diff --git a/sflphone-common/src/audio/alsa/alsalayer.cpp b/sflphone-common/src/audio/alsa/alsalayer.cpp
index 92f2fba69ae38af047b3890326e25b938f1b28a6..fcd22045b26058cafa629946837a5abface97557 100644
--- a/sflphone-common/src/audio/alsa/alsalayer.cpp
+++ b/sflphone-common/src/audio/alsa/alsalayer.cpp
@@ -37,14 +37,14 @@ AlsaLayer::AlsaLayer (ManagerImpl* manager)
, _is_running_capture (false)
, _is_open_playback (false)
, _is_open_capture (false)
- , _trigger_request (false)
- , _converterSamplingRate(0)
+ , _trigger_request (false)
+ , _audioThread(NULL)
{
_debug (" Constructor of AlsaLayer called\n");
/* Instanciate the audio thread */
- _audioThread = new AudioThread (this);
-
+ // _audioThread = new AudioThread (this);
+ // _audioThread = NULL;
_urgentRingBuffer.createReadPointer();
dcblocker = new DcBlocker();
@@ -60,7 +60,9 @@ AlsaLayer::~AlsaLayer (void)
delete _converter; _converter = NULL;
}
- delete dcblocker; dcblocker = NULL;
+ if(dcblocker) {
+ delete dcblocker; dcblocker = NULL;
+ }
}
bool
@@ -71,6 +73,7 @@ AlsaLayer::closeLayer()
try {
/* Stop the audio thread first */
if (_audioThread) {
+ _debug("Stop Audio Thread\n");
delete _audioThread;
_audioThread=NULL;
}
@@ -112,7 +115,7 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize,
_frameSize = frameSize;
- _audioPlugin = plugin;
+ _audioPlugin = std::string(plugin);
_debugAlsa (" Setting AlsaLayer: device in=%2d, out=%2d\n", _indexIn, _indexOut);
@@ -122,37 +125,76 @@ AlsaLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize,
_debugAlsa (" : sample rate=%5d, format=%s\n", _audioSampleRate, SFLDataFormatString);
+ _audioThread = NULL;
+
ost::MutexLock lock (_mutex);
std::string pcmp = buildDeviceTopo (plugin, indexOut, 0);
- std::string pcmc = buildDeviceTopo (PCM_PLUGHW, indexIn, 0);
+ std::string pcmc = buildDeviceTopo (plugin, indexIn, 0);
_converter = new SamplerateConverter (_audioSampleRate, _frameSize);
- return open_device (pcmp, pcmc, stream);
+ // open_device (pcmp, pcmc, stream);
+ return true; // open_device (pcmp, pcmc, stream);
}
void
AlsaLayer::startStream (void)
{
- _debug ("Start ALSA streams\n");
+ _debug ("AlsaLayer:: startStream\n");
+
+ std::string pcmp = buildDeviceTopo (_audioPlugin, _indexOut, 0);
+ std::string pcmc = buildDeviceTopo (_audioPlugin, _indexIn, 0);
+
+ if(!is_playback_open()){
+ open_device (pcmp, pcmc, SFL_PCM_PLAYBACK);
+ }
+
+ if(!is_capture_open()){
+ open_device (pcmp, pcmc, SFL_PCM_CAPTURE);
+ }
+
prepareCaptureStream ();
+ preparePlaybackStream ();
startCaptureStream ();
startPlaybackStream ();
_urgentRingBuffer.flush();
- _mainBuffer.flush();
+ _mainBuffer.flushAllBuffers();
_mainBuffer.flushDefault();
+ if(_audioThread == NULL) {
+ try {
+ _debug("Start Audio Thread\n");
+ _audioThread = new AudioThread (this);
+ _audioThread->start();
+ } catch (...) {
+ _debugException ("Fail to start audio thread\n");
+ }
+ }
+
}
void
AlsaLayer::stopStream (void)
{
- _debug ("AlsaLayer::stopStream :: Stop ALSA streams\n");
- stopCaptureStream ();
- //stopPlaybackStream ();
+ _debug ("AlsaLayer:: stopStream\n");
+
+ try {
+ /* Stop the audio thread first */
+ if (_audioThread) {
+ _debug("Stop Audio Thread\n");
+ delete _audioThread;
+ _audioThread=NULL;
+ }
+ } catch (...) {
+ _debugException ("! ARTP Exception: when stopping audiortp\n");
+ throw;
+ }
+
+ closeCaptureStream ();
+ closePlaybackStream ();
/* Flush the ring buffers */
flushUrgent ();
