From fa57bac526b99ee5a40cf41342149be7fcc45536 Mon Sep 17 00:00:00 2001
From: Emmanuel Milou
<emmanuel.milou@savoirfairelinux.comemmanuel.milou@savoirfairelinux.com>
Date: Tue, 27 Nov 2007 11:03:49 -0500
Subject: [PATCH] use sampling frequency and frame size from the user config
Sampling rate values are no more hardcoded. The sampling rate of the audio layer
and the frame size can be set in the user config file. The clock rate of the codec we use
in the rtp session is set with his actual value, but can be changed by modifying
the available codec in the user config file (only G711 for now)
---
src/audio/audiolayer.cpp | 9 +-
src/audio/audiolayer.h | 190 +++++----
src/audio/audiortp.cpp | 869 ++++++++++++++++++++-------------------
src/audio/audiortp.h | 152 ++++---
src/managerimpl.cpp | 9 +-
src/user_cfg.h | 4 +-
6 files changed, 640 insertions(+), 593 deletions(-)
diff --git a/src/audio/audiolayer.cpp b/src/audio/audiolayer.cpp
index 7754a34c73..45924b2c5c 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 3c45ed0bb8..548ef19793 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 fe6f0833a0..53efa0962d 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 4b01827692..2444be5676 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 0e92c0c971..de29ed38b9 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 0d9319d12b..53e51cdd56 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"
--
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