diff --git a/sflphone-common/src/audio/alsa/alsalayer.cpp b/sflphone-common/src/audio/alsa/alsalayer.cpp
index b8389aa4ad85a92d8b41b630d490fb298eeba488..ee5dc273501e0e12738c5f4f44226ba73356981e 100644
--- a/sflphone-common/src/audio/alsa/alsalayer.cpp
+++ b/sflphone-common/src/audio/alsa/alsalayer.cpp
@@ -886,6 +886,7 @@ void AlsaLayer::audioCallback (void)
memset (out, 0, playbackAvailBytes);
if (out) {
+ _debug("Write tone (normal %d) (playbackAvail %d)", normalAvailBytes, playbackAvailBytes);
tone->getNext (out, playbackAvailSmpl, spkrVolume);
write (out , playbackAvailBytes, _PlaybackHandle);
free (out);
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
index e2b30d630cd30753f66bf4f7409cc026d9c026c5..3f17b171d3002742067c34114b0bf0446ca74d00 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
@@ -747,9 +747,9 @@ void PulseLayer::writeToSpeaker (void)
// available bytes to be written in pulseaudio internal buffer
int writeableSize = pa_stream_writable_size (playback->pulseStream());
- if (writeableSize < 0)
+ if (writeableSize < 0) {
_warn ("Audio: playback error : %s", pa_strerror (writeableSize));
-
+ }
if (urgentAvailBytes > writeableSize) {
@@ -758,7 +758,6 @@ void PulseLayer::writeToSpeaker (void)
_urgentRingBuffer.Get (out, writeableSize, 100);
- // spkrFile->write ( (const char *) out, writeableSize);
pa_stream_write (playback->pulseStream(), out, writeableSize, NULL, 0, PA_SEEK_RELATIVE);
pa_xfree (out);
@@ -769,115 +768,88 @@ void PulseLayer::writeToSpeaker (void)
} else {
- // Get ringtone
- AudioLoop* tone = _manager->getTelephoneTone();
-
// We must test if data have been received from network in case of early media
normalAvailBytes = getMainBuffer()->availForGet();
// flush remaining samples in _urgentRingBuffer
flushUrgent();
- if ( (tone != 0) && (normalAvailBytes <= 0)) {
-
- if (playback->getStreamState() == PA_STREAM_READY) {
-
- out = (SFLDataFormat*) pa_xmalloc (writeableSize);
- memset (out, 0, writeableSize);
-
- int copied = tone->getNext (out, writeableSize / sizeof (SFLDataFormat), 100);
-
- //spkrFile->write ( (const char *) out, copied*sizeof (SFLDataFormat));
- pa_stream_write (playback->pulseStream(), out, copied * sizeof (SFLDataFormat), NULL, 0, PA_SEEK_RELATIVE);
-
- pa_xfree (out);
+ int _mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
- }
+ int maxNbBytesToGet = 0;
+ // test if audio resampling is needed
+ if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
+ double upsampleFactor = (double) _mainBufferSampleRate / _audioSampleRate;
+ maxNbBytesToGet = ( (double) writeableSize * upsampleFactor);
} else {
+ maxNbBytesToGet = writeableSize;
+ }
- int _mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
-
- int maxNbBytesToGet = 0;
-
- // test if audio resampling is needed
- if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
-
- // upsamplefactor is used to compute the number of bytes to get in the ring buffer
- double upsampleFactor = (double) _mainBufferSampleRate / _audioSampleRate;
-
- maxNbBytesToGet = ( (double) writeableSize * upsampleFactor);
-
- } else {
-
- // maxNbSamplesToGet = framesPerBuffer;
- maxNbBytesToGet = writeableSize;
-
- }
-
- byteToGet = (normalAvailBytes < (int) (maxNbBytesToGet)) ? normalAvailBytes : maxNbBytesToGet;
-
- if (byteToGet) {
-
- // Sending an odd number of byte breaks the audio!
- // TODO, find out where the problem occurs to get rid of this hack
- if ( (byteToGet%2) != 0)
- byteToGet = byteToGet-1;
-
- out = (SFLDataFormat*) pa_xmalloc (maxNbBytesToGet);
- memset (out, 0, maxNbBytesToGet);
-
- getMainBuffer()->getData (out, byteToGet, 100);
-
- // test if resampling is required
- if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
+ byteToGet = (normalAvailBytes < (int) (maxNbBytesToGet)) ? normalAvailBytes : maxNbBytesToGet;
- SFLDataFormat* rsmpl_out = (SFLDataFormat*) pa_xmalloc (writeableSize);
- memset (out, 0, writeableSize);
+ if (byteToGet) {
- // Do sample rate conversion
- int nb_sample_down = byteToGet / sizeof (SFLDataFormat);
+ // Sending an odd number of byte breaks the audio!
