diff --git a/sflphone-gtk/src/audioconf.c b/sflphone-gtk/src/audioconf.c
index 3e194d2f79565f8871dc01799dea0c0c6fcebe7d..18b1b4e7150151c1b4e34bc769402189b776d3c0 100644
--- a/sflphone-gtk/src/audioconf.c
+++ b/sflphone-gtk/src/audioconf.c
@@ -599,7 +599,9 @@ GtkWidget* codecs_box()
void
select_audio_manager( void )
{
+
g_print("audio manager selected\n");
+
if( !SHOW_ALSA_CONF && !gtk_toggle_button_get_active( GTK_TOGGLE_BUTTON(pulse) ) )
{
g_print(" display alsa conf panel\n");
diff --git a/src/audio/alsa-bak b/src/audio/alsa-bak
deleted file mode 100644
index b2733b244ee201771196b46d7946de829c9e16e1..0000000000000000000000000000000000000000
--- a/src/audio/alsa-bak
+++ /dev/null
@@ -1,633 +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 "alsalayer.h"
-
-// Constructor
-AlsaLayer::AlsaLayer( ManagerImpl* manager )
- : AudioLayer( manager )
- , _PlaybackHandle(NULL)
- , _CaptureHandle(NULL)
-{
- _debug(" Constructor of AlsaLayer called\n");
- _defaultVolume = 100;
-}
-
-// Destructor
-AlsaLayer::~AlsaLayer (void)
-{
- _debugAlsa("Close ALSA streams\n");
- closeCaptureStream();
- closePlaybackStream();
- deviceClosed = true;
-}
-
- bool
-AlsaLayer::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 AlsaLayer: 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
-AlsaLayer::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
-AlsaLayer::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 AlsaLayer::AlsaCallBack( snd_async_handler_t* pcm_callback )
-{
- ( ( AlsaLayer *)snd_async_handler_get_callback_private( pcm_callback )) -> playTones();
-}
-
- void
-AlsaLayer::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
-AlsaLayer::isStreamActive (void)
-{
- ost::MutexLock lock( _mutex );
- return (isPlaybackActive() && isCaptureActive());
-}
-
-
- int
-AlsaLayer::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
-AlsaLayer::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
-AlsaLayer::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
-AlsaLayer::getMic(void *buffer, int toCopy)
-{
- int res = 0 ;
- if( _CaptureHandle )
- {
- res = read( buffer, toCopy );
- adjustVolume( buffer , toCopy , SFL_PCM_CAPTURE );
- }
- return res ;
-}
-
-
- bool
-AlsaLayer::isStreamStopped (void)
-{
- ost::MutexLock lock( _mutex );
- return !(isStreamActive());
-}
-
-
-//////////////////////////////////////////////////////////////////////////////////////////////
-///////////////// ALSA PRIVATE FUNCTIONS ////////////////////////////////////////////////
-////////////////////////////////////////////////////////////////////////////////////////////
-
-
-
- void
-AlsaLayer::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
-AlsaLayer::isPlaybackActive(void) {
- ost::MutexLock guard( _mutex );
- if( _PlaybackHandle )
- return (snd_pcm_state(_PlaybackHandle) == SND_PCM_STATE_RUNNING ? true : false);
- else
- return false;
-}
-
-bool
-AlsaLayer::isCaptureActive(void) {
- ost::MutexLock guard( _mutex );
- if( _CaptureHandle )
- return (snd_pcm_state( _CaptureHandle) == SND_PCM_STATE_RUNNING ? true : false);
- else
- return false;
-}
-
-
- bool
-AlsaLayer::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 = 2048;
- snd_pcm_uframes_t buffer_size = period_size * 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( _CaptureHandle, 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( _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, 0 ) < 0){
- _debugAlsa("Error while opening playback device %s\n", pcm_p.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_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
-AlsaLayer::write(void* buffer, int length)
-{
- 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("EAGAIN (%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("ESTRPIPE (%s)\n", snd_strerror(err));
- //snd_pcm_resume( _PlaybackHandle );
- break;
- case -EBADFD:
- _debugAlsa("EBADFD (%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
-AlsaLayer::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
-AlsaLayer::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
