diff --git a/sflphone-common/src/audio/alsa/alsalayer.cpp b/sflphone-common/src/audio/alsa/alsalayer.cpp
index 84dac7b974c20456da7975d6d82abf3975fbd8c3..3114c6da5428889ede94132be1e70ad74cf552e0 100644
--- a/sflphone-common/src/audio/alsa/alsalayer.cpp
+++ b/sflphone-common/src/audio/alsa/alsalayer.cpp
@@ -830,7 +830,6 @@ void AlsaLayer::audioCallback (void)
AudioLoop *file_tone;
SFLDataFormat *out = NULL;
- SFLDataFormat *rsmpl_out = NULL;
notifyincomingCall();
@@ -934,20 +933,12 @@ void AlsaLayer::audioCallback (void)
out[i] = out[i] * spkrVolume / 100;
if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
+ SFLDataFormat *rsmpl_out = (SFLDataFormat*) malloc (playbackAvailBytes);
+ memset (out, 0, playbackAvailBytes);
- // Do sample rate conversion
- int nb_sample_down = toGet / sizeof (SFLDataFormat);
-
- if (rsmpl_out) {
- rsmpl_out = (SFLDataFormat*) malloc (playbackAvailBytes);
- memset (out, 0, playbackAvailBytes);
-
- int nbSample = _converter->upsampleData ( (SFLDataFormat*) out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, nb_sample_down);
- write (rsmpl_out, nbSample*sizeof (SFLDataFormat), _PlaybackHandle);
- free (rsmpl_out);
- }
-
- rsmpl_out = 0;
+ _converter->upsampleData ( (SFLDataFormat*) out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, toGet / sizeof (SFLDataFormat));
+ write (rsmpl_out, toGet, _PlaybackHandle);
+ free (rsmpl_out);
} else {
@@ -1035,14 +1026,11 @@ void AlsaLayer::audioCallback (void)
SFLDataFormat* rsmpl_out = (SFLDataFormat*) malloc (framesPerBufferAlsa * sizeof (SFLDataFormat));
- int nbSample = toPut / sizeof (SFLDataFormat);
- int nb_sample_up = nbSample;
-
- nbSample = _converter->downsampleData ( (SFLDataFormat*) in, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, nb_sample_up);
+ _converter->downsampleData ( (SFLDataFormat*) in, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, toPut / sizeof (SFLDataFormat));
- _audiofilter->processAudio (rsmpl_out, nbSample*sizeof (SFLDataFormat));
+ _audiofilter->processAudio (rsmpl_out, toPut);
- getMainBuffer()->putData (rsmpl_out, nbSample * sizeof (SFLDataFormat));
+ getMainBuffer()->putData (rsmpl_out, toPut);
free (rsmpl_out);
} else {
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
index 7b9b0c83dd27bd4bc6e08751305bde2d86bc53eb..b72ccbb64a13f5a8602ab5d6fd8b1ad97ddf21f2 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
@@ -188,7 +188,9 @@ void AudioRtpRecordHandler::initBuffers()
// initialize SampleRate converter using AudioLayer's sampling rate
// (internal buffers initialized with maximal sampling rate and frame size)
- _audioRtpRecord._converter = new SamplerateConverter ();
+ int rate = getCodecSampleRate();
+ int fs = getCodecFrameSize();
+ _audioRtpRecord._converter = new SamplerateConverter (rate, fs);
int nbSamplesMax = (int) ( (getCodecSampleRate() * getCodecFrameSize() / 1000));
_audioRtpRecord._micData = new SFLDataFormat[nbSamplesMax];
@@ -262,47 +264,35 @@ int AudioRtpRecordHandler::processDataEncode (void)
