.deviceBuffer;
|
|
channels = stream_.nDeviceChannels[1];
|
|
format = stream_.deviceFormat[1];
|
|
}
|
|
else {
|
|
buffer = stream_.userBuffer[1];
|
|
channels = stream_.nUserChannels[1];
|
|
format = stream_.userFormat;
|
|
}
|
|
|
|
// 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( RtError::WARNING );
|
|
goto tryOutput;
|
|
}
|
|
|
|
// Do byte swapping if necessary.
|
|
if ( stream_.doByteSwap[1] )
|
|
byteSwapBuffer( buffer, stream_.bufferSize * channels, format );
|
|
|
|
// Do buffer conversion if necessary.
|
|
if ( stream_.doConvertBuffer[1] )
|
|
convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
|
|
|
|
// Check stream latency
|
|
result = snd_pcm_delay( handle[1], &frames );
|
|
if ( result == 0 && frames > 0 ) stream_.latency[1] = frames;
|
|
}
|
|
|
|
tryOutput:
|
|
|
|
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;
|
|
}
|
|
|
|
// 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_.deviceInterleaved[0] )
|
|
result = snd_pcm_writei( handle[0], buffer, stream_.bufferSize );
|
|
else {
|
|
void *bufs[channels];
|
|
size_t offset = stream_.bufferSize * formatBytes( format );
|
|
for ( int i=0; i<channels; i++ )
|
|
bufs[i] = (void *) (buffer + (i * offset));
|
|
result = snd_pcm_writen( handle[0], bufs, stream_.bufferSize );
|
|
}
|
|
|
|
if ( result < (int) stream_.bufferSize ) {
|
|
// Either an error or underrun occured.
|
|
if ( result == -EPIPE ) {
|
|
snd_pcm_state_t state = snd_pcm_state( handle[0] );
|
|
if ( state == SND_PCM_STATE_XRUN ) {
|
|
apiInfo->xrun[0] = true;
|
|
result = snd_pcm_prepare( handle[0] );
|
|
if ( result < 0 ) {
|
|
errorStream_ << "RtApiAlsa::callbackEvent: error preparing device after underrun, " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
}
|
|
}
|
|
else {
|
|
errorStream_ << "RtApiAlsa::callbackEvent: error, current state is " << snd_pcm_state_name( state ) << ", " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
}
|
|
}
|
|
else {
|
|
errorStream_ << "RtApiAlsa::callbackEvent: audio write error, " << snd_strerror( result ) << ".";
|
|
errorText_ = errorStream_.str();
|
|
}
|
|
error( RtError::WARNING );
|
|
goto unlock;
|
|
}
|
|
|
|
// Check stream latency
|
|
result = snd_pcm_delay( handle[0], &frames );
|
|
if ( result == 0 && frames > 0 ) stream_.latency[0] = frames;
|
|
}
|
|
|
|
unlock:
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
|
|
RtApi::tickStreamTime();
|
|
if ( doStopStream == 1 ) this->stopStream();
|
|
}
|
|
|
|
extern "C" void *alsaCallbackHandler( void *ptr )
|
|
{
|
|
CallbackInfo *info = (CallbackInfo *) ptr;
|
|
RtApiAlsa *object = (RtApiAlsa *) info->object;
|
|
bool *isRunning = &info->isRunning;
|
|
|
|
while ( *isRunning == true ) {
|
|
pthread_testcancel();
|
|
object->callbackEvent();
|
|
}
|
|
|
|
pthread_exit( NULL );
|
|
}
|
|
|
|
//******************** End of __LINUX_ALSA__ *********************//
|
|
#endif
|
|
|
|
|
|
#if defined(__LINUX_OSS__)
|
|
|
|
#include <unistd.h>
|
|
#include <sys/ioctl.h>
|
|
#include <unistd.h>
|
|
#include <fcntl.h>
|
|
#include "soundcard.h"
|
|
#include <errno.h>
|
|
#include <math.h>
|
|
|
|
extern "C" void *ossCallbackHandler(void * ptr);
|
|
|
|
// 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()
|
|
{
|
|
// Nothing to do here.