@@ -227,38 +269,52 @@ void AlsaLayer::stopCaptureStream (void)
int err;
if (_CaptureHandle) {
- err = snd_pcm_drop (_CaptureHandle);
-
- stop_capture ();
+ _debug("AlsaLayer:: stop Alsa capture\n");
+ if((err = snd_pcm_drop (_CaptureHandle)) < 0)
+ _debug("AlsaLayer:: Error stopping ALSA capture: %s\n", snd_strerror(err));
+ else
+ stop_capture ();
}
}
void AlsaLayer::closeCaptureStream (void)
{
+ int err;
+
if (is_capture_prepared() == true && is_capture_running() == true)
stopCaptureStream ();
- if (is_capture_open())
- snd_pcm_close (_CaptureHandle);
-
- close_capture ();
+ if (is_capture_open()) {
+ _debug("AlsaLayer:: close ALSA capture\n");
+ if ((err = snd_pcm_close (_CaptureHandle)) < 0)
+ _debug("Error closing ALSA capture: %s\n", snd_strerror(err));
+ else
+ close_capture ();
+ }
}
void AlsaLayer::startCaptureStream (void)
{
- if (_CaptureHandle) {
- _debug ("Start the capture\n");
- snd_pcm_start (_CaptureHandle);
- start_capture();
+ int err;
+
+ if (_CaptureHandle && !is_capture_running()) {
+ _debug ("AlsaLayer:: start ALSA capture\n");
+ if((err = snd_pcm_start (_CaptureHandle)) < 0)
+ _debug("Error starting ALSA capture: %s\n", snd_strerror(err));
+ else
+ start_capture();
}
}
void AlsaLayer::prepareCaptureStream (void)
{
- if (is_capture_open()) {
- if (snd_pcm_prepare (_CaptureHandle) < 0)
- _debug ("");
+ int err;
+
+ if (is_capture_open() && !is_capture_prepared()) {
+ _debug("AlsaLayer:: prepare ALSA capture\n");
+ if ((err = snd_pcm_prepare (_CaptureHandle)) < 0)
+ _debug ("Error preparing ALSA capture: %s\n", snd_strerror(err));
else
prepare_capture ();
}
@@ -266,38 +322,57 @@ void AlsaLayer::prepareCaptureStream (void)
void AlsaLayer::stopPlaybackStream (void)
{
- if (_PlaybackHandle) {
- snd_pcm_drop (_PlaybackHandle);
- stop_playback ();
+ int err;
+
+ if (_PlaybackHandle && is_playback_running()) {
+ _debug("AlsaLayer:: stop ALSA playback\n");
+ if((err = snd_pcm_drop (_PlaybackHandle)) < 0)
+ _debug("Error stopping ALSA playback: %s\n", snd_strerror(err));
+ else
+ stop_playback ();
}
}
void AlsaLayer::closePlaybackStream (void)
{
+ int err;
+
if (is_playback_prepared() == true && is_playback_running() == true)
stopPlaybackStream ();
- if (is_playback_open())
- snd_pcm_close (_PlaybackHandle);
-
- close_playback ();
+ if (is_playback_open()) {
+ _debug("AlsaLayer:: close ALSA playback\n");
+ if ((err = snd_pcm_close (_PlaybackHandle)) < 0)
+ _debug("Error closing ALSA playback: %s\n", snd_strerror(err));
+ else
+ close_playback ();
+ }
}
void AlsaLayer::startPlaybackStream (void)
{
- if (_PlaybackHandle) {
- snd_pcm_start (_PlaybackHandle);
- start_playback();
+ int err;
+
+ if (_PlaybackHandle && !is_playback_running()) {
+ _debug ("AlsaLayer:: start ALSA playback\n");
+ if ((err = snd_pcm_start (_PlaybackHandle)) < 0)
+ _debug("Error starting ALSA playback: %s\n", snd_strerror(err));
+ else
+ start_playback();
}
}
void AlsaLayer::preparePlaybackStream (void)
{
- if (is_playback_open()) {
- if (snd_pcm_prepare (_PlaybackHandle) < 0) _debug ("Error preparing the device\n");
+ int err;
- prepare_playback ();
+ if (is_playback_open() && !is_playback_prepared()) {
+ _debug("AlsaLayer:: prepare playback stream\n");
+ if ((err = snd_pcm_prepare (_PlaybackHandle)) < 0)
+ _debug ("Error preparing the device: %s\n", snd_strerror(err));
+ else
+ prepare_playback ();
}
}
@@ -431,6 +506,8 @@ AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, int flag)
int err;