unsigned char *micDataEncoded = _audioRtpRecord._micDataEncoded;
SFLDataFormat *micDataConverted = _audioRtpRecord._micDataConverted;
- int codecFrameSize = getCodecFrameSize();
int codecSampleRate = getCodecSampleRate();
-
int mainBufferSampleRate = Manager::instance().getMainBuffer()->getInternalSamplingRate();
// compute codec framesize in ms
- float fixedCodecFramesize = computeCodecFrameSize (codecFrameSize, codecSampleRate);
-
+ 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 = computeNbByteAudioLayer (mainBufferSampleRate, fixedCodecFramesize);
-
- // available bytes inside ringbuffer
- int availBytesFromMic = Manager::instance().getMainBuffer()->availForGet (_id);
+ int bytesToGet = (int) ( ( (float) mainBufferSampleRate * fixedCodecFramesize * sizeof (SFLDataFormat)) / 1000.0);
- if (availBytesFromMic < bytesToGet)
+ if (Manager::instance().getMainBuffer()->availForGet (_id) < bytesToGet)
return 0;
- // Get bytes from micRingBuffer to data_from_mic
- int nbSample = Manager::instance().getMainBuffer()->getData (micData, bytesToGet, _id) / sizeof (SFLDataFormat);
+ int bytes = Manager::instance().getMainBuffer()->getData (micData, bytesToGet, _id);
+ if (bytes == 0)
+ return 0;
- // process mic fade in
if (!_audioRtpRecord._micFadeInComplete)
- _audioRtpRecord._micFadeInComplete = fadeIn (micData, nbSample, &_audioRtpRecord._micAmplFactor);
-
- if (nbSample == 0)
- return nbSample;
+ _audioRtpRecord._micFadeInComplete = fadeIn (micData, bytes / sizeof(SFLDataFormat), &_audioRtpRecord._micAmplFactor);
// nb bytes to be sent over RTP
- int compSize = 0;
+ int compSize;
// test if resampling is required
if (codecSampleRate != mainBufferSampleRate) {
-
- int nbSampleUp = nbSample;
-
- nbSample = _audioRtpRecord._converter->downsampleData (micData, micDataConverted, codecSampleRate, mainBufferSampleRate, nbSampleUp);
+ _audioRtpRecord._converter->downsampleData (micData, micDataConverted, codecSampleRate, mainBufferSampleRate, bytes / sizeof(SFLDataFormat));
_audioRtpRecord.audioProcessMutex.enter();
if (Manager::instance().audioPreference.getNoiseReduce()) {
- _audioRtpRecord._audioProcess->processAudio (micDataConverted, nbSample * sizeof (SFLDataFormat));
+ _audioRtpRecord._audioProcess->processAudio (micDataConverted, bytes);
}
if(Manager::instance().getEchoCancelState() == "enabled") {
@@ -313,22 +303,18 @@ int AudioRtpRecordHandler::processDataEncode (void)
_audioRtpRecord.audioCodecMutex.enter();
- compSize = _audioRtpRecord._audioCodec->encode (micDataEncoded, micData, nbSample * sizeof (SFLDataFormat));
+ compSize = _audioRtpRecord._audioCodec->encode (micDataEncoded, micData, bytes);
_audioRtpRecord.audioCodecMutex.leave();
- } else {
-
+ } else { // no resampling required
_audioRtpRecord.audioProcessMutex.enter();
- if (Manager::instance().audioPreference.getNoiseReduce()) {
- _audioRtpRecord._audioProcess->processAudio (micData, nbSample * sizeof (SFLDataFormat));
- }
+ if (Manager::instance().audioPreference.getNoiseReduce())
+ _audioRtpRecord._audioProcess->processAudio (micData, bytes);