-AlsaLayer::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 );
- _debug("Underrun\n");
- snd_pcm_prepare( _PlaybackHandle );
- //snd_pcm_start( _PlaybackHandle );
- }
- }
-}
-
- std::string
-AlsaLayer::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>
-AlsaLayer::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
-AlsaLayer::closeCaptureStream( void)
-{
- if(_CaptureHandle){
- snd_pcm_drop( _CaptureHandle );
- snd_pcm_close( _CaptureHandle );
- _CaptureHandle = 0;
- }
-}
-
- void
-AlsaLayer::closePlaybackStream( void)
-{
- if(_PlaybackHandle){
- snd_pcm_drop( _PlaybackHandle );
- snd_pcm_close( _PlaybackHandle );
- _PlaybackHandle = 0;
- }
-}
-
- bool
-AlsaLayer::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
-AlsaLayer::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*
-AlsaLayer::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/alsalayer.cpp b/src/audio/alsalayer.cpp
index 5d7e378837b6fda5bfd1b897e17160e78b27dbfa..3402cfe4db642a178d9601d2dd05a2831149c936 100644
--- a/src/audio/alsalayer.cpp
+++ b/src/audio/alsalayer.cpp
@@ -466,11 +466,6 @@ AlsaLayer::read( void* buffer, int toCopy)
}
-
- int
-AlsaLayer::putMain(void* buffer, int toCopy)
-{}
-
int
AlsaLayer::putInCache( char code, void *buffer, int toCopy )
{}
diff --git a/src/audio/alsalayer.h b/src/audio/alsalayer.h
index 1b31eb2f03a2c6286e56310b87327341924f3164..fb5def1fa6d4e8e81ebfad4e152dd1f876f08eb9 100644
--- a/src/audio/alsalayer.h
+++ b/src/audio/alsalayer.h
@@ -186,10 +186,6 @@ class AlsaLayer : public AudioLayer {
*/
int putInCache( char code, void *buffer, int toCopy );
- /**
- * UNUSED in ALSA layer
- */
- int putMain(void* buffer, int toCopy);
/**
* UNUSED in ALSA layer
diff --git a/src/audio/audiolayer.h b/src/audio/audiolayer.h
index 6e2b70879d2aed393ac3708ff6f8b13a9ae9951a..690542f5e08ddf0ba8eb006352289449c3ccd888 100644
--- a/src/audio/audiolayer.h
+++ b/src/audio/audiolayer.h
@@ -99,12 +99,6 @@ class AudioLayer {
*/
virtual void stopStream(void) = 0;
- /**
- * Check if both capture and playback are running
- * @return true if capture and playback are running
- * false otherwise
- */
- virtual bool isStreamActive(void) = 0;
/**
* Check if the capture is running
@@ -131,7 +125,6 @@ class AudioLayer {
*/
virtual int putUrgent(void* buffer, int toCopy) = 0;
- virtual int putMain( void* buffer, int toCopy) = 0;
virtual int putInCache(char code, void* buffer, int toCopy) = 0;
/**
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index 641eb4f841a5ebfc05ea3fb7c06828ca6a36d4fa..2c7da7d7e29938a5dee72d21d79af4b908069300 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -336,13 +336,9 @@ AudioRtpRTX::receiveSessionForSpkr (int& countTime)
#ifdef DATAFORMAT_IS_FLOAT
#else
#endif
-
- int layer = audiolayer->getLayerType();
- //_debug(" interface %i - ALSA = %i\n" , layer, ALSA);
- if( CHECK_INTERFACE( layer, ALSA ) )
- audiolayer->playSamples( spkrDataConverted, nbSample * sizeof(SFLDataFormat), true);
- else
- audiolayer->putMain( spkrDataConverted, nbSample * sizeof(SFLDataFormat) );
+
+ audiolayer->playSamples( spkrDataConverted, nbSample * sizeof(SFLDataFormat), true);
+
// 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/audiostream.cpp b/src/audio/audiostream.cpp
index 196a17e1086fb714ca9e2d511f6f2bcfa08703a0..0ad925c4e574e3119fa77b672c8d69f483744f49 100644
--- a/src/audio/audiostream.cpp
+++ b/src/audio/audiostream.cpp
@@ -47,6 +47,7 @@ AudioStream::disconnect( void )
{
_debug("Destroy audio streams\n");
pa_stream_disconnect( pulseStream() );
+ pa_stream_unref( pulseStream() );
}
void
@@ -74,7 +75,6 @@ AudioStream::stream_state_callback( pa_stream* s, void* user_data )
pa_stream*
AudioStream::createStream( pa_context* c )
{
- _debug("Creating %s stream...\n" , _streamDescription.c_str());
pa_stream* s;
pa_cvolume cv;
@@ -88,14 +88,14 @@ AudioStream::createStream( pa_context* c )
if( _streamType == PLAYBACK_STREAM ){
pa_buffer_attr* attributes;
- attributes->maxlength = 66500;
- attributes->tlength = 44100;
- attributes->prebuf = 10000;
