diff --git a/src/audio/audiolayer.cpp b/src/audio/audiolayer.cpp
deleted file mode 100644
index d799939335b1e22cda5456a85532f8bd12885c79..0000000000000000000000000000000000000000
--- a/src/audio/audiolayer.cpp
+++ /dev/null
@@ -1,664 +0,0 @@
-/*
- * Copyright (C) 2008 Savoir-Faire Linux inc.
- * Author: Yan Morin <yan.morin@savoirfairelinux.com>
- * Author: Jerome Oufella <jerome.oufella@savoirfairelinux.com>
- * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3 of the License, or
- * (at your option) any later version.
- *
- * This program is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
- */
-
-#include <cstdio>
-#include <cstdlib>
-
-#include "audiolayer.h"
-#include "../global.h"
-#include "../manager.h"
-#include "../user_cfg.h"
-
-#define PCM_PAUSE 1
-#define PCM_RESUME 0
-
-#ifdef SFL_TEST_SINE
-#include <cmath>
-#endif
-
- AudioLayer::AudioLayer(ManagerImpl* manager)
- : _defaultVolume(100)
- , _manager(manager)
- , _PlaybackHandle( NULL )
- , _CaptureHandle( NULL )
- , deviceClosed( true )
- , _urgentBuffer( SIZEBUF )
-{
-
- _inChannel = 1; // don't put in stereo
- _outChannel = 1; // don't put in stereo
- _echoTesting = false;
-
-}
-
-// Destructor
-AudioLayer::~AudioLayer (void)
-{
- _debugAlsa("Close ALSA streams\n");
- closeCaptureStream();
- closePlaybackStream();
- deviceClosed = true;
-}
-
-
- bool
-AudioLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize, int stream , std::string plugin)
-{
-
- if(deviceClosed == false)
- {
- if( stream == SFL_PCM_CAPTURE )
- closeCaptureStream();
- else if( stream == SFL_PCM_PLAYBACK)
- closePlaybackStream();
- else
- {
- closeCaptureStream();
- closePlaybackStream();
- }
- }
-
- _indexIn = indexIn;
- _indexOut = indexOut;
- _sampleRate = sampleRate;
- _frameSize = frameSize;
- _audioPlugin = plugin;
-
- _debugAlsa(" Setting audiolayer: device in=%2d, out=%2d\n", _indexIn, _indexOut);
- _debugAlsa(" : alsa plugin=%s\n", _audioPlugin.c_str());
- _debugAlsa(" : nb channel in=%2d, out=%2d\n", _inChannel, _outChannel);
- _debugAlsa(" : sample rate=%5d, format=%s\n", _sampleRate, SFLDataFormatString);
-
- ost::MutexLock lock( _mutex );
-
- std::string pcmp = buildDeviceTopo( plugin , indexOut , 0);
- std::string pcmc = buildDeviceTopo( PCM_PLUGHW , indexIn , 0);
- return open_device( pcmp , pcmc , stream);
-}
-
- void
-AudioLayer::startStream(void)
-{
- if( _CaptureHandle && _PlaybackHandle )
- {
- _talk = true ;
- _debugAlsa(" Start stream\n");
- int err;
- //ost::MutexLock lock( _mutex );
- snd_pcm_prepare( _CaptureHandle );
- snd_pcm_start( _CaptureHandle ) ;
-
- snd_pcm_prepare( _PlaybackHandle );
- if( err = snd_pcm_start( _PlaybackHandle) < 0 ) _debugAlsa(" Cannot start (%s)\n", snd_strerror(err));
- }
-}
-
- void
-AudioLayer::stopStream(void)
-{
- if( _CaptureHandle && _PlaybackHandle )
- {
- //ost::MutexLock lock( _mutex );
- _debugAlsa(" Stop Stream\n ");
- _talk = false;
- snd_pcm_drop( _CaptureHandle );
- snd_pcm_prepare( _CaptureHandle );
- snd_pcm_drop( _PlaybackHandle );
- snd_pcm_prepare( _PlaybackHandle );
- _urgentBuffer.flush();
- }
-}
-
-void AudioLayer::AlsaCallBack( snd_async_handler_t* pcm_callback )
-{
- ( ( AudioLayer *)snd_async_handler_get_callback_private( pcm_callback )) -> playTones();
-}
-
- void
-AudioLayer::fillHWBuffer( void)
-{
- unsigned char* data;
- int pcmreturn, l1, l2;
- short s1, s2;
- int periodSize = 128 ;
- int frames = periodSize >> 2 ;
- _debug("frames = %d\n");
-
- data = (unsigned char*)malloc(periodSize);
- for(l1 = 0; l1 < 100; l1++) {
- for(l2 = 0; l2 < frames; l2++) {
- s1 = 0;
- s2 = 0;
- data[4*l2] = (unsigned char)s1;
- data[4*l2+1] = s1 >> 8;
- data[4*l2+2] = (unsigned char)s2;
- data[4*l2+3] = s2 >> 8;
- }
- while ((pcmreturn = snd_pcm_mmap_writei(_PlaybackHandle, data, frames)) < 0) {
- snd_pcm_prepare(_PlaybackHandle);
- //_debugAlsa("< Buffer Underrun >\n");
- }
- }
-}
-
- bool
-AudioLayer::isStreamActive (void)
-{
- ost::MutexLock lock( _mutex );
- return (isPlaybackActive() && isCaptureActive());
-}
-
-
- int
-AudioLayer::playSamples(void* buffer, int toCopy, bool isTalking)
-{
- //ost::MutexLock lock( _mutex );
- if( isTalking )
- _talk = true;
- if ( _PlaybackHandle ){
- write( adjustVolume( buffer , toCopy , SFL_PCM_PLAYBACK ) , toCopy );
- }
- return 0;
-}
-
- int
-AudioLayer::putUrgent(void* buffer, int toCopy)
-{
- if ( _PlaybackHandle ){
- fillHWBuffer();
- int a = _urgentBuffer.AvailForPut();
- if( a >= toCopy ){
- return _urgentBuffer.Put( buffer , toCopy , _defaultVolume );
- } else {
- return _urgentBuffer.Put( buffer , a , _defaultVolume ) ;
- }
- }
- return 0;
-}
-
- int
-AudioLayer::canGetMic()
-{
- int avail;
- if ( _CaptureHandle ) {
- avail = snd_pcm_avail_update( _CaptureHandle );
- //printf("%d\n", avail );
- if(avail > 0)
- return avail;
- else
- return 0;
- }
- else
- return 0;
-}
-
- int
-AudioLayer::getMic(void *buffer, int toCopy)
-{
- int res = 0 ;
- if( _CaptureHandle )
- {