- if(Manager::instance().getEchoCancelState() == "enabled") {
+ if(Manager::instance().getEchoCancelState() == "enabled")
echoCanceller.getData(micData);
- }
-
#ifdef DUMP_PROCESS_DATA_ENCODE
teststream.write(reinterpret_cast<char *>(micData), nbSample * sizeof(SFLDataFormat));
@@ -337,10 +323,7 @@ int AudioRtpRecordHandler::processDataEncode (void)
_audioRtpRecord.audioProcessMutex.leave();
_audioRtpRecord.audioCodecMutex.enter();
-
- // no resampling required
- compSize = _audioRtpRecord._audioCodec->encode (micDataEncoded, micData, nbSample * sizeof (SFLDataFormat));
-
+ compSize = _audioRtpRecord._audioCodec->encode (micDataEncoded, micData, bytes);
_audioRtpRecord.audioCodecMutex.leave();
}
@@ -376,15 +359,14 @@ void AudioRtpRecordHandler::processDataDecode (unsigned char *spkrData, unsigned
// test if resampling is required
if (codecSampleRate != mainBufferSampleRate) {
// Do sample rate conversion
- nbSample = _audioRtpRecord._converter->upsampleData (spkrDataDecoded, spkrDataConverted, codecSampleRate, mainBufferSampleRate, nbSample);
+ _audioRtpRecord._converter->upsampleData (spkrDataDecoded, spkrDataConverted, codecSampleRate, mainBufferSampleRate, nbSample);
if(Manager::instance().getEchoCancelState() == "enabled") {
- echoCanceller.putData(spkrDataConverted, nbSample * sizeof(SFLDataFormat));
+ echoCanceller.putData(spkrDataConverted, expandedSize);
}
// put data in audio layer, size in byte
- Manager::instance().getMainBuffer()->putData (spkrDataConverted, nbSample * sizeof (SFLDataFormat), _id);
-
+ Manager::instance().getMainBuffer()->putData (spkrDataConverted, expandedSize, _id);
} else {
if(Manager::instance().getEchoCancelState() == "enabled") {
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h
index 07ed3df713b7537c1eec0533ecfd7bdc4cb9ec60..21a4e8a08177220cd592b47d6f8bfbb5a4070b69 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.h
@@ -174,14 +174,6 @@ class AudioRtpRecordHandler
return _audioRtpRecord._micDataEncoded;
}
- float computeCodecFrameSize (int codecSamplePerFrame, int codecClockRate) const {
- return ( (float) codecSamplePerFrame * 1000.0) / (float) codecClockRate;
- }
-
- int computeNbByteAudioLayer (int mainBufferSamplingRate, float codecFrameSize) const {
- return (int) ( ( (float) mainBufferSamplingRate * codecFrameSize * sizeof (SFLDataFormat)) / 1000.0);
- }
-
void init (void);
/**
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
index b4d38ba09eccaf25c432f7b1966bbc8c95015faa..59fa25b631b1575ef96a97de442b686df32e37c5 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
@@ -768,7 +768,6 @@ void PulseLayer::writeToSpeaker (void)
pa_stream_write (playback->pulseStream(), out, copied * sizeof (SFLDataFormat), NULL, 0, PA_SEEK_RELATIVE);
pa_xfree (out);
-
}
}
else {
@@ -794,11 +793,8 @@ void PulseLayer::writeToSpeaker (void)
byteToGet = (normalAvailBytes < (int) (maxNbBytesToGet)) ? normalAvailBytes : maxNbBytesToGet;
if (byteToGet) {
-
// Sending an odd number of byte breaks the audio!
