diff --git a/sflphone-common/src/audio/alsa/alsalayer.cpp b/sflphone-common/src/audio/alsa/alsalayer.cpp
index a636de6840b6ca57b68e1fe6fe916f2c493313e0..a24aeef2e82c5fbb958560dc866ef4f5f1ff1f58 100644
--- a/sflphone-common/src/audio/alsa/alsalayer.cpp
+++ b/sflphone-common/src/audio/alsa/alsalayer.cpp
@@ -33,8 +33,6 @@
#include "managerimpl.h"
-int framesPerBufferAlsa = 2048;
-
// Constructor
AlsaLayer::AlsaLayer (ManagerImpl* manager)
: AudioLayer (manager , ALSA)
@@ -428,7 +426,7 @@ bool AlsaLayer::alsa_set_params (snd_pcm_t *pcm_handle, int type, int rate)
/* Set sample rate. If we can't set to the desired exact value, we set to the nearest acceptable */
dir=0;
- rate = getSampleRate();
+ rate = _audioSampleRate;
exact_ivalue = rate;
@@ -825,12 +823,12 @@ AlsaLayer::soundCardGetIndex (std::string description)
void AlsaLayer::audioCallback (void)
{
- int toGet, urgentAvailBytes, normalAvailBytes;
+ int urgentAvailBytes;
unsigned short spkrVolume, micVolume;
AudioLoop *tone;
AudioLoop *file_tone;
-
- SFLDataFormat *out = NULL;
+ unsigned int _mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
+ bool resample = _audioSampleRate != _mainBufferSampleRate;
notifyincomingCall();
@@ -853,119 +851,68 @@ void AlsaLayer::audioCallback (void)
int playbackAvailBytes = playbackAvailSmpl*sizeof (SFLDataFormat);
if (urgentAvailBytes > 0) {
-
// Urgent data (dtmf, incoming call signal) come first.
- toGet = (urgentAvailBytes < (int) (playbackAvailBytes)) ? urgentAvailBytes : playbackAvailBytes;
- out = (SFLDataFormat*) malloc (toGet);
- memset (out, 0, toGet);
-
- if (out) {
- _urgentRingBuffer.Get (out, toGet);
- if (spkrVolume!=100)
- for (int i=0; i<toGet / sizeof (SFLDataFormat); i++)
- out[i] = out[i] * spkrVolume / 100;
-
- write (out, toGet, _PlaybackHandle);
- free (out);
- }
-
- out=NULL;
-
+ int toGet = urgentAvailBytes;
+ if (toGet > playbackAvailBytes)
+ toGet = playbackAvailBytes;
+ SFLDataFormat *out = (SFLDataFormat*) malloc (toGet);
+ _urgentRingBuffer.Get (out, toGet);
+ if (spkrVolume != 100)
+ for (unsigned int i=0; i < toGet / sizeof (SFLDataFormat); i++)
+ out[i] = out[i] * spkrVolume / 100;
+
+ write (out, toGet, _PlaybackHandle);
+ free (out);
// Consume the regular one as well (same amount of bytes)
getMainBuffer()->discard (toGet);
-
} else {
-
- normalAvailBytes = getMainBuffer()->availForGet();
-
- if (tone && (normalAvailBytes <= 0)) {
-
- out = (SFLDataFormat *) malloc (playbackAvailBytes);
- 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);
- }
-
- out = NULL;
-
- } else if (file_tone && !_RingtoneHandle && (normalAvailBytes <= 0)) {
-
- out = (SFLDataFormat *) malloc (playbackAvailBytes);
- memset (out, 0, playbackAvailBytes);
-
- if (out) {
- file_tone->getNext (out, playbackAvailSmpl, spkrVolume);
- write (out, playbackAvailBytes, _PlaybackHandle);
- free (out);
- }
-
- out = NULL;
-
+ // regular audio data
+ int toGet = getMainBuffer()->availForGet();
+
+ if (toGet <= 0) {
+ // no audio available, play tone or silence
+ SFLDataFormat *out = (SFLDataFormat *) malloc (playbackAvailBytes);
+
+ if (tone)
+ tone->getNext (out, playbackAvailSmpl, spkrVolume);
+ else if (file_tone && !_RingtoneHandle)
+ file_tone->getNext (out, playbackAvailSmpl, spkrVolume);
+ else
+ memset(out, 0, playbackAvailBytes);
+
+ write (out, playbackAvailBytes, _PlaybackHandle);
+ free (out);
} else {
- // If nothing urgent, play the regular sound samples
+ // play the regular sound samples
- int _mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
- int maxNbSamplesToGet = playbackAvailSmpl;
int maxNbBytesToGet = playbackAvailBytes;
-
- // Compute maximal value to get into the ring buffer
-
- if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
-
- double upsampleFactor = (double) _audioSampleRate / _mainBufferSampleRate;
- maxNbSamplesToGet = (int) ( (double) playbackAvailSmpl / upsampleFactor);
- maxNbBytesToGet = maxNbSamplesToGet * sizeof (SFLDataFormat);
-
- }
-
- toGet = (normalAvailBytes < (int) maxNbBytesToGet) ? normalAvailBytes : maxNbBytesToGet;
-
- out = (SFLDataFormat*) malloc (maxNbBytesToGet);
- memset (out, 0, maxNbBytesToGet);
-
- if (normalAvailBytes) {
- getMainBuffer()->getData (out, toGet);
- if (spkrVolume!=100)
- for (int i=0; i<toGet / sizeof (SFLDataFormat); i++)
- out[i] = out[i] * spkrVolume / 100;
-
- if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
- SFLDataFormat *rsmpl_out = (SFLDataFormat*) malloc (playbackAvailBytes);
- memset (out, 0, playbackAvailBytes);
-
- _converter->resample ( (SFLDataFormat*) out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, toGet / sizeof (SFLDataFormat));
- write (rsmpl_out, toGet, _PlaybackHandle);
- free (rsmpl_out);
-
- } else {
-
- write (out, toGet, _PlaybackHandle);
-
- }
- } else {
-
- if (!tone && !file_tone) {
-
- SFLDataFormat *zeros = (SFLDataFormat*) malloc (playbackAvailBytes);
-
- if (zeros) {
- bzero (zeros, playbackAvailBytes);
- write (zeros, playbackAvailBytes, _PlaybackHandle);
- free (zeros);
- }
-
- zeros = NULL;
- }
+ // Compute maximal value to get from the ring buffer
+ double resampleFactor = 1.0;
+ if (resample) {
+ resampleFactor = (double) _audioSampleRate / _mainBufferSampleRate;
+ maxNbBytesToGet = (double) toGet / resampleFactor;
}
- _urgentRingBuffer.Discard (toGet);
-
- free (out);
- out = NULL;
+ if (toGet > maxNbBytesToGet)
+ toGet = maxNbBytesToGet;
+
+ SFLDataFormat *out = (SFLDataFormat*) malloc (toGet);
+ getMainBuffer()->getData (out, toGet);
+ if (spkrVolume!=100)
+ for (unsigned int i=0; i<toGet / sizeof (SFLDataFormat); i++)
+ out[i] = out[i] * spkrVolume / 100;
+
+ if (resample) {
+ int inSamples = toGet / sizeof(SFLDataFormat);
+ int outSamples = inSamples * resampleFactor;
+ SFLDataFormat *rsmpl_out = (SFLDataFormat*) malloc (outSamples * sizeof(SFLDataFormat));
+ _converter->resample (out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, inSamples);
+ write (rsmpl_out, outSamples * sizeof(SFLDataFormat), _PlaybackHandle);
