diff --git a/sflphone-common/src/audio/echocancel.cpp b/sflphone-common/src/audio/echocancel.cpp
index d957853560520930b15f8d7c1d147f933baba81f..ea5441bb47937a63e56b78b61033231dc1e1d0ec 100644
--- a/sflphone-common/src/audio/echocancel.cpp
+++ b/sflphone-common/src/audio/echocancel.cpp
@@ -32,12 +32,15 @@
// #include <fstream>
+#define MICDELAY 32000
+
EchoCancel::EchoCancel(int smplRate, int frameLength) : _samplingRate(smplRate),
_frameLength(frameLength),
_smplPerFrame(0),
_smplPerSeg(0),
_nbSegmentPerFrame(0),
- _historyLength(0),
+ _micHistoryLength(0),
+ _spkrHistoryLength(0),
_spkrLevel(0),
_micLevel(0),
_spkrHistCnt(0),
@@ -62,11 +65,11 @@ EchoCancel::EchoCancel(int smplRate, int frameLength) : _samplingRate(smplRate),
micFile = new ofstream("micData", ofstream::binary);
echoFile = new ofstream("echoData", ofstream::binary);
spkrFile = new ofstream("spkrData", ofstream::binary);
+
+ micLevelData = new ofstream("micLevelData", ofstream::binary);
+ spkrLevelData = new ofstream("spkrLevelData", ofstream::binary);
*/
- micLearningData = new ofstream("micLearningData", ofstream::binary);
- spkrLearningData = new ofstream("spkrLearningData", ofstream::binary);
-
_micData = new RingBuffer(50000);
_spkrData = new RingBuffer(50000);
_spkrDataOut = new RingBuffer(50000);
@@ -80,7 +83,8 @@ EchoCancel::EchoCancel(int smplRate, int frameLength) : _samplingRate(smplRate),
_smplPerFrame = (_samplingRate * _frameLength) / MS_PER_SEC;
_smplPerSeg = (_samplingRate * SEGMENT_LENGTH) / MS_PER_SEC;
- _historyLength = ECHO_LENGTH / SEGMENT_LENGTH;
+ _micHistoryLength = MIC_LENGTH / SEGMENT_LENGTH;
+ _spkrHistoryLength = SPKR_LENGTH / SEGMENT_LENGTH;
_nbSegmentPerFrame = _frameLength / SEGMENT_LENGTH;
@@ -130,12 +134,13 @@ EchoCancel::~EchoCancel()
delete micFile;
delete spkrFile;
delete echoFile;
+
+ micLevelData->close();
+ spkrLevelData->close();
+ delete micLevelData;
+ delete spkrLevelData;
*/
- micLearningData->close();
- spkrLearningData->close();
- delete micLearningData;
- delete spkrLearningData;
}
@@ -156,7 +161,8 @@ void EchoCancel::reset()
_smplPerFrame = (_samplingRate * _frameLength) / MS_PER_SEC;
_smplPerSeg = (_samplingRate * SEGMENT_LENGTH) / MS_PER_SEC;
- _historyLength = ECHO_LENGTH / SEGMENT_LENGTH;
+ _micHistoryLength = MIC_LENGTH / SEGMENT_LENGTH;
+ _spkrHistoryLength = SPKR_LENGTH / SEGMENT_LENGTH;
_nbSegmentPerFrame = _frameLength / SEGMENT_LENGTH;
memset(_delayLineAmplify, 0, MAX_DELAY_LINE_AMPL*sizeof(float));
@@ -174,10 +180,12 @@ void EchoCancel::reset()
_spkrData->flushAll();
_spkrDataOut->flushAll();
- // SFLDataFormat delay[960];
- // memset(delay, 0, sizeof(SFLDataFormat));
+ /*
+ SFLDataFormat delay[MICDELAY];
+ memset(delay, 0, sizeof(SFLDataFormat));
- // _micData->Put(delay, 960*2);
+ _micData->Put(delay, MICDELAY*2);
+ */
speex_preprocess_state_destroy(_noiseState);
@@ -202,19 +210,21 @@ void EchoCancel::putData(SFLDataFormat *inputData, int nbBytes)
{
// std::cout << "putData nbBytes: " << nbBytes << std::endl;
- /*
+
if(_spkrStoped) {
_debug("EchoCancel: Flush data");
_micData->flushAll();
_spkrData->flushAll();
_spkrStoped = false;
}
- */
// Put data in speaker ring buffer
_spkrData->Put(inputData, nbBytes);
_spkrDataOut->Put(inputData, nbBytes);
+ // _debug("********************** signal event *****************");
+ // _event.signal();