if (flag == SFL_PCM_BOTH || flag == SFL_PCM_PLAYBACK) {
+
+ _debug("AlsaLayer:: open playback device\n");
// if((err = snd_pcm_open(&_PlaybackHandle, pcm_p.c_str(), SND_PCM_STREAM_PLAYBACK, 0 )) < 0){
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());
@@ -450,6 +527,8 @@ AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, int flag)
}
if (flag == SFL_PCM_BOTH || flag == SFL_PCM_CAPTURE) {
+
+ _debug("AlsaLayer:: open capture device\n");
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);
@@ -466,15 +545,17 @@ AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, int flag)
open_capture ();
- prepare_capture ();
+ // prepare_capture ();
}
/* Start the secondary audio thread for callbacks */
+ /*
try {
_audioThread->start();
} catch (...) {
_debugException ("Fail to start audio thread\n");
}
+ */
return true;
}
@@ -570,6 +651,8 @@ AlsaLayer::read (void* buffer, int toCopy)
void
AlsaLayer::handle_xrun_capture (void)
{
+ _debugAlsa("handle_xrun_capture\n");
+
snd_pcm_status_t* status;
snd_pcm_status_alloca (&status);
@@ -588,6 +671,8 @@ AlsaLayer::handle_xrun_capture (void)
void
AlsaLayer::handle_xrun_playback (void)
{
+ _debugAlsa("AlsaLayer:: handle_xrun_playback\n");
+
int state;
snd_pcm_status_t* status;
snd_pcm_status_alloca (&status);
@@ -729,6 +814,7 @@ void AlsaLayer::audioCallback (void)
int toGet, urgentAvailBytes, normalAvailBytes, maxBytes;
unsigned short spkrVolume, micVolume;
AudioLoop *tone;
+ AudioLoop *file_tone;
SFLDataFormat *out;
SFLDataFormat *rsmpl_out;
@@ -759,8 +845,10 @@ void AlsaLayer::audioCallback (void)
} else {
tone = _manager->getTelephoneTone();
- toGet = 940 ;
- maxBytes = toGet * sizeof (SFLDataFormat) ;
+ file_tone = _manager->getTelephoneFile();
+
+ toGet = framesPerBufferAlsa;
+ maxBytes = toGet * sizeof (SFLDataFormat);
if (tone != 0) {
@@ -771,10 +859,10 @@ void AlsaLayer::audioCallback (void)
free(out);
out = 0;
- } else if ( (tone=_manager->getTelephoneFile()) != 0) {
+ } else if (file_tone != 0) {
out = (SFLDataFormat*) malloc (maxBytes * sizeof (SFLDataFormat));
- tone->getNext (out, toGet, spkrVolume);
+ file_tone->getNext(out, toGet, spkrVolume);
write (out , maxBytes);
free(out);
@@ -822,7 +910,6 @@ void AlsaLayer::audioCallback (void)
// Do sample rate conversion
int nb_sample_down = toGet / sizeof(SFLDataFormat);
-
int nbSample = _converter->upsampleData((SFLDataFormat*)out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, nb_sample_down);
@@ -840,7 +927,11 @@ void AlsaLayer::audioCallback (void)
} else {
- bzero (out, maxNbBytesToGet);
+ if((tone == 0) && (file_tone == 0)) {
+
+ bzero (out, maxNbBytesToGet);
+ write (out, maxNbBytesToGet);
+ }
}
_urgentRingBuffer.Discard (toGet);
@@ -865,11 +956,11 @@ void AlsaLayer::audioCallback (void)
{
micAvailBytes = snd_pcm_avail_update(_CaptureHandle);
-
+ // _debug("micAvailBytes %i\n", micAvailBytes);
if(micAvailBytes > 0)
{
micAvailPut = _mainBuffer.availForPut();
- toPut = (micAvailBytes <= micAvailPut) ? micAvailBytes : micAvailPut;
+ toPut = (micAvailBytes <= framesPerBufferAlsa) ? micAvailBytes : framesPerBufferAlsa;
in = (SFLDataFormat*)malloc(toPut * sizeof(SFLDataFormat));
toPut = read (in, toPut* sizeof(SFLDataFormat));
@@ -886,6 +977,7 @@ void AlsaLayer::audioCallback (void)
int nbSample = toPut / sizeof(SFLDataFormat);
int nb_sample_up = nbSample;
+ // _debug("nb_sample_up %i\n", nb_sample_up);