+ free (rsmpl_out);
+ } else {
+ write (out, toGet, _PlaybackHandle);
+ }
+ free (out);
}
}
@@ -973,96 +920,68 @@ void AlsaLayer::audioCallback (void)
int ringtoneAvailSmpl = snd_pcm_avail_update (_RingtoneHandle);
int ringtoneAvailBytes = ringtoneAvailSmpl*sizeof (SFLDataFormat);
- out = (SFLDataFormat *) malloc (ringtoneAvailBytes);
-
- if (out) {
- file_tone->getNext (out, ringtoneAvailSmpl, spkrVolume);
- write (out, ringtoneAvailBytes, _RingtoneHandle);
- free (out);
- }
-
- out = NULL;
-
+ SFLDataFormat *out = (SFLDataFormat *) malloc (ringtoneAvailBytes);
+ file_tone->getNext (out, ringtoneAvailSmpl, spkrVolume);
+ write (out, ringtoneAvailBytes, _RingtoneHandle);
+ free (out);
} else if (_RingtoneHandle) {
int ringtoneAvailSmpl = snd_pcm_avail_update (_RingtoneHandle);
int ringtoneAvailBytes = ringtoneAvailSmpl*sizeof (SFLDataFormat);
- out = (SFLDataFormat *) malloc (ringtoneAvailBytes);
-
- if (out) {
- memset (out, 0, ringtoneAvailBytes);
- write (out, ringtoneAvailBytes, _RingtoneHandle);
- free (out);
- }
-
- out = NULL;
+ SFLDataFormat *out = (SFLDataFormat *) malloc (ringtoneAvailBytes);
+ memset (out, 0, ringtoneAvailBytes);
+ write (out, ringtoneAvailBytes, _RingtoneHandle);
+ free (out);
}
// Additionally handle the mic's audio stream
- int micAvailBytes;
- int toPut;
-
- SFLDataFormat* in = NULL;
-
if (!is_capture_running())
return;
- micAvailBytes = snd_pcm_avail_update (_CaptureHandle);
-
- if (micAvailBytes < 0)
- _debug ("Audio: Mic error: %s", snd_strerror (micAvailBytes));
- if (micAvailBytes <= 0)
+ int toPutSamples = snd_pcm_avail_update (_CaptureHandle);
+ if (toPutSamples <= 0) {
+ if (toPutSamples < 0)
+ _error ("Audio: Mic error: %s", snd_strerror (toPutSamples));
return;
+ }
- toPut = (micAvailBytes <= framesPerBufferAlsa) ? micAvailBytes : framesPerBufferAlsa;
- in = (SFLDataFormat*) malloc (toPut * sizeof (SFLDataFormat));
- toPut = read (in, toPut* sizeof (SFLDataFormat));
-
- adjustVolume (in, toPut, SFL_PCM_CAPTURE);
-
- int _mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
-
- if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
-
- SFLDataFormat* rsmpl_out = (SFLDataFormat*) malloc (framesPerBufferAlsa * sizeof (SFLDataFormat));
-
- _converter->resample ( (SFLDataFormat*) in, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, toPut / sizeof (SFLDataFormat));
-
- _audiofilter->processAudio (rsmpl_out, toPut);
-
- getMainBuffer()->putData (rsmpl_out, toPut);
-
+ const int framesPerBufferAlsa = 2048;
+ if (toPutSamples > framesPerBufferAlsa)
+ toPutSamples = framesPerBufferAlsa;
+
+ int toPutBytes = toPutSamples * sizeof(SFLDataFormat);
+ SFLDataFormat* in = (SFLDataFormat*) malloc (toPutBytes);
+ int bytes = read (in, toPutBytes);
+ if (toPutBytes != bytes) {
+ _error("ALSA MIC : Couldn't read!");
+ free(in);
+ return;
+ }
+ adjustVolume (in, toPutSamples, _manager->getSpkrVolume());
+
+ if (resample) {
+ int outSamples = toPutSamples * ((double) _audioSampleRate / _mainBufferSampleRate);
+ int outBytes = outSamples * sizeof (SFLDataFormat);
+ SFLDataFormat* rsmpl_out = (SFLDataFormat*) malloc (outBytes);
+ _converter->resample ( (SFLDataFormat*) in, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, toPutSamples);
+ _audiofilter->processAudio (rsmpl_out, outBytes);
+ getMainBuffer()->putData (rsmpl_out, outBytes);
free (rsmpl_out);
} else {
- SFLDataFormat* filter_out = (SFLDataFormat*) malloc (framesPerBufferAlsa * sizeof (SFLDataFormat));
-
- if (filter_out) {
- _audiofilter->processAudio (in, filter_out, toPut);
- // captureFile->write ( (const char *) filter_out, toPut);
- getMainBuffer()->putData (filter_out, toPut);
- free (filter_out);
- }
+ SFLDataFormat* filter_out = (SFLDataFormat*) malloc (toPutBytes);
+ _audiofilter->processAudio (in, filter_out, toPutBytes);
+ // captureFile->write ( (const char *) filter_out, toPut);
+ getMainBuffer()->putData (filter_out, toPutBytes);
+ free (filter_out);
}
free (in);
}
-void* AlsaLayer::adjustVolume (void* buffer , int len, int stream)
+void AlsaLayer::adjustVolume (SFLDataFormat *src , int samples, int volume)
{
- int vol = (stream == SFL_PCM_PLAYBACK)
- ? _manager->getSpkrVolume()
- : _manager->getMicVolume();
-
- SFLDataFormat *src = (SFLDataFormat*) buffer;
-
- if (vol != 100) {
- int i, size = len / sizeof (SFLDataFormat);
-
- for (i = 0 ; i < size ; i++) {
- src[i] = src[i] * vol / 100 ;
- }
- }
-
- return src ;
+ if (volume != 100)
+ for (int i = 0 ; i < samples; i++)
+ src[i] = src[i] * volume / 100 ;
}
diff --git a/sflphone-common/src/audio/alsa/alsalayer.h b/sflphone-common/src/audio/alsa/alsalayer.h
index c317700327a35d44b9c63a8a9c81157cf7e5d1f4..f35f322a4b521edaa2164e2bd2d9b923b29e5f41 100644
--- a/sflphone-common/src/audio/alsa/alsalayer.h
+++ b/sflphone-common/src/audio/alsa/alsalayer.h
@@ -292,7 +292,7 @@ class AlsaLayer : public AudioLayer
*/
void handle_xrun_playback (snd_pcm_t *handle);
- void* adjustVolume (void* buffer , int len, int stream);
+ void adjustVolume (SFLDataFormat* buffer , int samples, int volume);
/**
* Handles to manipulate playback stream
diff --git a/sflphone-common/src/audio/audioloop.cpp b/sflphone-common/src/audio/audioloop.cpp
index ebfc29edddeafdfaf4394bb1fff620faeb7f06a8..7bd8afe3987f90556066810ab97975eaab1114fb 100644
--- a/sflphone-common/src/audio/audioloop.cpp
+++ b/sflphone-common/src/audio/audioloop.cpp
@@ -46,20 +46,19 @@ AudioLoop::~AudioLoop()
_buffer = 0;
}
-int
-AudioLoop::getNext (SFLDataFormat* output, int nb, short volume)
+void
+AudioLoop::getNext (SFLDataFormat* output, int samples, short volume)
{
- int copied = 0;
int block;
int pos = _pos;
if(_size == 0) {
_error("AudioLoop: Error: Audio loop size is 0");
- return 0;
+ return;
}
- while (nb) {
- block = nb;
+ while (samples) {
+ block = samples;
if (block > (_size-pos)) {
block = _size-pos;
@@ -79,13 +78,9 @@ AudioLoop::getNext (SFLDataFormat* output, int nb, short volume)
// should adjust sound here, in output???