+
}
int EchoCancel::getData(SFLDataFormat *outputData)
@@ -235,11 +245,16 @@ void EchoCancel::process(SFLDataFormat *data, int nbBytes) {}
int EchoCancel::process(SFLDataFormat *inputData, SFLDataFormat *outputData, int nbBytes)
{
- /*
+
if(_spkrStoped) {
return 0;
}
- */
+
+ // _debug("****************************** wait signal ********************************");
+
+ // _event.wait();
+
+ // _debug("****************************** process ************************ ********");
int byteSize = _smplPerFrame*sizeof(SFLDataFormat);
@@ -260,16 +275,32 @@ int EchoCancel::process(SFLDataFormat *inputData, SFLDataFormat *outputData, int
// Init number of frame processed
int nbFrame = 0;
+
// Get data from mic and speaker internal buffer
- while((spkrAvail >= byteSize) && (micAvail >= byteSize)) {
+ // while((spkrAvail >= byteSize) && (micAvail >= byteSize)) {
+ if ((spkrAvail >= byteSize) && (micAvail >= byteSize)) {
+
+ _debug("----------- ProcessData ----------");
// get synchronized data
_spkrData->Get(_tmpSpkr, byteSize);
_micData->Get(_tmpMic, byteSize);
+ // micFile->write((const char *)_tmpMic, byteSize);
+ // spkrFile->write((const char *)_tmpSpkr, byteSize);
+
+ /*
+ while((computeAmplitudeLevel(_tmpSpkr, _smplPerFrame) < MIN_SIG_LEVEL) && (spkrAvail >= byteSize)) {
+ _spkrData->Get(_tmpSpkr, byteSize);
+ spkrAvail = _spkrData->AvailForGet();
+ }
+ */
+
// Processed echo cancellation
performEchoCancel(_tmpMic, _tmpSpkr, _tmpOut);
+ // echoFile->write((const char *)_tmpOut, byteSize);
+
// Remove noise
speex_preprocess_run(_noiseState, _tmpOut);
@@ -287,6 +318,10 @@ int EchoCancel::process(SFLDataFormat *inputData, SFLDataFormat *outputData, int
++nbFrame;
}
+
+
+ // _event.reset();
+
return nbFrame * _smplPerFrame;
}
@@ -306,13 +341,23 @@ void EchoCancel::setSamplingRate(int smplRate) {
void EchoCancel::performEchoCancel(SFLDataFormat *micData, SFLDataFormat *spkrData, SFLDataFormat *outputData) {
- for(int k = 0; k < _nbSegmentPerFrame; k++) {
+ // int tempmiclevel[_nbSegmentPerFrame];
+ // int tempspkrlevel[_nbSegmentPerFrame];
+ for(int k = 0; k < _nbSegmentPerFrame; k++) {
+ /*
updateEchoCancel(micData+(k*_smplPerSeg), spkrData+(k*_smplPerSeg));
- _spkrLevel = getMaxAmplitude(_avgSpkrLevelHist);
- _micLevel = getMaxAmplitude(_avgMicLevelHist);
+ _spkrLevel = getMaxAmplitude(_avgSpkrLevelHist, _spkrHistoryLength);
+ _micLevel = getMaxAmplitude(_avgMicLevelHist, _micHistoryLength);
+
+ // _debug("_spkrLevel: (max): %d", _spkrLevel);
+ // _debug("_micLevel: (min): %d", _micLevel);
+ tempspkrlevel[k] = _spkrLevel;
+ tempmiclevel[k] = _micLevel;
+ */
+ /*
if(_micLevel >= MIN_SIG_LEVEL) {
if(_spkrLevel < MIN_SIG_LEVEL) {
increaseFactor(0.02);
@@ -323,46 +368,73 @@ void EchoCancel::performEchoCancel(SFLDataFormat *micData, SFLDataFormat *spkrDa
else {
// decreaseFactor();
_amplFactor = 0.0;
+ _debug("CANCELECHO AAAAAA");
}
}
else {
if(_spkrLevel < MIN_SIG_LEVEL) {
increaseFactor(0.02);
+ // _amplFactor = 0.0;
+ // _debug("CANCELECHO BBBBBBB");
}
else {
// decreaseFactor();
_amplFactor = 0.0;
+ _debug("CANCELECHO CCCCCCCC");
}
}
+ */
+ /*
+ if(_spkrLevel >= MIN_SIG_LEVEL) {
+ _amplFactor = 0.0;
+ // _debug("------------------------------------------ echocancel");
+ }
+ else {
+ increaseFactor(0.02);
+ }
// lowpass filtering
float amplify = (_lastAmplFactor + _amplFactor) / 2;