nbSample = _converter->downsampleData ((SFLDataFormat*)in, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, nb_sample_up);
dcblocker->filter_signal(rsmpl_out, nbSample);
diff --git a/sflphone-common/src/audio/alsa/alsalayer.h b/sflphone-common/src/audio/alsa/alsalayer.h
index e1d4b189b78d3b79c7ae5194a3751e04fc4e7044..1f5167ede252b9516c55df316e10763c3dfcd036 100644
--- a/sflphone-common/src/audio/alsa/alsalayer.h
+++ b/sflphone-common/src/audio/alsa/alsalayer.h
@@ -265,13 +265,12 @@ class AlsaLayer : public AudioLayer {
bool _is_open_capture;
bool _trigger_request;
- AudioThread *_audioThread;
+ AudioThread* _audioThread;
/** Sample rate converter object */
- SamplerateConverter * _converter;
-
- int _converterSamplingRate;
+ SamplerateConverter* _converter;
+ // Allpass filter to remove DC offset
DcBlocker* dcblocker;
};
diff --git a/sflphone-common/src/audio/audiolayer.cpp b/sflphone-common/src/audio/audiolayer.cpp
index 9f1119ff694baf871e77d9fa9c17b4b13818a667..8966eb6aba086a30183d9dfa048ec3d42f5350f8 100644
--- a/sflphone-common/src/audio/audiolayer.cpp
+++ b/sflphone-common/src/audio/audiolayer.cpp
@@ -24,13 +24,13 @@ void AudioLayer::flushMain (void)
ost::MutexLock guard (_mutex);
// should pass call id
- _mainBuffer.flush();
+ _mainBuffer.flushAllBuffers();
}
void AudioLayer::flushUrgent (void)
{
ost::MutexLock guard (_mutex);
- _urgentRingBuffer.flush();
+ _urgentRingBuffer.flushAll();
}
void AudioLayer::flushMic (void)
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpSession.h b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
index bed84b1c3d3956e64170e2e2507f2dc14e0f9967..3ac8916b3aa4dbb20093b47d40500406647156ec 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpSession.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpSession.h
@@ -498,10 +498,12 @@ namespace sfl {
}
_ca->setRecordingSmplRate(_audiocodec->getClockRate());
-
+
_audiolayer->startStream();
static_cast<D*>(this)->startRunning();
+ _audiolayer->flushUrgent();
+
_debug ("Entering RTP mainloop for callid %s\n",_ca->getCallId().c_str());
while (!testCancel()) {
diff --git a/sflphone-common/src/audio/pulseaudio/audiostream.cpp b/sflphone-common/src/audio/pulseaudio/audiostream.cpp
index e30f40f78b200cdf0845cc5394b8c709a2a7af6e..deaab30f5b545228952e237fd4ba8c52b43f8ead 100644
--- a/sflphone-common/src/audio/pulseaudio/audiostream.cpp
+++ b/sflphone-common/src/audio/pulseaudio/audiostream.cpp
@@ -20,7 +20,7 @@
#include <audiostream.h>
#include "pulselayer.h"
-static pa_channel_map channel_map ;
+static pa_channel_map channel_map;
AudioStream::AudioStream (PulseLayerType * driver)
@@ -31,7 +31,7 @@ AudioStream::AudioStream (PulseLayerType * driver)
_volume(),
flag (PA_STREAM_AUTO_TIMING_UPDATE),
sample_spec(),
- _mainloop (driver->mainloop)
+ _mainloop (driver->mainloop)
{
sample_spec.format = PA_SAMPLE_S16LE;
sample_spec.rate = 44100;
@@ -99,7 +99,7 @@ AudioStream::disconnectStream (void)
{
_debug ("Destroy audio streams\n");
- pa_threaded_mainloop_lock (_mainloop);
+ // pa_threaded_mainloop_lock (_mainloop);
if (_audiostream) {
pa_stream_disconnect (_audiostream);
@@ -107,7 +107,7 @@ AudioStream::disconnectStream (void)
_audiostream = NULL;
}
- pa_threaded_mainloop_unlock (_mainloop);
+ // pa_threaded_mainloop_unlock (_mainloop);
return true;
}
@@ -136,7 +136,6 @@ AudioStream::stream_state_callback (pa_stream* s, void* user_data)
break;
case PA_STREAM_READY:
-
_debug ("Stream successfully created, connected to %s\n", pa_stream_get_device_name (s));