- res = read( buffer, toCopy );
- adjustVolume( buffer , toCopy , SFL_PCM_CAPTURE );
- }
- return res ;
-}
-
-
- bool
-AudioLayer::isStreamStopped (void)
-{
- ost::MutexLock lock( _mutex );
- return !(isStreamActive());
-}
-
-void
-AudioLayer::toggleEchoTesting() {
- ost::MutexLock lock( _mutex );
- _echoTesting = (_echoTesting == true) ? false : true;
-}
-
-
-//////////////////////////////////////////////////////////////////////////////////////////////
-///////////////// ALSA PRIVATE FUNCTIONS ////////////////////////////////////////////////
-////////////////////////////////////////////////////////////////////////////////////////////
-
-
-
- void
-AudioLayer::playTones( void )
-{
- int frames = _periodSize ;
- int maxBytes = frames * sizeof(SFLDataFormat) ;
- SFLDataFormat* out = (SFLDataFormat*)malloc(maxBytes * sizeof(SFLDataFormat));
- if( _talk ) {}
- else {
- AudioLoop *tone = _manager -> getTelephoneTone();
- int spkrVol = _manager -> getSpkrVolume();
- if( tone != 0 ){
- tone -> getNext( out , frames , spkrVol );
- write( out , maxBytes );
- }
- else if( ( tone=_manager->getTelephoneFile() ) != 0 ){
- tone ->getNext( out , frames , spkrVol );
- write( out , maxBytes );
- }
- }
- // free the temporary data buffer
- free( out ); out = 0;
-}
-
-bool
-AudioLayer::isPlaybackActive(void) {
- ost::MutexLock guard( _mutex );
- if( _PlaybackHandle )
- return (snd_pcm_state(_PlaybackHandle) == SND_PCM_STATE_RUNNING ? true : false);
- else
- return false;
-}
-
-bool
-AudioLayer::isCaptureActive(void) {
- ost::MutexLock guard( _mutex );
- if( _CaptureHandle )
- return (snd_pcm_state( _CaptureHandle) == SND_PCM_STATE_RUNNING ? true : false);
- else
- return false;
-}
-
-
- bool
-AudioLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
-{
- std::stringstream errMsg;
- int err;
- snd_pcm_hw_params_t* hwParams = NULL;
- snd_pcm_sw_params_t *swparams = NULL;
- unsigned int rate_in = getSampleRate();
- unsigned int rate_out = getSampleRate();
- int dir = 0;
- snd_pcm_uframes_t period_size_in = getFrameSize() * getSampleRate() / 1000 * 2 ;
- snd_pcm_uframes_t buffer_size_in = period_size_in * 4 ;
- snd_pcm_uframes_t threshold = 1024 ;
- snd_pcm_uframes_t period_size_out = getFrameSize() * getSampleRate() / 1000 * 2;//1024 ;
- snd_pcm_uframes_t buffer_size_out = period_size_out * 4 ;
-
- unsigned int buffer_time = 80000; //80ms
- unsigned int period_time = buffer_time / 4 ; //20ms
-
- if(flag == SFL_PCM_BOTH || flag == SFL_PCM_CAPTURE)
- {
- _debugAlsa("Opening capture device %s\n", pcm_c.c_str());
- if(err = snd_pcm_open(&_CaptureHandle, pcm_c.c_str(), SND_PCM_STREAM_CAPTURE, 0) < 0){
- _debugAlsa("Error while opening capture device %s\n", pcm_c.c_str());
- setErrorMessage( ALSA_CAPTURE_DEVICE );
- return false;
- }
-
- if( err = snd_pcm_hw_params_malloc( &hwParams ) < 0 ) {
- _debugAlsa(" Cannot allocate hardware parameter structure (%s)\n", snd_strerror(err));
- return false;
- }
- if( err = snd_pcm_hw_params_any(_CaptureHandle, hwParams) < 0) _debugAlsa(" Cannot initialize hardware parameter structure (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_access( _CaptureHandle, hwParams, SND_PCM_ACCESS_MMAP_INTERLEAVED) < 0) _debugAlsa(" Cannot set access type (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_format( _CaptureHandle, hwParams, SND_PCM_FORMAT_S16_LE) < 0) _debugAlsa(" Cannot set sample format (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_rate_near( _CaptureHandle, hwParams, &rate_in, &dir) < 0) _debugAlsa(" Cannot set sample rate (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_channels( _CaptureHandle, hwParams, 1) < 0) _debugAlsa(" Cannot set channel count (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_period_time_near( _CaptureHandle, hwParams, &period_time , &dir) < 0) _debugAlsa(" Cannot set period time (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_buffer_time_near( _CaptureHandle, hwParams, &buffer_time , &dir) < 0) _debugAlsa(" Cannot set buffer time (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_get_period_size( hwParams, &period_size_in , &dir) < 0) _debugAlsa(" Cannot get period size (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_get_buffer_size( hwParams, &buffer_size_in ) < 0) _debugAlsa(" Cannot get buffer size (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params( _CaptureHandle, hwParams ) < 0) _debugAlsa(" Cannot set hw parameters (%s)\n", snd_strerror(err));
- _debug("buffer size = %d\n" , period_size_in);
- _debug("period size = %d\n" , buffer_size_in);
- snd_pcm_hw_params_free( hwParams );
-
- snd_pcm_uframes_t val ;
- snd_pcm_sw_params_malloc( &swparams );
- snd_pcm_sw_params_current( _CaptureHandle, swparams );
-
- if( err = snd_pcm_sw_params_set_start_threshold( _CaptureHandle, swparams, period_size_out) < 0 ) _debugAlsa(" Cannot set start threshold (%s)\n", snd_strerror(err));
- snd_pcm_sw_params_get_start_threshold( swparams , &val);
