diff --git a/src/audio/audiolayer.cpp b/src/audio/audiolayer.cpp
index 7754a34c737b78f3a2c396197d839b1d13e53e51..45924b2c5c39212be8f48abfc66462e3ee729a4a 100644
--- a/src/audio/audiolayer.cpp
+++ b/src/audio/audiolayer.cpp
@@ -2,10 +2,11 @@
* Copyright (C) 2005 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author: Jerome Oufella <jerome.oufella@savoirfairelinux.com>
+ * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
+ * the Free Software Foundation; either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
@@ -108,13 +109,15 @@ AudioLayer::hasStream(void) {
void
-AudioLayer::openDevice (int indexIn, int indexOut, int sampleRate)
+AudioLayer::openDevice (int indexIn, int indexOut, int sampleRate, int frameSize)
{
closeStream();
_sampleRate = sampleRate;
-
+ _frameSize = frameSize;
int portaudioFramePerBuffer = FRAME_PER_BUFFER; //=FRAME_PER_BUFFER; //= paFramesPerBufferUnspecified;
+ //int portaudioFramePerBuffer = (int) (8000 * frameSize / 1000);
+//= paFramesPerBufferUnspecified;
int nbDevice = getDeviceCount();
if (nbDevice == 0) {
diff --git a/src/audio/audiolayer.h b/src/audio/audiolayer.h
index 3c45ed0bb8e71bd7ee082d87cc4b55c21d89e6ff..548ef197930a17bc7b7bccf6ea2744fe26eff3b4 100644
--- a/src/audio/audiolayer.h
+++ b/src/audio/audiolayer.h
@@ -2,10 +2,11 @@
* Copyright (C) 2004-2005 Savoir-Faire Linux inc.
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author: Jerome Oufella <jerome.oufella@savoirfairelinux.com>
+ * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
+ * the Free Software Foundation; either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
@@ -35,96 +36,103 @@ class RingBuffer;
class ManagerImpl;
class AudioLayer {
-public:
- AudioLayer(ManagerImpl* manager);
- ~AudioLayer(void);
-
- /*
- * @param indexIn
- * @param indexOut
- * @param sampleRate
- */
- void openDevice(int, int, int);
- void startStream(void);
- void stopStream(void);
- void sleep(int);
- bool hasStream(void);
- bool isStreamActive(void);
- bool isStreamStopped(void);
-
- void flushMain();
- int putMain(void* buffer, int toCopy);
- int putUrgent(void* buffer, int toCopy);
- int canGetMic();
- int getMic(void *, int);
- void flushMic();
-
- int audioCallback (const void *, void *, unsigned long,
- const PaStreamCallbackTimeInfo*, PaStreamCallbackFlags);
- int miniAudioCallback (const void *, void *, unsigned long,
- const PaStreamCallbackTimeInfo*, PaStreamCallbackFlags);
-
- void setErrorMessage(const std::string& error) { _errorMessage = error; }
- std::string getErrorMessage() { return _errorMessage; }
-
- /**
- * Get the sample rate of audiolayer
- * accessor only
- */
- unsigned int getSampleRate() { return _sampleRate; }
-
- int getDeviceCount();
- AudioDevice* getAudioDeviceInfo(int index, int ioDeviceMask);
-
- enum IODEVICE {InputDevice=0x01, OutputDevice=0x02 };
-
- /**
- * Toggle echo testing on/off
- */
- void toggleEchoTesting();
-
-private:
- void closeStream (void);
- RingBuffer _urgentRingBuffer;
- RingBuffer _mainSndRingBuffer;
- RingBuffer _micRingBuffer;
- ManagerImpl* _manager; // augment coupling, reduce indirect access
- // a audiolayer can't live without manager
-
- portaudio::MemFunCallbackStream<AudioLayer> *_stream;
-
- /**
- * Sample Rate of SFLphone : should be 8000 for 8khz
- * Added because we could change it in the futur
- */
- unsigned int _sampleRate;
-
- /**
- * Input channel (mic) should be 1 mono
- */
- unsigned int _inChannel; // mic
-
- /**
- * Output channel (stereo) should be 1 mono
- */
- unsigned int _outChannel; // speaker
-
- /**
- * Default volume for incoming RTP and Urgent sounds.
- */
- unsigned short _defaultVolume; // 100
-
- /**
- * Echo testing or not
- */
- bool _echoTesting;
-
- std::string _errorMessage;
- ost::Mutex _mutex;
-
- float *table_;
- int tableSize_;
- int leftPhase_;
+ public:
+ AudioLayer(ManagerImpl* manager);
+ ~AudioLayer(void);
+
+ /*
+ * @param indexIn
+ * @param indexOut
+ * @param sampleRate
+ * @param frameSize
+ */
+ void openDevice(int, int, int, int);
+ void startStream(void);
+ void stopStream(void);
+ void sleep(int);
+ bool hasStream(void);
+ bool isStreamActive(void);
+ bool isStreamStopped(void);
+
+ void flushMain();
+ int putMain(void* buffer, int toCopy);
+ int putUrgent(void* buffer, int toCopy);
+ int canGetMic();
+ int getMic(void *, int);
+ void flushMic();
+
+ int audioCallback (const void *, void *, unsigned long,
+ const PaStreamCallbackTimeInfo*, PaStreamCallbackFlags);
+ int miniAudioCallback (const void *, void *, unsigned long,
+ const PaStreamCallbackTimeInfo*, PaStreamCallbackFlags);
+
+ void setErrorMessage(const std::string& error) { _errorMessage = error; }
+ std::string getErrorMessage() { return _errorMessage; }
+
+ /**
+ * Get the sample rate of audiolayer
+ * accessor only
+ */
+ unsigned int getSampleRate() { return _sampleRate; }
+ unsigned int getFrameSize() { return _frameSize; }
+ int getDeviceCount();
+ AudioDevice* getAudioDeviceInfo(int index, int ioDeviceMask);
+
+ enum IODEVICE {InputDevice=0x01, OutputDevice=0x02 };
+
+ /**
+ * Toggle echo testing on/off
+ */
+ void toggleEchoTesting();
+
+ private:
+ void closeStream (void);
+ RingBuffer _urgentRingBuffer;
+ RingBuffer _mainSndRingBuffer;
+ RingBuffer _micRingBuffer;
+ ManagerImpl* _manager; // augment coupling, reduce indirect access
+ // a audiolayer can't live without manager
+
+ portaudio::MemFunCallbackStream<AudioLayer> *_stream;
+
+ /**
+ * Sample Rate SFLphone should send sound data to the sound card
+ * The value can be set in the user config file- now: 44100HZ
+ */
+ unsigned int _sampleRate;
+
+ /**
+ * Length of the sound frame we capture or read in ms
+ * The value can be set in the user config file - now: 20ms
+ */
+ unsigned int _frameSize;
+
+ /**
+ * Input channel (mic) should be 1 mono
+ */
+ unsigned int _inChannel; // mic
+
+ /**
+ * Output channel (stereo) should be 1 mono
+ */
+ unsigned int _outChannel; // speaker
+
+ /**
+ * Default volume for incoming RTP and Urgent sounds.