// pa_stream_cork( s, 0, NULL, NULL);
break;
@@ -187,9 +186,9 @@ AudioStream::createStream (pa_context* c)
if (_streamType == PLAYBACK_STREAM) {
attributes->maxlength = 16000;
attributes->tlength = 2048;
- attributes->prebuf = 1024;
+ attributes->prebuf = 2048;
attributes->minreq = 1024;
- attributes->fragsize = 1024;
+ attributes->fragsize = 4096;
pa_stream_connect_playback( s , NULL , attributes, PA_STREAM_INTERPOLATE_TIMING, &_volume, NULL);
// pa_stream_connect_playback (s , NULL , attributes, PA_STREAM_START_CORKED, &_volume, NULL);
} else if (_streamType == CAPTURE_STREAM) {
diff --git a/sflphone-common/src/audio/pulseaudio/audiostream.h b/sflphone-common/src/audio/pulseaudio/audiostream.h
index 3f6e25c66c27fb7221ea9740e4a1434432e121c0..40fa26e50e663300876f63478fcbe2d93c9941cc 100644
--- a/sflphone-common/src/audio/pulseaudio/audiostream.h
+++ b/sflphone-common/src/audio/pulseaudio/audiostream.h
@@ -183,6 +183,8 @@ class AudioStream {
ost::Mutex _mutex;
+ bool _stream_is_ready;
+
};
#endif // _AUDIO_STREAM_H
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
index cea5e4eeccff50329d87b621a2f4eef1561f00da..2ce375a3ca788d6e4d486fe37dd178190b3ed6cf 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
@@ -25,7 +25,7 @@ int framesPerBuffer = 2048;
static void playback_callback (pa_stream* s, size_t bytes, void* userdata)
{
- _debug("playback_callback\n");
+ // _debug("playback_callback\n");
assert (s && bytes);
assert (bytes > 0);
@@ -112,10 +112,12 @@ PulseLayer::closeLayer (void)
}
// playback->disconnectStream();
- closePlaybackStream();
+ // closePlaybackStream();
// record->disconnectStream();
- closeCaptureStream();
+ // closeCaptureStream();
+
+ disconnectAudioStream();
if (context) {
pa_context_disconnect (context);
@@ -154,6 +156,8 @@ PulseLayer::connectPulseAudioServer (void)
pa_threaded_mainloop_unlock (m);
+ _urgentRingBuffer.flushAll();
+
//serverinfo();
//muteAudioApps(99);
_debug ("Context creation done\n");
@@ -189,15 +193,15 @@ void PulseLayer::context_state_callback (pa_context* c, void* user_data)
default:
_debug (" Error : %s\n" , pa_strerror (pa_context_errno (c)));
- pulse->disconnectPulseAudioServer();
+ pulse->disconnectAudioStream();
exit (0);
break;
}
}
-bool PulseLayer::disconnectPulseAudioServer (void)
+bool PulseLayer::disconnectAudioStream (void)
{
- _debug (" PulseLayer::disconnectPulseAudioServer( void ) \n");
+ _debug (" PulseLayer::disconnectAudioStream( void ) \n");
closePlaybackStream();
@@ -212,7 +216,7 @@ bool PulseLayer::disconnectPulseAudioServer (void)
bool PulseLayer::createStreams (pa_context* c)
{
- _debug ("--------------------------------- PulseLayer::createStreams ---------------------\n");
+ _debug ("PulseLayer::createStreams\n");
PulseLayerType * playbackParam = new PulseLayerType();
playbackParam->context = c;
@@ -246,6 +250,8 @@ bool PulseLayer::createStreams (pa_context* c)
pa_threaded_mainloop_signal (m , 0);
+ _urgentRingBuffer.flushAll();
+
return true;
}
@@ -253,42 +259,23 @@ bool PulseLayer::createStreams (pa_context* c)
bool PulseLayer::openDevice (int indexIn UNUSED, int indexOut UNUSED, int sampleRate, int frameSize , int stream UNUSED, std::string plugin UNUSED)
{
-
- _debug ("PulseLayer::openDevice \n");
_audioSampleRate = sampleRate;
_frameSize = frameSize;
- /*
- m = pa_threaded_mainloop_new();
- assert (m);
- if (pa_threaded_mainloop_start (m) < 0) {
- _debug ("Failed starting the mainloop\n");