|
|
}
|
|
|
|
RtApiOss :: ~RtApiOss()
|
|
{
|
|
if ( stream_.state != STREAM_CLOSED ) closeStream();
|
|
}
|
|
|
|
unsigned int RtApiOss :: getDeviceCount( void )
|
|
{
|
|
int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
|
|
if ( mixerfd == -1 ) {
|
|
errorText_ = "RtApiOss::getDeviceCount: error opening '/dev/mixer'.";
|
|
error( RtError::WARNING );
|
|
return 0;
|
|
}
|
|
|
|
oss_sysinfo sysinfo;
|
|
if ( ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo ) == -1 ) {
|
|
close( mixerfd );
|
|
errorText_ = "RtApiOss::getDeviceCount: error getting sysinfo, OSS version >= 4.0 is required.";
|
|
error( RtError::WARNING );
|
|
return 0;
|
|
}
|
|
|
|
close( mixerfd );
|
|
return sysinfo.numaudios;
|
|
}
|
|
|
|
RtAudio::DeviceInfo RtApiOss :: getDeviceInfo( unsigned int device )
|
|
{
|
|
RtAudio::DeviceInfo info;
|
|
info.probed = false;
|
|
|
|
int mixerfd = open( "/dev/mixer", O_RDWR, 0 );
|
|
if ( mixerfd == -1 ) {
|
|
errorText_ = "RtApiOss::getDeviceInfo: error opening '/dev/mixer'.";
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
oss_sysinfo sysinfo;
|
|
int result = ioctl( mixerfd, SNDCTL_SYSINFO, &sysinfo );
|
|
if ( result == -1 ) {
|
|
close( mixerfd );
|
|
errorText_ = "RtApiOss::getDeviceInfo: error getting sysinfo, OSS version >= 4.0 is required.";
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
unsigned nDevices = sysinfo.numaudios;
|
|
if ( nDevices == 0 ) {
|
|
close( mixerfd );
|
|
errorText_ = "RtApiOss::getDeviceInfo: no devices found!";
|
|
error( RtError::INVALID_USE );
|
|
}
|
|
|
|
if ( device >= nDevices ) {
|
|
close( mixerfd );
|
|
errorText_ = "RtApiOss::getDeviceInfo: device ID is invalid!";
|
|
error( RtError::INVALID_USE );
|
|
}
|
|
|
|
oss_audioinfo ainfo;
|
|
ainfo.dev = device;
|
|
result = ioctl( mixerfd, SNDCTL_AUDIOINFO, &ainfo );
|
|
close( mixerfd );
|
|
if ( result == -1 ) {
|
|
errorStream_ << "RtApiOss::getDeviceInfo: error getting device (" << ainfo.name << ") info.";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
// Probe channels
|
|
if ( ainfo.caps & PCM_CAP_OUTPUT ) info.outputChannels = ainfo.max_channels;
|
|
if ( ainfo.caps & PCM_CAP_INPUT ) info.inputChannels = ainfo.max_channels;
|
|
if ( ainfo.caps & PCM_CAP_DUPLEX ) {
|
|
if ( info.outputChannels > 0 && info.inputChannels > 0 && ainfo.caps & PCM_CAP_DUPLEX )
|
|
info.duplexChannels = (info.outputChannels > info.inputChannels) ? info.inputChannels : info.outputChannels;
|
|
}
|
|
|
|
// Probe data formats ... do for input
|
|
unsigned long mask = ainfo.iformats;
|
|
if ( mask & AFMT_S16_LE || mask & AFMT_S16_BE )
|
|
info.nativeFormats |= RTAUDIO_SINT16;
|
|
if ( mask & AFMT_S8 )
|
|
info.nativeFormats |= RTAUDIO_SINT8;
|
|
if ( mask & AFMT_S32_LE || mask & AFMT_S32_BE )
|
|
info.nativeFormats |= RTAUDIO_SINT32;
|
|
if ( mask & AFMT_FLOAT )
|
|
info.nativeFormats |= RTAUDIO_FLOAT32;
|
|
if ( mask & AFMT_S24_LE || mask & AFMT_S24_BE )
|
|
info.nativeFormats |= RTAUDIO_SINT24;
|
|
|
|
// Check that we have at least one supported format
|
|
if ( info.nativeFormats == 0 ) {
|
|
errorStream_ << "RtApiOss::getDeviceInfo: device (" << ainfo.name << ") data format not supported by RtAudio.";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
return info;
|
|
}
|
|
|
|
// Probe the supported sample rates.
