diff --git a/sflphone-common/src/audio/alsa/alsalayer.cpp b/sflphone-common/src/audio/alsa/alsalayer.cpp
index ddd1303f39487f4008a2341c7e5cee6f5a24c446..2938219c8af8e04e96903b9fcd1d4725ad77623a 100644
--- a/sflphone-common/src/audio/alsa/alsalayer.cpp
+++ b/sflphone-common/src/audio/alsa/alsalayer.cpp
@@ -161,8 +161,8 @@ AlsaLayer::startStream (void)
startPlaybackStream ();
_urgentRingBuffer.flush();
- _mainBuffer.flushAllBuffers();
- _mainBuffer.flushDefault();
+ getMainBuffer()->flushAllBuffers();
+ getMainBuffer()->flushDefault();
if(_audioThread == NULL) {
try {
@@ -853,6 +853,10 @@ void AlsaLayer::audioCallback (void)
spkrVolume = _manager->getSpkrVolume();
micVolume = _manager->getMicVolume();
+ /*
+ int writeableSize = snd_pcm_avail_update(_PlaybackHandle);
+ _debug("writeableSize %i\n", writeableSize);
+ */
// AvailForGet tell the number of chars inside the buffer
// framePerBuffer are the number of data for one channel (left)
@@ -871,7 +875,7 @@ void AlsaLayer::audioCallback (void)
out=0;
// Consume the regular one as well (same amount of bytes)
- _mainBuffer.discard (toGet);
+ getMainBuffer()->discard (toGet);
} else {
@@ -924,14 +928,14 @@ void AlsaLayer::audioCallback (void)
maxNbBytesToGet = maxNbSamplesToGet * sizeof(SFLDataFormat);
- normalAvailBytes = _mainBuffer.availForGet();
+ normalAvailBytes = getMainBuffer()->availForGet();
toGet = (normalAvailBytes < (int)maxNbBytesToGet) ? normalAvailBytes : maxNbBytesToGet;
out = (SFLDataFormat*) malloc (maxNbBytesToGet);
if (normalAvailBytes) {
- _mainBuffer.getData(out, toGet, spkrVolume);
+ getMainBuffer()->getData(out, toGet, spkrVolume);
if (_mainBufferSampleRate && ((int)_audioSampleRate != _mainBufferSampleRate)) {
@@ -994,7 +998,7 @@ void AlsaLayer::audioCallback (void)
// _debug("micAvailBytes %i\n", micAvailBytes);
if(micAvailBytes > 0)
{
- micAvailPut = _mainBuffer.availForPut();
+ micAvailPut = getMainBuffer()->availForPut();
toPut = (micAvailBytes <= framesPerBufferAlsa) ? micAvailBytes : framesPerBufferAlsa;
in = (SFLDataFormat*)malloc(toPut * sizeof(SFLDataFormat));
toPut = read (in, toPut* sizeof(SFLDataFormat));
@@ -1017,14 +1021,14 @@ void AlsaLayer::audioCallback (void)
dcblocker->filter_signal(rsmpl_out, nbSample);
- _mainBuffer.putData(rsmpl_out, nbSample * sizeof (SFLDataFormat), 100);
+ getMainBuffer()->putData(rsmpl_out, nbSample * sizeof (SFLDataFormat), 100);
free(rsmpl_out);
rsmpl_out = 0;
} else {
- _mainBuffer.putData(in, toPut, 100);
+ getMainBuffer()->putData(in, toPut, 100);
}
}
free(in); in=0;
diff --git a/sflphone-common/src/audio/audiolayer.cpp b/sflphone-common/src/audio/audiolayer.cpp
index 8966eb6aba086a30183d9dfa048ec3d42f5350f8..0dcace767bab39182c745c4f3ac37ca000bc9fd7 100644
--- a/sflphone-common/src/audio/audiolayer.cpp
+++ b/sflphone-common/src/audio/audiolayer.cpp
@@ -24,7 +24,7 @@ void AudioLayer::flushMain (void)
ost::MutexLock guard (_mutex);
// should pass call id
- _mainBuffer.flushAllBuffers();
+ getMainBuffer()->flushAllBuffers();
}
void AudioLayer::flushUrgent (void)
@@ -36,7 +36,7 @@ void AudioLayer::flushUrgent (void)
void AudioLayer::flushMic (void)
{
ost::MutexLock guard (_mutex);
- _mainBuffer.flushDefault();
+ getMainBuffer()->flushDefault();
}