- attributes->minreq = 882;
- pa_stream_connect_playback( s , NULL , attributes ,
+ //attributes->maxlength = 66500;
+ //attributes->tlength = 44100;
+ //attributes->prebuf = 10000;
+ //attributes->minreq = 882;
+ pa_stream_connect_playback( s , NULL , NULL ,
PA_STREAM_INTERPOLATE_TIMING,
&_volume, NULL);
- //pa_cvolume_set(&cv, sample_spec.channels , _volume) , NULL );
+ //pa_cvolume_set(&cv, sample_spec.channels , PA_VOLUME_NORM) , NULL );
}
else if( _streamType == CAPTURE_STREAM ){
pa_stream_connect_record( s , NULL , NULL , PA_STREAM_START_CORKED );
diff --git a/src/audio/pulselayer.cpp b/src/audio/pulselayer.cpp
index 99e8c0fce15bfd3ddcd4e44737c2965fcaef7f67..9c0d401b74931525bf911a04400fe83ed3028909 100644
--- a/src/audio/pulselayer.cpp
+++ b/src/audio/pulselayer.cpp
@@ -37,6 +37,7 @@ PulseLayer::~PulseLayer (void)
delete playback;
delete record;
pa_context_disconnect(context);
+ pa_context_unref( context );
}
void
@@ -45,6 +46,7 @@ PulseLayer::closeLayer( void )
playback->disconnect();
record->disconnect();
pa_context_disconnect( context );
+ pa_context_unref( context );
}
void
@@ -160,11 +162,6 @@ PulseLayer::closePlaybackStream( void )
int
PulseLayer::playSamples(void* buffer, int toCopy, bool isTalking)
-{
-}
-
- int
-PulseLayer::putMain(void* buffer, int toCopy)
{
int a = _mainSndRingBuffer.AvailForPut();
if ( a >= toCopy ) {
@@ -238,11 +235,6 @@ PulseLayer::stopStream (void)
flushMic();
}
- bool
-PulseLayer::isStreamActive (void)
-{
-}
-
void
PulseLayer::audioCallback ( pa_stream* s, size_t bytes, void* userdata )
{
diff --git a/src/audio/pulselayer.h b/src/audio/pulselayer.h
index 63b951120a048f600dd61a82bf59e115e730d8a7..122b267753edb7ebfd70adc3d8e53042377cf647 100644
--- a/src/audio/pulselayer.h
+++ b/src/audio/pulselayer.h
@@ -70,8 +70,6 @@ class PulseLayer : public AudioLayer {
*/
void flushMain();
- int putMain(void* buffer, int toCopy);
-
int putUrgent(void* buffer, int toCopy);
/**
diff --git a/src/iaxvoiplink.cpp b/src/iaxvoiplink.cpp
index 6b76c9c123dc6bcdfae3761b2cdb33facedd8b45..1629c1a35c598cb31907e24878d76841c334a8ff 100644
--- a/src/iaxvoiplink.cpp
+++ b/src/iaxvoiplink.cpp
@@ -201,11 +201,9 @@ IAXVoIPLink::getEvent()
}
_mutexIAX.leaveMutex();
-
// Do the doodle-moodle to send audio from the microphone to the IAX channel.
sendAudioFromMic();
-
// Refresh registration.
if (_nextRefreshStamp && _nextRefreshStamp - 2 < time(NULL)) {
sendRegister();
@@ -622,7 +620,7 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
Manager::instance().peerAnsweredCall(id);
//audiolayer->flushMic();
- audiolayer->startStream();
+ //audiolayer->startStream();
// start audio here?
} else {
// deja connecté ?
@@ -639,7 +637,7 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
case IAX_EVENT_VOICE:
//_debug("Should have a decent value!!!!!! = %i\n" , call -> getAudioCodec());
//if( !audiolayer -> isCaptureActive())
- //audiolayer->startStream();
+ //audiolayer->startStream();
iaxHandleVoiceEvent(event, call);
break;
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index af9c12f6f6cd435f90193f6f505d9571885d86f3..6e86c6d8b48ea0da4a0f7518d4249f922a28a412 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -1224,14 +1224,14 @@ ManagerImpl::setInputAudioPlugin(const std::string& audioPlugin)
ManagerImpl::setOutputAudioPlugin(const std::string& audioPlugin)
{
int layer = _audiodriver -> getLayerType();
- _debug("Set output audio plugin\n");
- _audiodriver -> setErrorMessage( -1 );
- _audiodriver -> openDevice( _audiodriver -> getIndexIn(),
- _audiodriver -> getIndexOut(),
- _audiodriver -> getSampleRate(),
- _audiodriver -> getFrameSize(),
- SFL_PCM_BOTH,
- audioPlugin);
+ _debug("Set output audio plugin\n");
+ _audiodriver -> setErrorMessage( -1 );
+ _audiodriver -> openDevice( _audiodriver -> getIndexIn(),
+ _audiodriver -> getIndexOut(),
+ _audiodriver -> getSampleRate(),
+ _audiodriver -> getFrameSize(),
+ SFL_PCM_BOTH,
+ audioPlugin);
if( _audiodriver -> getErrorMessage() != -1)
notifyErrClient( _audiodriver -> getErrorMessage() );
// set config