+ if ( (byteToGet%2) != 0) {
+ byteToGet = byteToGet-1;
+ }
- int nbSample = _converter->upsampleData ( (SFLDataFormat*) out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, nb_sample_down);
+ out = (SFLDataFormat*) pa_xmalloc (maxNbBytesToGet);
+ memset (out, 0, maxNbBytesToGet);
- if ( (nbSample*sizeof (SFLDataFormat)) > (unsigned int) writeableSize)
- _warn ("Audio: Error: nbsbyte exceed buffer length");
+ getMainBuffer()->getData (out, byteToGet, 100);
- // spkrFile->write ( (const char *) out, nbSample*sizeof (SFLDataFormat));
- pa_stream_write (playback->pulseStream(), rsmpl_out, nbSample*sizeof (SFLDataFormat), NULL, 0, PA_SEEK_RELATIVE);
+ // test if resampling is required
+ if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
- pa_xfree (rsmpl_out);
+ SFLDataFormat* rsmpl_out = (SFLDataFormat*) pa_xmalloc (writeableSize);
+ memset (out, 0, writeableSize);
- } else {
+ // Do sample rate conversion
+ int nb_sample_down = byteToGet / sizeof (SFLDataFormat);
- // spkrFile->write ( (const char *) out, byteToGet);
- pa_stream_write (playback->pulseStream(), out, byteToGet, NULL, 0, PA_SEEK_RELATIVE);
+ int nbSample = _converter->upsampleData ( (SFLDataFormat*) out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, nb_sample_down);
+ if ( (nbSample*sizeof (SFLDataFormat)) > (unsigned int) writeableSize) {
+ _warn ("Audio: Error: nbsbyte exceed buffer length");
+ }
- }
+ // write resample data
+ pa_stream_write (playback->pulseStream(), rsmpl_out, nbSample*sizeof (SFLDataFormat), NULL, 0, PA_SEEK_RELATIVE);
- pa_xfree (out);
+ // free resampling buffer
+ pa_xfree (rsmpl_out);
} else {
+ // write origin data
+ pa_stream_write (playback->pulseStream(), out, byteToGet, NULL, 0, PA_SEEK_RELATIVE);
+ }
- if (tone == 0) {
+ // free audio buffer used to get data from
+ pa_xfree (out);
- SFLDataFormat* zeros = (SFLDataFormat*) pa_xmalloc (writeableSize);
- bzero (zeros, writeableSize);
+ } else {
- // spkrFile->write ( (const char *) zeros, writeableSize);
- pa_stream_write (playback->pulseStream(), zeros, writeableSize, NULL, 0, PA_SEEK_RELATIVE);
+ // Get ringtone
+ AudioLoop *fileToPlay = _manager->getTelephoneTone();
- pa_xfree (zeros);
+ // If audio stream is open and there is realy nothing to play (i.e. audio voce is late and audio layer is underrun)
+ if (fileToPlay == NULL) {
- }
- }
+ SFLDataFormat* zeros = (SFLDataFormat*) pa_xmalloc (writeableSize);
+ bzero (zeros, writeableSize);
+ _debug("write zeros");
+ pa_stream_write (playback->pulseStream(), zeros, writeableSize, NULL, 0, PA_SEEK_RELATIVE);
- _urgentRingBuffer.Discard (byteToGet);
+ pa_xfree (zeros);
+ }
}
+ _urgentRingBuffer.Discard (byteToGet);
+
}
}
@@ -910,13 +882,9 @@ void PulseLayer::readFromMic (void)
nbSample = _converter->downsampleData ( (SFLDataFormat *) data, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, nb_sample_up);
- // captureRsmplFile->write ((const char *)rsmpl_out, nbSample*sizeof(SFLDataFormat));
-
// remove dc offset
_audiofilter->processAudio (rsmpl_out, nbSample*sizeof (SFLDataFormat));
- // captureFilterFile->write ((const char *)rsmpl_out, nbSample*sizeof(SFLDataFormat));
-
getMainBuffer()->putData (rsmpl_out, nbSample*sizeof (SFLDataFormat), 100);
pa_xfree (rsmpl_out);
@@ -930,15 +898,10 @@ void PulseLayer::readFromMic (void)
// remove dc offset
_audiofilter->processAudio ( (SFLDataFormat *) data, filter_out, r);
- // captureFile->write ( (const char *) filter_out, r);
-
getMainBuffer()->putData (filter_out, r, 100);
pa_xfree (filter_out);
}
-
-
-
}
if (pa_stream_drop (record->pulseStream()) < 0) {
@@ -950,41 +913,41 @@ void PulseLayer::readFromMic (void)
void PulseLayer::ringtoneToSpeaker (void)
{
- AudioLoop* file_tone = _manager->getTelephoneFile();
+ AudioLoop* fileToPlay = _manager->getTelephoneFile();
SFLDataFormat* out;
int writableSize = pa_stream_writable_size (ringtone->pulseStream());