+ */
+ unsigned short _defaultVolume; // 100
+
+ /**
+ * Echo testing or not
+ */
+ bool _echoTesting;
+
+ std::string _errorMessage;
+ ost::Mutex _mutex;
+
+ float *table_;
+ int tableSize_;
+ int leftPhase_;
};
#endif // _AUDIO_LAYER_H_
diff --git a/src/audio/audiortp.cpp b/src/audio/audiortp.cpp
index fe6f0833a0d255cee8befe522f04861f1c0ca64b..53efa0962dd9fa11271441c6e17d01c71eb8f304 100644
--- a/src/audio/audiortp.cpp
+++ b/src/audio/audiortp.cpp
@@ -1,12 +1,13 @@
/*
* Copyright (C) 2004-2007 Savoir-Faire Linux inc.
+ * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
* Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
+ * the Free Software Foundation; either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
@@ -44,476 +45,478 @@
////////////////////////////////////////////////////////////////////////////////
AudioRtp::AudioRtp ()
{
- _RTXThread = 0;
+ _RTXThread = 0;
}
AudioRtp::~AudioRtp (void) {
- delete _RTXThread; _RTXThread = 0;
+ delete _RTXThread; _RTXThread = 0;
}
int
AudioRtp::createNewSession (SIPCall *ca) {
- ost::MutexLock m(_threadMutex);
-
- // something should stop the thread before...
- if ( _RTXThread != 0 ) {
- _debug("! ARTP Failure: Thread already exists..., stopping it\n");
- delete _RTXThread; _RTXThread = 0;
- //return -1;
- }
-
- // Start RTP Send/Receive threads
- _symmetric = Manager::instance().getConfigInt(SIGNALISATION,SYMMETRIC) ? true : false;
- _RTXThread = new AudioRtpRTX (ca, _symmetric);
-
- try {
- if (_RTXThread->start() != 0) {
- _debug("! ARTP Failure: unable to start RTX Thread\n");
- return -1;
- }
- } catch(...) {
- _debugException("! ARTP Failure: when trying to start a thread");
- throw;
- }
- return 0;
+ ost::MutexLock m(_threadMutex);
+
+ // something should stop the thread before...
+ if ( _RTXThread != 0 ) {
+ _debug("! ARTP Failure: Thread already exists..., stopping it\n");
+ delete _RTXThread; _RTXThread = 0;
+ //return -1;
+ }
+
+ // Start RTP Send/Receive threads
+ _symmetric = Manager::instance().getConfigInt(SIGNALISATION,SYMMETRIC) ? true : false;
+ _RTXThread = new AudioRtpRTX (ca, _symmetric);
+
+ try {
+ if (_RTXThread->start() != 0) {
+ _debug("! ARTP Failure: unable to start RTX Thread\n");
+ return -1;
+ }
+ } catch(...) {
+ _debugException("! ARTP Failure: when trying to start a thread");
+ throw;
+ }
+ return 0;
}
-
+
void
AudioRtp::closeRtpSession () {
- ost::MutexLock m(_threadMutex);
- // This will make RTP threads finish.
- // _debug("Stopping AudioRTP\n");
- try {
- delete _RTXThread; _RTXThread = 0;
- } catch(...) {
- _debugException("! ARTP Exception: when stopping audiortp\n");
- throw;
- }
+ ost::MutexLock m(_threadMutex);
+ // This will make RTP threads finish.
+ // _debug("Stopping AudioRTP\n");
+ try {
+ delete _RTXThread; _RTXThread = 0;
+ } catch(...) {
+ _debugException("! ARTP Exception: when stopping audiortp\n");
+ throw;
+ }
}
////////////////////////////////////////////////////////////////////////////////
// AudioRtpRTX Class //
////////////////////////////////////////////////////////////////////////////////
AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym)
- // : _fstream("/tmp/audio.dat", std::ofstream::binary)
- {
- setCancel(cancelDeferred);
- time = new ost::Time();
- _ca = sipcall;
- _sym = sym;
- // AudioRtpRTX should be close if we change sample rate
-
- _receiveDataDecoded = new int16[RTP_20S_48KHZ_MAX];
- _sendDataEncoded = new unsigned char[RTP_20S_8KHZ_MAX];
-
- // we estimate that the number of format after a conversion 8000->48000 is expanded to 6 times
- _dataAudioLayer = new SFLDataFormat[RTP_20S_48KHZ_MAX];
- _floatBuffer8000 = new float32[RTP_20S_8KHZ_MAX];
- _floatBuffer48000 = new float32[RTP_20S_48KHZ_MAX];
- _intBuffer8000 = new int16[RTP_20S_8KHZ_MAX];
-
- // TODO: Change bind address according to user settings.
- // TODO: this should be the local ip not the external (router) IP
- std::string localipConfig = _ca->getLocalIp(); // _ca->getLocalIp();
- ost::InetHostAddress local_ip(localipConfig.c_str());
-
- if (!_sym) {
- _sessionRecv = new ost::RTPSession(local_ip, _ca->getLocalAudioPort());
- _sessionSend = new ost::RTPSession(local_ip, _ca->getLocalAudioPort());
- _session = NULL;
- } else {
- _session = new ost::SymmetricRTPSession (local_ip, _ca->getLocalAudioPort());
- _sessionRecv = NULL;
- _sessionSend = NULL;
- }
-
- // libsamplerate-related
- _src_state_mic = src_new(SRC_SINC_BEST_QUALITY, 1, &_src_err);
- _src_state_spkr = src_new(SRC_SINC_BEST_QUALITY, 1, &_src_err);
-
+ // : _fstream("/tmp/audio.dat", std::ofstream::binary)
+{
+ setCancel(cancelDeferred);
+ time = new ost::Time();
+ _ca = sipcall;
+ _sym = sym;
+ // AudioRtpRTX should be close if we change sample rate
+
+ _codecSampleRate = _ca->getAudioCodec()->getClockRate();
+
+ // TODO: Change bind address according to user settings.