int AudioLayer::putUrgent (void* buffer, int toCopy)
@@ -60,13 +60,13 @@ int AudioLayer::putMain (void *buffer, int toCopy, CallID call_id)
int a;
ost::MutexLock guard (_mutex);
- a = _mainBuffer.availForPut(call_id);
+ a = getMainBuffer()->availForPut(call_id);
if (a >= toCopy) {
- return _mainBuffer.putData (buffer, toCopy, _defaultVolume, call_id);
+ return getMainBuffer()->putData (buffer, toCopy, _defaultVolume, call_id);
} else {
_debug ("Chopping sound, Ouch! RingBuffer full ?\n");
- return _mainBuffer.putData (buffer, a, _defaultVolume, call_id);
+ return getMainBuffer()->putData (buffer, a, _defaultVolume, call_id);
}
return 0;
diff --git a/sflphone-common/src/audio/audiolayer.h b/sflphone-common/src/audio/audiolayer.h
index 75921da755bb7f9ceb34489f5f2a10be3d85566d..2e3dbee8e2e9eb484ea13f16dadfb3b0c1086653 100644
--- a/sflphone-common/src/audio/audiolayer.h
+++ b/sflphone-common/src/audio/audiolayer.h
@@ -208,7 +208,12 @@ class AudioLayer {
*
* @return MainBuffer* a pointer to the MainBuffer instance
*/
- MainBuffer* getMainBuffer( void ) { return &_mainBuffer; }
+ MainBuffer* getMainBuffer( void ) { return _mainBuffer; }
+
+ /**
+ * Set the mainbuffer once the audiolayer is created
+ */
+ void setMainBuffer( MainBuffer* mainbuffer ) { _mainBuffer = mainbuffer; }
/**
* Default volume for incoming RTP and Urgent sounds.
@@ -256,7 +261,7 @@ class AudioLayer {
* Audio instances must be registered into the MainBuffer and bound together via the ManagerImpl.
*
*/
- MainBuffer _mainBuffer;
+ MainBuffer* _mainBuffer;
/**
* A pointer to the recordable instance (may be a call or a conference)
diff --git a/sflphone-common/src/audio/pulseaudio/audiostream.cpp b/sflphone-common/src/audio/pulseaudio/audiostream.cpp
index 614d2e5c27490bee99880ecae0a9116fd80d610c..565ef8b78349b8a2d5f303e4a59166c05cc61cd2 100644
--- a/sflphone-common/src/audio/pulseaudio/audiostream.cpp
+++ b/sflphone-common/src/audio/pulseaudio/audiostream.cpp
@@ -192,30 +192,23 @@ AudioStream::createStream (pa_context* c)
// parameters are defined as number of bytes
// 2048 bytes (1024 int16) is 20 ms at 44100 Hz
if (_streamType == PLAYBACK_STREAM) {
-
+
+ // 20 ms framesize TODO: take framesize value from config
attributes->maxlength = (uint32_t) -1;
- attributes->tlength = pa_usec_to_bytes(20 * PA_USEC_PER_MSEC, &sample_spec);
+ attributes->tlength = pa_usec_to_bytes(50 * PA_USEC_PER_MSEC, &sample_spec);
attributes->prebuf = (uint32_t) -1;
attributes->minreq = (uint32_t) -1;
attributes->fragsize = (uint32_t) -1;
-
- _debug("tlength: %i\n", pa_usec_to_bytes(20 * PA_USEC_PER_MSEC, &sample_spec));
- /*
- attributes->maxlength = 88200;
- attributes->tlength = 4096;
- attributes->prebuf = 4096;
- attributes->minreq = 2048;
- attributes->fragsize = 4096;
- */
pa_stream_connect_playback( s , NULL , attributes, PA_STREAM_ADJUST_LATENCY, &_volume, NULL);
} else if (_streamType == CAPTURE_STREAM) {
-
+
+ // 20 ms framesize TODO: take framesize value from config
attributes->maxlength = (uint32_t) -1;
attributes->tlength = (uint32_t) -1;
attributes->prebuf = (uint32_t) -1;
attributes->minreq = (uint32_t) -1;
- attributes->fragsize = 4096;
+ attributes->fragsize = pa_usec_to_bytes(50 * PA_USEC_PER_MSEC, &sample_spec);