- if (file_tone) {
+ if (fileToPlay) {
if (ringtone->getStreamState() != PA_STREAM_READY) {
- _error("PulseAudio: Error: Ringtone stream not in state ready");
- return;
+ _error("PulseAudio: Error: Ringtone stream not in state ready");
+ return;
}
out = (SFLDataFormat *) pa_xmalloc (writableSize);
if(out == NULL) {
- _error("PulseAudio: Error: Could not allocate memory for buffer");
- return;
+ _error("PulseAudio: Error: Could not allocate memory for buffer");
+ return;
}
memset (out, 0, writableSize);
- int copied = file_tone->getNext (out, writableSize/sizeof (SFLDataFormat), 100);
-
+ int copied = fileToPlay->getNext (out, writableSize/sizeof (SFLDataFormat), 100);
if(copied == 0) {
- copied = writableSize/sizeof(SFLDataFormat);
+ copied = writableSize/sizeof(SFLDataFormat);
}
pa_stream_write (ringtone->pulseStream(), out, copied*sizeof (SFLDataFormat), NULL, 0, PA_SEEK_RELATIVE);
pa_xfree (out);
- } else {
+ }
+ else {
if (ringtone->getStreamState() != PA_STREAM_READY) {
- _error("PulseAudio: Error: Ringtone stream not in state ready");
- return;
+ _error("PulseAudio: Error: Ringtone stream not in state ready");
+ return;
}
out = (SFLDataFormat*) pa_xmalloc (writableSize);
@@ -998,8 +961,6 @@ void PulseLayer::ringtoneToSpeaker (void)
pa_xfree (out);
}
-
-
}
static void retrieve_server_info (pa_context *c UNUSED, const pa_server_info *i, void *userdata UNUSED)
diff --git a/sflphone-common/src/audio/sound/audiofile.cpp b/sflphone-common/src/audio/sound/audiofile.cpp
index e49b2947036a22b495d329baf13b6f599cc65d19..b847d6bd2226033f5aeecce25f31cc7534ff7b33 100644
--- a/sflphone-common/src/audio/sound/audiofile.cpp
+++ b/sflphone-common/src/audio/sound/audiofile.cpp
@@ -56,6 +56,8 @@ RawFile::~RawFile()
// load file in mono format
void RawFile::loadFile (const std::string& name, sfl::AudioCodec* codec, unsigned int sampleRate = 8000) throw(AudioFileException)
{
+ _debug("RawFile: Load new file %s", name.c_str());
+
audioCodec = codec;
// if the filename was already load, with the same samplerate
@@ -437,6 +439,8 @@ void WaveFile::openExistingWaveFile (const std::string& fileName, int audioSampl
void WaveFile::loadFile (const std::string& filename, sfl::AudioCodec * /*codec*/, unsigned int sampleRate) throw(AudioFileException)
{
+ _debug("WaveFile: Load new file %s", filename.c_str());
+
try {
openFile (filename, sampleRate);
}
diff --git a/sflphone-common/src/managerimpl.cpp b/sflphone-common/src/managerimpl.cpp
index 215e06ee490eecc569eb2ebb9618a105b3cbcfb4..80d13a72fc1709ac94c8ba3d0858a568b4ae643a 100644
--- a/sflphone-common/src/managerimpl.cpp
+++ b/sflphone-common/src/managerimpl.cpp
@@ -2306,7 +2306,6 @@ void ManagerImpl::ringtone (const AccountID& accountID)
std::string ringchoice;
sfl::AudioCodec *codecForTone;
int layer, samplerate;
- bool loadFile;
_debug ("Manager: Ringtone");
@@ -2358,22 +2357,23 @@ void ManagerImpl::ringtone (const AccountID& accountID)
std::string wave (".wav");
size_t found = ringchoice.find (wave);
- if (found != std::string::npos)
- _audiofile = static_cast<AudioFile *> (new WaveFile());
- else
- _audiofile = static_cast<AudioFile *> (new RawFile());
+ try {
+ if (found != std::string::npos) {
+ _audiofile = static_cast<AudioFile *> (new WaveFile());
+ }
+ else {
+ _audiofile = static_cast<AudioFile *> (new RawFile());
+ }
- _debug ("Manager: ringChoice: %s, codecForTone: %d, samplerate %d", ringchoice.c_str(), codecForTone->getPayloadType(), samplerate);
+ _debug ("Manager: ringChoice: %s, codecForTone: %d, samplerate %d", ringchoice.c_str(), codecForTone->getPayloadType(), samplerate);
- if (_audiofile) {
_audiofile->loadFile (ringchoice, codecForTone, samplerate);
}
+ catch (AudioFileException &e) {
+ _error("Manager: Exception: %s", e.what());
+ }
- _toneMutex.leaveMutex();
-
-
- _toneMutex.enterMutex();
_audiofile->start();
_toneMutex.leaveMutex();