- _debug("Start threshold = %d\n" ,val);
- if( err = snd_pcm_sw_params_set_avail_min( _CaptureHandle, swparams, period_size_out) < 0) _debugAlsa(" Cannot set min avail (%s)\n" , snd_strerror(err));
- snd_pcm_sw_params_get_avail_min( swparams , &val);
- _debug("Min available = %d\n" ,val);
- if( err = snd_pcm_sw_params( _CaptureHandle, swparams ) < 0 ) _debugAlsa(" Cannot set sw parameters (%s)\n", snd_strerror(err));
- snd_pcm_sw_params_free( swparams );
- deviceClosed = false;
- }
-
- if(flag == SFL_PCM_BOTH || flag == SFL_PCM_PLAYBACK)
- {
-
- _debugAlsa(" Opening playback device %s\n", pcm_p.c_str());
- if(err = snd_pcm_open(&_PlaybackHandle, pcm_p.c_str(), SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK ) < 0){
- _debugAlsa("Error while opening playback device %s\n", pcm_c.c_str());
- setErrorMessage( ALSA_PLAYBACK_DEVICE );
- return false;
- }
- if( err = snd_pcm_hw_params_malloc( &hwParams ) < 0 ) {
- _debugAlsa(" Cannot allocate hardware parameter structure (%s)\n", snd_strerror(err));
- return false;
- }
- if( err = snd_pcm_hw_params_any( _PlaybackHandle,hwParams) < 0) _debugAlsa(" Cannot initialize hardware parameter structure (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_access( _PlaybackHandle, hwParams, SND_PCM_ACCESS_MMAP_INTERLEAVED) < 0) _debugAlsa(" Cannot set access type (%s)\n", snd_strerror(err));
- //if( err = snd_pcm_hw_params_set_access( _PlaybackHandle, hwParams, SND_PCM_ACCESS_RW_INTERLEAVED) < 0) _debugAlsa(" Cannot set access type (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_format( _PlaybackHandle, hwParams, SND_PCM_FORMAT_S16_LE) < 0) _debugAlsa(" Cannot set sample format (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_rate( _PlaybackHandle, hwParams, rate_out, dir) < 0) _debugAlsa(" Cannot set sample rate (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_channels( _PlaybackHandle, hwParams, 1) < 0) _debugAlsa(" Cannot set channel count (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_buffer_time_near( _PlaybackHandle, hwParams, &buffer_time , &dir) < 0) _debugAlsa(" Cannot set buffer time (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_set_period_time_near( _PlaybackHandle, hwParams, &period_time , &dir) < 0) _debugAlsa(" Cannot set period time (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_get_period_size( hwParams, &_periodSize , &dir) < 0) _debugAlsa(" Cannot get period size (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params_get_buffer_size( hwParams, &buffer_size_out ) < 0) _debugAlsa(" Cannot get buffer size (%s)\n", snd_strerror(err));
- if( err = snd_pcm_hw_params( _PlaybackHandle, hwParams ) < 0) _debugAlsa(" Cannot set hw parameters (%s)\n", snd_strerror(err));
-
-
- snd_pcm_hw_params_free( hwParams );
-
- snd_pcm_uframes_t val ;
- snd_pcm_sw_params_malloc( &swparams );
- snd_pcm_sw_params_current( _PlaybackHandle, swparams );
-
- if( err = snd_pcm_sw_params_set_start_threshold( _PlaybackHandle, swparams, period_size_out) < 0 ) _debugAlsa(" Cannot set start threshold (%s)\n", snd_strerror(err));
- snd_pcm_sw_params_get_start_threshold( swparams , &val);
- _debug("Start threshold = %d\n" ,val);
- //if( err = snd_pcm_sw_params_set_stop_threshold( _PlaybackHandle, swparams, buffer_size_out ) < 0 ) _debugAlsa(" Cannot set stop threshold (%s)\n", snd_strerror(err));
- //snd_pcm_sw_params_get_stop_threshold( swparams , &val);
- //_debug("Stop threshold = %d\n" ,val);
- if( err = snd_pcm_sw_params_set_avail_min( _PlaybackHandle, swparams, period_size_out) < 0) _debugAlsa(" Cannot set min avail (%s)\n" , snd_strerror(err));
- if( err = snd_pcm_sw_params_set_xfer_align( _PlaybackHandle, swparams, 1) < 0) _debugAlsa(" Cannot set xfer align (%s)\n" , snd_strerror(err));
- snd_pcm_sw_params_get_avail_min( swparams , &val);
- _debug("Min available = %d\n" ,val);
- //if( err = snd_pcm_sw_params_set_silence_threshold( _PlaybackHandle, swparams, period_size_out) < 0) _debugAlsa(" Cannot set silence threshold (%s)\n" , snd_strerror(err));
- //snd_pcm_sw_params_get_silence_threshold( swparams , &val);
- // _debug("Silence threshold = %d\n" ,val);
- if( err = snd_pcm_sw_params( _PlaybackHandle, swparams ) < 0 ) _debugAlsa(" Cannot set sw parameters (%s)\n", snd_strerror(err));
- snd_pcm_sw_params_free( swparams );
-
- if ( err = snd_async_add_pcm_handler( &_AsyncHandler, _PlaybackHandle , AlsaCallBack, this ) < 0) _debugAlsa(" Unable to install the async callback handler (%s)\n", snd_strerror(err));
- deviceClosed = false;
- }
- //fillHWBuffer();
-
- _talk = false;
- return true;
-}
-
-//TODO first frame causes broken pipe (underrun) because not enough data are send --> make the handle wait to be ready
- int
-AudioLayer::write(void* buffer, int length)
-{
- //if(snd_pcm_state( _PlaybackHandle ) == SND_PCM_STATE_XRUN)
- //handle_xrun_playback();
- //_debugAlsa("avail = %d - toWrite = %d\n" , snd_pcm_avail_update( _PlaybackHandle ) , length / 2);
-