+ // TODO: this should be the local ip not the external (router) IP
+ std::string localipConfig = _ca->getLocalIp(); // _ca->getLocalIp();
+ ost::InetHostAddress local_ip(localipConfig.c_str());
+
+ if (!_sym) {
+ _sessionRecv = new ost::RTPSession(local_ip, _ca->getLocalAudioPort());
+ _sessionSend = new ost::RTPSession(local_ip, _ca->getLocalAudioPort());
+ _session = NULL;
+ } else {
+ _session = new ost::SymmetricRTPSession (local_ip, _ca->getLocalAudioPort());
+ _sessionRecv = NULL;
+ _sessionSend = NULL;
+ }
+
+ // libsamplerate-related
+ // Set the converter type for the upsampling and the downsampling
+ // interpolator SRC_SINC_BEST_QUALITY
+ _src_state_mic = src_new(SRC_SINC_BEST_QUALITY, 1, &_src_err);
+ _src_state_spkr = src_new(SRC_SINC_BEST_QUALITY, 1, &_src_err);
+
}
AudioRtpRTX::~AudioRtpRTX () {
- _start.wait();
-
- try {
- this->terminate();
- } catch(...) {
- _debugException("! ARTP: Thread destructor didn't terminate correctly");
- throw;
- }
- //_debug("terminate audiortprtx ended...\n");
- _ca = 0;
-
- if (!_sym) {
- delete _sessionRecv; _sessionRecv = NULL;
- delete _sessionSend; _sessionSend = NULL;
- } else {
- delete _session; _session = NULL;
- }
-
- delete [] _intBuffer8000; _intBuffer8000 = NULL;
- delete [] _floatBuffer48000; _floatBuffer48000 = NULL;
- delete [] _floatBuffer8000; _floatBuffer8000 = NULL;
- delete [] _dataAudioLayer; _dataAudioLayer = NULL;
-
- delete [] _sendDataEncoded; _sendDataEncoded = NULL;
- delete [] _receiveDataDecoded; _receiveDataDecoded = NULL;
-
- delete time; time = NULL;
-
- // libsamplerate-related
- _src_state_mic = src_delete(_src_state_mic);
- _src_state_spkr = src_delete(_src_state_spkr);
+ _start.wait();
+
+ try {
+ this->terminate();
+ } catch(...) {
+ _debugException("! ARTP: Thread destructor didn't terminate correctly");
+ throw;
+ }
+ //_debug("terminate audiortprtx ended...\n");
+ _ca = 0;
+
+ if (!_sym) {
+ delete _sessionRecv; _sessionRecv = NULL;
+ delete _sessionSend; _sessionSend = NULL;
+ } else {
+ delete _session; _session = NULL;
+ }
+
+ delete [] _intBufferDown; _intBufferDown = NULL;
+ delete [] _floatBufferUp; _floatBufferUp = NULL;
+ delete [] _floatBufferDown; _floatBufferDown = NULL;
+ delete [] _dataAudioLayer; _dataAudioLayer = NULL;
+
+ delete [] _sendDataEncoded; _sendDataEncoded = NULL;
+ delete [] _receiveDataDecoded; _receiveDataDecoded = NULL;
+
+ delete time; time = NULL;
+
+ // libsamplerate-related
+ _src_state_mic = src_delete(_src_state_mic);
+ _src_state_spkr = src_delete(_src_state_spkr);
}
-void
-AudioRtpRTX::initAudioRtpSession (void)
+ void
+AudioRtpRTX::initBuffers()
{
- try {
- if (_ca == 0) { return; }
-
- //_debug("Init audio RTP session\n");
- ost::InetHostAddress remote_ip(_ca->getRemoteIp().c_str());
- if (!remote_ip) {
- _debug("! ARTP Thread Error: Target IP address [%s] is not correct!\n", _ca->getRemoteIp().data());
- return;
- }
-
- // Initialization
- if (!_sym) {
- _sessionRecv->setSchedulingTimeout (10000);
- _sessionRecv->setExpireTimeout(1000000);
-
- _sessionSend->setSchedulingTimeout(10000);
- _sessionSend->setExpireTimeout(1000000);
- } else {
- _session->setSchedulingTimeout(10000);
- _session->setExpireTimeout(1000000);
- }
-
- if (!_sym) {
- if ( !_sessionRecv->addDestination(remote_ip, (unsigned short) _ca->getRemoteAudioPort()) ) {
- _debug("AudioRTP Thread Error: could not connect to port %d\n", _ca->getRemoteAudioPort());
- return;
- }
- if (!_sessionSend->addDestination (remote_ip, (unsigned short) _ca->getRemoteAudioPort())) {
- _debug("! ARTP Thread Error: could not connect to port %d\n", _ca->getRemoteAudioPort());
- return;
- }
-
- AudioCodec* audiocodec = _ca->getAudioCodec();
- bool payloadIsSet = false;
- if (audiocodec) {
- if (audiocodec->hasDynamicPayload()) {
- payloadIsSet = _sessionRecv->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) audiocodec->getPayload(), audiocodec->getClockRate()));
- } else {
- payloadIsSet= _sessionRecv->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) audiocodec->getPayload()));
- payloadIsSet = _sessionSend->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) audiocodec->getPayload()));
- }
- }
- _sessionSend->setMark(true);
- } else {
-
- //_debug("AudioRTP Thread: Added session destination %s:%d\n", remote_ip.getHostname(), (unsigned short) _ca->getRemoteSdpAudioPort());
-
- if (!_session->addDestination (remote_ip, (unsigned short) _ca->getRemoteAudioPort())) {
- return;
- }
-
- AudioCodec* audiocodec = _ca->getAudioCodec();
- bool payloadIsSet = false;
- if (audiocodec) {
- if (audiocodec->hasDynamicPayload()) {
- payloadIsSet = _session->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) audiocodec->getPayload(), audiocodec->getClockRate()));
- } else {
- payloadIsSet = _session->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) audiocodec->getPayload()));
- }
- }
- }
- } catch(...) {
- _debugException("! ARTP Failure: initialisation failed");
- throw;
- }
+ int nbSamplesMax = (int) (_layerSampleRate * _layerFrameSize /1000);
+ _dataAudioLayer = new SFLDataFormat[nbSamplesMax];
+ _receiveDataDecoded = new int16[nbSamplesMax];
+ _floatBufferDown = new float32[nbSamplesMax];
+ _floatBufferUp = new float32[nbSamplesMax];
+ _sendDataEncoded = new unsigned char[nbSamplesMax];
