diff --git a/src/audio/alsalayer.cpp b/src/audio/alsalayer.cpp
index 33a6baae1a42e3211d962116247fdcf44128fae0..a6964f2a9f70aafad3bab0f07f65ba6c9e4d5cc6 100644
--- a/src/audio/alsalayer.cpp
+++ b/src/audio/alsalayer.cpp
@@ -31,6 +31,14 @@ int framesPerBufferAlsa = 2048;
, _inChannel()
, _outChannel()
, IDSoundCards()
+ , _is_prepared_playback (false)
+ , _is_running_playback (false)
+ , _is_open_playback (false)
+ , _is_prepared_capture (false)
+ , _is_running_capture (false)
+ , _is_open_capture (false)
+ , _trigger_request (false)
+
{
_debug(" Constructor of AlsaLayer called\n");
/* Instanciate the audio thread */
@@ -40,6 +48,7 @@ int framesPerBufferAlsa = 2048;
// Destructor
AlsaLayer::~AlsaLayer (void)
{
+ _debug("Destructor of AlsaLayer called\n");
closeLayer();
}
@@ -48,35 +57,31 @@ AlsaLayer::closeLayer()
{
_debugAlsa("Close ALSA streams\n");
- ost::MutexLock guard(_mutex);
-
if (_audioThread)
{
+ _debug("Try to stop audio thread\n");
_audioThread->stop();
delete _audioThread; _audioThread=NULL;
}
+
closeCaptureStream();
closePlaybackStream();
-
- deviceClosed = true;
}
bool
AlsaLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize, int stream , std::string plugin)
{
-
- if(deviceClosed == false)
+ /* Close the devices before open it */
+ if (stream == SFL_PCM_BOTH && is_capture_open() == true && is_playback_open() == true)
{
- if( stream == SFL_PCM_CAPTURE )
- closeCaptureStream();
- else if( stream == SFL_PCM_PLAYBACK)
- closePlaybackStream();
- else
- {
- closeCaptureStream();
- closePlaybackStream();
- }
+ closeCaptureStream();
+ closePlaybackStream();
}
+ else if ((stream == SFL_PCM_CAPTURE || stream == SFL_PCM_BOTH) && is_capture_open() == true)
+ closeCaptureStream ();
+ else if ((stream == SFL_PCM_PLAYBACK || stream == SFL_PCM_BOTH) && is_playback_open () == true)
+ closePlaybackStream ();
+
_indexIn = indexIn;
_indexOut = indexOut;
@@ -92,17 +97,7 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize,
_debugAlsa(" : sample rate=%5d, format=%s\n", _sampleRate, SFLDataFormatString);
ost::MutexLock lock( _mutex );
-
- /*void **hint;
- int r = snd_device_name_hint(-1, "pcm", &hint);
-
- while( *hint ){
- printf("%s\n", snd_device_name_get_hint(*hint, "DESC"));
- ++hint;
- }*/
-
-
std::string pcmp = buildDeviceTopo( plugin , indexOut , 0);
std::string pcmc = buildDeviceTopo( PCM_PLUGHW , indexIn , 0);
@@ -112,44 +107,22 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize,
void
AlsaLayer::startStream(void)
{
- ost::MutexLock guard(_mutex);
- int err;
-
- if( _CaptureHandle && _PlaybackHandle )
- {
- _debugAlsa(" Start stream\n");
- 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));
- }
+ _debug ("Start ALSA streams\n");
+ prepareCaptureStream ();
+ startCaptureStream ();
+ startPlaybackStream ();
}
void
AlsaLayer::stopStream(void)
{
- ost::MutexLock guard(_mutex);
-
- if( _CaptureHandle && _PlaybackHandle )
- {
- //ost::MutexLock lock( _mutex );
- _debugAlsa(" Stop Stream\n ");
- snd_pcm_drop( _CaptureHandle );
- snd_pcm_prepare( _CaptureHandle );
- snd_pcm_drop( _PlaybackHandle );
- snd_pcm_prepare( _PlaybackHandle );
- flushUrgent();
- flushMain();
- }
-}
+ _debug ("Stop ALSA streams\n");
+ stopCaptureStream ();
+ //stopPlaybackStream ();
-
- bool
-AlsaLayer::isStreamActive (void)
-{
