diff --git a/sflphone-common/src/audio/alsa/alsalayer.cpp b/sflphone-common/src/audio/alsa/alsalayer.cpp
index 3114c6da5428889ede94132be1e70ad74cf552e0..9291dc1f27b9760bb7fedf7d756597ef6830064a 100644
--- a/sflphone-common/src/audio/alsa/alsalayer.cpp
+++ b/sflphone-common/src/audio/alsa/alsalayer.cpp
@@ -936,7 +936,7 @@ void AlsaLayer::audioCallback (void)
SFLDataFormat *rsmpl_out = (SFLDataFormat*) malloc (playbackAvailBytes);
memset (out, 0, playbackAvailBytes);
- _converter->upsampleData ( (SFLDataFormat*) out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, toGet / sizeof (SFLDataFormat));
+ _converter->resample ( (SFLDataFormat*) out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, toGet / sizeof (SFLDataFormat));
write (rsmpl_out, toGet, _PlaybackHandle);
free (rsmpl_out);
@@ -1026,7 +1026,7 @@ void AlsaLayer::audioCallback (void)
SFLDataFormat* rsmpl_out = (SFLDataFormat*) malloc (framesPerBufferAlsa * sizeof (SFLDataFormat));
- _converter->downsampleData ( (SFLDataFormat*) in, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, toPut / sizeof (SFLDataFormat));
+ _converter->resample ( (SFLDataFormat*) in, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, toPut / sizeof (SFLDataFormat));
_audiofilter->processAudio (rsmpl_out, toPut);
diff --git a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
index b72ccbb64a13f5a8602ab5d6fd8b1ad97ddf21f2..e4c741de1bdfd6360105f28e30a0f96afd204d1c 100644
--- a/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
+++ b/sflphone-common/src/audio/audiortp/AudioRtpRecordHandler.cpp
@@ -287,7 +287,7 @@ int AudioRtpRecordHandler::processDataEncode (void)
// test if resampling is required
if (codecSampleRate != mainBufferSampleRate) {
- _audioRtpRecord._converter->downsampleData (micData, micDataConverted, codecSampleRate, mainBufferSampleRate, bytes / sizeof(SFLDataFormat));
+ _audioRtpRecord._converter->resample (micData, micDataConverted, codecSampleRate, mainBufferSampleRate, bytes / sizeof(SFLDataFormat));
_audioRtpRecord.audioProcessMutex.enter();
@@ -359,7 +359,7 @@ void AudioRtpRecordHandler::processDataDecode (unsigned char *spkrData, unsigned
// test if resampling is required
if (codecSampleRate != mainBufferSampleRate) {
// Do sample rate conversion
- _audioRtpRecord._converter->upsampleData (spkrDataDecoded, spkrDataConverted, codecSampleRate, mainBufferSampleRate, nbSample);
+ _audioRtpRecord._converter->resample (spkrDataDecoded, spkrDataConverted, codecSampleRate, mainBufferSampleRate, nbSample);
if(Manager::instance().getEchoCancelState() == "enabled") {
echoCanceller.putData(spkrDataConverted, expandedSize);
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
index 59fa25b631b1575ef96a97de442b686df32e37c5..e7f081197628f83dd6e412e475c9fefe582e221a 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
@@ -806,7 +806,7 @@ void PulseLayer::writeToSpeaker (void)
SFLDataFormat* rsmpl_out = (SFLDataFormat*) pa_xmalloc (writeableSize);
memset (out, 0, writeableSize);
- _converter->upsampleData ( (SFLDataFormat*) out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, byteToGet / sizeof (SFLDataFormat));
+ _converter->resample ( (SFLDataFormat*) out, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, byteToGet / sizeof (SFLDataFormat));
if ( byteToGet > (unsigned int) writeableSize)
_warn ("Audio: Error: nbsbyte exceed buffer length");
@@ -868,7 +868,7 @@ void PulseLayer::readFromMic (void)
SFLDataFormat* rsmpl_out = (SFLDataFormat*) pa_xmalloc (readableSize);
memset (rsmpl_out, 0, readableSize);
- _converter->downsampleData ( (SFLDataFormat *) data, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, r / sizeof (SFLDataFormat));
+ _converter->resample ( (SFLDataFormat *) data, rsmpl_out, _mainBufferSampleRate, _audioSampleRate, r / sizeof (SFLDataFormat));
// remove dc offset