pos = (pos + block) % _size;
- nb -= block;
-
- copied += block;
+ samples -= block;
}
_pos = pos;
-
- return copied;
}
diff --git a/sflphone-common/src/audio/audioloop.h b/sflphone-common/src/audio/audioloop.h
index 2eb1c7bc4e86c1e6e366250529888dedad5d8a78..5e6616e710c825fba5c2243adcdf642c84f89318 100644
--- a/sflphone-common/src/audio/audioloop.h
+++ b/sflphone-common/src/audio/audioloop.h
@@ -60,9 +60,8 @@ class AudioLoop
* @param output The data buffer
* @param nb of int16 to send
* @param volume The volume
- * @return the number of int16 sent (nb*2)
*/
- int getNext (SFLDataFormat* output, int nb, short volume=100);
+ void getNext (SFLDataFormat* output, int samples, short volume=100);
/**
* Reset the pointer position
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
index c27b1533a3fa905e186bfad51f2d260bba709510..8a5503ecf1f781cf765286a18a6059ea077f741b 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
@@ -42,18 +42,10 @@ static const SFLDataFormat initFadeinFactor = 32000;
AudioRtpRecord::AudioRtpRecord () : _audioCodec (NULL)
, _hasDynamicPayloadType (false)
- , _micData (NULL)
- , _micDataConverted (NULL)
- , _micDataEncoded (NULL)
- , _spkrDataDecoded (NULL)
- , _spkrDataConverted (NULL)
, _converter (NULL)
, _codecSampleRate (0)
, _codecFrameSize (0)
- , _micFadeInComplete (false)
- , _spkrFadeInComplete (false)
, _micAmplFactor (initFadeinFactor)
- , _spkrAmplFactor (initFadeinFactor)
, _audioProcess (NULL)
, _noiseSuppress (NULL)
, _callId ("")
@@ -65,35 +57,10 @@ AudioRtpRecord::AudioRtpRecord () : _audioCodec (NULL)
AudioRtpRecord::~AudioRtpRecord()
{
- _debug ("AudioRtpRecord: Delete audio rtp internal data");
-
- delete [] _micData;
-
- delete [] _micDataConverted;
-
- delete [] _micDataEncoded;
-
- delete [] _micDataEchoCancelled;
-
- delete [] _spkrDataDecoded;
-
- delete [] _spkrDataConverted;
-
delete _converter;
-
- audioCodecMutex.enter();
-
delete _audioCodec;
-
- audioCodecMutex.leave();
-
- audioProcessMutex.enter();
-
delete _audioProcess;
-
delete _noiseSuppress;
-
- audioProcessMutex.leave();
}
@@ -124,93 +91,41 @@ void AudioRtpRecordHandler::updateRtpMedia (AudioCodec *audioCodec)
{
int lastSamplingRate = _audioRtpRecord._codecSampleRate;
- _audioRtpRecord.audioCodecMutex.enter();
-
- if (_audioRtpRecord._audioCodec) {
- delete _audioRtpRecord._audioCodec;
- _audioRtpRecord._audioCodec = NULL;
- }
-
- _audioRtpRecord._audioCodec = audioCodec;
- _audioRtpRecord._codecPayloadType = audioCodec->getPayloadType();
- _audioRtpRecord._codecSampleRate = audioCodec->getClockRate();
- _audioRtpRecord._codecFrameSize = audioCodec->getFrameSize();
- _audioRtpRecord._hasDynamicPayloadType = audioCodec->hasDynamicPayload();
-
- _audioRtpRecord.audioCodecMutex.leave();
+ setRtpMedia(audioCodec);
Manager::instance().audioSamplingRateChanged(_audioRtpRecord._codecSampleRate);
- if (lastSamplingRate != _audioRtpRecord._codecSampleRate)
- updateNoiseSuppress();
-}
-
-void AudioRtpRecordHandler::init()
-{
+ if (lastSamplingRate != _audioRtpRecord._codecSampleRate) {
+ _debug ("AudioRtpSession: Update noise suppressor with sampling rate %d and frame size %d", getCodecSampleRate(), getCodecFrameSize());
+ initNoiseSuppress();
+ }
}
void AudioRtpRecordHandler::initBuffers()
{
- int codecSampleRate = _audioRtpRecord._codecSampleRate;
-
// Set sampling rate, main buffer choose the highest one
// Manager::instance().getMainBuffer()->setInternalSamplingRate (codecSampleRate);
- Manager::instance().audioSamplingRateChanged(codecSampleRate);
+ Manager::instance().audioSamplingRateChanged(_audioRtpRecord._codecSampleRate);
// initialize SampleRate converter using AudioLayer's sampling rate
// (internal buffers initialized with maximal sampling rate and frame size)
- int rate = getCodecSampleRate();
- _audioRtpRecord._converter = new SamplerateConverter (rate);
-
- int nbSamplesMax = (int) ( (getCodecSampleRate() * getCodecFrameSize() / 1000));
- _audioRtpRecord._micData = new SFLDataFormat[nbSamplesMax];
- _audioRtpRecord._micDataConverted = new SFLDataFormat[nbSamplesMax];
- _audioRtpRecord._micDataEchoCancelled = new SFLDataFormat[nbSamplesMax];
- _audioRtpRecord._micDataEncoded = new unsigned char[nbSamplesMax * 2];
- _audioRtpRecord._spkrDataConverted = new SFLDataFormat[nbSamplesMax];
- _audioRtpRecord._spkrDataDecoded = new SFLDataFormat[nbSamplesMax];
+ delete _audioRtpRecord._converter;
+ _audioRtpRecord._converter = new SamplerateConverter (getCodecSampleRate());
}
void AudioRtpRecordHandler::initNoiseSuppress()
{
_audioRtpRecord.audioProcessMutex.enter();
- NoiseSuppress *noiseSuppress = new NoiseSuppress (getCodecFrameSize(), getCodecSampleRate());
- AudioProcessing *processing = new AudioProcessing (noiseSuppress);
+ delete _audioRtpRecord._audioProcess;
+ delete _audioRtpRecord._noiseSuppress;
- _audioRtpRecord._noiseSuppress = noiseSuppress;
- _audioRtpRecord._audioProcess = processing;
+ _audioRtpRecord._noiseSuppress = new NoiseSuppress (getCodecFrameSize(), getCodecSampleRate());
+ _audioRtpRecord._audioProcess = new AudioProcessing (_audioRtpRecord._noiseSuppress);
_audioRtpRecord.audioProcessMutex.leave();
}
-void AudioRtpRecordHandler::updateNoiseSuppress()
-{
-
- _audioRtpRecord.audioProcessMutex.enter();
-
- if (_audioRtpRecord._audioProcess)
- delete _audioRtpRecord._audioProcess;
-
- _audioRtpRecord._audioProcess = NULL;
-
- if (_audioRtpRecord._noiseSuppress)
- delete _audioRtpRecord._noiseSuppress;
-
- _audioRtpRecord._noiseSuppress = NULL;
-
- _debug ("AudioRtpSession: Update noise suppressor with sampling rate %d and frame size %d", getCodecSampleRate(), getCodecFrameSize());
-
- NoiseSuppress *noiseSuppress = new NoiseSuppress (getCodecFrameSize(), getCodecSampleRate());
- AudioProcessing *processing = new AudioProcessing (noiseSuppress);
-
- _audioRtpRecord._noiseSuppress = noiseSuppress;
- _audioRtpRecord._audioProcess = processing;
-
- _audioRtpRecord.audioProcessMutex.leave();
-
-}
-
void AudioRtpRecordHandler::putDtmfEvent (int digit)
{
sfl::DtmfEvent *dtmf = new sfl::DtmfEvent();
@@ -230,73 +145,54 @@ std::ofstream teststream("test_process_data_encode.raw");
int AudioRtpRecordHandler::processDataEncode (void)
{
- SFLDataFormat *micData = _audioRtpRecord._micData;
- unsigned char *micDataEncoded = _audioRtpRecord._micDataEncoded;
- SFLDataFormat *micDataConverted = _audioRtpRecord._micDataConverted;
+ SFLDataFormat *micData = _audioRtpRecord.decData;
+ unsigned char *micDataEncoded = _audioRtpRecord.encodedData;