|
|
info.sampleRates.clear();
|
|
if ( ainfo.nrates ) {
|
|
for ( unsigned int i=0; i<ainfo.nrates; i++ ) {
|
|
for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
|
|
if ( ainfo.rates[i] == SAMPLE_RATES[k] ) {
|
|
info.sampleRates.push_back( SAMPLE_RATES[k] );
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
else {
|
|
// Check min and max rate values;
|
|
for ( unsigned int k=0; k<MAX_SAMPLE_RATES; k++ ) {
|
|
if ( ainfo.min_rate <= (int) SAMPLE_RATES[k] && ainfo.max_rate >= (int) SAMPLE_RATES[k] )
|
|
info.sampleRates.push_back( SAMPLE_RATES[k] );
|
|
}
|
|
}
|
|
|
|
if ( info.sampleRates.size() == 0 ) {
|
|
errorStream_ << "RtApiOss::getDeviceInfo: no supported sample rates found for device (" << ainfo.name << ").";
|
|
errorText_ = errorStream_.str();
|
|
error( RtError::WARNING );
|
|
}
|
|
else {
|
|
info.probed = true;
|
|
info.name = ainfo.name;
|
|
}
|
|
|
|
return info;
|
|
}
|
|
|
|
|
|
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;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
// 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;
|
|
}
|
|
|
|
int flags = 0;
|
|
OssHandle *handle = (OssHandle *) stream_.apiHandle;
|
|
if ( mode == OUTPUT )
|
|
flags |= O_WRONLY;
|
|
else { // mode == INPUT
|
|
if (stream_.mode == OUTPUT && stream_.device[0] == device) {
|
|
// We just set the same device for playback ... close and reopen for duplex (OSS only).
|
|
close( handle->id[0] );
|
|
handle->id[0] = 0;
|
|
if ( !( ainfo.caps & PCM_CAP_DUPLEX ) ) {
|
|
errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support duplex mode.";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
// Check that the number previously set channels is the same.
|
|
if ( stream_.nUserChannels[0] != channels ) {
|
|
errorStream_ << "RtApiOss::probeDeviceOpen: input/output channels must be equal for OSS duplex device (" << ainfo.name << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
flags |= O_RDWR;
|
|
}
|
|
else
|
|
flags |= O_RDONLY;
|
|
}
|
|
|
|
// 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;
|
|
}
|
|
|
|
// 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;
|
|
}
|
|
|
|
// Set the number of channels.
|
|
int deviceChannels = channels + firstChannel;
|
|
result = ioctl( fd, SNDCTL_DSP_CHANNELS, &deviceChannels );
|
|
if ( result == -1 || deviceChannels < (int)(channels + firstChannel) ) {
|
|
close( fd );
|
|
errorStream_ << "RtApiOss::probeDeviceOpen: error setting channel parameters on device (" << ainfo.name << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
stream_.nDeviceChannels[mode] = deviceChannels;
|
|
|
|
// Get 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;
|
|
}
|
|
|
|
// Determine how to set the device format.
|
|
stream_.userFormat = format;
|
|
int deviceFormat = -1;
|
|
stream_.doByteSwap[mode] = false;
|
|
if ( format == RTAUDIO_SINT8 ) {
|
|
if ( mask & AFMT_S8 ) {
|
|
deviceFormat = AFMT_S8;
|
|
stream_.deviceFormat[mode] = RTAUDIO_SINT8;
|
|
}
|
|
}
|
|
else if ( format == RTAUDIO_SINT16 ) {
|
|
if ( mask & AFMT_S16_NE ) {
|
|
deviceFormat = AFMT_S16_NE;
|
|
stream_.deviceFormat[mode] = RTAUDIO_SINT16;
|
|
}
|
|
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;
|
|
}
|
|
}
|
|
|
|
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;
|
|
|
|
// Save buffer size (in sample frames).
|
|
*bufferSize = ossBufferBytes / ( formatBytes(stream_.deviceFormat[mode]) * deviceChannels );
|
|
stream_.bufferSize = *bufferSize;
|
|
|
|
// Set the sample rate.
|
|
int srate = sampleRate;
|
|
result = ioctl( fd, SNDCTL_DSP_SPEED, &srate );
|
|
if ( result == -1 ) {
|
|
close( fd );
|
|
errorStream_ << "RtApiOss::probeDeviceOpen: error setting sample rate (" << sampleRate << ") on device (" << ainfo.name << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
|
|
// Verify the sample rate setup worked.
|
|
if ( abs( srate - sampleRate ) > 100 ) {
|
|
close( fd );
|
|
errorStream_ << "RtApiOss::probeDeviceOpen: device (" << ainfo.name << ") does not support sample rate (" << sampleRate << ").";
|
|
errorText_ = errorStream_.str();
|
|
return FAILURE;
|
|
}
|
|
stream_.sampleRate = sampleRate;
|
|
|
|
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;
|
|
}
|
|
|
|
// 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] )
|
|
stream_.doConvertBuffer[mode] = true;
|
|
if ( stream_.nUserChannels[mode] < stream_.nDeviceChannels[mode] )
|
|
stream_.doConvertBuffer[mode] = true;
|
|
if ( stream_.userInterleaved != stream_.deviceInterleaved[mode] &&
|
|
stream_.nUserChannels[mode] > 1 )
|
|
stream_.doConvertBuffer[mode] = true;
|
|
|
|
// Allocate the stream handles if necessary and then save.