_audiofilter->processAudio (rsmpl_out, r);
diff --git a/sflphone-common/src/audio/samplerateconverter.cpp b/sflphone-common/src/audio/samplerateconverter.cpp
index 585217a4596baec7b54361235f29bcfc87fe7cf8..2ea2e275c966c479baef7bb584466f9e89fcc954 100644
--- a/sflphone-common/src/audio/samplerateconverter.cpp
+++ b/sflphone-common/src/audio/samplerateconverter.cpp
@@ -31,35 +31,23 @@
#include "samplerateconverter.h"
#include "manager.h"
-SamplerateConverter::SamplerateConverter (int freq , int fs) :
- _floatBufferDownMic (NULL)
- , _floatBufferUpMic (NULL)
- , _src_state_mic (NULL)
- , _floatBufferDownSpkr (NULL)
- , _floatBufferUpSpkr (NULL)
- , _src_state_spkr (NULL)
+SamplerateConverter::SamplerateConverter (int freq , int fs)
{
int err;
- _src_state_mic = src_new (SRC_LINEAR, 1, &err);
- _src_state_spkr = src_new (SRC_LINEAR, 1, &err);
+ _src_state = src_new (SRC_LINEAR, 1, &err);
int nbSamplesMax = (int) ( (freq * fs) / 1000);
- _floatBufferDownMic = new float32[nbSamplesMax];
- _floatBufferUpMic = new float32[nbSamplesMax];
- _floatBufferDownSpkr = new float32[nbSamplesMax];
- _floatBufferUpSpkr = new float32[nbSamplesMax];
+ _floatBufferIn = new float32[nbSamplesMax];
+ _floatBufferOut = new float32[nbSamplesMax];
}
SamplerateConverter::~SamplerateConverter (void)
{
- delete [] _floatBufferUpMic;
- delete [] _floatBufferDownMic;
- delete [] _floatBufferUpSpkr;
- delete [] _floatBufferDownSpkr;
+ delete [] _floatBufferIn;
+ delete [] _floatBufferOut;
- src_delete (_src_state_mic);
- src_delete (_src_state_spkr);
+ src_delete (_src_state);
}
void
@@ -72,50 +60,30 @@ SamplerateConverter::Short2FloatArray (const short *in, float *out, int len)
out[len] = (float) in[len] * .000030517578125f;
}
-
-//TODO Add ifdef for int16 or float32 type
-void SamplerateConverter::upsampleData (SFLDataFormat* dataIn , SFLDataFormat* dataOut, int samplerate1 , int samplerate2 , int nbSamples)
-{
- double upsampleFactor = (double) samplerate2 / samplerate1 ;
-
- 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
-void SamplerateConverter::downsampleData (SFLDataFormat* dataIn , SFLDataFormat* dataOut , int samplerate1 , int samplerate2 , int nbSamples)
+void SamplerateConverter::resample (SFLDataFormat* dataIn , SFLDataFormat* dataOut , int samplerate1 , int samplerate2 , int nbSamples)
{
- double downsampleFactor = (double) samplerate1 / samplerate2;
+ double sampleFactor;
+ if (samplerate1 > samplerate2)
+ sampleFactor = (double) samplerate1 / samplerate2;
+ else
+ sampleFactor = (double) samplerate2 / samplerate1;
- if (downsampleFactor == 1)
+ if (sampleFactor == 1)
return;
SRC_DATA src_data;
- src_data.data_in = _floatBufferUpMic;
- src_data.data_out = _floatBufferDownMic;
+ src_data.data_in = _floatBufferIn;
+ src_data.data_out = _floatBufferOut;
src_data.input_frames = nbSamples;
src_data.output_frames = nbSamples;
- src_data.src_ratio = downsampleFactor;
+ src_data.src_ratio = sampleFactor;
src_data.end_of_input = 0; // More data will come
- Short2FloatArray (dataIn , _floatBufferUpMic, nbSamples);
- src_process (_src_state_mic, &src_data);
+ Short2FloatArray (dataIn , _floatBufferIn, nbSamples);
+ src_process (_src_state, &src_data);
assert(nbSamples == src_data.output_frames_gen);
- src_float_to_short_array (_floatBufferDownMic , dataOut , nbSamples);
+ src_float_to_short_array (_floatBufferOut, dataOut , nbSamples);
}
diff --git a/sflphone-common/src/audio/samplerateconverter.h b/sflphone-common/src/audio/samplerateconverter.h
index 21204ebd7d5e42c6b3db172c4d8fce298cea285d..18f095d735283364790dc46ca904c83d4bb8ad8f 100644
--- a/sflphone-common/src/audio/samplerateconverter.h
+++ b/sflphone-common/src/audio/samplerateconverter.h
@@ -53,22 +53,13 @@ class SamplerateConverter
~SamplerateConverter (void);
/**
- * Upsample from the samplerate1 to the samplerate2