+ SFLDataFormat *micDataConverted = _audioRtpRecord.resampledData;
int codecSampleRate = getCodecSampleRate();
int mainBufferSampleRate = Manager::instance().getMainBuffer()->getInternalSamplingRate();
- // compute codec framesize in ms
- float fixedCodecFramesize = ((float)getCodecFrameSize() * 1000.0) / (float)codecSampleRate;
- // compute nb of byte to get coresponding to 20 ms at audio layer frame size (44.1 khz)
- int bytesToGet = (int) ( ( (float) mainBufferSampleRate * fixedCodecFramesize * sizeof (SFLDataFormat)) / 1000.0);
+ double resampleFactor = (double)mainBufferSampleRate / codecSampleRate;
- if (Manager::instance().getMainBuffer()->availForGet (id_) < bytesToGet)
+ // compute nb of byte to get coresponding to 1 audio frame
+ int samplesToGet = resampleFactor * getCodecFrameSize();
+ int bytesToGet = samplesToGet * sizeof (SFLDataFormat);
+
+ if (Manager::instance().getMainBuffer()->availForGet (id_) < bytesToGet) {
+ _error("%s : not enough data available", __PRETTY_FUNCTION__);
return 0;
+ }
int bytes = Manager::instance().getMainBuffer()->getData (micData, bytesToGet, id_);
- if (bytes == 0)
+ if (bytes != bytesToGet) {
+ _error("%s : asked %d bytes from mainbuffer, got %d", __PRETTY_FUNCTION__, bytesToGet, bytes);
return 0;
+ }
- if (!_audioRtpRecord._micFadeInComplete)
- _audioRtpRecord._micFadeInComplete = fadeIn (micData, bytes / sizeof(SFLDataFormat), &_audioRtpRecord._micAmplFactor);
-
- // nb bytes to be sent over RTP
- int compSize;
-
- // test if resampling is required
- if (codecSampleRate != mainBufferSampleRate) {
- _audioRtpRecord._converter->resample (micData, micDataConverted, codecSampleRate, mainBufferSampleRate, bytes / sizeof(SFLDataFormat));
-
- _audioRtpRecord.audioProcessMutex.enter();
-
- if (Manager::instance().audioPreference.getNoiseReduce()) {
- _audioRtpRecord._audioProcess->processAudio (micDataConverted, bytes);
- }
-
- if(Manager::instance().getEchoCancelState() == "enabled") {
- echoCanceller.getData(micData);
- }
-
- _audioRtpRecord.audioProcessMutex.leave();
-
- _audioRtpRecord.audioCodecMutex.enter();
-
- compSize = _audioRtpRecord._audioCodec->encode (micDataEncoded, micData, bytes);
-
- _audioRtpRecord.audioCodecMutex.leave();
+ fadeIn (micData, bytesToGet / sizeof(SFLDataFormat), &_audioRtpRecord._micAmplFactor);
- } else { // no resampling required
- _audioRtpRecord.audioProcessMutex.enter();
+ if(Manager::instance().getEchoCancelState() == "enabled")
+ echoCanceller.getData(micData);
- if (Manager::instance().audioPreference.getNoiseReduce())
- _audioRtpRecord._audioProcess->processAudio (micData, bytes);
+ _audioRtpRecord.audioProcessMutex.enter();
+ if (Manager::instance().audioPreference.getNoiseReduce())
+ _audioRtpRecord._audioProcess->processAudio (micData, bytesToGet);
+ _audioRtpRecord.audioProcessMutex.leave();
- if(Manager::instance().getEchoCancelState() == "enabled")
- echoCanceller.getData(micData);
-
#ifdef DUMP_PROCESS_DATA_ENCODE
- teststream.write(reinterpret_cast<char *>(micData), nbSample * sizeof(SFLDataFormat));
+ teststream.write(reinterpret_cast<char *>(micData), bytesToGet);
#endif
-
- _audioRtpRecord.audioProcessMutex.leave();
- _audioRtpRecord.audioCodecMutex.enter();
- compSize = _audioRtpRecord._audioCodec->encode (micDataEncoded, micData, bytes);
- _audioRtpRecord.audioCodecMutex.leave();
+ SFLDataFormat *out = micData;
+ if (codecSampleRate != mainBufferSampleRate) {
+ out = micDataConverted;
+ _audioRtpRecord._converter->resample (micData, micDataConverted, codecSampleRate, mainBufferSampleRate, samplesToGet);
}
+ _audioRtpRecord.audioCodecMutex.enter();
+ int compSize = _audioRtpRecord._audioCodec->encode (micDataEncoded, out, getCodecFrameSize());
+ _audioRtpRecord.audioCodecMutex.leave();
+
return compSize;
}
@@ -304,8 +200,8 @@ void AudioRtpRecordHandler::processDataDecode (unsigned char *spkrData, unsigned
{
int codecSampleRate = getCodecSampleRate();
- SFLDataFormat *spkrDataDecoded = _audioRtpRecord._spkrDataConverted;
- SFLDataFormat *spkrDataConverted = _audioRtpRecord._spkrDataDecoded;
+ SFLDataFormat *spkrDataDecoded = _audioRtpRecord.decData;
+ SFLDataFormat *spkrDataConverted = _audioRtpRecord.resampledData;
int mainBufferSampleRate = Manager::instance().getMainBuffer()->getInternalSamplingRate();
@@ -316,50 +212,38 @@ void AudioRtpRecordHandler::processDataDecode (unsigned char *spkrData, unsigned
_audioRtpRecord.audioCodecMutex.leave();
- // buffer _receiveDataDecoded ----> short int or int16, coded on 2 bytes
- int nbSample = expandedSize / sizeof (SFLDataFormat);
+ int inSamples = expandedSize / sizeof (SFLDataFormat);
- if (!_audioRtpRecord._spkrFadeInComplete) {
- _audioRtpRecord._spkrFadeInComplete = fadeIn (spkrDataDecoded, nbSample, &_audioRtpRecord._micAmplFactor);
- }
+ fadeIn (spkrDataDecoded, inSamples, &_audioRtpRecord._micAmplFactor);
// Normalize incomming signal
- gainController.process(spkrDataDecoded, nbSample);
+ gainController.process(spkrDataDecoded, inSamples);
+ SFLDataFormat *out = spkrDataDecoded;
+ int outSamples = inSamples;
// test if resampling is required
if (codecSampleRate != mainBufferSampleRate) {
// Do sample rate conversion
- _audioRtpRecord._converter->resample (spkrDataDecoded, spkrDataConverted, codecSampleRate, mainBufferSampleRate, nbSample);
-
- if(Manager::instance().getEchoCancelState() == "enabled") {
- echoCanceller.putData(spkrDataConverted, expandedSize);
- }
-
- // put data in audio layer, size in byte
- Manager::instance().getMainBuffer()->putData (spkrDataConverted, expandedSize, id_);
-
- } else {
- if(Manager::instance().getEchoCancelState() == "enabled") {
- echoCanceller.putData(spkrDataDecoded, expandedSize);
- }
- // put data in audio layer, size in byte
- Manager::instance().getMainBuffer()->putData (spkrDataDecoded, expandedSize, id_);
+ outSamples = ((float) inSamples * ( (float) mainBufferSampleRate / (float) codecSampleRate));
+ _audioRtpRecord._converter->resample (spkrDataDecoded, spkrDataConverted, codecSampleRate, mainBufferSampleRate, inSamples);
+ out = spkrDataConverted;
}
+
+ if(Manager::instance().getEchoCancelState() == "enabled")
+ echoCanceller.putData(out, outSamples * sizeof (SFLDataFormat));
+ Manager::instance().getMainBuffer()->putData (out, outSamples * sizeof (SFLDataFormat), id_);
}
-bool AudioRtpRecordHandler::fadeIn (SFLDataFormat *audio, int size, SFLDataFormat *factor)
+void AudioRtpRecordHandler::fadeIn (SFLDataFormat *audio, int size, SFLDataFormat *factor)