|
|
if ( stream_.apiHandle == 0 ) {
|
|
try {
|
|
handle = new OssHandle;
|
|
}
|
|
catch ( std::bad_alloc& ) {
|
|
errorText_ = "RtApiOss::probeDeviceOpen: error allocating OssHandle memory.";
|
|
goto error;
|
|
}
|
|
|
|
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] ) {
|
|
|
|
bool makeBuffer = true;
|
|
bufferBytes = stream_.nDeviceChannels[mode] * formatBytes( stream_.deviceFormat[mode] );
|
|
if ( mode == INPUT ) {
|
|
if ( stream_.mode == OUTPUT && stream_.deviceBuffer ) {
|
|
unsigned long bytesOut = stream_.nDeviceChannels[0] * formatBytes( stream_.deviceFormat[0] );
|
|
if ( bufferBytes <= bytesOut ) makeBuffer = false;
|
|
}
|
|
}
|
|
|
|
if ( makeBuffer ) {
|
|
bufferBytes *= *bufferSize;
|
|
if ( stream_.deviceBuffer ) free( stream_.deviceBuffer );
|
|
stream_.deviceBuffer = (char *) calloc( bufferBytes, 1 );
|
|
if ( stream_.deviceBuffer == NULL ) {
|
|
errorText_ = "RtApiOss::probeDeviceOpen: error allocating device buffer memory.";
|
|
goto error;
|
|
}
|
|
}
|
|
}
|
|
|
|
stream_.device[mode] = device;
|
|
stream_.state = STREAM_STOPPED;
|
|
|
|
// Setup the buffer conversion information structure.
|
|
if ( stream_.doConvertBuffer[mode] ) setConvertInfo( mode, firstChannel );
|
|
|
|
// Setup thread if necessary.
|
|
if ( stream_.mode == OUTPUT && mode == INPUT ) {
|
|
// We had already set up an output stream.
|
|
stream_.mode = DUPLEX;
|
|
if ( stream_.device[0] == device ) handle->id[0] = fd;
|
|
}
|
|
else {
|
|
stream_.mode = mode;
|
|
|
|
// Setup callback thread.
|
|
stream_.callbackInfo.object = (void *) this;
|
|
|
|
// Set the thread attributes for joinable and realtime scheduling
|
|
// priority. The higher priority will only take affect if the
|
|
// program is run as root or suid.
|
|
pthread_attr_t attr;
|
|
pthread_attr_init( &attr );
|
|
pthread_attr_setdetachstate( &attr, PTHREAD_CREATE_JOINABLE );
|
|
#ifdef SCHED_RR // Undefined with some OSes (eg: NetBSD 1.6.x with GNU Pthread)
|
|
if ( options && options->flags & RTAUDIO_SCHEDULE_REALTIME ) {
|
|
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;
|
|
pthread_attr_setschedparam( &attr, ¶m );
|
|
pthread_attr_setschedpolicy( &attr, SCHED_RR );
|
|
}
|
|
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 ) {
|
|
stream_.callbackInfo.isRunning = false;
|
|
errorText_ = "RtApiOss::error creating callback thread!";
|
|
goto error;
|
|
}
|
|
}
|
|
|
|
return SUCCESS;
|
|
|
|
error:
|
|
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;
|
|
}
|
|
|
|
for ( int i=0; i<2; i++ ) {
|
|
if ( stream_.userBuffer[i] ) {
|
|
free( stream_.userBuffer[i] );
|
|
stream_.userBuffer[i] = 0;
|
|
}
|
|
}
|
|
|
|
if ( stream_.deviceBuffer ) {
|
|
free( stream_.deviceBuffer );
|
|
stream_.deviceBuffer = 0;
|
|
}
|
|
|
|
return FAILURE;
|
|
}
|
|
|
|
void RtApiOss :: closeStream()
|
|
{
|
|
if ( stream_.state == STREAM_CLOSED ) {
|
|
errorText_ = "RtApiOss::closeStream(): no open stream to close!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
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_.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 ) {
|
|
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;
|
|
}
|
|
|
|
for ( int i=0; i<2; i++ ) {
|
|
if ( stream_.userBuffer[i] ) {
|
|
free( stream_.userBuffer[i] );
|
|
stream_.userBuffer[i] = 0;
|
|
}
|
|
}
|
|
|
|
if ( stream_.deviceBuffer ) {
|
|
free( stream_.deviceBuffer );
|
|
stream_.deviceBuffer = 0;
|
|
}
|
|
|
|
stream_.mode = UNINITIALIZED;
|
|
stream_.state = STREAM_CLOSED;
|
|
}
|
|
|
|
void RtApiOss :: startStream()
|
|
{
|
|
verifyStream();
|
|
if ( stream_.state == STREAM_RUNNING ) {
|
|
errorText_ = "RtApiOss::startStream(): the stream is already running!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
|
|
stream_.state = STREAM_RUNNING;
|
|
|
|
// 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 RtApiOss :: stopStream()
|
|
{
|
|
verifyStream();
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
errorText_ = "RtApiOss::stopStream(): the stream is already stopped!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
|
|
// The state might change while waiting on a mutex.