+ _intBufferDown = new int16[nbSamplesMax];
}
+ void
+AudioRtpRTX::initAudioRtpSession (void)
+{
+ try {
+ if (_ca == 0) { return; }
+
+ //_debug("Init audio RTP session\n");
+ ost::InetHostAddress remote_ip(_ca->getRemoteIp().c_str());
+ if (!remote_ip) {
+ _debug("! ARTP Thread Error: Target IP address [%s] is not correct!\n", _ca->getRemoteIp().data());
+ return;
+ }
+
+ // Initialization
+ if (!_sym) {
+ _sessionRecv->setSchedulingTimeout (10000);
+ _sessionRecv->setExpireTimeout(1000000);
+
+ _sessionSend->setSchedulingTimeout(10000);
+ _sessionSend->setExpireTimeout(1000000);
+ } else {
+ _session->setSchedulingTimeout(10000);
+ _session->setExpireTimeout(1000000);
+ }
+
+ if (!_sym) {
+ if ( !_sessionRecv->addDestination(remote_ip, (unsigned short) _ca->getRemoteAudioPort()) ) {
+ _debug("AudioRTP Thread Error: could not connect to port %d\n", _ca->getRemoteAudioPort());
+ return;
+ }
+ if (!_sessionSend->addDestination (remote_ip, (unsigned short) _ca->getRemoteAudioPort())) {
+ _debug("! ARTP Thread Error: could not connect to port %d\n", _ca->getRemoteAudioPort());
+ return;
+ }
+
+ AudioCodec* audiocodec = _ca->getAudioCodec();
+ bool payloadIsSet = false;
+ if (audiocodec) {
+ if (audiocodec->hasDynamicPayload()) {
+ payloadIsSet = _sessionRecv->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) audiocodec->getPayload(), audiocodec->getClockRate()));
+ } else {
+ payloadIsSet= _sessionRecv->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) audiocodec->getPayload()));
+ payloadIsSet = _sessionSend->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) audiocodec->getPayload()));
+ }
+ }
+ _sessionSend->setMark(true);
+ } else {
+
+ //_debug("AudioRTP Thread: Added session destination %s:%d\n", remote_ip.getHostname(), (unsigned short) _ca->getRemoteSdpAudioPort());
+
+ if (!_session->addDestination (remote_ip, (unsigned short) _ca->getRemoteAudioPort())) {
+ return;
+ }
+
+ AudioCodec* audiocodec = _ca->getAudioCodec();
+ bool payloadIsSet = false;
+ if (audiocodec) {
+ if (audiocodec->hasDynamicPayload()) {
+ payloadIsSet = _session->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) audiocodec->getPayload(), audiocodec->getClockRate()));
+ } else {
+ payloadIsSet = _session->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) audiocodec->getPayload()));
+ }
+ }
+ }
+ } catch(...) {
+ _debugException("! ARTP Failure: initialisation failed");
+ throw;
+ }
+}
+
void
AudioRtpRTX::sendSessionFromMic(int timestamp)
{
- // STEP:
- // 1. get data from mic
- // 2. convert it to int16 - good sample, good rate
- // 3. encode it
- // 4. send it
- try {
- timestamp += time->getSecond();
- if (_ca==0) { _debug(" !ARTP: No call associated (mic)\n"); return; } // no call, so we do nothing
- AudioLayer* audiolayer = Manager::instance().getAudioDriver();
- if (!audiolayer) { _debug(" !ARTP: No audiolayer available for mic\n"); return; }
-
- AudioCodec* audiocodec = _ca->getAudioCodec();
- if (!audiocodec) { _debug(" !ARTP: No audiocodec available for mic\n"); return; }
-
- // we have to get 20ms of data from the mic *20/1000 = /50
- // rate/50 shall be lower than RTP_20S_48KHZ_MAX
- int maxBytesToGet = audiolayer->getSampleRate()/50*sizeof(SFLDataFormat);
-
- // available bytes inside ringbuffer
- int availBytesFromMic = audiolayer->canGetMic();
-
- // take the lowest
- int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
- //_debug("available = %d, maxBytesToGet = %d\n", availBytesFromMic, maxBytesToGet);
-
- // Get bytes from micRingBuffer to data_from_mic
- int nbSample = audiolayer->getMic(_dataAudioLayer, bytesAvail) / sizeof(SFLDataFormat);
-
- int16* toSIP = NULL;
- if (audiolayer->getSampleRate() != audiocodec->getClockRate() && nbSample) {
- SRC_DATA src_data;
- #ifdef DATAFORMAT_IS_FLOAT
- src_data.data_in = _dataAudioLayer;
- #else
- src_short_to_float_array(_dataAudioLayer, _floatBuffer48000, nbSample);
- src_data.data_in = _floatBuffer48000;
- #endif
-
- double factord = (double) audiocodec->getClockRate() / audiolayer->getSampleRate();
-
- src_data.src_ratio = factord;
- src_data.input_frames = nbSample;
- src_data.output_frames = (int) floor(factord * nbSample);
- src_data.data_out = _floatBuffer8000;
- src_data.end_of_input = 0; /* More data to come */
-
- src_process(_src_state_mic, &src_data);
-
- nbSample = src_data.output_frames_gen;
-
- /* WRITE IN A FILE FOR DEBUG */
- //_fstream.write((char *) _floatBuffer48000, src_data.output_frames_gen * sizeof(float));
- //_fstream.flush();
-
-
- //if (nbSample > RTP_20S_8KHZ_MAX) { _debug("Alert from mic, nbSample %d is bigger than expected %d\n", nbSample, RTP_20S_8KHZ_MAX); }
-
- src_float_to_short_array (_floatBuffer8000, _intBuffer8000, nbSample);
- toSIP = _intBuffer8000;
- } else {
- #ifdef DATAFORMAT_IS_FLOAT
- // convert _receiveDataDecoded to float inside _receiveData
- src_float_to_short_array(_dataAudioLayer, _intBuffer8000, nbSample);
- toSIP = _intBuffer8000;
- //if (nbSample > RTP_20S_8KHZ_MAX) { _debug("Alert from mic, nbSample %d is bigger than expected %d\n", nbSample, RTP_20S_8KHZ_MAX); }
- #else
- toSIP = _dataAudioLayer; // int to int
- #endif
- }
-
- if ( nbSample < (RTP_20S_8KHZ_MAX - 10) ) { // if only 10 is missing, it's ok
- // fill end with 0...