{
- // if factor reach 0, this function should no be called anymore
+ // if factor reach 0, this function should have no effect
if (*factor <= 0)
- return true;
+ return;
while (size)
audio[--size] /= *factor;
*factor /= FADEIN_STEP_SIZE;
-
- return *factor <= 0;
}
}
-
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h
index 87bd6e4814e0021486d3de23beef46d0d1c039a7..c9d64998660d61db534146a7dd44c854a8c90453 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h
@@ -95,21 +95,15 @@ class AudioRtpRecord
ost::Mutex audioCodecMutex;
int _codecPayloadType;
bool _hasDynamicPayloadType;
- SFLDataFormat *_micData;
- SFLDataFormat *_micDataConverted;
- SFLDataFormat *_micDataEchoCancelled;
- unsigned char *_micDataEncoded;
- SFLDataFormat *_spkrDataDecoded;
- SFLDataFormat *_spkrDataConverted;
+ SFLDataFormat decData[DEC_BUFFER_SIZE];
+ SFLDataFormat resampledData[DEC_BUFFER_SIZE];
+ unsigned char encodedData[DEC_BUFFER_SIZE];
SamplerateConverter *_converter;
int _codecSampleRate;
int _codecFrameSize;
int _converterSamplingRate;
EventQueue _eventQueue;
- bool _micFadeInComplete;
- bool _spkrFadeInComplete;
SFLDataFormat _micAmplFactor;
- SFLDataFormat _spkrAmplFactor;
AudioProcessing *_audioProcess;
NoiseSuppress *_noiseSuppress;
ost::Mutex audioProcessMutex;
@@ -162,30 +156,14 @@ class AudioRtpRecordHandler
return _audioRtpRecord._eventQueue.size();
}
- SFLDataFormat *getMicData (void) {
- return _audioRtpRecord._micData;
+ const unsigned char *getMicDataEncoded (void) const {
+ return _audioRtpRecord.encodedData;
}
- SFLDataFormat *getMicDataConverted (void) const {
- return _audioRtpRecord._micDataConverted;
- }
-
- unsigned char *getMicDataEncoded (void) const {
- return _audioRtpRecord._micDataEncoded;
- }
-
- void init (void);
-
- /**
- * Allocate memory for RTP buffers and fill them with zeros
- * @prereq Session codec needs to be initialized prior calling this method
- */
void initBuffers (void);
void initNoiseSuppress (void);
- void updateNoiseSuppress (void);
-
/**
* Encode audio data from mainbuffer
*/
@@ -199,7 +177,7 @@ class AudioRtpRecordHandler
/**
* Ramp In audio data to avoid audio click from peer
*/
- bool fadeIn (SFLDataFormat *audio, int size, SFLDataFormat *factor);
+ void fadeIn (SFLDataFormat *audio, int size, SFLDataFormat *factor);
void setDtmfPayloadType(unsigned int payloadType) {
_audioRtpRecord._dtmfPayloadType = payloadType;
diff --git a/sflphone-common/src/audio/codecs/audiocodec.h b/sflphone-common/src/audio/codecs/audiocodec.h
index 7f443db054fd4447269073c76bcc60068c1e9727..aa05c008e4f2d2a532d863168de087e93dee07f5 100644
--- a/sflphone-common/src/audio/codecs/audiocodec.h
+++ b/sflphone-common/src/audio/codecs/audiocodec.h
@@ -38,6 +38,11 @@
#include "Codec.h"
+// We assume all decoders will be fed 20ms of audio or less
+// And we'll resample them to 44.1kHz or less
+// Also assume mono
+#define DEC_BUFFER_SIZE ((44100 * 20) / 1000)
+
namespace ost {
class PayloadFormat;
class DynamicPayloadFormat;
diff --git a/sflphone-common/src/audio/codecs/audiocodecfactory.cpp b/sflphone-common/src/audio/codecs/audiocodecfactory.cpp
index 2142e5bc1be5e213d91adad47b0e86e87fc42740..9c429336aba12fce69d78c9ce5cd964c8fa219aa 100644
--- a/sflphone-common/src/audio/codecs/audiocodecfactory.cpp
+++ b/sflphone-common/src/audio/codecs/audiocodecfactory.cpp
@@ -388,7 +388,6 @@ bool AudioCodecFactory::isCodecLoaded (int payload)
std::vector <std::string> AudioCodecFactory::getCodecSpecifications (const int32_t& payload)
{
-
_debug ("CodecDescriptor: Gathering codec specifications for payload %i", payload);
std::vector<std::string> v;
@@ -408,5 +407,4 @@ std::vector <std::string> AudioCodecFactory::getCodecSpecifications (const int32
ss.str ("");
return v;
-
}
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
index dbc3457b04215abe4fa9cad72cf2d1f8350c726d..db24a223a154ee47cd6953739bd9d60f6cd8ce36 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
@@ -719,132 +719,86 @@ void PulseLayer::setNoiseSuppressState (bool state)
void PulseLayer::writeToSpeaker (void)
{
- /** Bytes available in the urgent ringbuffer ( reserved for DTMF ) */
- int urgentAvailBytes;
- /** Bytes available in the regular ringbuffer ( reserved for voice ) */
- int normalAvailBytes;
- /** Bytes to get in the ring buffer **/
- int byteToGet;
-
notifyincomingCall();
- SFLDataFormat* out;// = (SFLDataFormat*)pa_xmalloc(framesPerBuffer);
- urgentAvailBytes = _urgentRingBuffer.AvailForGet();
-
// available bytes to be written in pulseaudio internal buffer
- int writeableSize = pa_stream_writable_size (playback->pulseStream());
-
- if (writeableSize < 0) {
- _warn ("Audio: playback error : %s", pa_strerror (writeableSize));
- }
-
- AudioLoop *toneToPlay = _manager->getTelephoneTone();
-
- if (urgentAvailBytes > writeableSize) {
-
- out = (SFLDataFormat*) pa_xmalloc (writeableSize);
- memset (out, 0, writeableSize);
-
- _urgentRingBuffer.Get (out, writeableSize);
-
- pa_stream_write (playback->pulseStream(), out, writeableSize, NULL, 0, PA_SEEK_RELATIVE);
-
+ int writeableSizeBytes = pa_stream_writable_size (playback->pulseStream());
+ if (writeableSizeBytes < 0) {
+ _error ("Audio: playback error : %s", pa_strerror (writeableSizeBytes));
+ return;
+ }
+
+ int urgentBytes = _urgentRingBuffer.AvailForGet();
+ if (urgentBytes > writeableSizeBytes)
+ urgentBytes = writeableSizeBytes;
+ if (urgentBytes) {
+ SFLDataFormat *out = (SFLDataFormat*) pa_xmalloc (urgentBytes);
+ _urgentRingBuffer.Get (out, urgentBytes);
+ pa_stream_write (playback->pulseStream(), out, urgentBytes, NULL, 0, PA_SEEK_RELATIVE);
pa_xfree (out);
-
// Consume the regular one as well (same amount of bytes)
- getMainBuffer()->discard (writeableSize);
-
-
- }
- else if (toneToPlay != 0) {
-
- if (playback->getStreamState() == PA_STREAM_READY) {
-
- out = (SFLDataFormat*) pa_xmalloc (writeableSize);
- memset (out, 0, writeableSize);
-
- int copied = toneToPlay->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);
- }
+ getMainBuffer()->discard (urgentBytes);
+ return;
}
- else {
-
- // We must test if data have been received from network in case of early media
- normalAvailBytes = getMainBuffer()->availForGet();
-
- // flush remaining samples in _urgentRingBuffer
- flushUrgent();
-
- 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;
- }
-
- byteToGet = (normalAvailBytes < (int) (maxNbBytesToGet)) ? normalAvailBytes : maxNbBytesToGet;
-
- if (byteToGet) {
- // Sending an odd number of byte breaks the audio!