- }
- */
-
- /*
- // Instanciate a context
- if (! (context = pa_context_new (pa_threaded_mainloop_get_api (m) , "SFLphone")))
- _debug ("Error while creating the context\n");
-
- assert (context);
- */
-
- // connectPulseAudioServer();
-
- // startStream();
+ _urgentRingBuffer.flushAll();
_converter = new SamplerateConverter (_audioSampleRate, _frameSize);
- _debug ("Connection Done!! \n");
-
return true;
}
void PulseLayer::closeCaptureStream (void)
{
if (record) {
+
+ pa_threaded_mainloop_lock (m);
delete record;
+ pa_threaded_mainloop_unlock (m);
record=NULL;
}
}
@@ -296,7 +283,10 @@ void PulseLayer::closeCaptureStream (void)
void PulseLayer::closePlaybackStream (void)
{
if (playback) {
+
+ pa_threaded_mainloop_lock (m);
delete playback;
+ pa_threaded_mainloop_unlock (m);
playback=NULL;
}
}
@@ -323,9 +313,11 @@ void PulseLayer::startStream (void)
if(!is_started) {
- _debug ("------------------------ PulseLayer::Start stream ---------------\n");
+ _debug ("PulseLayer::Start Stream\n");
if (!m) {
+
+ _debug("Creating PulseAudio MainLoop\n");
m = pa_threaded_mainloop_new();
assert (m);
@@ -336,28 +328,22 @@ void PulseLayer::startStream (void)
if (!context) {
+ _debug("Creating new PulseAudio Context\n");
+ pa_threaded_mainloop_lock (m);
// Instanciate a context
if (! (context = pa_context_new (pa_threaded_mainloop_get_api (m) , "SFLphone")))
_debug ("Error while creating the context\n");
+ pa_threaded_mainloop_unlock (m);
assert (context);
}
_urgentRingBuffer.flush();
-
_mainBuffer.flushAllBuffers();
+ // Create Streams
connectPulseAudioServer();
- // pa_threaded_mainloop_lock (m);
-
- // pa_stream_cork (playback->pulseStream(), 0, NULL, NULL);
- // pa_stream_cork (record->pulseStream(), 0, NULL, NULL);
-
- // pa_threaded_mainloop_unlock (m);
-
- // createStreams(context);
-
is_started = true;
}
@@ -369,32 +355,30 @@ PulseLayer::stopStream (void)
if(is_started) {
- _debug ("------------------------ PulseLayer::Stop stream ---------------\n");
+ _debug ("PulseLayer::Stop Audio Stream\n");
pa_stream_flush (playback->pulseStream(), NULL, NULL);
pa_stream_flush (record->pulseStream(), NULL, NULL);
- flushMic();
- flushMain();
- flushUrgent();
-
- // closeCaptureStream();
- // closePlaybackStream();
+ disconnectAudioStream();
- disconnectPulseAudioServer();
+ if (m) {
+ pa_threaded_mainloop_stop (m);
+ }
- // pa_threaded_mainloop_lock (m);
-
- // pa_stream_cork (playback->pulseStream(), 1, NULL, NULL);
- // pa_stream_cork (record->pulseStream(), 1, NULL, NULL);
- // pa_threaded_mainloop_unlock (m);
+ _debug("Disconnecting PulseAudio context\n");
if (context) {
+
+ pa_threaded_mainloop_lock (m);
pa_context_disconnect (context);
pa_context_unref (context);
+ pa_threaded_mainloop_unlock (m);
context = NULL;
}
+ _debug("Freeing Pulseaudio mainloop\n");
+
if (m) {
pa_threaded_mainloop_free (m);
m = NULL;
@@ -428,6 +412,8 @@ void PulseLayer::processPlaybackData (void)
// Handle the data for the speakers
if ( playback &&(playback->pulseStream()) && (pa_stream_get_state (playback->pulseStream()) == PA_STREAM_READY)) {
+ // _debug("PulseLayer::processPlaybackData()\n");
+
// If the playback buffer is full, we don't overflow it; wait for it to have free space
if (pa_stream_writable_size (playback->pulseStream()) == 0)
return;
@@ -462,12 +448,12 @@ void PulseLayer::writeToSpeaker (void)
if (urgentAvailBytes > 0) {
- // Urgent data (dtmf, incoming call signal) come first.