- if ( (byteToGet%2) != 0) {
- byteToGet = byteToGet-1;
- }
+ byteToGet &= ~1;
out = (SFLDataFormat*) pa_xmalloc (maxNbBytesToGet);
memset (out, 0, maxNbBytesToGet);
@@ -807,29 +803,22 @@ void PulseLayer::writeToSpeaker (void)
// test if resampling is required
if (_mainBufferSampleRate && ( (int) _audioSampleRate != _mainBufferSampleRate)) {
-
SFLDataFormat* rsmpl_out = (SFLDataFormat*) pa_xmalloc (writeableSize);
memset (out, 0, writeableSize);
- // Do sample rate conversion
- int nb_sample_down = byteToGet / sizeof (SFLDataFormat);
+ _converter->upsampleData ( (SFLDataFormat*) out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, byteToGet / sizeof (SFLDataFormat));
- int nbSample = _converter->upsampleData ( (SFLDataFormat*) out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, nb_sample_down);
-
- if ( (nbSample*sizeof (SFLDataFormat)) > (unsigned int) writeableSize) {
+ if ( byteToGet > (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_stream_write (playback->pulseStream(), rsmpl_out, byteToGet, NULL, 0, PA_SEEK_RELATIVE);
- // free resampling buffer
pa_xfree (rsmpl_out);
} else {
// write origin data
pa_stream_write (playback->pulseStream(), out, byteToGet, NULL, 0, PA_SEEK_RELATIVE);
- }
+ }
// free audio buffer used to get data from
pa_xfree (out);
@@ -855,7 +844,6 @@ void PulseLayer::writeToSpeaker (void)
_urgentRingBuffer.Discard (byteToGet);
}
-
}
void PulseLayer::readFromMic (void)
@@ -880,16 +868,12 @@ void PulseLayer::readFromMic (void)
SFLDataFormat* rsmpl_out = (SFLDataFormat*) pa_xmalloc (readableSize);
memset (rsmpl_out, 0, readableSize);
- int nbSample = r / sizeof (SFLDataFormat);
-
- int nb_sample_up = nbSample;
-
- nbSample = _converter->downsampleData ( (SFLDataFormat *) data, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, nb_sample_up);
+ _converter->downsampleData ( (SFLDataFormat *) data, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, r / sizeof (SFLDataFormat));
// remove dc offset
- _audiofilter->processAudio (rsmpl_out, nbSample*sizeof (SFLDataFormat));
+ _audiofilter->processAudio (rsmpl_out, r);
- getMainBuffer()->putData (rsmpl_out, nbSample*sizeof (SFLDataFormat));
+ getMainBuffer()->putData (rsmpl_out, r);
pa_xfree (rsmpl_out);
diff --git a/sflphone-common/src/audio/samplerateconverter.cpp b/sflphone-common/src/audio/samplerateconverter.cpp
index e9a75b21fdd315646571070f9f165958095685d3..585217a4596baec7b54361235f29bcfc87fe7cf8 100644
--- a/sflphone-common/src/audio/samplerateconverter.cpp
+++ b/sflphone-common/src/audio/samplerateconverter.cpp
@@ -31,67 +31,19 @@
#include "samplerateconverter.h"
#include "manager.h"
-SamplerateConverter::SamplerateConverter (int freq , int fs)
- : _frequence (freq)
- , _framesize (fs)
- , _floatBufferDownMic (NULL)
+SamplerateConverter::SamplerateConverter (int freq , int fs) :
+ _floatBufferDownMic (NULL)
, _floatBufferUpMic (NULL)
, _src_state_mic (NULL)
, _floatBufferDownSpkr (NULL)
, _floatBufferUpSpkr (NULL)
, _src_state_spkr (NULL)
- , _src_err (0)
{
- init();
-}
-
-SamplerateConverter::~SamplerateConverter (void)
-{
-
- if (_floatBufferUpMic) {
- delete [] _floatBufferUpMic;
- _floatBufferUpMic = NULL;
- }
-
- if (_floatBufferDownMic) {
- delete [] _floatBufferDownMic;
- _floatBufferDownMic = NULL;
- }
-
- if (_floatBufferUpSpkr) {
- delete [] _floatBufferUpSpkr;
- _floatBufferUpSpkr = NULL;
- }
-
- if (_floatBufferDownSpkr) {
- delete [] _floatBufferDownSpkr;
- _floatBufferDownSpkr = NULL;
- }
-
- // libSamplerateConverter-related