- snd_pcm_uframes_t frames = snd_pcm_bytes_to_frames( _PlaybackHandle, length);
- int err = snd_pcm_mmap_writei( _PlaybackHandle , buffer , frames );
- switch(err) {
- case -EAGAIN:
- _debugAlsa(" (%s)\n", snd_strerror( err ));
- snd_pcm_resume( _PlaybackHandle );
- break;
- case -EPIPE:
- _debugAlsa(" UNDERRUN (%s)\n", snd_strerror(err));
- handle_xrun_playback();
- snd_pcm_mmap_writei( _PlaybackHandle , buffer , frames );
- break;
- case -ESTRPIPE:
- _debugAlsa(" (%s)\n", snd_strerror(err));
- snd_pcm_resume( _PlaybackHandle );
- break;
- case -EBADFD:
- _debugAlsa(" (%s)\n", snd_strerror( err ));
- break;
- }
-
- if( err >=0 && err < frames )
- _debugAlsa("Short write : %d out of %d\n", err , frames);
-
- return ( err > 0 )? err : 0 ;
-}
-
- int
-AudioLayer::read( void* buffer, int toCopy)
-{
- if(deviceClosed || _CaptureHandle == NULL)
- return 0;
- int err;
- if(snd_pcm_state( _CaptureHandle ) == SND_PCM_STATE_XRUN)
- {
- snd_pcm_prepare( _CaptureHandle );
- snd_pcm_start( _CaptureHandle );
- }
- snd_pcm_uframes_t frames = snd_pcm_bytes_to_frames( _CaptureHandle, toCopy );
- if( err = snd_pcm_mmap_readi( _CaptureHandle, buffer, frames) < 0 ) {
- switch(err){
- case EPERM:
- _debugAlsa(" Capture EPERM (%s)\n", snd_strerror(err));
- snd_pcm_prepare( _CaptureHandle);
- snd_pcm_start( _CaptureHandle );
- break;
- case -ESTRPIPE:
- _debugAlsa(" Capture ESTRPIPE (%s)\n", snd_strerror(err));
- snd_pcm_resume( _CaptureHandle);
- break;
- case -EAGAIN:
- _debugAlsa(" Capture EAGAIN (%s)\n", snd_strerror(err));
- break;
- case -EBADFD:
- _debugAlsa(" Capture EBADFD (%s)\n", snd_strerror(err));
- break;
- case -EPIPE:
- _debugAlsa(" Capture EPIPE (%s)\n", snd_strerror(err));
- handle_xrun_capture();
- break;
- }
- return 0;
- }
-
- return toCopy;
-
-}
-
- void
-AudioLayer::handle_xrun_capture( void )
-{
- snd_pcm_status_t* status;
- snd_pcm_status_alloca( &status );
-
- int res = snd_pcm_status( _CaptureHandle, status );
- if( res <= 0){
- if(snd_pcm_status_get_state(status) == SND_PCM_STATE_XRUN ){
- snd_pcm_drop( _CaptureHandle );
- snd_pcm_prepare( _CaptureHandle );
- snd_pcm_start( _CaptureHandle );
- }
- }
- else
- _debugAlsa(" Get status failed\n");
-}
-
- void
-AudioLayer::handle_xrun_playback( void )
-{
- int state;
- snd_pcm_status_t* status;
- snd_pcm_status_alloca( &status );
-
- if( state = snd_pcm_status( _PlaybackHandle, status ) < 0 ) _debugAlsa(" Error: Cannot get playback handle status (%s)\n" , snd_strerror( state ) );
- else
- {
- state = snd_pcm_status_get_state( status );
- if( state == SND_PCM_STATE_XRUN )
- {
- snd_pcm_drop( _PlaybackHandle );
- snd_pcm_prepare( _PlaybackHandle );
- //snd_pcm_start( _PlaybackHandle );
- }
- }
-}
-
- std::string
-AudioLayer::buildDeviceTopo( std::string plugin, int card, int subdevice )
-{
- std::string pcm = plugin;
- std::stringstream ss,ss1;
- if( pcm == "default" || pcm == "pulse")
- return pcm;
- ss << card;
- pcm.append(":");
- pcm.append(ss.str());
- if( subdevice != 0 ){
- pcm.append(",");
- ss1 << subdevice;
- pcm.append(ss1.str());
- }
- return pcm;
-}
-
- std::vector<std::string>
-AudioLayer::getSoundCardsInfo( int stream )
-{
- std::vector<std::string> cards_id;
- HwIDPair p;
-
- snd_ctl_t* handle;
- snd_ctl_card_info_t *info;
- snd_pcm_info_t* pcminfo;
- snd_ctl_card_info_alloca( &info );
- snd_pcm_info_alloca( &pcminfo );
-
- int numCard = -1 ;
- int err;
- std::string description;
-
- if(snd_card_next( &numCard ) < 0 || numCard < 0)
- return cards_id;
-
- while(numCard >= 0){
- std::stringstream ss;
- ss << numCard;
- std::string name= "hw:";
- name.append(ss.str());
-
- if( snd_ctl_open( &handle, name.c_str(), 0) == 0 ){
- if( snd_ctl_card_info( handle, info) == 0){
- snd_pcm_info_set_device( pcminfo , 0);
- snd_pcm_info_set_stream( pcminfo, ( stream == SFL_PCM_CAPTURE )? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK );
- if( snd_ctl_pcm_info ( handle ,pcminfo ) < 0) _debugAlsa(" Cannot get info\n");
- else{
- _debugAlsa("card %i : %s [%s]\n",
- numCard,
- snd_ctl_card_info_get_id(info),
- snd_ctl_card_info_get_name( info ));
- description = snd_ctl_card_info_get_name( info );
- description.append(" - ");
- description.append(snd_pcm_info_get_name( pcminfo ));
- cards_id.push_back( description );
- // The number of the sound card is associated with a string description
- p = HwIDPair( numCard , description );
- IDSoundCards.push_back( p );
- }
- }
- snd_ctl_close( handle );
- }
- if ( snd_card_next( &numCard ) < 0 ) {
- break;
- }
- }
- return cards_id;
-}
-
- void
-AudioLayer::closeCaptureStream( void)
-{
- if(_CaptureHandle){
- snd_pcm_drop( _CaptureHandle );
- snd_pcm_close( _CaptureHandle );
- _CaptureHandle = 0;
- }
-}
-
- void
-AudioLayer::closePlaybackStream( void)
-{
- if(_PlaybackHandle){
- snd_pcm_drop( _PlaybackHandle );
- snd_pcm_close( _PlaybackHandle );
- _PlaybackHandle = 0;
- }
-}
-
- bool
-AudioLayer::soundCardIndexExist( int card , int stream )
-{