- byteToGet &= ~1;
-
- out = (SFLDataFormat*) pa_xmalloc (maxNbBytesToGet);
- memset (out, 0, maxNbBytesToGet);
-
- getMainBuffer()->getData (out, byteToGet);
-
- // test if resampling is required
- if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
- SFLDataFormat* rsmpl_out = (SFLDataFormat*) pa_xmalloc (writeableSize);
- memset (out, 0, writeableSize);
- _converter->resample ( (SFLDataFormat*) out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, byteToGet / sizeof (SFLDataFormat));
-
- if ( byteToGet > (unsigned int) writeableSize)
- _warn ("Audio: Error: nbsbyte exceed buffer length");
-
- pa_stream_write (playback->pulseStream(), rsmpl_out, byteToGet, NULL, 0, PA_SEEK_RELATIVE);
+ AudioLoop *toneToPlay = _manager->getTelephoneTone();
+ if (toneToPlay) {
+ if (playback->getStreamState() == PA_STREAM_READY) {
+ SFLDataFormat *out = (SFLDataFormat*) pa_xmalloc (writeableSizeBytes);
+ toneToPlay->getNext (out, writeableSizeBytes / sizeof (SFLDataFormat), 100);
+ //spkrFile->write ( (const char *) out, writeableSize);
+ pa_stream_write (playback->pulseStream(), out, writeableSizeBytes, NULL, 0, PA_SEEK_RELATIVE);
+ pa_xfree (out);
+ }
+ return;
+ }
- pa_xfree (rsmpl_out);
+ flushUrgent(); // flush remaining samples in _urgentRingBuffer
- } else {
- // write origin data
- pa_stream_write (playback->pulseStream(), out, byteToGet, NULL, 0, PA_SEEK_RELATIVE);
- }
+ int availSamples = getMainBuffer()->availForGet() / sizeof(SFLDataFormat);
+ if (availSamples == 0) {
+ // play silence
+ SFLDataFormat* zeros = (SFLDataFormat*) pa_xmalloc0(writeableSizeBytes);
+ pa_stream_write (playback->pulseStream(), zeros, writeableSizeBytes, NULL, 0, PA_SEEK_RELATIVE);
+ pa_xfree (zeros);
+ return;
+ }
- // free audio buffer used to get data from
- pa_xfree (out);
+ unsigned int mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
+ bool resample = _audioSampleRate != mainBufferSampleRate;
- } else {
+ // how much samples we can write in the output
+ int outSamples = writeableSizeBytes / sizeof(SFLDataFormat);
- // Get ringtone
- // AudioLoop *fileToPlay = _manager->getTelephoneTone();
+ // how much samples we want to read from the buffer
+ int inSamples = outSamples;
- // If audio stream is open and there is realy nothing to play (i.e. audio voce is late and audio layer is underrun)
- if (toneToPlay == NULL) {
+ double resampleFactor = 1.;
+ if (resample) {
+ resampleFactor = (double) _audioSampleRate / mainBufferSampleRate;
+ inSamples = (double) inSamples / resampleFactor;
+ }
- SFLDataFormat* zeros = (SFLDataFormat*) pa_xmalloc (writeableSize);
- bzero (zeros, writeableSize);
+ if (inSamples > availSamples)
+ inSamples = availSamples;
+ int outBytes = (double)inSamples * resampleFactor * sizeof(SFLDataFormat);
- pa_stream_write (playback->pulseStream(), zeros, writeableSize, NULL, 0, PA_SEEK_RELATIVE);
+ int inBytes = inSamples * sizeof (SFLDataFormat);
+ SFLDataFormat *out = (SFLDataFormat*) pa_xmalloc (inBytes);
+ getMainBuffer()->getData (out, inBytes);
- pa_xfree (zeros);
+ // test if resampling is required
+ if (resample) {
+ SFLDataFormat* rsmpl_out = (SFLDataFormat*) pa_xmalloc (outBytes);
+ _converter->resample (out, rsmpl_out, mainBufferSampleRate, _audioSampleRate, inSamples);
+ pa_stream_write (playback->pulseStream(), rsmpl_out, outBytes, NULL, 0, PA_SEEK_RELATIVE);
+ pa_xfree (rsmpl_out);
- }
- }
+ } else {
+ pa_stream_write (playback->pulseStream(), out, inBytes, NULL, 0, PA_SEEK_RELATIVE);
+ }
- _urgentRingBuffer.Discard (byteToGet);
-
- }
+ pa_xfree (out);
}
void PulseLayer::readFromMic (void)
@@ -852,51 +806,37 @@ void PulseLayer::readFromMic (void)
const char* data = NULL;
size_t r;
- int readableSize = pa_stream_readable_size (record->pulseStream());
-
+ unsigned int mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
+ bool resample = _audioSampleRate != mainBufferSampleRate;
if (pa_stream_peek (record->pulseStream() , (const void**) &data , &r) < 0 || !data) {
- _warn ("Audio: Error capture stream peek failed: %s" , pa_strerror (pa_context_errno (context)));
+ _error("Audio: Error capture stream peek failed: %s" , pa_strerror (pa_context_errno (context)));
+ goto end;
}
- if (data != 0) {
-
- int _mainBufferSampleRate = getMainBuffer()->getInternalSamplingRate();
-
- // test if resampling is required
- if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
-
- SFLDataFormat* rsmpl_out = (SFLDataFormat*) pa_xmalloc (readableSize);
- memset (rsmpl_out, 0, readableSize);
-
- _converter->resample ( (SFLDataFormat *) data, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, r / sizeof (SFLDataFormat));
-
- // remove dc offset
- _audiofilter->processAudio (rsmpl_out, r);
+ if (resample) {
+ int inSamples = r / sizeof(SFLDataFormat);
+ double resampleFactor = (double) _audioSampleRate / mainBufferSampleRate;
+ int outSamples = (double) inSamples * resampleFactor;
+ int outBytes = outSamples * sizeof(SFLDataFormat);
- getMainBuffer()->putData (rsmpl_out, r);
-
- pa_xfree (rsmpl_out);
-
- } else {
-
-
- SFLDataFormat* filter_out = (SFLDataFormat*) pa_xmalloc (r);
- memset (filter_out, 0, r);
-
- // remove dc offset
- _audiofilter->processAudio ( (SFLDataFormat *) data, filter_out, r);
-
- getMainBuffer()->putData (filter_out, r);
-
- pa_xfree (filter_out);
- }
- }
-
- if (pa_stream_drop (record->pulseStream()) < 0) {
- _warn ("Audio: Error: capture stream drop failed: %s" , pa_strerror (pa_context_errno (context)));
- }
+ SFLDataFormat* rsmpl_out = (SFLDataFormat*) pa_xmalloc (outBytes);
+ _converter->resample ( (SFLDataFormat *) data, rsmpl_out, mainBufferSampleRate, _audioSampleRate, inSamples);
+ // remove dc offset
+ _audiofilter->processAudio (rsmpl_out, outBytes);
+ getMainBuffer()->putData (rsmpl_out, outBytes);
+ pa_xfree (rsmpl_out);
+ } else {
+ SFLDataFormat* filter_out = (SFLDataFormat*) pa_xmalloc (r);
+ // remove dc offset
+ _audiofilter->processAudio ( (SFLDataFormat *) data, filter_out, r);
+ getMainBuffer()->putData (filter_out, r);
+ pa_xfree (filter_out);
+ }
+end:
+ if (pa_stream_drop (record->pulseStream()) < 0)
+ _error ("Audio: Error: capture stream drop failed: %s" , pa_strerror (pa_context_errno (context)));
}
@@ -923,12 +863,8 @@ void PulseLayer::ringtoneToSpeaker (void)
memset (out, 0, writableSize);
- int copied = fileToPlay->getNext (out, writableSize/sizeof (SFLDataFormat), 100);
- if(copied == 0) {
- copied = writableSize/sizeof(SFLDataFormat);
- }
-
- pa_stream_write (ringtone->pulseStream(), out, copied*sizeof (SFLDataFormat), NULL, 0, PA_SEEK_RELATIVE);
+ fileToPlay->getNext (out, writableSize/sizeof (SFLDataFormat), 100);