+ * resample from the samplerate1 to the samplerate2
* @param data The data buffer
- * @param SamplerateConverter1 The lower sample rate
- * @param SamplerateConverter2 The higher sample rate
+ * @param SamplerateConverter1 The original sample rate
+ * @param SamplerateConverter2 The desired sample rate
* @param nbSamples The number of samples to process
*/
- void upsampleData (SFLDataFormat* dataIn , SFLDataFormat* dataOut , int samplerate1 , int samplerate2 , int nbSamples);
-
- /**
- * Downsample from the samplerate1 to the samplerate2
- * @param data The data buffer
- * @param SamplerateConverter1 The lower sample rate
- * @param SamplerateConverter2 The higher sample rate
- * @param nbSamples The number of samples to process
- */
- void downsampleData (SFLDataFormat* dataIn , SFLDataFormat* dataOut , int samplerate1 , int samplerate2 , int nbSamples);
+ void resample (SFLDataFormat* dataIn , SFLDataFormat* dataOut , int samplerate1 , int samplerate2 , int nbSamples);
/**
* Convert short table to floats for audio processing
@@ -86,17 +77,10 @@ class SamplerateConverter
// Assignment Operator
SamplerateConverter& operator= (const SamplerateConverter& rh);
- /** Downsampled/Upsampled float buffers for the mic data processing */
- float32* _floatBufferDownMic;
- float32* _floatBufferUpMic;
- /** libSamplerateConverter converter for outgoing voice */
- SRC_STATE* _src_state_mic;
+ float32* _floatBufferIn;
+ float32* _floatBufferOut;
- /** Downsampled/Upsampled float buffers for the speaker data processing */
- float32* _floatBufferDownSpkr;
- float32* _floatBufferUpSpkr;
- /** libSamplerateConverter converter for incoming voice */
- SRC_STATE* _src_state_spkr;
+ SRC_STATE* _src_state;
};
#endif //_SAMPLE_RATE_H
diff --git a/sflphone-common/src/audio/sound/audiofile.cpp b/sflphone-common/src/audio/sound/audiofile.cpp
index 1c78ab490e5e07bfb32c911279e69dbedd7b9797..69761cdab7e81de9a408d695dbc494f1c5eef8b1 100644
--- a/sflphone-common/src/audio/sound/audiofile.cpp
+++ b/sflphone-common/src/audio/sound/audiofile.cpp
@@ -250,9 +250,9 @@ WaveFile::WaveFile (const std::string& fileName, unsigned int audioSamplingRate)
}
if (srate < audioSamplingRate)
- _converter.upsampleData (tempBuffer, _buffer, srate, audioSamplingRate, nbSamples);
+ _converter.resample (tempBuffer, _buffer, srate, audioSamplingRate, nbSamples);
else if (srate > audioSamplingRate)
- _converter.downsampleData (tempBuffer, _buffer, audioSamplingRate, srate, nbSamples);
+ _converter.resample (tempBuffer, _buffer, audioSamplingRate, srate, nbSamples);
delete[] tempBuffer;
} else {
diff --git a/sflphone-common/src/iax/iaxvoiplink.cpp b/sflphone-common/src/iax/iaxvoiplink.cpp
index 6ec4923767b3708e804e3ce964bd91f6a7b409d9..e6f6212ce61ab1856395967686c1e658e755fccb 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
- converter->downsampleData (micData , micDataConverted , (int) audioCodec->getClockRate(), _mainBufferSampleRate, bytes);
+ converter->resample (micData , micDataConverted , (int) audioCodec->getClockRate(), _mainBufferSampleRate, bytes);
compSize = audioCodec->encode (micDataEncoded, micDataConverted , bytes);
} else {
compSize = audioCodec->encode (micDataEncoded, micData, bytes);
@@ -963,7 +963,7 @@ IAXVoIPLink::iaxHandleVoiceEvent (iax_event* event, IAXCall* call)
expandedSize = audioCodec->decode (spkrDataDecoded , data , size);
if (audioCodec->getClockRate() && ((int) audioCodec->getClockRate() != _mainBufferSampleRate)) {
- converter->upsampleData (spkrDataDecoded, spkrDataConverted, audioCodec->getClockRate(), _mainBufferSampleRate, expandedSize);
+ converter->resample (spkrDataDecoded, spkrDataConverted, audioCodec->getClockRate(), _mainBufferSampleRate, expandedSize);
audiolayer->getMainBuffer()->putData (spkrDataConverted, expandedSize, call->getCallId());
} else {