- snd_ctl_t* handle;
- snd_pcm_info_t *pcminfo;
- snd_pcm_info_alloca( &pcminfo );
- std::string name = "hw:";
- std::stringstream ss;
- ss << card ;
- name.append(ss.str());
- if(snd_ctl_open( &handle, name.c_str(), 0) == 0 ){
- snd_pcm_info_set_stream( pcminfo , ( stream == SFL_PCM_PLAYBACK )? SND_PCM_STREAM_PLAYBACK : SND_PCM_STREAM_CAPTURE );
- if( snd_ctl_pcm_info( handle , pcminfo ) < 0) return false;
- else
- return true;
- }
- else
- return false;
-}
-
- int
-AudioLayer::soundCardGetIndex( std::string description )
-{
- int i;
- for( i = 0 ; i < IDSoundCards.size() ; i++ )
- {
- HwIDPair p = IDSoundCards[i];
- if( p.second == description )
- return p.first ;
- }
- // else return the default one
- return 0;
-}
-
- void*
-AudioLayer::adjustVolume( void* buffer , int len, int stream )
-{
- int vol;
- if( stream == SFL_PCM_PLAYBACK )
- vol = _manager->getSpkrVolume();
- else
- vol = _manager->getMicVolume();
-
- SFLDataFormat* src = (SFLDataFormat*) buffer;
- if( vol != 100 )
- {
- int size = len / sizeof(SFLDataFormat);
- int i;
- for( i = 0 ; i < size ; i++ ){
- src[i] = src[i] * vol / 100 ;
- }
- }
- return src ;
-}
diff --git a/src/audio/audiolayer.h b/src/audio/audiolayer.h
index af585c00afb4e63928988d53e373211a20ae2195..a0083f99d85038994c0ec31096319abd11dfec0b 100644
--- a/src/audio/audiolayer.h
+++ b/src/audio/audiolayer.h
@@ -129,6 +129,8 @@ class AudioLayer {
*/
virtual int putUrgent(void* buffer, int toCopy) = 0;
+ virtual int putMain( void* buffer, int toCopy) = 0;
+
/**
* Query the capture device for number of bytes available in the hardware ring buffer
* @return int The number of bytes available
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index e98458bdc36af9fdd2098d8ffe81bb465eb94dcf..513b29db5814b09c303411ff0593f7aae070c6ba 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -377,8 +377,13 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
toAudioLayer = _dataAudioLayer;
#endif
-
- audiolayer->playSamples(toAudioLayer, nbSample * sizeof(SFLDataFormat), true);
+ // int layer = audiolayer->getLayerType();
+ // _debug(" interface %i\n" , layer);
+//#if CHECK_INTERFACE( layer , ALSA )
+ // audiolayer->playSamples(toAudioLayer, nbSample * sizeof(SFLDataFormat), true);
+//#else
+ audiolayer->putMain( toAudioLayer, nbSample * sizeof(SFLDataFormat) );
+//#endif
// Notify (with a beep) an incoming call when there is already a call
countTime += time->getSecond();
if (Manager::instance().incomingCallWaiting() > 0) {
diff --git a/src/audio/pulselayer.cpp b/src/audio/pulselayer.cpp
index db76eb4586e4551aa975a0d98efb98673528046c..6e1be72dc20e747842d1b9640b7d5723d51c069b 100644
--- a/src/audio/pulselayer.cpp
+++ b/src/audio/pulselayer.cpp
@@ -23,67 +23,85 @@ static pa_context *context = NULL;
static pa_stream* playback = NULL;
static pa_stream* record = NULL;
static pa_mainloop_api *mainloop_api = NULL;
-static pa_sample_spec sample_spec;
+static pa_threaded_mainloop *m = NULL;
+
+static pa_sample_spec sample_spec ;
+
static pa_channel_map channel_map;
-PulseLayer::PulseLayer(ManagerImpl* manager)
+static std::string pcm_p, pcm_r;
+
+int framesPerBuffer = 882;
+
+ PulseLayer::PulseLayer(ManagerImpl* manager)
: AudioLayer( manager , PULSEAUDIO )
+ , _mainSndRingBuffer( SIZEBUF )
{
_debug("Pulse audio constructor: Create context\n");
- create_context();
}
// Destructor
PulseLayer::~PulseLayer (void)
{
assert(mainloop_api);
- mainloop_api->quit( mainloop_api, 0 );
- // pa_stream_flush();
- // pa_stream_disconnect();
+ //mainloop_api->quit( mainloop_api, 0 );
+ pa_stream_flush( playback , NULL, NULL);
+ pa_stream_disconnect( playback );
+ pa_context_disconnect(context);
}
-void
-PulseLayer::create_context( void )
+ void
+PulseLayer::connectPulseServer( void )
{
pa_context_flags_t flag = PA_CONTEXT_NOAUTOSPAWN ;
int ret = 1;
- // Instanciate a mainloop object
- pa_mainloop *m;
- if(!( m=pa_mainloop_new()))
- _debug("Error while creating the mainloop\n");
- mainloop_api = pa_mainloop_get_api( m );
+ pa_threaded_mainloop_lock( m );
- // Instanciate a context
- if( !(context = pa_context_new( mainloop_api , "SFLphone" )))
- _debug("Error while creating the context\n");
- pa_context_ref( context );
+ _debug("Connect the context to the server\n");
+ pa_context_connect( context, NULL , flag , NULL );
- pa_context_set_state_callback(context, context_state_callback, NULL);
+ pa_context_set_state_callback(context, context_state_callback, this);
+ pa_threaded_mainloop_wait( m );
- pa_context_connect( context, NULL , flag , NULL );
-
// Run the main loop
- pa_mainloop_run(m , &ret);
+ if( pa_context_get_state( context ) != PA_CONTEXT_READY ){
+ _debug("Error connecting to pulse audio server\n");
+ pa_threaded_mainloop_unlock( m );
+ }
+
+ pa_threaded_mainloop_unlock( m );
_debug("Context creation done\n");
}
-void
+ void
PulseLayer::stream_state_callback( pa_stream* s, void* user_data )
{
_debug("The state of the stream changed\n");
+ assert(s);
