diff --git a/sflphone-common/src/audio/alsa/alsalayer.cpp b/sflphone-common/src/audio/alsa/alsalayer.cpp
index b7f95cb122af7084dacb46057188ec32fc135be3..2602b3f4f00552a3611788577c49c3658667187a 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);
@@ -122,6 +125,8 @@ 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);
@@ -137,6 +142,18 @@ void
AlsaLayer::startStream (void)
{
_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 ();
_debug("------------------------------\n");
preparePlaybackStream ();
@@ -150,14 +167,39 @@ AlsaLayer::startStream (void)
_mainBuffer.flush();
_mainBuffer.flushDefault();
+
+ _debug("audiothread %p\n", _audioThread);
+ 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\n");
- stopCaptureStream ();
- stopPlaybackStream ();
+
+ 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 ();
@@ -245,12 +287,11 @@ void AlsaLayer::closeCaptureStream (void)
{
int err;
- _debug("AlsaLayer:: close ALSA capture\n");
-
if (is_capture_prepared() == true && is_capture_running() == true)
stopCaptureStream ();
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
@@ -306,6 +347,7 @@ void AlsaLayer::closePlaybackStream (void)
stopPlaybackStream ();
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
@@ -469,6 +511,8 @@ 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());
@@ -488,6 +532,8 @@ 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);
@@ -508,11 +554,13 @@ AlsaLayer::open_device (std::string pcm_p, std::string pcm_c, int flag)
}
/* Start the secondary audio thread for callbacks */
+ /*
try {
_audioThread->start();
} catch (...) {
_debugException ("Fail to start audio thread\n");
}
+ */
return true;
}
@@ -608,6 +656,8 @@ 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);
@@ -626,6 +676,8 @@ AlsaLayer::handle_xrun_capture (void)
void
AlsaLayer::handle_xrun_playback (void)
{
+ _debugAlsa("handle_xrun_playback\n");
+
int state;
snd_pcm_status_t* status;
snd_pcm_status_alloca (&status);
@@ -767,6 +819,7 @@ void AlsaLayer::audioCallback (void)
int toGet, urgentAvailBytes, normalAvailBytes, maxBytes;
unsigned short spkrVolume, micVolume;
AudioLoop *tone;
+ AudioLoop *file_tone;
SFLDataFormat *out;
SFLDataFormat *rsmpl_out;
@@ -797,8 +850,10 @@ 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) {
@@ -809,7 +864,7 @@ 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);
@@ -846,6 +901,8 @@ void AlsaLayer::audioCallback (void)
normalAvailBytes = _mainBuffer.availForGet();
toGet = (normalAvailBytes < (int)maxNbBytesToGet) ? normalAvailBytes : maxNbBytesToGet;
+
+ _debug("maxNbByteToGet %i\n", maxNbBytesToGet);
out = (SFLDataFormat*) malloc (maxNbBytesToGet);
if (normalAvailBytes) {
@@ -860,7 +917,7 @@ void AlsaLayer::audioCallback (void)
// Do sample rate conversion
int nb_sample_down = toGet / sizeof(SFLDataFormat);
-
+ _debug("nb_sample_down %i\n", nb_sample_down);
int nbSample = _converter->upsampleData((SFLDataFormat*)out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, nb_sample_down);
@@ -878,7 +935,12 @@ void AlsaLayer::audioCallback (void)
} else {
- bzero (out, maxNbBytesToGet);
+ if((tone == 0) && (file_tone == 0)) {
+
+ _debug("Write Zeros ......................\n");
+ bzero (out, maxNbBytesToGet);
+ write (out, maxNbBytesToGet);
+ }
}
_urgentRingBuffer.Discard (toGet);
@@ -903,11 +965,11 @@ 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));
@@ -924,6 +986,7 @@ 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;
};