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
return;
|
|
}
|
|
|
|
int result = 0;
|
|
OssHandle *handle = (OssHandle *) stream_.apiHandle;
|
|
if ( stream_.mode == OUTPUT || stream_.mode == DUPLEX ) {
|
|
|
|
// Flush the output with zeros a few times.
|
|
char *buffer;
|
|
int samples;
|
|
RtAudioFormat format;
|
|
|
|
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( RtError::WARNING );
|
|
}
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
if ( stream_.mode == INPUT || ( stream_.mode == DUPLEX && handle->id[0] != handle->id[1] ) ) {
|
|
result = ioctl( handle->id[1], SNDCTL_DSP_HALT, 0 );
|
|
if ( result == -1 ) {
|
|
errorStream_ << "RtApiOss::stopStream: system error stopping input callback procedure on device (" << stream_.device[0] << ").";
|
|
errorText_ = errorStream_.str();
|
|
goto unlock;
|
|
}
|
|
}
|
|
|
|
unlock:
|
|
stream_.state = STREAM_STOPPED;
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
|
|
if ( result != -1 ) return;
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
|
|
void RtApiOss :: abortStream()
|
|
{
|
|
verifyStream();
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
errorText_ = "RtApiOss::abortStream(): the stream is already stopped!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
MUTEX_LOCK( &stream_.mutex );
|
|
|
|
// The state might change while waiting on a mutex.
|
|
if ( stream_.state == STREAM_STOPPED ) {
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
return;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
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;
|
|
}
|
|
}
|
|
|
|
unlock:
|
|
stream_.state = STREAM_STOPPED;
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
|
|
if ( result != -1 ) return;
|
|
error( RtError::SYSTEM_ERROR );
|
|
}
|
|
|
|
void RtApiOss :: callbackEvent()
|
|
{
|
|
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 );
|
|
}
|
|
|
|
if ( stream_.state == STREAM_CLOSED ) {
|
|
errorText_ = "RtApiOss::callbackEvent(): the stream is closed ... this shouldn't happen!";
|
|
error( RtError::WARNING );
|
|
return;
|
|
}
|
|
|
|
// 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 );
|
|
|
|
// The state might change while waiting on a mutex.
|
|
if ( stream_.state == STREAM_STOPPED ) goto unlock;
|
|
|
|
int result;
|
|
char *buffer;
|
|
int samples;
|
|
RtAudioFormat format;
|
|
|
|
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] );
|
|
samples = stream_.bufferSize * stream_.nDeviceChannels[0];
|
|
format = stream_.deviceFormat[0];
|
|
}
|
|
else {
|
|
buffer = stream_.userBuffer[0];
|
|
samples = stream_.bufferSize * stream_.nUserChannels[0];
|
|
format = stream_.userFormat;
|
|
}
|
|
|
|
// Do byte swapping if necessary.
|
|
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) );
|
|
|
|
if ( result == -1 ) {
|
|
// We'll assume this is an underrun, though there isn't a
|
|
// specific means for determining that.
|
|
handle->xrun[0] = true;
|
|
errorText_ = "RtApiOss::callbackEvent: audio write error.";
|
|
error( RtError::WARNING );
|
|
// Continue on to input section.
|
|
}
|
|
}
|
|
|
|
if ( stream_.mode == INPUT || stream_.mode == DUPLEX ) {
|
|
|
|
// Setup parameters.