_lastAmplFactor = _amplFactor;
-
amplifySignal(micData+(k*_smplPerSeg), outputData+(k*_smplPerSeg), amplify);
+
+ */
+ amplifySignal(micData+(k*_smplPerSeg), outputData+(k*_smplPerSeg), 1.0);
}
+
+ // micLevelData->write((const char *)tempmiclevel, sizeof(int)*_nbSegmentPerFrame);
+ // spkrLevelData->write((const char *)tempspkrlevel, sizeof(int)*_nbSegmentPerFrame);
}
void EchoCancel::updateEchoCancel(SFLDataFormat *micData, SFLDataFormat *spkrData) {
- int micLvl = computeAmplitudeLevel(micData);
- int spkrLvl = computeAmplitudeLevel(spkrData);
+ int micLvl = computeAmplitudeLevel(micData, _smplPerSeg);
+ int spkrLvl = computeAmplitudeLevel(spkrData, _smplPerSeg);
// Add 1 to make sure we are not dividing by 0
_avgMicLevelHist[_micHistCnt++] = micLvl+1;
_avgSpkrLevelHist[_spkrHistCnt++] = spkrLvl+1;
+
+ // spkrLevelData->write((const char *)_spkrAdaptArray, _spkrAdaptSize*sizeof(int));
+ // micLevelData->write((const char *)_micAdaptArray, _micAdaptSize*sizeof(int));
- if(_micHistCnt >= _historyLength)
+
+ // _debug("micLevel: %d", micLvl);
+ // _debug("spkrLevel: %d", spkrLvl);
+
+ if(_micHistCnt >= _micHistoryLength)
_micHistCnt = 0;
- if(_spkrHistCnt >= _historyLength)
+ if(_spkrHistCnt >= _spkrHistoryLength)
_spkrHistCnt = 0;
+ /*
// if adaptation done, stop here
- if(_adaptDone)
+ // if(_adaptDone)
+ if(true)
return;
// start learning only if there is data played through speaker
@@ -383,7 +455,7 @@ void EchoCancel::updateEchoCancel(SFLDataFormat *micData, SFLDataFormat *spkrDat
if(_adaptCnt > _spkrAdaptSize) {
int k = _adaptCnt - _spkrAdaptSize;
_correlationArray[k] = performCorrelation(_spkrAdaptArray, _micAdaptArray+k, _correlationSize);
- _debug("EchoCancel: Correlation: %d", _correlationArray[k]);
+ // _debug("EchoCancel: Correlation: %d", _correlationArray[k]);
}
_adaptCnt++;
@@ -394,18 +466,16 @@ void EchoCancel::updateEchoCancel(SFLDataFormat *micData, SFLDataFormat *spkrDat
_adaptDone = true;
_amplDelayIndexOut = 0;// getMaximumIndex(_correlationArray, _correlationSize);
_debug("EchoCancel: Echo length %d", _amplDelayIndexOut);
- spkrLearningData->write((const char *)_spkrAdaptArray, _spkrAdaptSize*sizeof(int));
- micLearningData->write((const char *)_micAdaptArray, _micAdaptSize*sizeof(int));
}
-
+ */
}
-int EchoCancel::computeAmplitudeLevel(SFLDataFormat *data) {
+int EchoCancel::computeAmplitudeLevel(SFLDataFormat *data, int size) {
int level = 0;
- for(int i = 0; i < _smplPerSeg; i++) {
+ for(int i = 0; i < size; i++) {
if(data[i] >= 0.0)
level += (int)data[i];
else
@@ -418,11 +488,11 @@ int EchoCancel::computeAmplitudeLevel(SFLDataFormat *data) {
}
-int EchoCancel::getMaxAmplitude(int *data) {
+int EchoCancel::getMaxAmplitude(int *data, int size) {
SFLDataFormat level = 0.0;
- for(int i = 0; i < _historyLength; i++) {
+ for(int i = 0; i < size; i++) {
if(data[i] >= level)
level = data[i];
}
@@ -433,17 +503,19 @@ int EchoCancel::getMaxAmplitude(int *data) {
void EchoCancel::amplifySignal(SFLDataFormat *micData, SFLDataFormat *outputData, float amplify) {
+ for(int i = 0; i < _smplPerSeg; i++)
+ outputData[i] = micData[i];
// Use delayed amplification factor due to sound card latency
- for(int i = 0; i < _smplPerSeg; i++) {
- outputData[i] = (SFLDataFormat)(((float)micData[i])*_delayLineAmplify[_amplDelayIndexOut]);
- }
-