+ pa_stream_write (ringtone->pulseStream(), out, writableSize, NULL, 0, PA_SEEK_RELATIVE);
pa_xfree (out);
}
@@ -940,10 +876,6 @@ void PulseLayer::ringtoneToSpeaker (void)
}
out = (SFLDataFormat*) pa_xmalloc (writableSize);
- if(out == NULL) {
- _error("PulseAudio: Error: Could not allocate memory for buffer");
- return;
- }
memset (out, 0, writableSize);
pa_stream_write (ringtone->pulseStream(), out, writableSize, NULL, 0, PA_SEEK_RELATIVE);
diff --git a/sflphone-common/src/audio/samplerateconverter.cpp b/sflphone-common/src/audio/samplerateconverter.cpp
index 411c20389c39b4b6e415f93a8ef3ca5085522e1c..ca4e54def666fd2ef60f2a90695b2b55afef214b 100644
--- a/sflphone-common/src/audio/samplerateconverter.cpp
+++ b/sflphone-common/src/audio/samplerateconverter.cpp
@@ -66,8 +66,8 @@ void SamplerateConverter::resample (SFLDataFormat* dataIn , SFLDataFormat* dataO
{
assert(outputFreq <= _maxFreq);
- double sampleFactor = (double) inputFreq / outputFreq;
- if (sampleFactor == 1)
+ double sampleFactor = (double) outputFreq / inputFreq;
+ if (sampleFactor == 1.0)
return;
int outSamples = nbSamples * sampleFactor;
diff --git a/sflphone-common/src/audio/sound/audiofile.cpp b/sflphone-common/src/audio/sound/audiofile.cpp
index b10382945d52fd9da91ec7af22011fa8d13f206c..9c8d4d25ddc41425fd5a700131b4499349bf00db 100644
--- a/sflphone-common/src/audio/sound/audiofile.cpp
+++ b/sflphone-common/src/audio/sound/audiofile.cpp
@@ -241,19 +241,9 @@ WaveFile::WaveFile (const std::string& fileName, unsigned int audioSamplingRate)
}
if (srate != audioSamplingRate) {
- nbSamples = (int) ( (float) nbSamples * ( (float) audioSamplingRate / (float) srate));
-
- _buffer = new SFLDataFormat[nbSamples];
- if (_buffer == NULL) {
- delete[] tempBuffer;
- throw AudioFileException("Could not allocate buffer for audio");
- }
-
- if (srate < audioSamplingRate)
- _converter.resample (tempBuffer, _buffer, srate, audioSamplingRate, nbSamples);
- else if (srate > audioSamplingRate)
- _converter.resample (tempBuffer, _buffer, audioSamplingRate, srate, nbSamples);
-
+ int outSamples = ((float) nbSamples * ( (float) audioSamplingRate / (float) srate));
+ _buffer = new SFLDataFormat[outSamples];
+ _converter.resample (tempBuffer, _buffer, srate, audioSamplingRate, nbSamples);
delete[] tempBuffer;
} else {
_buffer = tempBuffer;
diff --git a/sflphone-common/src/iax/iaxvoiplink.cpp b/sflphone-common/src/iax/iaxvoiplink.cpp
index ba70c8628ea47d4b11ab77f959e1dd38b567a45c..9c597dfb5652766a861025286a3d26c755fd2577 100644
--- a/sflphone-common/src/iax/iaxvoiplink.cpp
+++ b/sflphone-common/src/iax/iaxvoiplink.cpp
@@ -62,11 +62,6 @@ IAXVoIPLink::IAXVoIPLink (const std::string& accountID) : VoIPLink ()
, _regSession (NULL)
, _nextRefreshStamp (0)
, audiolayer (NULL)
- , micData (NULL)
- , micDataConverted (NULL)
- , micDataEncoded (NULL)
- , spkrDataDecoded (NULL)
- , spkrDataConverted (NULL)
, converter (NULL)
, converterSamplingRate (0)
, urlhook (NULL)
@@ -78,17 +73,6 @@ IAXVoIPLink::IAXVoIPLink (const std::string& accountID) : VoIPLink ()
srand (time (NULL));
converter = new SamplerateConverter (44100);
-
- // int nbSamplesMax = (int) (converter->getFrequence() * converter->getFramesize() / 1000);
- int nbSamplesMax = (44100 * 20) / 1000;
-
- micData = new SFLDataFormat[nbSamplesMax];
- micDataConverted = new SFLDataFormat[nbSamplesMax];
- micDataEncoded = new unsigned char[nbSamplesMax];
-
- spkrDataConverted = new SFLDataFormat[nbSamplesMax];
- spkrDataDecoded = new SFLDataFormat[nbSamplesMax];
-
urlhook = new UrlHook ();
}
@@ -102,11 +86,6 @@ IAXVoIPLink::~IAXVoIPLink()
terminate();
delete converter;
- delete [] micData;
- delete [] micDataConverted;
- delete [] micDataEncoded;
- delete [] spkrDataDecoded;
- delete [] spkrDataConverted;
}
bool
@@ -282,53 +261,50 @@ IAXVoIPLink::sendAudioFromMic (void)
// do not use the current ID in Manager (it may refer to a conference also)
// currentCall = getIAXCall (Manager::instance().getCurrentCallId());
- CallMap::iterator iter_call = _callMap.begin();
-
- while (iter_call != _callMap.end()) {
- IAXCall *currentCall = (IAXCall*) iter_call->second;
- iter_call++;
-
+ for (CallMap::iterator iter = _callMap.begin(); iter != _callMap.end() ; ++iter) {
+ IAXCall *currentCall = (IAXCall*) iter->second;
if (!currentCall || currentCall->getState() != Call::Active)
continue;
AudioCodecType codecType = currentCall->getAudioCodec();
sfl::AudioCodec *audioCodec = static_cast<sfl::AudioCodec *>(currentCall->getAudioCodecFactory().getCodec (codecType));
- // Send sound here
-
if (!audioCodec || !audiolayer)
continue;
Manager::instance().getMainBuffer()->setInternalSamplingRate (audioCodec->getClockRate());
- int _mainBufferSampleRate = audiolayer->getMainBuffer()->getInternalSamplingRate();
+ unsigned int mainBufferSampleRate = audiolayer->getMainBuffer()->getInternalSamplingRate();
// we have to get 20ms of data from the mic *20/1000 = /50
// rate/50 shall be lower than IAX__20S_48KHZ_MAX
- int needed = _mainBufferSampleRate * audiolayer->getFrameSize() / 1000 * sizeof (SFLDataFormat);
- if (audiolayer->getMainBuffer()->availForGet (currentCall->getCallId()) < needed)
+ int bytesNeeded = mainBufferSampleRate * 20 / 1000 * sizeof (SFLDataFormat);
+ if (audiolayer->getMainBuffer()->availForGet (currentCall->getCallId()) < bytesNeeded)
continue;
// Get bytes from micRingBuffer to data_from_mic
- int bytes = audiolayer->getMainBuffer()->getData (micData, needed, currentCall->getCallId()) / sizeof (SFLDataFormat);
+ int bytes = audiolayer->getMainBuffer()->getData (decData, bytesNeeded, currentCall->getCallId());
+ int samples = bytes / sizeof(SFLDataFormat);
int compSize;
- if (audioCodec->getClockRate() && ((int) audioCodec->getClockRate() != _mainBufferSampleRate)) {
- // resample
- converter->resample (micData , micDataConverted , (int) audioCodec->getClockRate(), _mainBufferSampleRate, bytes);
- compSize = audioCodec->encode (micDataEncoded, micDataConverted , bytes);
+ unsigned int audioRate = audioCodec->getClockRate();
+ int outSamples;
+ if (audioRate != mainBufferSampleRate) {
+ int outSamples = ((float) samples * ( (float) mainBufferSampleRate / (float) audioRate));
+ converter->resample (decData , resampledData , audioRate, mainBufferSampleRate, samples);
+ compSize = audioCodec->encode (encodedData, resampledData , outSamples * sizeof(SFLDataFormat));
} else {
- compSize = audioCodec->encode (micDataEncoded, micData, bytes);
+ outSamples = samples;
+ compSize = audioCodec->encode (encodedData, decData, bytes);
}
// Send it out!
_mutexIAX.enterMutex();
// Make sure the session and the call still exists.