|
|
if ( stream_.doConvertBuffer[1] ) {
|
|
buffer = stream_.deviceBuffer;
|
|
samples = stream_.bufferSize * stream_.nDeviceChannels[1];
|
|
format = stream_.deviceFormat[1];
|
|
}
|
|
else {
|
|
buffer = stream_.userBuffer[1];
|
|
samples = stream_.bufferSize * stream_.nUserChannels[1];
|
|
format = stream_.userFormat;
|
|
}
|
|
|
|
// Read samples from device.
|
|
result = read( handle->id[1], buffer, samples * formatBytes(format) );
|
|
|
|
if ( result == -1 ) {
|
|
// We'll assume this is an overrun, though there isn't a
|
|
// specific means for determining that.
|
|
handle->xrun[1] = true;
|
|
errorText_ = "RtApiOss::callbackEvent: audio read error.";
|
|
error( RtError::WARNING );
|
|
goto unlock;
|
|
}
|
|
|
|
// Do byte swapping if necessary.
|
|
if ( stream_.doByteSwap[1] )
|
|
byteSwapBuffer( buffer, samples, format );
|
|
|
|
// Do buffer conversion if necessary.
|
|
if ( stream_.doConvertBuffer[1] )
|
|
convertBuffer( stream_.userBuffer[1], stream_.deviceBuffer, stream_.convertInfo[1] );
|
|
}
|
|
|
|
unlock:
|
|
MUTEX_UNLOCK( &stream_.mutex );
|
|
|
|
RtApi::tickStreamTime();
|
|
if ( doStopStream == 1 ) this->stopStream();
|
|
}
|
|
|
|
extern "C" void *ossCallbackHandler( void *ptr )
|
|
{
|
|
CallbackInfo *info = (CallbackInfo *) ptr;
|
|
RtApiOss *object = (RtApiOss *) info->object;
|
|
bool *isRunning = &info->isRunning;
|
|
|
|
while ( *isRunning == true ) {
|
|
pthread_testcancel();
|
|
object->callbackEvent();
|
|
}
|
|
|
|
pthread_exit( NULL );
|
|
}
|
|
|
|
//******************** End of __LINUX_OSS__ *********************//
|
|
#endif
|
|
|
|
|
|
// *************************************************** //
|
|
//
|
|
// Protected common (OS-independent) RtAudio methods.
|
|
//
|
|
// *************************************************** //
|
|
|
|
// This method can be modified to control the behavior of error
|
|
// message printing.
|
|
void RtApi :: error( RtError::Type type )
|
|
{
|
|
errorStream_.str(""); // clear the ostringstream
|
|
if ( type == RtError::WARNING && showWarnings_ == true )
|
|
std::cerr << '\n' << errorText_ << "\n\n";
|
|
else
|
|
throw( RtError( errorText_, type ) );
|
|
}
|
|
|
|
void RtApi :: verifyStream()
|
|
{
|
|
if ( stream_.state == STREAM_CLOSED ) {
|
|
errorText_ = "RtApi:: a stream is not open!";
|
|
error( RtError::INVALID_USE );
|
|
}
|
|
}
|
|
|
|
void RtApi :: clearStreamInfo()
|
|
{
|
|
stream_.mode = UNINITIALIZED;
|
|
stream_.state = STREAM_CLOSED;
|
|
stream_.sampleRate = 0;
|
|
stream_.bufferSize = 0;
|
|
stream_.nBuffers = 0;
|
|
stream_.userFormat = 0;
|
|
stream_.userInterleaved = true;
|
|
stream_.streamTime = 0.0;
|
|
stream_.apiHandle = 0;
|
|
stream_.deviceBuffer = 0;
|
|
stream_.callbackInfo.callback = 0;
|
|
stream_.callbackInfo.userData = 0;
|
|
stream_.callbackInfo.isRunning = false;
|
|
for ( int i=0; i<2; i++ ) {
|
|
stream_.device[i] = 11111;
|
|
stream_.doConvertBuffer[i] = false;
|
|
stream_.deviceInterleaved[i] = true;
|
|
stream_.doByteSwap[i] = false;
|
|
stream_.nUserChannels[i] = 0;
|
|
stream_.nDeviceChannels[i] = 0;
|
|
stream_.channelOffset[i] = 0;
|
|
stream_.deviceFormat[i] = 0;
|
|
stream_.latency[i] = 0;
|
|
stream_.userBuffer[i] = 0;
|
|
stream_.convertInfo[i].channels = 0;
|
|
stream_.convertInfo[i].inJump = 0;
|
|
stream_.convertInfo[i].outJump = 0;
|
|
stream_.convertInfo[i].inFormat = 0;
|
|
stream_.convertInfo[i].outFormat = 0;
|
|
stream_.convertInfo[i].inOffset.clear();
|
|
stream_.convertInfo[i].outOffset.clear();
|
|
}
|
|
}
|
|
|
|
unsigned int RtApi :: formatBytes( RtAudioFormat format )
|
|
{
|
|
if ( format == RTAUDIO_SINT16 )
|
|
return 2;
|
|
else if ( format == RTAUDIO_SINT24 || format == RTAUDIO_SINT32 ||
|
|
format == RTAUDIO_FLOAT32 )
|
|
return 4;
|
|
else if ( format == RTAUDIO_FLOAT64 )
|
|
return 8;
|
|
else if ( format == RTAUDIO_SINT8 )
|
|
return 1;
|
|
|
|
errorText_ = "RtApi::formatBytes: undefined format.";
|
|
error( RtError::WARNING );
|
|
|
|
return 0;
|
|
}
|
|
|
|
void RtApi :: setConvertInfo( StreamMode mode, unsigned int firstChannel )
|
|
{
|
|
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
|
|
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];
|
|
}
|
|
|
|
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 { // 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;
|
|
}
|
|
}
|
|
}
|
|
|
|
// Add channel offset.