- //_debug("begin: %p, nbSample: %d\n", toSIP, nbSample);
- //_debug("has to fill: %d chars at %p\n", (RTP_20S_8KHZ_MAX-nbSample)*sizeof(int16), toSIP + nbSample);
- memset(toSIP + nbSample, 0, (RTP_20S_8KHZ_MAX-nbSample)*sizeof(int16));
- nbSample = RTP_20S_8KHZ_MAX;
- }
- //_debug("AR: Nb sample: %d int, [0]=%d [1]=%d [2]=%d\n", nbSample, toSIP[0], toSIP[1], toSIP[2]);
-
- // for the mono: range = 0 to RTP_FRAME2SEND * sizeof(int16)
- // codecEncode(char *dest, int16* src, size in bytes of the src)
- int compSize = audiocodec->codecEncode(_sendDataEncoded, toSIP, nbSample*sizeof(int16));
-
- // encode divise by two
- // Send encoded audio sample over the network
- if (compSize > RTP_20S_8KHZ_MAX) { _debug("! ARTP: %d should be %d\n", compSize, RTP_20S_8KHZ_MAX);}
- if (!_sym) {
- _sessionSend->putData(timestamp, _sendDataEncoded, compSize);
- } else {
- _session->putData(timestamp, _sendDataEncoded, compSize);
- }
- toSIP = NULL;
- } catch(...) {
- _debugException("! ARTP: sending failed");
- throw;
- }
+ // STEP:
+ // 1. get data from mic
+ // 2. convert it to int16 - good sample, good rate
+ // 3. encode it
+ // 4. send it
+ try {
+ int16* toSIP = NULL;
+
+ timestamp += time->getSecond();
+ if (_ca==0) { _debug(" !ARTP: No call associated (mic)\n"); return; } // no call, so we do nothing
+ AudioLayer* audiolayer = Manager::instance().getAudioDriver();
+ if (!audiolayer) { _debug(" !ARTP: No audiolayer available for mic\n"); return; }
+
+ AudioCodec* audiocodec = _ca->getAudioCodec();
+ if (!audiocodec) { _debug(" !ARTP: No audiocodec available for mic\n"); return; }
+
+ // we have to get 20ms of data from the mic *20/1000 = /50
+ int maxBytesToGet = _layerSampleRate * _layerFrameSize * sizeof(SFLDataFormat) / 1000;
+
+ // available bytes inside ringbuffer
+ int availBytesFromMic = audiolayer->canGetMic();
+
+ // take the lowest
+ int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;
+
+ // Get bytes from micRingBuffer to data_from_mic
+ int nbSample = audiolayer->getMic(_dataAudioLayer, bytesAvail) / sizeof(SFLDataFormat);
+ int nb_sample_up = nbSample;
+ int nbSamplesMax = _layerFrameSize * audiocodec->getClockRate() / 1000;
+
+ nbSample = reSampleData(audiocodec->getClockRate(), nb_sample_up, DOWN_SAMPLING);
+
+ toSIP = _intBufferDown;
+
+ if ( nbSample < nbSamplesMax - 10 ) { // if only 10 is missing, it's ok
+ // fill end with 0...
+ //_debug("begin: %p, nbSample: %d\n", toSIP, nbSample);
+ memset(toSIP + nbSample, 0, (nbSamplesMax-nbSample)*sizeof(int16));
+ nbSample = nbSamplesMax;
+ }
+ //_debug("AR: Nb sample: %d int, [0]=%d [1]=%d [2]=%d\n", nbSample, toSIP[0], toSIP[1], toSIP[2]);
+
+ // for the mono: range = 0 to RTP_FRAME2SEND * sizeof(int16)
+ // codecEncode(char *dest, int16* src, size in bytes of the src)
+ int compSize = audiocodec->codecEncode(_sendDataEncoded, toSIP, nbSample*sizeof(int16));
+
+ // encode divise by two
+ // Send encoded audio sample over the network
+ if (compSize > nbSamplesMax) { _debug("! ARTP: %d should be %d\n", compSize, nbSamplesMax);}
+ if (!_sym) {
+ _sessionSend->putData(timestamp, _sendDataEncoded, compSize);
+ } else {
+ _session->putData(timestamp, _sendDataEncoded, compSize);
+ }
+ toSIP = NULL;
+ } catch(...) {
+ _debugException("! ARTP: sending failed");
+ throw;
+ }
}
-void
+
+
+ void
AudioRtpRTX::receiveSessionForSpkr (int& countTime)
{
- if (_ca == 0) { return; }
- try {
- AudioLayer* audiolayer = Manager::instance().getAudioDriver();
- if (!audiolayer) { return; }
-
- const ost::AppDataUnit* adu = NULL;
- // Get audio data stream
-
- if (!_sym) {
- adu = _sessionRecv->getData(_sessionRecv->getFirstTimestamp());
- } else {
- adu = _session->getData(_session->getFirstTimestamp());
- }
- if (adu == NULL) {
- return;
- }
-
- int payload = adu->getType(); // codec type
- unsigned char* data = (unsigned char*)adu->getData(); // data in char
- unsigned int size = adu->getSize(); // size in char
-
- if ( size > RTP_20S_8KHZ_MAX ) {
- _debug("We have received from RTP a packet larger than expected: %s VS %s\n", size, RTP_20S_8KHZ_MAX);
- _debug("The packet size has been cropped\n");
- size=RTP_20S_8KHZ_MAX;
- }
-
- // NOTE: L'audio rendu ici (dans data/size) est parfait.
-
- // Decode data with relevant codec
- AudioCodec* audiocodec = _ca->getCodecMap().getCodec((CodecType)payload);
- if (audiocodec != NULL) {
- // codecDecode(int16 *dest, char* src, size in bytes of the src)
- // decode multiply by two, so the number of byte should be double
- // size shall be RTP_FRAME2SEND or lower
- int expandedSize = audiocodec->codecDecode(_receiveDataDecoded, data, size);
- int nbInt16 = expandedSize / sizeof(int16);
- if (nbInt16 > RTP_20S_8KHZ_MAX) {
- _debug("We have decoded an RTP packet larger than expected: %s VS %s. Cropping.\n", nbInt16, RTP_20S_8KHZ_MAX);
- nbInt16=RTP_20S_8KHZ_MAX;
- }
-
- // NOTE: l'audio arrivé ici (dans _receiveDataDecoded/expandedSize) est parfait.
-
- SFLDataFormat* toAudioLayer;
- int nbSample = nbInt16;
- // 48000 / 8000 = 6, the number of samples for the maximum rate conversion.
- int nbSampleMaxRate = nbInt16 * 6; // TODO: change it
-
- // We assume over here that we pass from a lower rate to a higher one. Bad bad.
- if ( audiolayer->getSampleRate() != audiocodec->getClockRate() && nbSample) {
- // Do sample rate conversion
-
- double factord = (double) audiolayer->getSampleRate() / audiocodec->getClockRate();
-
- // SRC_DATA from samplerate.h
- SRC_DATA src_data;
- src_data.data_in = _floatBuffer8000;
- src_data.data_out = _floatBuffer48000;
- src_data.input_frames = nbSample;
- src_data.output_frames = (int) floor(factord * nbSample);
- src_data.src_ratio = factord;
- src_data.end_of_input = 0; /* More data will come */
- src_short_to_float_array(_receiveDataDecoded, _floatBuffer8000, nbSample);
-
- // NOTE: L'audio arrivé ici (dans _floatBuffer8000/nbSample*sizeof(float) est parfait.