- //_debug("Play urgent!: %i\e" , urgentAvail);
+
toGet = (urgentAvailBytes < (int) (framesPerBuffer * sizeof (SFLDataFormat))) ? urgentAvailBytes : framesPerBuffer * sizeof (SFLDataFormat);
out = (SFLDataFormat*) pa_xmalloc (toGet * sizeof (SFLDataFormat));
_urgentRingBuffer.Get (out, toGet, 100);
pa_stream_write (playback->pulseStream(), out, toGet, NULL, 0, PA_SEEK_RELATIVE);
+
// Consume the regular one as well (same amount of bytes)
_mainBuffer.discard (toGet);
@@ -476,26 +462,35 @@ void PulseLayer::writeToSpeaker (void)
} else {
AudioLoop* tone = _manager->getTelephoneTone();
+ AudioLoop* file_tone = _manager->getTelephoneFile();
if (tone != 0) {
- toGet = framesPerBuffer;
- out = (SFLDataFormat*) pa_xmalloc (toGet * sizeof (SFLDataFormat));
- tone->getNext (out, toGet , 100);
- pa_stream_write (playback->pulseStream(), out, toGet * sizeof (SFLDataFormat), NULL, 0, PA_SEEK_RELATIVE);
+ if (playback->getStreamState() == PA_STREAM_READY)
+ {
+
+ toGet = framesPerBuffer;
+ out = (SFLDataFormat*) pa_xmalloc (toGet * sizeof (SFLDataFormat));
+ tone->getNext (out, toGet , 100);
+ pa_stream_write (playback->pulseStream(), out, toGet * sizeof (SFLDataFormat), NULL, 0, PA_SEEK_RELATIVE);
- pa_xfree (out);
+ pa_xfree (out);
+ }
}
- if ( (tone=_manager->getTelephoneFile()) != 0) {
+ if (file_tone != 0) {
+
+ if (playback->getStreamState() == PA_STREAM_READY)
+ {
- toGet = framesPerBuffer;
- toPlay = ( (int) (toGet * sizeof (SFLDataFormat)) > framesPerBuffer) ? framesPerBuffer : toGet * sizeof (SFLDataFormat) ;
- out = (SFLDataFormat*) pa_xmalloc (toPlay);
- tone->getNext (out, toPlay/2 , 100);
- pa_stream_write (playback->pulseStream(), out, toPlay, NULL, 0, PA_SEEK_RELATIVE);
+ toGet = framesPerBuffer;
+ toPlay = ( (int) (toGet * sizeof (SFLDataFormat)) > framesPerBuffer) ? framesPerBuffer : toGet * sizeof (SFLDataFormat);
+ out = (SFLDataFormat*) pa_xmalloc (toPlay);
+ file_tone->getNext(out, toPlay/2 , 100);
+ pa_stream_write (playback->pulseStream(), out, toPlay, NULL, 0, PA_SEEK_RELATIVE);
- pa_xfree (out);
+ pa_xfree (out);
+ }
} else {
@@ -549,10 +544,11 @@ void PulseLayer::writeToSpeaker (void)
} else {
- _debug("send zeros......................\n");
+ if((tone == 0) && (file_tone == 0)) {
- bzero (out, maxNbBytesToGet);
- pa_stream_write (playback->pulseStream(), out, maxNbBytesToGet, NULL, 0, PA_SEEK_RELATIVE);
+ bzero (out, maxNbBytesToGet);
+ pa_stream_write (playback->pulseStream(), out, maxNbBytesToGet, NULL, 0, PA_SEEK_RELATIVE);
+ }
}
_urgentRingBuffer.Discard(toGet);
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.h b/sflphone-common/src/audio/pulseaudio/pulselayer.h
index 151fdec00591edae24a8bc1c52d408792663f4bc..e4a4981e570ff6af9a774e82e9a7b396082d270a 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.h
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.h
@@ -172,7 +172,7 @@ class PulseLayer : public AudioLayer {
/**
* Close the connection with the local pulseaudio server
*/
- bool disconnectPulseAudioServer( void );
+ bool disconnectAudioStream( void );
/**
* Get some information about the pulseaudio server
diff --git a/sflphone-common/src/managerimpl.cpp b/sflphone-common/src/managerimpl.cpp
index 4055a143f154e96379c5d56054ac22e76aa11eed..5491f2fe9d44ecc3c7ad645ba6bda9daf1dc3a07 100644
--- a/sflphone-common/src/managerimpl.cpp
+++ b/sflphone-common/src/managerimpl.cpp
@@ -372,6 +372,7 @@ ManagerImpl::answerCall (const CallID& call_id)
}
+
if (!getAccountLink (account_id)->answer (call_id)) {
// error when receiving...