|
|
if ( firstChannel > 0 ) {
|
|
if ( stream_.deviceInterleaved[mode] ) {
|
|
if ( mode == OUTPUT ) {
|
|
for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
|
|
stream_.convertInfo[mode].outOffset[k] += firstChannel;
|
|
}
|
|
else {
|
|
for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
|
|
stream_.convertInfo[mode].inOffset[k] += firstChannel;
|
|
}
|
|
}
|
|
else {
|
|
if ( mode == OUTPUT ) {
|
|
for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
|
|
stream_.convertInfo[mode].outOffset[k] += ( firstChannel * stream_.bufferSize );
|
|
}
|
|
else {
|
|
for ( int k=0; k<stream_.convertInfo[mode].channels; k++ )
|
|
stream_.convertInfo[mode].inOffset[k] += ( firstChannel * stream_.bufferSize );
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
void RtApi :: convertBuffer( char *outBuffer, char *inBuffer, ConvertInfo &info )
|
|
{
|
|
// This function does format conversion, input/output channel compensation, and
|
|
// data interleaving/deinterleaving. 24-bit integers are assumed to occupy
|
|
// the upper three bytes of a 32-bit integer.
|
|
|
|
// 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 ) );
|
|
|
|
int j;
|
|
if (info.outFormat == RTAUDIO_FLOAT64) {
|
|
Float64 scale;
|
|
Float64 *out = (Float64 *)outBuffer;
|
|
|
|
if (info.inFormat == RTAUDIO_SINT8) {
|
|
signed char *in = (signed char *)inBuffer;
|
|
scale = 1.0 / 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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
else if (info.inFormat == RTAUDIO_SINT24) {
|
|
Int32 *in = (Int32 *)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]] & 0x00ffffff);
|
|
out[info.outOffset[j]] += 0.5;
|
|
out[info.outOffset[j]] *= scale;
|
|
}
|
|
in += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
else if (info.inFormat == RTAUDIO_FLOAT64) {
|
|
// Channel compensation and/or (de)interleaving only.
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
}
|
|
else if (info.outFormat == RTAUDIO_FLOAT32) {
|
|
Float32 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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
else if (info.inFormat == RTAUDIO_SINT24) {
|
|
Int32 *in = (Int32 *)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]] & 0x00ffffff);
|
|
out[info.outOffset[j]] += 0.5;
|
|
out[info.outOffset[j]] *= scale;
|
|
}
|
|
in += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
else if (info.inFormat == RTAUDIO_FLOAT32) {
|
|
// Channel compensation and/or (de)interleaving only.
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
else if (info.inFormat == RTAUDIO_SINT24) {
|
|
Int32 *in = (Int32 *)inBuffer;
|
|
for (unsigned int i=0; i<stream_.bufferSize; i++) {
|
|
for (j=0; j<info.channels; j++) {
|
|
out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
|
|
out[info.outOffset[j]] <<= 8;
|
|
}
|
|
in += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
else if (info.inFormat == RTAUDIO_SINT32) {
|
|
// Channel compensation and/or (de)interleaving only.