- //ost::MutexLock lock( _mutex );
- return (isPlaybackActive() && isCaptureActive());
+ /* Flush the ring buffers */
+ flushUrgent();
+ flushMain();
}
int
@@ -179,38 +152,86 @@ AlsaLayer::getMic(void *buffer, int toCopy)
return res ;
}
-
- bool
-AlsaLayer::isStreamStopped (void)
-{
- //ost::MutexLock lock( _mutex );
- return !(isStreamActive());
-}
-
-
void AlsaLayer::reducePulseAppsVolume( void ){}
void AlsaLayer::restorePulseAppsVolume( void ){}
//////////////////////////////////////////////////////////////////////////////////////////////
///////////////// ALSA PRIVATE FUNCTIONS ////////////////////////////////////////////////
////////////////////////////////////////////////////////////////////////////////////////////
+
+void AlsaLayer::stopCaptureStream (void)
+{
+ if(_CaptureHandle){
+ snd_pcm_drop (_CaptureHandle);
+ stop_capture ();
+ }
+}
-bool
-AlsaLayer::isPlaybackActive(void) {
- //ost::MutexLock guard( _mutex );
- if( _PlaybackHandle )
- return (snd_pcm_state(_PlaybackHandle) == SND_PCM_STATE_RUNNING ? true : false);
- else
- return false;
+void AlsaLayer::closeCaptureStream (void)
+{
+ if( is_capture_prepared() == true && is_capture_running() == true )
+ stopCaptureStream ();
+ if (is_capture_open())
+ snd_pcm_close (_CaptureHandle);
+
+ close_capture ();
}
-bool
-AlsaLayer::isCaptureActive(void) {
- //ost::MutexLock guard( _mutex );
- if( _CaptureHandle )
- return (snd_pcm_state( _CaptureHandle) == SND_PCM_STATE_RUNNING ? true : false);
- else
- return false;
+void AlsaLayer::startCaptureStream (void)
+{
+ if(_CaptureHandle){
+ snd_pcm_start (_CaptureHandle);
+ start_capture();
+ }
+}
+
+void AlsaLayer::prepareCaptureStream (void)
+{
+ int err;
+
+ err = snd_pcm_prepare (_CaptureHandle);
+ if( err<0 )
+ _debug("Error preparing the device\n");
+
+ prepare_capture ();
+}
+
+void AlsaLayer::stopPlaybackStream (void)
+{
+ if( _PlaybackHandle){
+ snd_pcm_drop (_PlaybackHandle);
+ stop_playback ();
+ }
+}
+
+
+void AlsaLayer::closePlaybackStream (void)
+{
+ if( is_playback_prepared() == true && is_playback_running() == true )
+ stopPlaybackStream ();
+ if (is_playback_open())
+ snd_pcm_close (_PlaybackHandle);
+
+ close_playback ();
+}
+
+void AlsaLayer::startPlaybackStream (void)
+{
+ if( _PlaybackHandle){
+ snd_pcm_start (_PlaybackHandle);
+ start_playback();
+ }
+}
+
+void AlsaLayer::preparePlaybackStream (void)
+{
+ int err;
+
+ err = snd_pcm_prepare (_PlaybackHandle);
+ if( err<0 )
+ _debug("Error preparing the device\n");
+
+ prepare_playback ();
}
bool AlsaLayer::alsa_set_params( snd_pcm_t *pcm_handle, int type, int rate ){
@@ -311,12 +332,8 @@ bool AlsaLayer::alsa_set_params( snd_pcm_t *pcm_handle, int type, int rate ){
return false;
}
- if( type == 1 ){
-
- }
snd_pcm_sw_params_free( swparams );
- deviceClosed = false;
return true;
}
@@ -340,6 +357,8 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
snd_pcm_close( _PlaybackHandle );
return false;
}
+
+ open_playback ();
}
if(flag == SFL_PCM_BOTH || flag == SFL_PCM_CAPTURE)
@@ -354,9 +373,10 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
snd_pcm_close( _CaptureHandle );
return false;
}
- if( (err = snd_pcm_start( _CaptureHandle ))<0) {
- _debugAlsa( "snd_pcm_start failed\n");
- }
+
+ open_capture ();
+
+ startCaptureStream ();
}
/* Start the secondary audio thread for callbacks */