-
- src_process(_src_state_spkr, &src_data);
-
- // Truncate number of samples if too high (ouch!)
- nbSample = ( src_data.output_frames_gen > RTP_20S_48KHZ_MAX) ? RTP_20S_48KHZ_MAX : src_data.output_frames_gen;
-
- #ifdef DATAFORMAT_IS_FLOAT
- toAudioLayer = _floatBuffer48000;
- #else
- src_float_to_short_array(_floatBuffer48000, _dataAudioLayer, nbSample);
- toAudioLayer = _dataAudioLayer;
- #endif
+ if (_ca == 0) { return; }
+ try {
+ AudioLayer* audiolayer = Manager::instance().getAudioDriver();
+ if (!audiolayer) { return; }
+
+ const ost::AppDataUnit* adu = NULL;
+ // Get audio data stream
+
+ if (!_sym) {
+ adu = _sessionRecv->getData(_sessionRecv->getFirstTimestamp());
+ } else {
+ adu = _session->getData(_session->getFirstTimestamp());
+ }
+ if (adu == NULL) {
+ return;
+ }
+
+ int payload = adu->getType(); // codec type
+ unsigned char* data = (unsigned char*)adu->getData(); // data in char
+ unsigned int size = adu->getSize(); // size in char
+
+
+ // Decode data with relevant codec
+ AudioCodec* audiocodec = _ca->getCodecMap().getCodec((CodecType)payload);
+ _codecSampleRate = audiocodec->getClockRate();
+ int max = (int)(_codecSampleRate * _layerFrameSize);
+
+ if ( size > max ) {
+ _debug("We have received from RTP a packet larger than expected: %s VS %s\n", size, max);
+ _debug("The packet size has been cropped\n");
+ size=max;
+ }
+
+
+ if (audiocodec != NULL) {
+ int expandedSize = audiocodec->codecDecode(_receiveDataDecoded, data, size);
+ //buffer _receiveDataDecoded ----> short int or int16, coded on 2 bytes
+ int nbInt16 = expandedSize / sizeof(int16);
+ //nbInt16 represents the number of samples we just decoded
+ if (nbInt16 > max) {
+ _debug("We have decoded an RTP packet larger than expected: %s VS %s. Cropping.\n", nbInt16, max);
+ nbInt16=max;
+ }
+
+ SFLDataFormat* toAudioLayer;
+ int nbSample = nbInt16;
+
+ // Do sample rate conversion
+ int nb_sample_down = nbSample;
+ nbSample = reSampleData(_codecSampleRate , nb_sample_down, UP_SAMPLING);
+#ifdef DATAFORMAT_IS_FLOAT
+ toAudioLayer = _floatBufferUp;
+#else
+ toAudioLayer = _dataAudioLayer;
+#endif
+
+
+ audiolayer->putMain(toAudioLayer, nbSample * sizeof(SFLDataFormat));
+
+ // Notify (with a beep) an incoming call when there is already a call
+ countTime += time->getSecond();
+ if (Manager::instance().incomingCallWaiting() > 0) {
+ countTime = countTime % 500; // more often...
+ if (countTime == 0) {
+ Manager::instance().notificationIncomingCall();
+ }
+ }
+
+ } else {
+ countTime += time->getSecond();
+ }
+
+ delete adu; adu = NULL;
+ } catch(...) {
+ _debugException("! ARTP: receiving failed");
+ throw;
+ }
+}
-
- } else {
- nbSample = nbInt16;
- #ifdef DATAFORMAT_IS_FLOAT
- // convert _receiveDataDecoded to float inside _receiveData
- src_short_to_float_array(_receiveDataDecoded, _floatBuffer8000, nbSample);
- toAudioLayer = _floatBuffer8000;
- #else
- toAudioLayer = _receiveDataDecoded; // int to int
- #endif
- }
-
- audiolayer->putMain(toAudioLayer, nbSample * sizeof(SFLDataFormat));
- //_debug("ARTP: %d\n", nbSample * sizeof(SFLDataFormat));
-
- // Notify (with a beep) an incoming call when there is already a call
- countTime += time->getSecond();
- if (Manager::instance().incomingCallWaiting() > 0) {
- countTime = countTime % 500; // more often...
- if (countTime == 0) {
- Manager::instance().notificationIncomingCall();
- }
- }
-
- } else {
- countTime += time->getSecond();
- }
-
- delete adu; adu = NULL;
- } catch(...) {
- _debugException("! ARTP: receiving failed");
- throw;
- }
+int
+AudioRtpRTX::reSampleData(int sampleRate_codec, int nbSamples, int status)
+{
+ if(status==UP_SAMPLING)
+ return upSampleData(sampleRate_codec, nbSamples);
+ else if(status==DOWN_SAMPLING)
+ return downSampleData(sampleRate_codec, nbSamples);
+ else
+ return 0;
}
+////////////////////////////////////////////////////////////////////
+//////////// RESAMPLING FUNCTIONS /////////////////////////////////
+//////////////////////////////////////////////////////////////////
+
+ int
+AudioRtpRTX::upSampleData(int sampleRate_codec, int nbSamples)
+{
+ double upsampleFactor = (double) _layerSampleRate / sampleRate_codec;
+ int nbSamplesMax = (int) (_layerSampleRate * _layerFrameSize /1000);
+ if( upsampleFactor != 1 )
+ {
+ SRC_DATA src_data;
+ src_data.data_in = _floatBufferDown;
+ src_data.data_out = _floatBufferUp;
+ 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
+ src_short_to_float_array(_receiveDataDecoded, _floatBufferDown, nbSamples);
+ src_process(_src_state_spkr, &src_data);
+ nbSamples = ( src_data.output_frames_gen > nbSamplesMax) ? nbSamplesMax : src_data.output_frames_gen;
+ src_float_to_short_array(_floatBufferUp, _dataAudioLayer, nbSamples);
+ }
+
+ return nbSamples;
+}
+
+ int
+AudioRtpRTX::downSampleData(int sampleRate_codec, int nbSamples)
+{
+ double downsampleFactor = (double) sampleRate_codec / _layerSampleRate;
+ int nbSamplesMax = (int) (sampleRate_codec * _layerFrameSize / 1000);
+ if ( downsampleFactor != 1)
+ {
+ SRC_DATA src_data;
+ src_data.data_in = _floatBufferUp;
+ src_data.data_out = _floatBufferDown;
+ 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
+ src_short_to_float_array(_dataAudioLayer, _floatBufferUp, nbSamples);
+ src_process(_src_state_mic, &src_data);
+ nbSamples = ( src_data.output_frames_gen > nbSamplesMax) ? nbSamplesMax : src_data.output_frames_gen;
+ src_float_to_short_array(_floatBufferDown, _intBufferDown, nbSamples);
+ }
+ return nbSamples;
+
+}
+
+//////////////////////// END RESAMPLING //////////////////////////////////////////////////////
+
void
AudioRtpRTX::run () {
- //mic, we receive from soundcard in stereo, and we send encoded
- //encoding before sending
- AudioLayer *audiolayer = Manager::instance().getAudioDriver();
-
- try {
- // Init the session
- initAudioRtpSession();
-
- // start running the packet queue scheduler.