+ switch(pa_stream_get_state(s)){
+ case PA_STREAM_CREATING:
+ case PA_STREAM_TERMINATED:
+ _debug("Stream is creating...\n");
+ break;
+ case PA_STREAM_READY:
+ _debug("Stream successfully created\n");
+ break;
+ case PA_STREAM_FAILED:
+ default:
+ _debug("Stream error: %s\n" , pa_strerror(pa_context_errno(pa_stream_get_context(s))));
+ break;
+ }
}
-void
+ void
PulseLayer::context_state_callback( pa_context* c, void* user_data )
{
_debug("The state of the context changed\n");
- sample_spec.format = PA_SAMPLE_S16LE;
- sample_spec.channels = 1;
- channel_map.channels = 1;
- pa_stream_flags_t flag = PA_STREAM_START_CORKED ;
- assert(c);
+ PulseLayer* pulse = (PulseLayer*)user_data;
+ //pa_stream_flags_t flag = PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE ;
+ assert(c && m);
switch(pa_context_get_state(c)){
case PA_CONTEXT_CONNECTING:
case PA_CONTEXT_AUTHORIZING:
@@ -93,20 +111,8 @@ PulseLayer::context_state_callback( pa_context* c, void* user_data )
case PA_CONTEXT_READY:
pa_cvolume cv;
assert(c && !playback && !record);
+ pulse->createStreams( c );
_debug("Connection to PulseAudio server established\n");
- playback = pa_stream_new( c, "playback stream" , &sample_spec, &channel_map);
- record = pa_stream_new( c, "capture stream" , &sample_spec, &channel_map);
-
- assert(playback);
- assert(record);
-
- // Set up the parameters required to open a (Callback)Stream:
-
- pa_stream_set_state_callback(playback, stream_state_callback, NULL);
- // Transferring Data - Asynchronous Mode
- pa_stream_set_write_callback(playback, audioCallback, NULL);
- pa_stream_connect_playback( playback, NULL , NULL , flag , NULL, NULL );
-
break;
case PA_CONTEXT_TERMINATED:
_debug("Context terminated\n");
@@ -118,70 +124,91 @@ PulseLayer::context_state_callback( pa_context* c, void* user_data )
}
}
+ void
+PulseLayer::createStreams( pa_context* c )
+{
+ _debug( " Create Audio Streams \n ");
+ //pa_stream_flags_t flag = PA_STREAM_AUTO_TIMING_UPDATE;
+ pa_stream_flags_t flag = PA_STREAM_FIX_RATE;
+ sample_spec.format = PA_SAMPLE_S16LE;
+ sample_spec.rate = 44100;
+ sample_spec.channels = 1;
+ channel_map.channels = 1;
+
+ if( !pa_sample_spec_valid( &sample_spec ) ){
+ _debug("Invalid sample specifications\n");
+ exit(0);
+ }
+
+ if(!(playback = pa_stream_new( c, "SFLphone out" , &sample_spec, &channel_map))){
+ _debug("Playback: pa_stream_new() failed : %s\n" , pa_strerror( pa_context_errno( c)));
+ exit(0);
+ }
+
+ if(!(record = pa_stream_new( c, "SFLphone Mic" , &sample_spec, &channel_map))){
+ _debug("Capture: pa_stream_new() failed : %s\n" , pa_strerror( pa_context_errno( c)));
+ exit(0);
+ }
+
+ assert(playback);
+ assert(record);
+ assert(m);
+
+ // Set up the parameters required to open a (Callback)Stream:
+
+ pa_stream_connect_playback( playback, NULL , NULL , flag , NULL, NULL );
+ pa_stream_set_state_callback(playback, stream_state_callback, NULL);
+ // Transferring Data - Asynchronous Mode
+ pa_stream_set_write_callback(playback, audioCallback, this);
- bool
+ pa_threaded_mainloop_signal(m , 0);
+
+ pa_stream_set_state_callback(record, stream_state_callback, NULL);
+ // Transferring Data - Asynchronous Mode
+ pa_stream_set_read_callback(record, audioCallback, this);
+ pa_stream_connect_record( record, NULL , NULL , flag );
+
+}
+
+ bool
PulseLayer::openDevice(int indexIn, int indexOut, int sampleRate, int frameSize , int stream, std::string plugin)
{
_sampleRate = sampleRate;
_frameSize = frameSize;
- //sample_spec.rate = sampleRate;
- //sample_spec.format = PA_SAMPLE_S16LE;
- //sample_spec.channels = 1;
- //channel_map.channels = 1;
- //pa_stream_flags_t flag = PA_STREAM_START_CORKED ;
-
_debug(" Setting PulseLayer: device in=%2d, out=%2d\n", indexIn, indexOut);
_debug(" : nb channel in=%2d, out=%2d\n", _inChannel, _outChannel);
_debug(" : sample rate=%5d\n", _sampleRate );
- /*
- assert(context);
- switch(pa_context_get_state(context)){
- case PA_CONTEXT_CONNECTING:
- case PA_CONTEXT_AUTHORIZING:
- case PA_CONTEXT_SETTING_NAME:
- _debug("Waiting....\n");
- break;
- case PA_CONTEXT_READY:
- pa_cvolume cv;
- assert(context && !playback && !record);
- _debug("Connection to PulseAudio server established\n");
- playback = pa_stream_new( context, "playback stream" , &sample_spec, &channel_map);
- record = pa_stream_new( context, "capture stream" , &sample_spec, &channel_map);
+ m = pa_threaded_mainloop_new();
+ assert(m);
- assert(playback);
- assert(record);
+ if( pa_threaded_mainloop_start( m ) < 0 ){
+ _debug("Failed starting the mainloop\n");
+ }
- // Set up the parameters required to open a (Callback)Stream:
+ // Instanciate a context
+ if( !(context = pa_context_new( pa_threaded_mainloop_get_api( m ) , "SFLphone" )))
+ _debug("Error while creating the context\n");
- pa_stream_set_state_callback(playback, stream_state_callback, NULL);
- // Transferring Data - Asynchronous Mode
- pa_stream_set_write_callback(playback, audioCallback, NULL);