@@ -369,33 +389,34 @@ AlsaLayer::open_device(std::string pcm_p, std::string pcm_c, int flag)
int
AlsaLayer::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_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_writei( _PlaybackHandle , buffer , frames );
- break;
- case -ESTRPIPE:
- _debugAlsa(" ESTRPIPE(%s)\n", snd_strerror(err));
- snd_pcm_resume( _PlaybackHandle );
- break;
- case -EBADFD:
- _debugAlsa(" (%s)\n", snd_strerror( err ));
- break;
+
+ if (_trigger_request == true)
+ {
+ _trigger_request = false;
+ startPlaybackStream ();
}
- if( ( err >=0 ) && ( err < (int)frames ) )
- _debugAlsa("Short write : %d out of %d\n", err , (int)frames);
+ snd_pcm_uframes_t frames = snd_pcm_bytes_to_frames( _PlaybackHandle, length);
+ int err;
+ if ((err=snd_pcm_writei( _PlaybackHandle , buffer , frames ))<0)
+ {
+ switch(err) {
+ case -EPIPE:
+ case -ESTRPIPE:
+ case -EIO:
+ //_debugAlsa(" XRUN playback ignored (%s)\n", snd_strerror(err));
+ handle_xrun_playback();
+ if (snd_pcm_writei( _PlaybackHandle , buffer , frames )<0)
+ _debugAlsa ("XRUN handling failed\n");
+ _trigger_request = true;
+ break;
+ default:
+ //_debugAlsa ("Write error unknown - dropping frames **********************************: %s\n", snd_strerror(err));
+ stopPlaybackStream ();
+ break;
+ }
+ }
return ( err > 0 )? err : 0 ;
}
@@ -405,29 +426,29 @@ AlsaLayer::read( void* buffer, int toCopy)
{
ost::MutexLock lock( _mutex );
- if(deviceClosed || _CaptureHandle == NULL)
- return 0;
+ int samples;
- int err;
if(snd_pcm_state( _CaptureHandle ) == SND_PCM_STATE_XRUN)
{
- snd_pcm_prepare( _CaptureHandle );
- snd_pcm_start( _CaptureHandle );
+ prepareCaptureStream ();
+ startCaptureStream ();
}
+
snd_pcm_uframes_t frames = snd_pcm_bytes_to_frames( _CaptureHandle, toCopy );
- if(( err = snd_pcm_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 -EAGAIN:
- _debugAlsa(" Capture EAGAIN (%s)\n", snd_strerror(err));
- break;
+ if(( samples = snd_pcm_readi( _CaptureHandle, buffer, frames)) < 0 ) {
+ switch (samples)
+ {
case -EPIPE:
- _debugAlsa(" Capture EPIPE (%s)\n", snd_strerror(err));
+ case -ESTRPIPE:
+ case -EIO:
+ //_debugAlsa(" XRUN capture ignored (%s)\n", snd_strerror(samples));
handle_xrun_capture();
+ samples = snd_pcm_readi( _CaptureHandle, buffer, frames);
+ if (samples<0) samples=0;
+ break;
+ default:
+ //_debugAlsa ("Error when capturing data ***********************************************: %s\n", snd_strerror(samples));
+ stopCaptureStream();
break;
}
return 0;
@@ -446,9 +467,9 @@ AlsaLayer::handle_xrun_capture( void )
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 );
+ stop_capture ();
+ prepare_capture ();
+ start_capture ();
}
}
else
@@ -468,9 +489,9 @@ AlsaLayer::handle_xrun_playback( void )
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 );
+ stopPlaybackStream ();
+ preparePlaybackStream ();
+ _trigger_request = true;
}
}
}
@@ -545,25 +566,7 @@ AlsaLayer::getSoundCardsInfo( int stream )
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 )
@@ -653,8 +656,8 @@ void AlsaLayer::audioCallback (void)
}
// Additionally handle the mic's audio stream
- micAvailPut = _micRingBuffer.AvailForPut();
- toPut = (micAvailPut <= (int)(framesPerBufferAlsa * sizeof(SFLDataFormat))) ? micAvailPut : framesPerBufferAlsa * sizeof(SFLDataFormat);