- if (_src_state_mic) {
- _src_state_mic = src_delete (_src_state_mic);
- _src_state_mic = NULL;
- }
-
- if (_src_state_spkr) {
- _src_state_spkr = src_delete (_src_state_spkr);
- _src_state_spkr = NULL;
- }
-}
+ int err;
+ _src_state_mic = src_new (SRC_LINEAR, 1, &err);
+ _src_state_spkr = src_new (SRC_LINEAR, 1, &err);
-void SamplerateConverter::init (void)
-{
-
- // libSamplerateConverter-related
- // Set the converter type for the upsampling and the downsampling
- // interpolator SRC_SINC_BEST_QUALITY
- // interpolator SRC_SINC_FASTEST
- // interpolator SRC_LINEAR
- _src_state_mic = src_new (SRC_LINEAR, 1, &_src_err);
- _src_state_spkr = src_new (SRC_LINEAR, 1, &_src_err);
-
- int nbSamplesMax = (int) ( (_frequence * _framesize) / 1000);
+ int nbSamplesMax = (int) ( (freq * fs) / 1000);
_floatBufferDownMic = new float32[nbSamplesMax];
_floatBufferUpMic = new float32[nbSamplesMax];
@@ -99,78 +51,71 @@ void SamplerateConverter::init (void)
_floatBufferUpSpkr = new float32[nbSamplesMax];
}
+SamplerateConverter::~SamplerateConverter (void)
+{
+ delete [] _floatBufferUpMic;
+ delete [] _floatBufferDownMic;
+ delete [] _floatBufferUpSpkr;
+ delete [] _floatBufferDownSpkr;
+
+ src_delete (_src_state_mic);
+ src_delete (_src_state_spkr);
+}
+
void
SamplerateConverter::Short2FloatArray (const short *in, float *out, int len)
{
// factor is 1/(2^15), used to rescale the short int range to the
// [-1.0 - 1.0] float range.
-#define S2F_FACTOR .000030517578125f;
- while (len) {
- len--;
- out[len] = (float) in[len] * S2F_FACTOR;
- }
+ while (len--)
+ out[len] = (float) in[len] * .000030517578125f;
}
//TODO Add ifdef for int16 or float32 type
-int SamplerateConverter::upsampleData (SFLDataFormat* dataIn , SFLDataFormat* dataOut, int samplerate1 , int samplerate2 , int nbSamples)
+void SamplerateConverter::upsampleData (SFLDataFormat* dataIn , SFLDataFormat* dataOut, int samplerate1 , int samplerate2 , int nbSamples)
{
-
double upsampleFactor = (double) samplerate2 / samplerate1 ;
- int nbSamplesMax = (int) (samplerate2 * _framesize / 1000);
-
- if (upsampleFactor != 1 && dataIn != NULL) {
- SRC_DATA src_data;
- src_data.data_in = _floatBufferDownSpkr;
- src_data.data_out = _floatBufferUpSpkr;
- src_data.input_frames = nbSamples;
- src_data.output_frames = (int) floor (upsampleFactor * nbSamples);
- src_data.src_ratio = upsampleFactor;
- src_data.end_of_input = 0; // More data will come
- // _debug(" upsample %d %d %f %d" , src_data.input_frames , src_data.output_frames, src_data.src_ratio , nbSamples);
- // Override libsamplerate conversion function
- Short2FloatArray (dataIn , _floatBufferDownSpkr, nbSamples);
- //src_short_to_float_array (dataIn , _floatBufferDownSpkr, nbSamples);
- //_debug("upsample %d %f %d" , src_data.output_frames, src_data.src_ratio , nbSamples);
- src_process (_src_state_spkr, &src_data);
- // _debug(" upsample %d %d %d" , samplerate1, samplerate2 , nbSamples);
- nbSamples = (src_data.output_frames_gen > nbSamplesMax) ? nbSamplesMax : src_data.output_frames_gen;
- src_float_to_short_array (_floatBufferUpSpkr, dataOut, nbSamples);
- //_debug("upsample %d %d %d" , samplerate1, samplerate2 , nbSamples);
- }
-
- return nbSamples;
+ if (upsampleFactor == 1)
+ return;
+
+ SRC_DATA src_data;
+ src_data.data_in = _floatBufferDownSpkr;
+ src_data.data_out = _floatBufferUpSpkr;
+ src_data.input_frames = nbSamples;
+ src_data.output_frames = nbSamples;
+ src_data.src_ratio = upsampleFactor;
+ src_data.end_of_input = 0; // More data will come