removeCallAccount (call_id);
@@ -427,10 +428,12 @@ ManagerImpl::hangupCall (const CallID& call_id)
int nbCalls = getCallList().size();
+ // _debug("nbCalls: %i\n", nbCalls);
+
audiolayer = getAudioDriver();
// stop streams
- if (! (nbCalls >= 1))
+ if (nbCalls <= 1)
{
_debug(" hangupCall: stop audio stream, ther is only %i call(s) remaining\n", nbCalls);
audiolayer->stopStream();
@@ -775,7 +778,20 @@ ManagerImpl::refuseCall (const CallID& id)
AccountID accountid;
bool returnValue;
- stopTone (true);
+ stopTone (false);
+
+
+ int nbCalls = getCallList().size();
+
+ // AudioLayer* audiolayer = getAudioDriver();
+
+ if (nbCalls <= 1)
+ {
+ _debug(" hangupCall: stop audio stream, ther is only %i call(s) remaining\n", nbCalls);
+
+ AudioLayer* audiolayer = getAudioDriver();
+ audiolayer->stopStream();
+ }
/* Direct IP to IP call */
@@ -807,6 +823,8 @@ ManagerImpl::refuseCall (const CallID& id)
switchCall ("");
}
+
+
return returnValue;
}
@@ -1683,6 +1701,8 @@ ManagerImpl::playDtmf (char code, bool isTalking)
// Put buffer to urgentRingBuffer
// put the size in bytes...
// so size * 1 channel (mono) * sizeof (bytes for the data)
+ audiolayer->flushUrgent();
+
audiolayer->startStream();
audiolayer->putUrgent (buf, size * sizeof (SFLDataFormat));
}
@@ -1895,6 +1915,18 @@ ManagerImpl::peerHungupCall (const CallID& call_id)
if (isCurrentCall(call_id))
{
stopTone (true);
+
+ int nbCalls = getCallList().size();
+
+ // stop streams
+ if (nbCalls <= 1)
+ {
+ _debug(" hangupCall: stop audio stream, ther is only %i call(s) remaining\n", nbCalls);
+
+ AudioLayer* audiolayer = getAudioDriver();
+ audiolayer->stopStream();
+ }
+
switchCall ("");
}
}
@@ -1998,6 +2030,12 @@ bool ManagerImpl::playATone (Tone::TONEID toneId)
audiolayer = getAudioDriver();
+
+ if (audiolayer) {
+ audiolayer->startStream();
+ audiolayer->flushUrgent();
+ }
+
if (_telephoneTone != 0) {
_toneMutex.enterMutex();
_telephoneTone->setCurrentTone (toneId);
@@ -2007,6 +2045,7 @@ bool ManagerImpl::playATone (Tone::TONEID toneId)
nbSamples = audioloop->getSize();
SFLDataFormat buf[nbSamples];
+
if (audiolayer) {
audiolayer->putUrgent (buf, nbSamples);
} else
@@ -2130,12 +2169,8 @@ ManagerImpl::ringtone()
_audiofile.start();
_toneMutex.leaveMutex();
- if (CHECK_INTERFACE (layer, ALSA)) {
- //ringback();
-
- } else {
- audiolayer->startStream();
- }
+ audiolayer->startStream();
+
} else {
ringback();
}
@@ -2186,6 +2221,7 @@ void ManagerImpl::notificationIncomingCall (void)
SFLDataFormat buf[nbSampling];
tone.getNext (buf, tone.getSize());
/* Put the data in the urgent ring buffer */
+ audiolayer->flushUrgent();
audiolayer->putUrgent (buf, sizeof (SFLDataFormat) *nbSampling);
}
}
diff --git a/sflphone-common/test/audiolayerTest.cpp b/sflphone-common/test/audiolayerTest.cpp
index 92648bc20166977365ad0d5b0a34b7954f9e575e..1cdb792598329c7cbf492b68d5bc4cd4435a89c3 100644
--- a/sflphone-common/test/audiolayerTest.cpp
+++ b/sflphone-common/test/audiolayerTest.cpp
@@ -169,7 +169,7 @@ void AudioLayerTest::testPulseConnect()
CPPUNIT_ASSERT (_pulselayer->getRecordStream()->getStreamState() == 1);
// usleep(1000000);
- CPPUNIT_ASSERT (_pulselayer->disconnectPulseAudioServer() == true);
+ CPPUNIT_ASSERT (_pulselayer->disconnectAudioStream() == true);
}