+ //micAvailPut = _micRingBuffer.AvailForPut();
+ //toPut = (micAvailPut <= (int)(framesPerBufferAlsa * sizeof(SFLDataFormat))) ? micAvailPut : framesPerBufferAlsa * 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);
+ //_micRingBuffer.Put(in, toPut, micVolume);
}
diff --git a/src/audio/alsalayer.h b/src/audio/alsalayer.h
index 9f8ffe0283620a163499612fa7e1d865f2b0f9c7..b64d1e66810879a14f235425fe165ce2b02bc706 100644
--- a/src/audio/alsalayer.h
+++ b/src/audio/alsalayer.h
@@ -80,34 +80,6 @@ class AlsaLayer : public AudioLayer {
void stopStream(void);
/**
- * Check if the playback is running
- * @return true if the state of the playback handle equals SND_PCM_STATE_RUNNING
- * false otherwise
- */
- bool isPlaybackActive( void );
-
- /**
- * Check if the capture is running
- * @return true if the state of the capture handle equals SND_PCM_STATE_RUNNING
- * false otherwise
- */
- bool isCaptureActive( void );
-
- /**
- * Check if both capture and playback are running
- * @return true if capture and playback are running
- * false otherwise
- */
- bool isStreamActive(void);
-
- /**
- * Check if both capture and playback are stopped
- * @return true if capture and playback are stopped
- * false otherwise
- */
- bool isStreamStopped(void);
-
- /**
* Query the capture device for number of bytes available in the hardware ring buffer
* @return int The number of bytes available
*/
@@ -191,17 +163,44 @@ class AlsaLayer : public AudioLayer {
// Assignment Operator
AlsaLayer& operator=( const AlsaLayer& rh);
+ bool _is_prepared_playback;
+ bool _is_prepared_capture;
+ bool _is_running_capture;
+ bool _is_running_playback;
+ bool _is_open_playback;
+ bool _is_open_capture;
+ bool _trigger_request;
+
+ bool is_playback_prepared (void) { return _is_prepared_playback; }
+ bool is_capture_prepared (void) { return _is_prepared_capture; }
+ void prepare_playback (void) { _is_prepared_playback = true; }
+ void prepare_capture (void) { _is_prepared_capture = true; }
+ bool is_capture_running (void) { return _is_running_capture; }
+ bool is_playback_running (void) { return _is_running_playback; }
+ void start_playback (void) { _is_running_playback = true; }
+ void stop_playback (void) { _is_running_playback = false; _is_prepared_playback = false; }
+ void start_capture (void) { _is_running_capture = true; }
+ void stop_capture (void) { _is_running_capture = false; _is_prepared_capture = false; }
+ void close_playback (void) { _is_open_playback = false; }
+ void close_capture (void) { _is_open_capture = false; }
+ void open_playback (void) { _is_open_playback = true; }
+ void open_capture (void) { _is_open_capture = true; }
+ bool is_capture_open (void) { return _is_open_capture; }
+ bool is_playback_open (void) { return _is_open_playback; }
+
/**
* Drop the pending frames and close the capture device
* ALSA Library API
*/
void closeCaptureStream( void );
+ void stopCaptureStream( void );
+ void startCaptureStream( void );
+ void prepareCaptureStream( void );
- /**
- * Drop the pending frames and close the playback device
- * ALSA Library API
- */
void closePlaybackStream( void );
+ void stopPlaybackStream( void );
+ void startPlaybackStream( void );
+ void preparePlaybackStream( void );
/**
* Open the specified device.