- pa_stream_connect_playback( playback, NULL , NULL , flag , NULL, NULL );
+ assert(context);
- break;
- case PA_CONTEXT_TERMINATED:
- _debug("Context terminated\n");
- break;
- case PA_CONTEXT_FAILED:
- default:
- _debug(" Error : %s" , pa_strerror(pa_context_errno(context)));
- exit(1);
- }*/
+ connectPulseServer();
+
+ _debug("Connection Done!! \n");
}
-void
+ void
PulseLayer::closeCaptureStream( void )
{
}
-void
+ void
PulseLayer::closePlaybackStream( void )
{
}
-int
+ int
PulseLayer::playSamples(void* buffer, int toCopy, bool isTalking)
{
}
@@ -189,6 +216,15 @@ PulseLayer::playSamples(void* buffer, int toCopy, bool isTalking)
int
PulseLayer::putMain(void* buffer, int toCopy)
{
+ int a = _mainSndRingBuffer.AvailForPut();
+ if ( a >= toCopy ) {
+ return _mainSndRingBuffer.Put(buffer, toCopy, 100);
+ } else {
+ _debug("Chopping sound, Ouch! RingBuffer full ?\n");
+ return _mainSndRingBuffer.Put(buffer, a, 100);
+ }
+
+ return 0;
}
void
@@ -204,11 +240,13 @@ PulseLayer::putUrgent(void* buffer, int toCopy)
int
PulseLayer::canGetMic()
{
+ return 882;
}
int
PulseLayer::getMic(void *buffer, int toCopy)
{
+ return 160;
}
void
@@ -221,29 +259,89 @@ PulseLayer::isStreamStopped (void)
{
}
- void
+ void
PulseLayer::startStream (void)
{
_debug("Start stream\n");
+ //pa_stream_cork( playback , 0, NULL, NULL);
}
- void
+ void
PulseLayer::stopStream (void)
{
_debug("Stop stream\n");
+ pa_stream_drop( playback );
}
- bool
+ bool
PulseLayer::isStreamActive (void)
{
}
-
void
-PulseLayer::audioCallback ( pa_stream* s, size_t bytes, void* user_data )
+PulseLayer::audioCallback ( pa_stream* s, size_t bytes, void* userdata )
{
- _debug("Audio callback: New data may be written to the stream\n");
+ pa_threaded_mainloop_signal( m , 0);
+
+ assert( s && bytes );
+
+ PulseLayer* pulse = (PulseLayer*) userdata;
+ pulse->write();
+ //if(pa_stream_get_state(s) == PA_STREAM_READY )
+ //{
+ //if( bytes > 0 ){
+ // pa_stream_write( s, user_data, bytes, pa_xfree, 0, PA_SEEK_RELATIVE );
+ //}
+ //}
// pa_stream_write
- // pa_stream_peek ( to read the next fragment from the buffer ) / pa_stream_drop( to remove the data from the buffer )
+ // // pa_stream_peek ( to read the next fragment from the buffer ) / pa_stream_drop( to remove the data from the buffer )
+ //
+ // int toPut;
+ // int urgentAvail; // number of data right and data left
+ // int micAvailPut;
+ //
+ // // AvailForGet tell the number of chars inside the buffer
+ // // framePerBuffer are the number of data for one channel (left)
+ // urgentAvail = _urgentRingBuffer.AvailForGet();
+ // if (urgentAvail > 0) {
+ // // Urgent data (dtmf, incoming call signal) come first.
+ // toGet = (urgentAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? urgentAvail : framesPerBuffer * sizeof(SFLDataFormat);
+ // _urgentRingBuffer.Get(out, toGet, spkrVolume);
+ // // Consume the regular one as well (same amount of bytes)
+ // _mainSndRingBuffer.Discard(toGet);
+ // } else {
+ // AudioLoop* tone = _manager->getTelephoneTone();
+ // if ( tone != 0) {
+ // tone->getNext(out, framesPerBuffer, spkrVolume);
+ // } else if ( (tone=_manager->getTelephoneFile()) != 0 ) {
+ // tone->getNext(out, framesPerBuffer, spkrVolume);
+ // } else {
+
+ // }
+ // }
+ //
+ // // Additionally handle the mic's audio stream
+ // micAvailPut = _micRingBuffer.AvailForPut();
+ // toPut = (micAvailPut <= (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? micAvailPut : framesPerBuffer * sizeof(SFLDataFormat);
+ // //_debug("AL: Nb sample: %d char, [0]=%f [1]=%f [2]=%f\n", toPut, in[0], in[1], in[2]);
+ // _micRingBuffer.Put(in, toPut, micVolume);
+
}
+ void
+PulseLayer::write( void )
+{
+ int toGet;
+ int normalAvail; // number of data right and data left
+ SFLDataFormat* out = (SFLDataFormat*)malloc(framesPerBuffer * sizeof(SFLDataFormat));
+ normalAvail = _mainSndRingBuffer.AvailForGet();
+ toGet = (normalAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? normalAvail : framesPerBuffer * sizeof(SFLDataFormat);
+ if (toGet) {
+ _mainSndRingBuffer.Get(out, toGet, 100);
+ _debug("Write %i bytes\n" , toGet);
+ pa_stream_write( playback , out , toGet , pa_xfree, 0 , PA_SEEK_RELATIVE);
+ _mainSndRingBuffer.Discard(toGet);
+ } else {
+ bzero(out, framesPerBuffer * sizeof(SFLDataFormat));
+ }
+}
diff --git a/src/audio/pulselayer.h b/src/audio/pulselayer.h
index a4e1b3bb81d896da584e77eb132c1884b9460d59..a63ce2319b635c524061e1722e5c4a3c658dd173 100644
--- a/src/audio/pulselayer.h
+++ b/src/audio/pulselayer.h
@@ -74,7 +74,7 @@ class PulseLayer : public AudioLayer {
*/
int playSamples(void* buffer, int toCopy, bool isTalking) ;
- static void audioCallback ( pa_stream* s, size_t bytes, void* user_data );
+ static void audioCallback ( pa_stream* s, size_t bytes, void* userdata );
static void stream_state_callback( pa_stream* s, void* user_data );
static void context_state_callback( pa_context* c, void* user_data );