+
+ Short2FloatArray (dataIn , _floatBufferDownSpkr, nbSamples);
+ src_process (_src_state_spkr, &src_data);
+
+ assert(nbSamples == src_data.output_frames_gen);
+ src_float_to_short_array (_floatBufferUpSpkr, dataOut, nbSamples);
}
//TODO Add ifdef for int16 or float32 type
-int SamplerateConverter::downsampleData (SFLDataFormat* dataIn , SFLDataFormat* dataOut , int samplerate1 , int samplerate2 , int nbSamples)
+void SamplerateConverter::downsampleData (SFLDataFormat* dataIn , SFLDataFormat* dataOut , int samplerate1 , int samplerate2 , int nbSamples)
{
-
double downsampleFactor = (double) samplerate1 / samplerate2;
- int nbSamplesMax = (int) (samplerate1 * _framesize / 1000);
-
- if (downsampleFactor != 1) {
- SRC_DATA src_data;
- src_data.data_in = _floatBufferUpMic;
- src_data.data_out = _floatBufferDownMic;
- src_data.input_frames = nbSamples;
- src_data.output_frames = (int) floor (downsampleFactor * nbSamples);
- src_data.src_ratio = downsampleFactor;
- src_data.end_of_input = 0; // More data will come
- //_debug("downsample %d %f %d" , src_data.output_frames, src_data.src_ratio , nbSamples);
- // Override libsamplerate conversion function
- Short2FloatArray (dataIn , _floatBufferUpMic, nbSamples);
- //src_short_to_float_array (dataIn, _floatBufferUpMic, nbSamples);
- //_debug("downsample %d %f %d" , src_data.output_frames, src_data.src_ratio , nbSamples);
- src_process (_src_state_mic, &src_data);
- //_debug("downsample %d %f %d" , src_data.output_frames, src_data.src_ratio , nbSamples);
- nbSamples = (src_data.output_frames_gen > nbSamplesMax) ? nbSamplesMax : src_data.output_frames_gen;
- //_debug("downsample %d %f %d" , src_data.output_frames, src_data.src_ratio , nbSamples);
- src_float_to_short_array (_floatBufferDownMic , dataOut , nbSamples);
- }
-
- return nbSamples;
+ if (downsampleFactor == 1)
+ return;
+
+ SRC_DATA src_data;
+ src_data.data_in = _floatBufferUpMic;
+ src_data.data_out = _floatBufferDownMic;
+ src_data.input_frames = nbSamples;
+ src_data.output_frames = nbSamples;
+ src_data.src_ratio = downsampleFactor;
+ src_data.end_of_input = 0; // More data will come
+
+ Short2FloatArray (dataIn , _floatBufferUpMic, nbSamples);
+ src_process (_src_state_mic, &src_data);
+
+ assert(nbSamples == src_data.output_frames_gen);
+
+ src_float_to_short_array (_floatBufferDownMic , dataOut , nbSamples);
}
diff --git a/sflphone-common/src/audio/samplerateconverter.h b/sflphone-common/src/audio/samplerateconverter.h
index a6c3b50ef7c08ef7212ce90e4ff45b5c868155ad..21204ebd7d5e42c6b3db172c4d8fce298cea285d 100644
--- a/sflphone-common/src/audio/samplerateconverter.h
+++ b/sflphone-common/src/audio/samplerateconverter.h
@@ -47,7 +47,7 @@ class SamplerateConverter
* internal buffer size. Converter must be reinitialized
* every time these parameters change
*/
- SamplerateConverter (int freq=44100, int frameSize=20);
+ SamplerateConverter (int freq, int frameSize);
/** Destructor */
~SamplerateConverter (void);
@@ -58,9 +58,8 @@ class SamplerateConverter
* @param SamplerateConverter1 The lower sample rate
* @param SamplerateConverter2 The higher sample rate
* @param nbSamples The number of samples to process
- * @return int The number of samples after the operation
*/
- int upsampleData (SFLDataFormat* dataIn , SFLDataFormat* dataOut , int samplerate1 , int samplerate2 , int nbSamples);
+ void upsampleData (SFLDataFormat* dataIn , SFLDataFormat* dataOut , int samplerate1 , int samplerate2 , int nbSamples);
/**