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
}
|
|
else if (info.outFormat == RTAUDIO_SINT24) {
|
|
Int32 *out = (Int32 *)outBuffer;
|
|
if (info.inFormat == RTAUDIO_SINT8) {
|
|
signed char *in = (signed char *)inBuffer;
|
|
for (unsigned int i=0; i<stream_.bufferSize; i++) {
|
|
for (j=0; j<info.channels; j++) {
|
|
out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
|
|
out[info.outOffset[j]] <<= 16;
|
|
}
|
|
in += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
else if (info.inFormat == RTAUDIO_SINT16) {
|
|
Int16 *in = (Int16 *)inBuffer;
|
|
for (unsigned int i=0; i<stream_.bufferSize; i++) {
|
|
for (j=0; j<info.channels; j++) {
|
|
out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
|
|
out[info.outOffset[j]] <<= 8;
|
|
}
|
|
in += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
else if (info.inFormat == RTAUDIO_SINT24) {
|
|
// Channel compensation and/or (de)interleaving only.
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
else if (info.inFormat == RTAUDIO_SINT32) {
|
|
Int32 *in = (Int32 *)inBuffer;
|
|
for (unsigned int i=0; i<stream_.bufferSize; i++) {
|
|
for (j=0; j<info.channels; j++) {
|
|
out[info.outOffset[j]] = (Int32) in[info.inOffset[j]];
|
|
out[info.outOffset[j]] >>= 8;
|
|
}
|
|
in += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
else if (info.inFormat == RTAUDIO_SINT16) {
|
|
// Channel compensation and/or (de)interleaving only.
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
else if (info.inFormat == RTAUDIO_SINT24) {
|
|
Int32 *in = (Int32 *)inBuffer;
|
|
for (unsigned int i=0; i<stream_.bufferSize; i++) {
|
|
for (j=0; j<info.channels; j++) {
|
|
out[info.outOffset[j]] = (Int16) ((in[info.inOffset[j]] >> 8) & 0x0000ffff);
|
|
}
|
|
in += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
}
|
|
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 *)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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
else if (info.inFormat == RTAUDIO_SINT24) {
|
|
Int32 *in = (Int32 *)inBuffer;
|
|
for (unsigned int i=0; i<stream_.bufferSize; i++) {
|
|
for (j=0; j<info.channels; j++) {
|
|
out[info.outOffset[j]] = (signed char) ((in[info.inOffset[j]] >> 16) & 0x000000ff);
|
|
}
|
|
in += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
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 += info.inJump;
|
|
out += info.outJump;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
//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;
|
|
|
|
ptr = buffer;
|
|
if ( format == RTAUDIO_SINT16 ) {
|
|
for ( unsigned int i=0; i<samples; i++ ) {
|
|
// Swap 1st and 2nd bytes.
|
|
val = *(ptr);
|
|
*(ptr) = *(ptr+1);
|
|
*(ptr+1) = val;
|
|
|
|
// Increment 2 bytes.
|
|
ptr += 2;
|
|
}
|
|
}
|
|
else if ( format == RTAUDIO_SINT24 ||
|
|
format == RTAUDIO_SINT32 ||
|
|
format == RTAUDIO_FLOAT32 ) {
|
|
for ( unsigned int i=0; i<samples; i++ ) {
|
|
// Swap 1st and 4th bytes.
|
|
val = *(ptr);
|
|
*(ptr) = *(ptr+3);
|
|
*(ptr+3) = val;
|
|
|
|
// Swap 2nd and 3rd bytes.
|
|
ptr += 1;
|
|
val = *(ptr);
|
|
*(ptr) = *(ptr+1);
|
|
*(ptr+1) = val;
|
|
|
|
// Increment 3 more bytes.
|
|
ptr += 3;
|
|
}
|
|
}
|
|
else if ( format == RTAUDIO_FLOAT64 ) {
|
|
for ( unsigned int i=0; i<samples; i++ ) {
|
|
// Swap 1st and 8th bytes
|
|
val = *(ptr);
|
|
*(ptr) = *(ptr+7);
|
|
*(ptr+7) = val;
|
|
|
|
// Swap 2nd and 7th bytes
|
|
ptr += 1;
|
|
val = *(ptr);
|
|
*(ptr) = *(ptr+5);
|
|
*(ptr+5) = val;
|
|
|
|
// Swap 3rd and 6th bytes
|
|
ptr += 1;
|
|
val = *(ptr);
|
|
*(ptr) = *(ptr+3);
|
|
*(ptr+3) = val;
|
|
|
|
// Swap 4th and 5th bytes
|
|
ptr += 1;
|
|
val = *(ptr);
|
|
*(ptr) = *(ptr+1);
|
|
*(ptr+1) = val;
|
|
|
|
// 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
|
|
|
|
#endif
|