diff --git a/src/audio/audiolayer.cpp b/src/audio/audiolayer.cpp
index a94eaf2595d22324cd47f96b44cf2e6fbdab26f7..3eea4392c737b51c3f954b438ca3dac7b57a30b9 100644
--- a/src/audio/audiolayer.cpp
+++ b/src/audio/audiolayer.cpp
@@ -31,6 +31,12 @@ void AudioLayer::flushUrgent (void)
_urgentRingBuffer.flush();
}
+void AudioLayer::flushMic (void)
+{
+ ost::MutexLock guard(_mutex);
+ _micRingBuffer.flush();
+}
+
int AudioLayer::putUrgent(void* buffer, int toCopy)
{
int a;
diff --git a/src/audio/audiolayer.h b/src/audio/audiolayer.h
index 6a558c9081f480a6a91be05fce050a68e422a77f..91aaafe78a2a584fd927b8030447dec25e501f36 100644
--- a/src/audio/audiolayer.h
+++ b/src/audio/audiolayer.h
@@ -60,7 +60,6 @@ class AudioLayer {
, _micRingBuffer( SIZEBUF )
, _defaultVolume(100)
, _talk ( false )
- , deviceClosed ( true )
, _indexIn ( 0 )
, _indexOut ( 0 )
, _sampleRate ( 0 )
@@ -77,7 +76,7 @@ class AudioLayer {
/**
* Destructor
*/
- ~AudioLayer(void){}
+ virtual ~AudioLayer(void) {}
virtual void closeLayer( void ) = 0;
@@ -110,14 +109,6 @@ class AudioLayer {
*/
virtual void stopStream(void) = 0;
-
- /**
- * Check if the capture is running
- * @return true if the state of the capture handle equals SND_PCM_STATE_RUNNING
- * false otherwise
- */
- virtual bool isCaptureActive( void ) = 0;
-
/**
* Query the capture device for number of bytes available in the hardware ring buffer
* @return int The number of bytes available
@@ -194,6 +185,11 @@ class AudioLayer {
void flushUrgent (void);
+ /**
+ * Flush the mic ringbuffer
+ */
+ void flushMic();
+
/**
* Write accessor to the error state
@@ -282,13 +278,6 @@ class AudioLayer {
*/
bool _talk;
- /**
- * Enable to determine if the devices are opened or not
- * true if the devices are closed
- * false otherwise
- */
- bool deviceClosed;
-
/**
* Number of audio cards on which capture stream has been opened
*/
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index 31f90fe56c662c8c159ced5cd5d77da3fcd7fb6e..ed25feec437a4373e2dc787453a348e3e9bb510a 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -60,7 +60,6 @@ AudioRtp::createNewSession (SIPCall *ca) {
_debug("! ARTP Failure: Thread already exists..., stopping it\n");
_debug("**********************************************************\n");
delete _RTXThread; _RTXThread = 0;
- //return -1;
}
// Start RTP Send/Receive threads
@@ -263,7 +262,7 @@ AudioRtpRTX::sendSessionFromMic(int timestamp)
int nb_sample_up = nbSample;
int nbSamplesMax = _layerFrameSize * _audiocodec->getClockRate() / 1000;
- //_debug("resample data\n");
+ //_debug("resample data = %i\n", nb_sample_up);
nbSample = reSampleData(_audiocodec->getClockRate(), nb_sample_up, DOWN_SAMPLING);
if ( nbSample < nbSamplesMax - 10 ) { // if only 10 is missing, it's ok
diff --git a/src/audio/pulselayer.cpp b/src/audio/pulselayer.cpp
index 4004cd557f43aa6ab6b3798c6ab4121566129042..9843666c645c3bef501f25beb4b22785e55542bc 100644
--- a/src/audio/pulselayer.cpp
+++ b/src/audio/pulselayer.cpp
@@ -198,12 +198,6 @@ PulseLayer::getMic(void *buffer, int toCopy)
return 0;
}
- void
-PulseLayer::flushMic()
-{
- _micRingBuffer.flush();
-}
-
void
PulseLayer::startStream (void)
{
@@ -221,6 +215,8 @@ PulseLayer::stopStream (void)
pa_stream_flush( playback->pulseStream(), NULL, NULL );
pa_stream_flush( record->pulseStream(), NULL, NULL );
flushMic();
+ flushMain();
+ flushUrgent();
}
@@ -273,7 +269,7 @@ void PulseLayer::writeToSpeaker( void )
if (urgentAvail > 0) {
// Urgent data (dtmf, incoming call signal) come first.