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
index 4c5dc0e64574bc6636a699e0210fedce817c9654..47eebfe2d9b64302fd083f4e321741d0928cee42 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
@@ -43,18 +43,19 @@ static void capture_callback (pa_stream* s, size_t bytes, void* userdata)
// static_cast<PulseLayer*> (userdata)->processData();
}
-
+/*
static void stream_suspended_callback (pa_stream *s UNUSED, void *userdata UNUSED)
{
_debug("PulseLayer::Stream Suspended\n");
}
+*/
-
+/*
static void stream_moved_callback(pa_stream *s UNUSED, void *userdata UNUSED)
{
_debug("PulseLayer::Stream Moved\n");
}
-
+*/
static void playback_underflow_callback (pa_stream* s, void* userdata UNUSED)
{
@@ -357,14 +358,11 @@ void PulseLayer::startStream (void)
// Create Streams
connectPulseAudioServer();
- // _urgentRingBuffer.flushAll();
- // _mainBuffer.flushAllBuffers();
-
is_started = true;
}
_urgentRingBuffer.flushAll();
- _mainBuffer.flushAllBuffers();
+ getMainBuffer()->flushAllBuffers();
}
@@ -378,12 +376,11 @@ PulseLayer::stopStream (void)
pa_stream_flush (playback->pulseStream(), NULL, NULL);
pa_stream_flush (record->pulseStream(), NULL, NULL);
- disconnectAudioStream();
-
if (m) {
pa_threaded_mainloop_stop (m);
}
+ disconnectAudioStream();
_debug("Disconnecting PulseAudio context\n");
@@ -484,15 +481,9 @@ void PulseLayer::writeToSpeaker (void)
int urgentAvailBytes;
/** Bytes available in the regular ringbuffer ( reserved for voice ) */
int normalAvailBytes;
+ /** Bytes to get in the ring buffer **/
int byteToGet;
- int toPlay;
- // const pa_timing_info* info = pa_stream_get_timing_info(s);
-
- // _debug(" pa write_index: %i", info->write_index);
- // _debug(" pa write_index_corupt (if not 0): %i" info->write_index_corrupt);
- // _debug(" pa read_index: %i", info->read_index);
- // _debug(" pa read_index_corrupt (if not 0): %i", info->read_index_corrupt);
SFLDataFormat* out;// = (SFLDataFormat*)pa_xmalloc(framesPerBuffer);
urgentAvailBytes = _urgentRingBuffer.AvailForGet();
@@ -513,7 +504,7 @@ void PulseLayer::writeToSpeaker (void)
pa_stream_write (playback->pulseStream(), out, writeableSize, pa_xfree, 0, PA_SEEK_RELATIVE);
// Consume the regular one as well (same amount of bytes)
- _mainBuffer.discard (writeableSize);
+ getMainBuffer()->discard (writeableSize);
} else {
@@ -552,22 +543,16 @@ void PulseLayer::writeToSpeaker (void)
} else {
int _mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
- // int maxNbSamplesToGet = 0;
+
int maxNbBytesToGet = 0;
- _debug("--------------- Playback -----------\n");
- _debug("writeableSize: %i\n", writeableSize);
// 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;
- _debug("upsampleFactor: %f\n", upsampleFactor);
- // maxNbSamplesToGet is the number of sample to get in the ring buffer which,
- // once resampled, will not be over the writeableSize
- // maxNbSamplesToGet = (int) ((double) framesPerBuffer * upsampleFactor);
maxNbBytesToGet = ((double) writeableSize * upsampleFactor);
} else {
@@ -577,23 +562,19 @@ void PulseLayer::writeToSpeaker (void)
}
- // maxNbBytesToGet = maxNbSamplesToGet * sizeof(SFLDataFormat);
- _debug("maxNbBytesToGet: %i\n", maxNbBytesToGet);
-
out = (SFLDataFormat*) pa_xmalloc (maxNbBytesToGet);