- //_debug("AudioRTP Thread started\n");
- if (!_sym) {
- _sessionRecv->startRunning();
- _sessionSend->startRunning();
- } else {
- _session->startRunning();
- //_debug("Session is now: %d active\n", _session->isActive());
- }
-
- int timestamp = 0; // for mic
- int countTime = 0; // for receive
- // TODO: get frameSize from user config
- int frameSize = 20; // 20ms frames
- TimerPort::setTimer(frameSize);
-
- audiolayer->flushMic();
- audiolayer->startStream();
- _start.post();
- _debug("- ARTP Action: Start\n");
- while (!testCancel()) {
- ////////////////////////////
- // Send session
- ////////////////////////////
- sendSessionFromMic(timestamp);
- timestamp += RTP_20S_8KHZ_MAX;
-
- ////////////////////////////
- // Recv session
- ////////////////////////////
- receiveSessionForSpkr(countTime);
-
- // Let's wait for the next transmit cycle
- Thread::sleep(TimerPort::getTimer());
- TimerPort::incTimer(frameSize); // 'frameSize' ms
- }
- //_debug("stop stream for audiortp loop\n");
- audiolayer->stopStream();
- } catch(std::exception &e) {
- _start.post();
- _debug("! ARTP: Stop %s\n", e.what());
- throw;
- } catch(...) {
- _start.post();
- _debugException("* ARTP Action: Stop");
- throw;
- }
+ //mic, we receive from soundcard in stereo, and we send encoded
+ //encoding before sending
+ AudioLayer *audiolayer = Manager::instance().getAudioDriver();
+
+ _layerFrameSize = audiolayer->getFrameSize(); // en ms
+ _layerSampleRate = audiolayer->getSampleRate();
+ initBuffers();
+ int step = (int)(_layerFrameSize * _codecSampleRate / 1000);
+
+ try {
+ // Init the session
+ initAudioRtpSession();
+
+ // start running the packet queue scheduler.
+ //_debug("AudioRTP Thread started\n");
+ if (!_sym) {
+ _sessionRecv->startRunning();
+ _sessionSend->startRunning();
+ } else {
+ _session->startRunning();
+ //_debug("Session is now: %d active\n", _session->isActive());
+ }
+
+ int timestamp = 0; // for mic
+ int countTime = 0; // for receive
+ TimerPort::setTimer(_layerFrameSize);
+
+ audiolayer->flushMic();
+ audiolayer->startStream();
+ _start.post();
+ _debug("- ARTP Action: Start\n");
+ while (!testCancel()) {
+ ////////////////////////////
+ // Send session
+ ////////////////////////////
+ sendSessionFromMic(timestamp);
+ timestamp += step;
+ ////////////////////////////
+ // Recv session
+ ////////////////////////////
+ receiveSessionForSpkr(countTime);
+
+ // Let's wait for the next transmit cycle
+ Thread::sleep(TimerPort::getTimer());
+ TimerPort::incTimer(_layerFrameSize); // 'frameSize' ms
+ }
+ //_debug("stop stream for audiortp loop\n");
+ audiolayer->stopStream();
+ } catch(std::exception &e) {
+ _start.post();
+ _debug("! ARTP: Stop %s\n", e.what());
+ throw;
+ } catch(...) {
+ _start.post();
+ _debugException("* ARTP Action: Stop");
+ throw;
+ }
}
diff --git a/src/audio/audiortp.h b/src/audio/audiortp.h
index 4b018276929b5a2c63d95bcb79027a1ffc3c8580..2444be567611e17b4858ef8321baed96a7c91500 100644
--- a/src/audio/audiortp.h
+++ b/src/audio/audiortp.h
@@ -1,12 +1,13 @@
/*
* Copyright (C) 2004-2007 Savoir-Faire Linux inc.
+ * Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
* Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
* Author: Yan Morin <yan.morin@savoirfairelinux.com>
* Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
+ * the Free Software Foundation; either version 3 of the License, or
* (at your option) any later version.
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
@@ -33,81 +34,108 @@
#include "../global.h"
-/** maximum bytes inside an incoming packet
- * 8000 sampling/s * 20s/1000 = 160
- */
-#define RTP_20S_8KHZ_MAX 160
-#define RTP_20S_48KHZ_MAX 960
+#define UP_SAMPLING 0
+#define DOWN_SAMPLING 1
+
+
class SIPCall;
///////////////////////////////////////////////////////////////////////////////
// Two pair of sockets
///////////////////////////////////////////////////////////////////////////////
class AudioRtpRTX : public ost::Thread, public ost::TimerPort {
-public:
- AudioRtpRTX (SIPCall *, bool);
- ~AudioRtpRTX();
-
- ost::Time *time; // For incoming call notification
- virtual void run ();
-
-private:
- SIPCall* _ca;
- ost::RTPSession *_sessionSend;
- ost::RTPSession *_sessionRecv;
- ost::SymmetricRTPSession *_session;
- ost::Semaphore _start;
- bool _sym;
-
- /** When we receive data, we decode it inside this buffer */
- int16* _receiveDataDecoded;
- /** When we send data, we encode it inside this buffer*/
- unsigned char* _sendDataEncoded;
-
- /** After that we send the data inside this buffer if there is a format conversion or rate conversion */
- /** Also use for getting mic-ringbuffer data */
- SFLDataFormat* _dataAudioLayer;
-
- /** Buffer for 8000hz samples in conversion */
- float32* _floatBuffer8000;
- /** Buffer for 48000hz samples in conversion */
- float32* _floatBuffer48000;
-
- /** Buffer for 8000Hz samples for mic conversion */
- int16* _intBuffer8000;
-
- /** Debugging output file */
- //std::ofstream _fstream;
-
- /** libsamplerate converter for incoming voice */
- SRC_STATE* _src_state_spkr;
-
- /** libsamplerate converter for outgoing voice */
- SRC_STATE* _src_state_mic;
-
- /** libsamplerate error */
- int _src_err;
-
- void initAudioRtpSession(void);
- void sendSessionFromMic(int);
- void receiveSessionForSpkr(int&);
+ public:
+ AudioRtpRTX (SIPCall *, bool);