- //_debug("Play urgent!: %i\n" , urgentAvail);
+ //_debug("Play urgent!: %i\e" , urgentAvail);
toGet = (urgentAvail < (int)(framesPerBuffer * sizeof(SFLDataFormat))) ? urgentAvail : framesPerBuffer * sizeof(SFLDataFormat);
out = (SFLDataFormat*)pa_xmalloc(toGet * sizeof(SFLDataFormat) );
_urgentRingBuffer.Get(out, toGet, 100);
diff --git a/src/audio/pulselayer.h b/src/audio/pulselayer.h
index ae1844b7861edcca8e6eb23849b526126697d726..a1948652411aa9fd45c8c06653c74b339194b385 100644
--- a/src/audio/pulselayer.h
+++ b/src/audio/pulselayer.h
@@ -57,16 +57,6 @@ class PulseLayer : public AudioLayer {
void stopStream(void);
- /**
- * UNUSED in pulseaudio layer
- */
- bool isCaptureActive( void ) { return true; }
-
- /**
- * UNUSED in pulseaudio layer
- */
- bool isStreamActive (void);
-
/**
* Query the capture device for number of bytes available in the hardware ring buffer
* @return int The number of bytes available
@@ -81,11 +71,6 @@ class PulseLayer : public AudioLayer {
*/
int getMic(void *, int);
- /**
- * Flush the mic ringbuffer
- */
- void flushMic();
-
static void overflow ( pa_stream* s, void* userdata );
static void underflow ( pa_stream* s, void* userdata );
static void stream_state_callback( pa_stream* s, void* user_data );
diff --git a/src/eventthread.cpp b/src/eventthread.cpp
index c6389ecad181eb010b564ab3e970747e7b498df5..045d62dd63bb24dc663b5f1ea664975805c00730 100644
--- a/src/eventthread.cpp
+++ b/src/eventthread.cpp
@@ -59,5 +59,10 @@ void AudioThread::run (void)
void AudioThread::stop (void)
{
- exit();
+ try{
+ terminate();
+ } catch(...){
+ _debugException("! ARTP: Thread destructor didn't terminate correctly");
+ throw;
+ }
}
diff --git a/src/iaxvoiplink.cpp b/src/iaxvoiplink.cpp
index 6a553427c61a7002f11c92319ba1590b7f3affc3..8fc1c5f308539f677d871d1d2996994c788c0fbc 100644
--- a/src/iaxvoiplink.cpp
+++ b/src/iaxvoiplink.cpp
@@ -285,7 +285,7 @@ IAXVoIPLink::sendAudioFromMic(void)
_mutexIAX.enterMutex();
// Make sure the session and the call still exists.
if (currentCall->getSession()) {
- if ( iax_send_voice(currentCall->getSession(), currentCall->getFormat(), micDataEncoded, compSize, nbSample) == -1) {
+ if (iax_send_voice(currentCall->getSession(), currentCall->getFormat(), micDataEncoded, compSize, nbSample) == -1) {
_debug("IAX: Error sending voice data.\n");
}
}
@@ -540,8 +540,6 @@ IAXVoIPLink::iaxOutgoingInvite(IAXCall* call)
std::string username, strNum;
char *lang=NULL;
int wait, audio_format_preferred, audio_format_capability;
- char user[username.length()+1];
- char num[strNum.length()+1];
IAXAccount *account;
newsession = iax_session_new();
@@ -554,16 +552,14 @@ IAXVoIPLink::iaxOutgoingInvite(IAXCall* call)
account = dynamic_cast<IAXAccount*> (getAccountPtr());
username = account->getUsername();
strNum = username + ":" + account->getPassword() + "@" + account->getHostname() + "/" + call->getPeerNumber();
- strcpy(user, username.c_str());
- strcpy(num, strNum.c_str());
wait = 0;
/** @todo Make preference dynamic, and configurable */
audio_format_preferred = call->getFirstMatchingFormat(call->getSupportedFormat());
audio_format_capability = call->getSupportedFormat();
- _debug("IAX New call: %s\n", num);