* Downsample from the samplerate1 to the samplerate2
@@ -68,9 +67,8 @@ class SamplerateConverter
* @param SamplerateConverter1 The lower sample rate
* @param SamplerateConverter2 The higher sample rate
* @param nbSamples The number of samples to process
- * @return int The number of samples after the operation
*/
- int downsampleData (SFLDataFormat* dataIn , SFLDataFormat* dataOut , int samplerate1 , int samplerate2 , int nbSamples);
+ void downsampleData (SFLDataFormat* dataIn , SFLDataFormat* dataOut , int samplerate1 , int samplerate2 , int nbSamples);
/**
* Convert short table to floats for audio processing
@@ -88,12 +86,6 @@ class SamplerateConverter
// Assignment Operator
SamplerateConverter& operator= (const SamplerateConverter& rh);
- void init (void);
-
- /** Audio layer caracteristics */
- int _frequence;
- int _framesize;
-
/** Downsampled/Upsampled float buffers for the mic data processing */
float32* _floatBufferDownMic;
float32* _floatBufferUpMic;
@@ -105,10 +97,6 @@ class SamplerateConverter
float32* _floatBufferUpSpkr;
/** libSamplerateConverter converter for incoming voice */
SRC_STATE* _src_state_spkr;
- /** libSamplerateConverter error */
- int _src_err;
-
-
};
#endif //_SAMPLE_RATE_H
diff --git a/sflphone-common/src/audio/sound/audiofile.cpp b/sflphone-common/src/audio/sound/audiofile.cpp
index 311e0ee914f6f04b8ea574c89ccc3eb64e859401..5214504beac6059c39a9a80d2ac9a1bb70d48f44 100644
--- a/sflphone-common/src/audio/sound/audiofile.cpp
+++ b/sflphone-common/src/audio/sound/audiofile.cpp
@@ -397,20 +397,19 @@ void WaveFile::openExistingWaveFile (const std::string& fileName, int audioSampl
while (remainingSamples > 0) {
int toProcess = remainingSamples > nbSampleMax ? nbSampleMax : remainingSamples;
- int nbSamplesConverted = 0;
if (srate < audioSamplingRate) {
- nbSamplesConverted = _converter.upsampleData (in, out, srate, audioSamplingRate, toProcess);
+ _converter.upsampleData (in, out, srate, audioSamplingRate, toProcess);
} else if (srate > audioSamplingRate) {
- nbSamplesConverted = _converter.downsampleData (in, out, audioSamplingRate, srate, toProcess);
+ _converter.downsampleData (in, out, audioSamplingRate, srate, toProcess);
}
// nbSamplesConverted = nbSamplesConverted*2;
in += toProcess;
- out += nbSamplesConverted;
+ out += toProcess;
remainingSamples -= toProcess;
- totalprocessed += nbSamplesConverted;
+ totalprocessed += toProcess;
}
}
diff --git a/sflphone-common/src/iax/iaxvoiplink.cpp b/sflphone-common/src/iax/iaxvoiplink.cpp
index 84838945edb0eb302819ce79d25dbd926984f1f3..6ec4923767b3708e804e3ce964bd91f6a7b409d9 100644
--- a/sflphone-common/src/iax/iaxvoiplink.cpp
+++ b/sflphone-common/src/iax/iaxvoiplink.cpp
@@ -315,7 +315,7 @@ IAXVoIPLink::sendAudioFromMic (void)
int compSize;
if (audioCodec->getClockRate() && ((int) audioCodec->getClockRate() != _mainBufferSampleRate)) {
// resample
- bytes = converter->downsampleData (micData , micDataConverted , (int) audioCodec->getClockRate(), _mainBufferSampleRate, bytes);
+ converter->downsampleData (micData , micDataConverted , (int) audioCodec->getClockRate(), _mainBufferSampleRate, bytes);
compSize = audioCodec->encode (micDataEncoded, micDataConverted , bytes);
} else {
compSize = audioCodec->encode (micDataEncoded, micData, bytes);
@@ -928,67 +928,47 @@ IAXVoIPLink::iaxHandleVoiceEvent (iax_event* event, IAXCall* call)
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
-
- 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
- // - subclass holds the voiceformat property.