- if (currentCall->getSession() && (micDataEncoded != NULL) && (bytes > 0)) {
- if (iax_send_voice (currentCall->getSession(), currentCall->getFormat(), micDataEncoded, compSize, bytes) == -1) {
- _debug ("IAX: Error sending voice data.");
- }
+ if (currentCall->getSession() && (bytes > 0)) {
+ if (iax_send_voice (currentCall->getSession(), currentCall->getFormat(), encodedData, compSize, outSamples) == -1)
+ _error ("IAX: Error sending voice data.");
}
_mutexIAX.leaveMutex();
@@ -479,13 +455,12 @@ IAXVoIPLink::hangup (const std::string& id) throw (VoipLinkException)
throw VoipLinkException("Could not find call");
}
- std::string reason = "Dumped Call";
CHK_VALID_CALL;
Manager::instance().getMainBuffer()->unBindAll (call->getCallId());
_mutexIAX.enterMutex();
- iax_hangup (call->getSession(), (char*) reason.c_str());
+ iax_hangup (call->getSession(), (char*) "Dumped Call");
_mutexIAX.leaveMutex();
call->setSession (NULL);
@@ -505,7 +480,6 @@ IAXVoIPLink::peerHungup (const std::string& id) throw (VoipLinkException)
throw VoipLinkException("Could not find call");
}
- std::string reason = "Dumped Call";
CHK_VALID_CALL;
Manager::instance().getMainBuffer()->unBindAll (call->getCallId());
@@ -589,12 +563,11 @@ bool
IAXVoIPLink::refuse (const std::string& id)
{
IAXCall* call = getIAXCall (id);
- std::string reason = "Call rejected manually.";
CHK_VALID_CALL;
_mutexIAX.enterMutex();
- iax_reject (call->getSession(), (char*) reason.c_str());
+ iax_reject (call->getSession(), (char*) "Call rejected manually.");
_mutexIAX.leaveMutex();
removeCall (id);
@@ -745,9 +718,8 @@ IAXVoIPLink::iaxHandleCallEvent (iax_event* event, IAXCall* call)
/*
_debug("IAXVoIPLink::iaxHandleCallEvent, peer hangup have been called");
- std::string reason = "Dumped Call";
_mutexIAX.enterMutex();
- iax_hangup(call->getSession(), (char*)reason.c_str());
+ iax_hangup(call->getSession(), (char*)"Dumped Call");
_mutexIAX.leaveMutex();
call->setSession(NULL);
audiolayer->stopStream();
@@ -905,39 +877,23 @@ IAXVoIPLink::iaxHandleCallEvent (iax_event* event, IAXCall* call)
void
IAXVoIPLink::iaxHandleVoiceEvent (iax_event* event, IAXCall* call)
{
-
- unsigned char *data;
- unsigned int size, max, nbInt16;
- int expandedSize, nbSample_;
- sfl::AudioCodec *audioCodec;
-
- if (!call) {
- return;
- }
-
if (!event->datalen) {
// Skip this empty packet.
//_debug("IAX: Skipping empty jitter-buffer interpolated packet");
return;
}
- AudioCodecType audioCodecType = call->getAudioCodec();
- audioCodec = static_cast<sfl::AudioCodec *>(call->getAudioCodecFactory().getCodec (audioCodecType));
-
- if (!audioCodec)
- return;
-
if (audiolayer) {
_debug ("IAX: incoming audio, but no sound card open");
return;
}
- Manager::instance().getMainBuffer ()->setInternalSamplingRate (audioCodec->getClockRate ());
-
- // If we receive datalen == 0, some things of the jitter buffer in libiax2/iax.c
- // were triggered
+ sfl::AudioCodec *audioCodec = static_cast<sfl::AudioCodec *>(call->getAudioCodecFactory().getCodec (call->getAudioCodec()));
+ if (!audioCodec)
+ return;
- int _mainBufferSampleRate = Manager::instance().getMainBuffer()->getInternalSamplingRate();
+ Manager::instance().getMainBuffer ()->setInternalSamplingRate (audioCodec->getClockRate ());
+ unsigned int mainBufferSampleRate = Manager::instance().getMainBuffer()->getInternalSamplingRate();
// On-the-fly codec changing (normally, when we receive a full packet)
// as per http://tools.ietf.org/id/draft-guy-iax-03.txt
@@ -948,26 +904,26 @@ IAXVoIPLink::iaxHandleVoiceEvent (iax_event* event, IAXCall* call)
call->setFormat (event->subclass);
}
- data = (unsigned char*) event->data;
-
- size = event->datalen;
+ unsigned char *data = (unsigned char*) event->data;
+ unsigned int size = event->datalen;
// Decode data with relevant codec
- max = (int) (audioCodec->getClockRate() * audiolayer->getFrameSize() / 1000);
+ unsigned int max = audioCodec->getClockRate() * audiolayer->getFrameSize() / 1000;
if (size > max) {
_debug ("The size %d is bigger than expected %d. Packet cropped. Ouch!", size, max);
size = max;
}
- expandedSize = audioCodec->decode (spkrDataDecoded , data , size);
-
- if (audioCodec->getClockRate() && ((int) audioCodec->getClockRate() != _mainBufferSampleRate)) {
- converter->resample (spkrDataDecoded, spkrDataConverted, audioCodec->getClockRate(), _mainBufferSampleRate, expandedSize);
- audiolayer->getMainBuffer()->putData (spkrDataConverted, expandedSize, call->getCallId());
-
+ int expandedSize = audioCodec->decode (decData, data , size);
+ unsigned int audioRate = audioCodec->getClockRate();
+ int inSamples = expandedSize / sizeof(SFLDataFormat);
+ if (audioRate != mainBufferSampleRate) {
+ int outSize = expandedSize * (mainBufferSampleRate / audioRate);
+ converter->resample (decData, resampledData, mainBufferSampleRate, audioRate, inSamples);
+ audiolayer->getMainBuffer()->putData (resampledData, outSize, call->getCallId());
} else {
- audiolayer->getMainBuffer()->putData (spkrDataDecoded, expandedSize, call->getCallId());
+ audiolayer->getMainBuffer()->putData (decData, expandedSize, call->getCallId());
}
}
@@ -1034,7 +990,6 @@ int IAXVoIPLink::processIAXMsgCount (int msgcount)
// msgcount >= 256 => msgcount/256 old messages , msgcount%256 new messages (RULES)
return msgcount%256;
-
}
@@ -1043,7 +998,6 @@ IAXVoIPLink::iaxHandlePrecallEvent (iax_event* event)
{
IAXCall* call = NULL;
std::string id;
- std::string reason = "Error ringing user.";
switch (event->etype) {
@@ -1117,10 +1071,8 @@ IAXVoIPLink::iaxHandlePrecallEvent (iax_event* event)
} else {
// reject call, unable to add it
- iax_reject (event->session, (char*) reason.c_str());
-
+ iax_reject (event->session, (char*) "Error ringing user.");
delete call;
- call = NULL;
}
break;
@@ -1150,7 +1102,6 @@ IAXVoIPLink::iaxHandlePrecallEvent (iax_event* event)
void IAXVoIPLink::updateAudiolayer (void)
{
_mutexIAX.enterMutex();
- audiolayer = NULL;
audiolayer = Manager::instance().getAudioDriver();
_mutexIAX.leaveMutex();
}
diff --git a/sflphone-common/src/iax/iaxvoiplink.h b/sflphone-common/src/iax/iaxvoiplink.h
index 88da9088d337fe3a74b9d03d42f2e63d0c0639ef..245c09fa6cb1dcbc412766b83c2be820fe20a68f 100644
--- a/sflphone-common/src/iax/iaxvoiplink.h
+++ b/sflphone-common/src/iax/iaxvoiplink.h
@@ -306,14 +306,10 @@ class IAXVoIPLink : public VoIPLink
/** Connection to audio card/device */
AudioLayer* audiolayer;
- /** Mic-data related buffers */
- SFLDataFormat* micData;
- SFLDataFormat* micDataConverted;
- unsigned char* micDataEncoded;
-
- /** Speaker-data related buffers */
- SFLDataFormat* spkrDataDecoded;
- SFLDataFormat* spkrDataConverted;
+ /** encoder/decoder/resampler buffers */
+ SFLDataFormat decData[DEC_BUFFER_SIZE];
+ SFLDataFormat resampledData[DEC_BUFFER_SIZE];
+ unsigned char encodedData[DEC_BUFFER_SIZE];
/** Sample rate converter object */
SamplerateConverter* converter;