- iax_call(newsession, user, user, num, lang, wait, audio_format_preferred, audio_format_capability);
+ _debug("IAX New call: %s\n", strNum.c_str());
+ iax_call(newsession, username.c_str(), username.c_str(), strNum.c_str(), lang, wait, audio_format_preferred, audio_format_capability);
return true;
}
@@ -653,8 +649,8 @@ IAXVoIPLink::iaxHandleCallEvent(iax_event* event, IAXCall* call)
case IAX_EVENT_VOICE:
//_debug("Should have a decent value!!!!!! = %i\n" , call -> getAudioCodec());
//TODO Check this method
- if( !audiolayer -> isCaptureActive())
- audiolayer->startStream();
+ //if( !audiolayer -> is_capture_running())
+ //audiolayer->startStream();
iaxHandleVoiceEvent(event, call);
break;
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 56f5cb18c8f28d8af32673c0c2d450ca5654e0af..3a06ab61923c0d8b445d8c23afa0fce4a4a5e295 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -1590,17 +1590,20 @@ ManagerImpl::initAudioDriver(void)
void
ManagerImpl::selectAudioDriver (void)
{
- int layer = _audiodriver->getLayerType();
+ int layer, numCardIn, numCardOut, sampleRate, frameSize;
+ std::string alsaPlugin;
+
+ layer = _audiodriver->getLayerType();
_debug("Audio layer type: %i\n" , layer);
- std::string alsaPlugin = getConfigString( AUDIO , ALSA_PLUGIN );
- int numCardIn = getConfigInt( AUDIO , ALSA_CARD_ID_IN );
- int numCardOut = getConfigInt( AUDIO , ALSA_CARD_ID_OUT );
- int sampleRate = getConfigInt( AUDIO , ALSA_SAMPLE_RATE );
+ alsaPlugin = getConfigString( AUDIO , ALSA_PLUGIN );
+ numCardIn = getConfigInt( AUDIO , ALSA_CARD_ID_IN );
+ numCardOut = getConfigInt( AUDIO , ALSA_CARD_ID_OUT );
+ sampleRate = getConfigInt( AUDIO , ALSA_SAMPLE_RATE );
if (sampleRate <=0 || sampleRate > 48000) {
sampleRate = 44100;
}
- int frameSize = getConfigInt( AUDIO , ALSA_FRAME_SIZE );
+ frameSize = getConfigInt( AUDIO , ALSA_FRAME_SIZE );
if( !_audiodriver -> soundCardIndexExist( numCardIn , SFL_PCM_CAPTURE ) )
{
@@ -1614,8 +1617,7 @@ ManagerImpl::selectAudioDriver (void)
numCardOut = ALSA_DFT_CARD_ID ;
setConfig( AUDIO , ALSA_CARD_ID_OUT , ALSA_DFT_CARD_ID );
}
-
-
+
if(CHECK_INTERFACE( layer , ALSA ))
{
delete _audiodriver;
diff --git a/src/samplerateconverter.cpp b/src/samplerateconverter.cpp
index 3c6d0ed90db1902bc1e1ee0f29224f1e767453b4..52d047ca4cb5c5da507b1d91b4a7851b49cea9b9 100644
--- a/src/samplerateconverter.cpp
+++ b/src/samplerateconverter.cpp
@@ -78,6 +78,7 @@ void SamplerateConverter::init( void ) {
int SamplerateConverter::upsampleData( SFLDataFormat* dataIn , SFLDataFormat* dataOut, int samplerate1 , int samplerate2 , int nbSamples ){
double upsampleFactor = (double)samplerate2 / samplerate1 ;
+ //_debug("factor = %f\n" , upsampleFactor);
int nbSamplesMax = (int) ( samplerate2 * getFramesize() / 1000 );
if( upsampleFactor != 1 && dataIn != NULL )
{
@@ -106,7 +107,7 @@ int SamplerateConverter::downsampleData( SFLDataFormat* dataIn , SFLDataFormat*
double downsampleFactor = (double)samplerate1 / samplerate2;
//_debug("factor = %f\n" , downsampleFactor);
int nbSamplesMax = (int) ( samplerate1 * getFramesize() / 1000 );
- if ( downsampleFactor != 1)
+ if ( downsampleFactor != 1 )
{
SRC_DATA src_data;
src_data.data_in = _floatBufferUpMic;