// do not increment amplitude array if adaptation is not done
- if (_adaptDone) {
+ // if (_adaptDone) {
+ /*
+ if (true) {
for(int i = 0; i < _smplPerSeg; i++) {
outputData[i] = (SFLDataFormat)(((float)micData[i])*_delayLineAmplify[_amplDelayIndexOut]);
}
+ _amplDelayIndexOut++;
+ _delayLineAmplify[_amplDelayIndexIn++] = amplify;
}
else {
for(int i = 0; i < _smplPerSeg; i++) {
@@ -452,15 +524,12 @@ void EchoCancel::amplifySignal(SFLDataFormat *micData, SFLDataFormat *outputData
return;
}
-
- _amplDelayIndexOut++;
- _delayLineAmplify[_amplDelayIndexIn++] = amplify;
-
if(_amplDelayIndexOut >= MAX_DELAY_LINE_AMPL)
_amplDelayIndexOut = 0;
if(_amplDelayIndexIn >= MAX_DELAY_LINE_AMPL)
_amplDelayIndexIn = 0;
+ */
}
diff --git a/sflphone-common/src/audio/echocancel.h b/sflphone-common/src/audio/echocancel.h
index e397f49e22c14cbc0dc19b565d88a9314b30ce12..abf5ba8d222cb5f3e0ba0a1845fe012c9862241a 100644
--- a/sflphone-common/src/audio/echocancel.h
+++ b/sflphone-common/src/audio/echocancel.h
@@ -31,9 +31,10 @@
#ifndef ECHOCANCEL_H
#define ECHOCANCEL_H
-#include "audioprocessing.h"
+#include <cc++/thread.h>
#include <speex/speex_preprocess.h>
+#include "audioprocessing.h"
#include "ringbuffer.h"
// Number of ms in sec
@@ -44,9 +45,11 @@
// Length of the echo tail in ms
#define ECHO_LENGTH 50
+#define SPKR_LENGTH 50
+#define MIC_LENGTH 50
// Voice level threashold
-#define MIN_SIG_LEVEL 1000
+#define MIN_SIG_LEVEL 75
// Delay between mic and speaker
#define DELAY_AMPLIFY 60
@@ -129,12 +132,12 @@ class EchoCancel : public Algorithm {
* Compute the average amplitude of the signal.
* \param data must be of SEGMENT_LENGTH long.
*/
- int computeAmplitudeLevel(SFLDataFormat *data);
+ int computeAmplitudeLevel(SFLDataFormat *data, int size);
/**
* Return the max amplitude provided any of _avgSpkrLevelHist or _avgMicLevelHist
*/
- int getMaxAmplitude(int *data);
+ int getMaxAmplitude(int *data, int size);
/**
* Apply gain factor on input buffer and copy result in output buffer.
@@ -211,7 +214,9 @@ class EchoCancel : public Algorithm {
* Number of segment considered in history
* Mainly used to compute signal level
*/
- int _historyLength;
+ int _micHistoryLength;
+
+ int _spkrHistoryLength;
/**
* Current playback level
@@ -294,20 +299,20 @@ class EchoCancel : public Algorithm {
int _processedByte;
- /*
ofstream *micFile;
ofstream *spkrFile;
ofstream *echoFile;
- */
- ofstream *micLearningData;
- ofstream *spkrLearningData;
+ ofstream *micLevelData;
+ ofstream *spkrLevelData;
/**
* Noise reduction processing state
*/
SpeexPreprocessState *_noiseState;
+ ost::Event _event;
+
};
#endif
diff --git a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
index 16cf22bb1ded8750ea134b198068b29a85aad69e..1c76ef3a40753b280af0c1bfac139ef0ed4ec5b1 100644
--- a/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
+++ b/sflphone-common/src/audio/pulseaudio/pulselayer.cpp
@@ -951,7 +951,6 @@ void PulseLayer::readFromMic (void)
// echo cancellation processing
int sampleready = _echoCanceller->processAudio(rsmpl_out, echoCancelledMic, nbSample*sizeof(SFLDataFormat));
- _debug("sampleready: %d", sampleready);
// getMainBuffer()->putData ( (void*) rsmpl_out, nbSample*sizeof (SFLDataFormat), 100);
if(sampleready)
getMainBuffer()->putData ( echoCancelledMic, sampleready*sizeof (SFLDataFormat), 100);