- normalAvailBytes = _mainBuffer.availForGet();
- _debug("normalAvailBytes: %i\n", normalAvailBytes);
+ normalAvailBytes = getMainBuffer()->availForGet();
byteToGet = (normalAvailBytes < (int)(maxNbBytesToGet)) ? normalAvailBytes : maxNbBytesToGet;
- _debug("byteToGet: %i\n", byteToGet);
if (byteToGet) {
- // TODO, find out where this problem occurs to get rid of this hack
+ // 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;
- _mainBuffer.getData (out, byteToGet, 100);
+ getMainBuffer()->getData (out, byteToGet, 100);
// test if resampling is required
if (_mainBufferSampleRate && ((int)_audioSampleRate != _mainBufferSampleRate)) {
@@ -602,12 +583,8 @@ void PulseLayer::writeToSpeaker (void)
// Do sample rate conversion
int nb_sample_down = byteToGet / sizeof(SFLDataFormat);
- _debug("nbSampleDown: %i\n", nb_sample_down);
-
int nbSample = _converter->upsampleData((SFLDataFormat*)out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, nb_sample_down);
- _debug("nbSample converted: %i\n", nbSample);
- _debug("bytes to be written: %i\n", nbSample*sizeof(SFLDataFormat));
pa_stream_write (playback->pulseStream(), rsmpl_out, nbSample*sizeof(SFLDataFormat), NULL, 0, PA_SEEK_RELATIVE);
@@ -623,24 +600,18 @@ void PulseLayer::writeToSpeaker (void)
} else {
if((tone == 0) && (file_tone == 0)) {
-
- // _debug("maxNbBytesToGet: %i\n", maxNbBytesToGet);
SFLDataFormat* zeros = (SFLDataFormat*)pa_xmalloc (writeableSize);
bzero (zeros, writeableSize);
pa_stream_write(playback->pulseStream(), zeros, writeableSize, pa_xfree, 0, PA_SEEK_RELATIVE);
- // pa_xfree (zeros);
-
-
}
}
_urgentRingBuffer.Discard(byteToGet);
- // pa_xfree (out);
}
}
@@ -652,8 +623,6 @@ void PulseLayer::readFromMic (void)
const char* data = NULL;
size_t r;
- // _debug("--------------- Capture -----------\n");
-
if (pa_stream_peek (record->pulseStream() , (const void**) &data , &r) < 0 || !data) {
_debug("pa_stream_peek() failed: %s\n" , pa_strerror( pa_context_errno( context) ));
}
@@ -671,24 +640,20 @@ void PulseLayer::readFromMic (void)
int nbSample = r / sizeof(SFLDataFormat);
int nb_sample_up = nbSample;
- // _debug("nbSample from mic: %i\n", nbSample);
-
nbSample = _converter->downsampleData ((SFLDataFormat*)data, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, nb_sample_up);
// remove dc offset
dcblocker->filter_signal( rsmpl_out, nbSample );
- // _debug("nbSample copied: %i\n", nbSample);
-
- _mainBuffer.putData ( (void*) rsmpl_out, nbSample*sizeof(SFLDataFormat), 100);
+ getMainBuffer()->putData ( (void*) rsmpl_out, nbSample*sizeof(SFLDataFormat), 100);
pa_xfree (rsmpl_out);
} else {
// no resampling required
- _mainBuffer.putData ( (void*) data, r, 100);
+ getMainBuffer()->putData ( (void*) data, r, 100);
}
diff --git a/sflphone-common/src/managerimpl.cpp b/sflphone-common/src/managerimpl.cpp
index 18aca9f6969729b377ecbf08122f1f04f4d7dd39..6f8304964d4f0af7c4670b3f5444394d57d104bd 100644
--- a/sflphone-common/src/managerimpl.cpp
+++ b/sflphone-common/src/managerimpl.cpp
@@ -1707,9 +1707,8 @@ ManagerImpl::playDtmf (char code, bool isTalking)
// 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));
+ audiolayer->startStream();
+ audiolayer->putUrgent (buf, size * sizeof (SFLDataFormat));
}
else {