-
- if (event->subclass && event->subclass != call->getFormat()) {
- _debug ("iaxHandleVoiceEvent: no format found in call setting it to %i", event->subclass);
- call->setFormat (event->subclass);
- }
-
- data = (unsigned char*) event->data;
-
- size = event->datalen;
-
- // Decode data with relevant codec
- max = (int) (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);
-
- nbInt16 = expandedSize/sizeof (int16);
+ Manager::instance().getMainBuffer ()->setInternalSamplingRate (audioCodec->getClockRate ());
- if (nbInt16 > max) {
- _debug ("We have decoded an IAX VOICE packet larger than expected: %i VS %i. Cropping.", nbInt16, max);
- nbInt16 = max;
- }
+ // If we receive datalen == 0, some things of the jitter buffer in libiax2/iax.c
+ // were triggered
- nbSample_ = nbInt16;
+ int _mainBufferSampleRate = Manager::instance().getMainBuffer()->getInternalSamplingRate();
- // test if resampling is required
+ // On-the-fly codec changing (normally, when we receive a full packet)
+ // as per http://tools.ietf.org/id/draft-guy-iax-03.txt
+ // - subclass holds the voiceformat property.
- if (audioCodec->getClockRate() && ((int) audioCodec->getClockRate() != _mainBufferSampleRate)) {
+ if (event->subclass && event->subclass != call->getFormat()) {
+ _debug ("iaxHandleVoiceEvent: no format found in call setting it to %i", event->subclass);
+ call->setFormat (event->subclass);
+ }
- // resample
- nbInt16 = converter->upsampleData (spkrDataDecoded, spkrDataConverted, audioCodec->getClockRate(), _mainBufferSampleRate, nbSample_);
+ data = (unsigned char*) event->data;
- /* Write the data to the mic ring buffer */
- audiolayer->getMainBuffer()->putData (spkrDataConverted, nbInt16 * sizeof (SFLDataFormat), call->getCallId());
+ size = event->datalen;
- } else {
+ // Decode data with relevant codec
+ max = (int) (audioCodec->getClockRate() * audiolayer->getFrameSize() / 1000);
- /* Write the data to the mic ring buffer */
- audiolayer->getMainBuffer()->putData (spkrDataDecoded, nbInt16 * sizeof (SFLDataFormat), call->getCallId());
+ 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);
- } else {
- _debug ("IAX: incoming audio, but no sound card open");
- }
+ if (audioCodec->getClockRate() && ((int) audioCodec->getClockRate() != _mainBufferSampleRate)) {
+ converter->upsampleData (spkrDataDecoded, spkrDataConverted, audioCodec->getClockRate(), _mainBufferSampleRate, expandedSize);
+ audiolayer->getMainBuffer()->putData (spkrDataConverted, expandedSize, call->getCallId());
+ } else {
+ audiolayer->getMainBuffer()->putData (spkrDataDecoded, expandedSize, call->getCallId());
+ }
}
/**