@@ -123,12 +123,16 @@ class PulseLayer : public AudioLayer {
*/
void closeCaptureStream( void );
+ void write( void );
+ void createStreams( pa_context* c );
/**
* Drop the pending frames and close the playback device
*/
void closePlaybackStream( void );
- void create_context( void );
+ void connectPulseServer( void );
+
+ RingBuffer _mainSndRingBuffer;
//pa_stream* playback;
//pa_stream* record;
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index b100deefd92b1434af35e073187a7028d147c81d..08f0360d4f35879c86b950b657e165a5d9042c39 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -715,8 +715,14 @@ ManagerImpl::playATone(Tone::TONEID toneId) {
unsigned int nbSampling = audioloop->getSize();
AudioLayer* audiolayer = getAudioDriver();
SFLDataFormat buf[nbSampling];
- if ( audiolayer ) {
+ if ( audiolayer ) {
+ int layer = audiolayer->getLayerType();
+#if CHECK_INTERFACE( layer , ALSA )
audiolayer->putUrgent( buf, nbSampling );
+#else
+ // Pulseaudio code
+ // startStream()?;
+#endif
}
else
return false;
@@ -734,7 +740,11 @@ ManagerImpl::stopTone(bool stopAudio=true) {
if (stopAudio) {
AudioLayer* audiolayer = getAudioDriver();
- //if (audiolayer) { audiolayer->stopStream(); }
+ int layer = audiolayer->getLayerType();
+#if CHECK_INTERFACE( layer , ALSA )
+#else
+ //if (audiolayer) { audiolayer->stopStream(); }
+#endif
}
_toneMutex.enterMutex();
@@ -799,6 +809,7 @@ ManagerImpl::ringtone()
}
AudioLayer* audiolayer = getAudioDriver();
+ int layer = audiolayer->getLayerType();
if (audiolayer==0) { return; }
int sampleRate = audiolayer->getSampleRate();
AudioCodec* codecForTone = _codecDescriptorMap.getFirstCodecAvailable();
@@ -810,10 +821,15 @@ ManagerImpl::ringtone()
_toneMutex.enterMutex();
_audiofile.start();
_toneMutex.leaveMutex();
+#if CHECK_INTERFACE( layer, ALSA )
int size = _audiofile.getSize();
SFLDataFormat output[ size ];
_audiofile.getNext(output, size , 100);
audiolayer->putUrgent( output , size );
+#else
+ // pulseaudio code
+ startStream();
+#endif
} else {
ringback();
}
@@ -852,6 +868,7 @@ ManagerImpl::notificationIncomingCall(void) {
AudioLayer* audiolayer = getAudioDriver();
if (audiolayer != 0) {
+ int layer = audiolayer->getLayerType();
unsigned int samplerate = audiolayer->getSampleRate();
std::ostringstream frequency;
frequency << "440/" << FRAME_PER_BUFFER;
@@ -860,7 +877,11 @@ ManagerImpl::notificationIncomingCall(void) {
unsigned int nbSampling = tone.getSize();
SFLDataFormat buf[nbSampling];
tone.getNext(buf, tone.getSize());
+#if CHECK_INTERFACE( layer , ALSA )
audiolayer->playSamples(buf, sizeof(SFLDataFormat)*nbSampling, true);
+#else
+ audiolayer->putUrgent( buf, sizeof(SFLDataFormat)*nbSampling );
+#endif
}
}
@@ -1158,6 +1179,8 @@ ManagerImpl::getOutputAudioPluginList(void)
void
ManagerImpl::setInputAudioPlugin(const std::string& audioPlugin)
{
+ int layer = _audiodriver -> getLayerType();
+#if CHECK_INTERFACE( layer , ALSA )
_debug("Set input audio plugin\n");
_audiodriver -> setErrorMessage( -1 );
_audiodriver -> openDevice( _audiodriver -> getIndexIn(),
@@ -1168,6 +1191,8 @@ ManagerImpl::setInputAudioPlugin(const std::string& audioPlugin)
audioPlugin);
if( _audiodriver -> getErrorMessage() != -1)
notifyErrClient( _audiodriver -> getErrorMessage() );
+#else
+#endif
}
/**
@@ -1176,6 +1201,8 @@ ManagerImpl::setInputAudioPlugin(const std::string& audioPlugin)
void
ManagerImpl::setOutputAudioPlugin(const std::string& audioPlugin)
{
+ int layer = _audiodriver -> getLayerType();
+#if CHECK_INTERFACE( layer , ALSA )
_debug("Set output audio plugin\n");
_audiodriver -> setErrorMessage( -1 );
_audiodriver -> openDevice( _audiodriver -> getIndexIn(),
@@ -1188,6 +1215,8 @@ ManagerImpl::setOutputAudioPlugin(const std::string& audioPlugin)
notifyErrClient( _audiodriver -> getErrorMessage() );
// set config
setConfig( AUDIO , ALSA_PLUGIN , audioPlugin );
+#else
+#endif
}
/**
@@ -1206,6 +1235,8 @@ ManagerImpl::getAudioOutputDeviceList(void)
void
ManagerImpl::setAudioOutputDevice(const int index)
{
+ int layer = _audiodriver -> getLayerType();
+#if CHECK_INTERFACE( layer , ALSA )
_debug("Set audio output device: %i\n", index);
_audiodriver -> setErrorMessage( -1 );
_audiodriver->openDevice(_audiodriver->getIndexIn(),
@@ -1218,6 +1249,8 @@ ManagerImpl::setAudioOutputDevice(const int index)
notifyErrClient( _audiodriver -> getErrorMessage() );
// set config
setConfig( AUDIO , ALSA_CARD_ID_OUT , index );
+#else
+#endif
}
/**
@@ -1236,6 +1269,8 @@ ManagerImpl::getAudioInputDeviceList(void)
void
ManagerImpl::setAudioInputDevice(const int index)
{
+ int layer = _audiodriver -> getLayerType();
+#if CHECK_INTERFACE( layer , ALSA )
_debug("Set audio input device %i\n", index);
_audiodriver -> setErrorMessage( -1 );
_audiodriver->openDevice(index,
@@ -1248,6 +1283,8 @@ ManagerImpl::setAudioInputDevice(const int index)
notifyErrClient( _audiodriver -> getErrorMessage() );
// set config
setConfig( AUDIO , ALSA_CARD_ID_IN , index );
+#else
+#endif
}
/**