+ ~AudioRtpRTX();
+
+ ost::Time *time; // For incoming call notification
+ virtual void run ();
+
+ private:
+ SIPCall* _ca;
+ ost::RTPSession *_sessionSend;
+ ost::RTPSession *_sessionRecv;
+ ost::SymmetricRTPSession *_session;
+ ost::Semaphore _start;
+ bool _sym;
+
+ /** When we receive data, we decode it inside this buffer */
+ int16* _receiveDataDecoded;
+ /** Buffers used for send data from the mic */
+ unsigned char* _sendDataEncoded;
+ int16* _intBufferDown;
+
+ /** After that we send the data inside this buffer if there is a format conversion or rate conversion */
+ /** Also use for getting mic-ringbuffer data */
+ SFLDataFormat* _dataAudioLayer;
+
+ /** Buffers used for sample rate conversion */
+ float32* _floatBufferDown;
+ float32* _floatBufferUp;
+
+ /** Debugging output file */
+ //std::ofstream _fstream;
+
+ /** libsamplerate converter for incoming voice */
+ SRC_STATE* _src_state_spkr;
+ /** libsamplerate converter for outgoing voice */
+ SRC_STATE* _src_state_mic;
+ /** libsamplerate error */
+ int _src_err;
+
+ // Variables to process audio stream
+ int _layerSampleRate; // sample rate for playing sound (typically 44100HZ)
+ int _codecSampleRate; // sample rate of the codec we use to encode and decode (most of time 8000HZ)
+ int _layerFrameSize; // length of the sound frame we capture in ms(typically 20ms)
+
+ void initAudioRtpSession(void);
+ /**
+ * Get the data from the mic, encode it and send it through the RTP session
+ */
+ void sendSessionFromMic(int);
+ /**
+ * Get the data from the RTP packets, decode it and send it to the sound card
+ */
+ void receiveSessionForSpkr(int&);
+ /**
+ * Init the buffers used for processing sound data
+ */
+ void initBuffers(void);
+ /**
+ * Call the appropriate function, up or downsampling
+ */
+ int reSampleData(int, int ,int);
+ /**
+ * Upsample the data from the clock rate of the codec to the sample rate of the layer
+ * @param int The clock rate of the codec
+ * @param int The number of samples we have in the buffer
+ * @return int The number of samples after the operation
+ */
+ int upSampleData(int, int);
+ /**
+ * Downsample the data from the sample rate of the layer to the clock rate of the codec
+ * @param int The clock rate of the codec
+ * @param int The number of samples we have in the buffer
+ * @return int The number of samples after the operation
+ */
+ int downSampleData(int, int);
};
///////////////////////////////////////////////////////////////////////////////
// Main class rtp
///////////////////////////////////////////////////////////////////////////////
class AudioRtp {
-public:
- AudioRtp();
- ~AudioRtp();
+ public:
+ AudioRtp();
+ ~AudioRtp();
- int createNewSession (SIPCall *);
- void closeRtpSession ();
+ int createNewSession (SIPCall *);
+ void closeRtpSession ();
-private:
- AudioRtpRTX* _RTXThread;
- bool _symmetric;
- ost::Mutex _threadMutex;
+ private:
+ AudioRtpRTX* _RTXThread;
+ bool _symmetric;
+ ost::Mutex _threadMutex;
};
#endif // __AUDIO_RTP_H__
diff --git a/src/managerimpl.cpp b/src/managerimpl.cpp
index 0e92c0c9711f5953afcf2e193ad1e8def3230385..de29ed38b9968313c6eb80652b8edb6338132156 100644
--- a/src/managerimpl.cpp
+++ b/src/managerimpl.cpp
@@ -1050,7 +1050,8 @@ ManagerImpl::initConfigFile (void)
//fill_config_int(DRIVER_NAME, DFT_DRIVER_STR);
fill_config_int(DRIVER_NAME_IN, DFT_DRIVER_STR);
fill_config_int(DRIVER_NAME_OUT, DFT_DRIVER_STR);
- fill_config_int(DRIVER_SAMPLE_RATE, DRIVER_SAMPLE_RATE_DEFAULT);
+ fill_config_int(DRIVER_SAMPLE_RATE, DFT_SAMPLE_RATE);
+ fill_config_int(DRIVER_FRAME_SIZE, DFT_FRAME_SIZE);
fill_config_str(CODEC1, DFT_CODEC);
fill_config_str(CODEC2, DFT_CODEC);
fill_config_str(CODEC3, DFT_CODEC);
@@ -1118,6 +1119,7 @@ ManagerImpl::selectAudioDriver (void)
if (sampleRate <=0 || sampleRate > 48000) {
sampleRate = 8000;
}
+ int frameSize = getConfigInt(AUDIO, DRIVER_FRAME_SIZE);
// this is when no audio device in/out are set
// or the audio device in/out are set to 0
@@ -1128,7 +1130,7 @@ ManagerImpl::selectAudioDriver (void)
//}
_debugInit(" AudioLayer Opening Device");
_audiodriver->setErrorMessage("");
- _audiodriver->openDevice(noDeviceIn, noDeviceOut, sampleRate);
+ _audiodriver->openDevice(noDeviceIn, noDeviceOut, sampleRate, frameSize);
}
/**
@@ -1787,10 +1789,11 @@ ManagerImpl::removeAccount(const AccountID& accountID)
std::string
ManagerImpl::getDefaultAccount()
{
+
std::string id;
id = getConfigString(PREFERENCES, "DefaultAccount");
_debug("Default Account = %s\n",id.c_str());
- return id;
+ return id;
}
void
diff --git a/src/user_cfg.h b/src/user_cfg.h
index 0d9319d12b616a73abd31e8d5ff1594370a9219d..53e51cdd561df403ad334ae85bf76c3eb9f62b0a 100644
--- a/src/user_cfg.h
+++ b/src/user_cfg.h
@@ -62,7 +62,7 @@
#define DRIVER_NAME_IN "Drivers.driverNameIn"
#define DRIVER_NAME_OUT "Drivers.driverNameOut"
#define DRIVER_SAMPLE_RATE "Drivers.sampleRate"
-#define DRIVER_SAMPLE_RATE_DEFAULT "8000"
+#define DRIVER_FRAME_SIZE "Drivers.framesize"
#define NB_CODEC "Codecs.nbCodec"
#define CODEC1 "Codecs.codec1"
#define CODEC2 "Codecs.codec2"
@@ -102,6 +102,8 @@
#define DFT_SKIN "metal"
#define DFT_ZONE "North America"
#define DFT_VOICEMAIL "888"
+#define DFT_FRAME_SIZE "20"
+#define DFT_SAMPLE_RATE "44100"
// zeroconfig default value
#ifdef USE_ZEROCONF
#define CONFIG_ZEROCONF_DEFAULT_STR "1"