_debug(" playDtmf: Error cannot play dtmf\n");
@@ -1920,22 +1919,22 @@ 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 ("");
}
}
+ 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();
+ }
+
/* Direct IP to IP call */
if (getConfigFromCall (call_id) == Call::IPtoIP) {
SIPVoIPLink::instance (AccountNULL)->hangup (call_id);
@@ -2039,8 +2038,8 @@ bool ManagerImpl::playATone (Tone::TONEID toneId)
if (audiolayer) {
- audiolayer->startStream();
audiolayer->flushUrgent();
+ audiolayer->startStream();
}
if (_telephoneTone != 0) {
@@ -3072,12 +3071,15 @@ ManagerImpl::initAudioDriver (void)
if (getConfigInt (PREFERENCES , CONFIG_AUDIO) == ALSA) {
_audiodriver = new AlsaLayer (this);
+ _audiodriver->setMainBuffer(&_mainBuffer);
} else if (getConfigInt (PREFERENCES , CONFIG_AUDIO) == PULSEAUDIO) {
if (app_is_running ("pulseaudio") == 0) {
_audiodriver = new PulseLayer (this);
+ _audiodriver->setMainBuffer(&_mainBuffer);
} else {
_audiodriver = new AlsaLayer (this);
setConfig (PREFERENCES, CONFIG_AUDIO, ALSA);
+ _audiodriver->setMainBuffer(&_mainBuffer);
}
} else
_debug ("Error - Audio API unknown\n");
@@ -3176,7 +3178,7 @@ void ManagerImpl::switchAudioManager (void)
_debug ("Deleting current layer... \n");
- //_audiodriver->closeLayer();
+ // _audiodriver->closeLayer();
delete _audiodriver;
_audiodriver = NULL;
@@ -3186,11 +3188,13 @@ void ManagerImpl::switchAudioManager (void)
case ALSA:
_debug ("Creating Pulseaudio layer...\n");
_audiodriver = new PulseLayer (this);
+ _audiodriver->setMainBuffer(&_mainBuffer);
break;
case PULSEAUDIO:
_debug ("Creating ALSA layer...\n");
_audiodriver = new AlsaLayer (this);
+ _audiodriver->setMainBuffer(&_mainBuffer);
break;
default:
@@ -3209,12 +3213,16 @@ void ManagerImpl::switchAudioManager (void)
_debug ("has current call: %i \n", hasCurrentCall());
+ if(hasCurrentCall())
+ _audiodriver->startStream();
+
+
// need to stop audio streams if there is currently no call
- if ( (type != PULSEAUDIO) && (!hasCurrentCall())) {
+ // if ( (type != PULSEAUDIO) && (!hasCurrentCall())) {
// _debug("There is currently a call!!\n");
- _audiodriver->stopStream();
+ // _audiodriver->stopStream();
- }
+ // }
}
/**
diff --git a/sflphone-common/src/managerimpl.h b/sflphone-common/src/managerimpl.h
index 19904a204ca692ef02e3dedf4a7254214d6ab123..9689044a087e78186e9150744e70c96b063430b9 100644
--- a/sflphone-common/src/managerimpl.h
+++ b/sflphone-common/src/managerimpl.h
@@ -43,6 +43,8 @@
#include "audio/sound/dtmf.h" // DTMF class contained by value here
#include "audio/codecs/codecDescriptor.h" // CodecDescriptor class contained by value here
+#include "audio/mainbuffer.h"
+
class AudioLayer;
class GuiFramework;
class TelephoneTone;
@@ -1191,6 +1193,17 @@ class ManagerImpl {
* Unload the account (delete them)
*/
void unloadAccountMap();
+
+
+ /**
+ * Instance of the MainBuffer for the whole application
+ *
+ * In order to send signal to other parts of the application, one must pass through the mainbuffer.
+ * Audio instances must be registered into the MainBuffer and bound together via the ManagerImpl.
+ *
+ */
+ MainBuffer _mainBuffer;
+
public:
@@ -1252,6 +1265,8 @@ class ManagerImpl {
//
ConferenceMap _conferencemap;
+
+
private:
// Copy Constructor