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Commit 4b951b37 authored by Alexandre Savard's avatar Alexandre Savard
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[#3443] Fix G722 codec timestamp incrementation

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sflphone-common/src/audio/audiortp/AudioRtpSession.h
+ 82
− 53
View file @ 4b951b37
......@@ -50,6 +50,9 @@
#include <ccrtp/rtp.h>
#include <cc++/numbers.h> // ost::Time
// Frequency (in packet number)
#define RTP_TIMESTAMP_RESET_FREQ 100
namespace sfl {
static const int schedulingTimeout = 100000;
......@@ -96,11 +99,11 @@ namespace sfl {
void putDtmfEvent(int digit);
/**
* Send DTMF over RTP (RFC2833). The timestamp and sequence number must be
* incremented as if it was microphone audio. This function change the payload type of the rtp session,
* send the appropriate DTMF digit using this payload, discard coresponding data from mainbuffer and get
* back the codec payload for further audio processing.
*/
* Send DTMF over RTP (RFC2833). The timestamp and sequence number must be
* incremented as if it was microphone audio. This function change the payload type of the rtp session,
* send the appropriate DTMF digit using this payload, discard coresponding data from mainbuffer and get
* back the codec payload for further audio processing.
*/
void sendDtmfEvent(sfl::DtmfEvent *dtmf);
inline float computeCodecFrameSize (int codecSamplePerFrame, int codecClockRate) {
......@@ -206,6 +209,17 @@ namespace sfl {
*/
int _timestamp;
/**
* Timestamp incrementation value based on codec period length (framesize)
* except for G722 which require a 8 kHz incrementation.
*/
int _timestampIncrement;
/**
* Timestamp reset freqeuncy specified in number of packet sent
*/
short _timestampCount;
/**
* Time counter used to trigger incoming call notification
*/
......@@ -215,7 +229,7 @@ namespace sfl {
* EventQueue used to store list of DTMF-
*/
EventQueue _eventQueue;
protected:
SIPCall * _ca;
};
......@@ -238,6 +252,8 @@ namespace sfl {
_manager(manager),
_converterSamplingRate(0),
_timestamp(0),
_timestampIncrement(0),
_timestampCount(0),
_countNotificationTime(0),
_ca (sipcall)
{
......@@ -329,18 +345,26 @@ namespace sfl {
_codecSampleRate = _audiocodec->getClockRate();
_codecFrameSize = _audiocodec->getFrameSize();
// G722 requires timestamp to be incremented at 8 kHz
if (_audiocodec->getPayload() == 9)
_timestampIncrement = 160;
else
_timestampIncrement = _codecFrameSize;
_debug("RTP: Codec sampling rate: %d", _codecSampleRate);
_debug("RTP: Codec frame size: %d", _codecFrameSize);
_debug("RTP: RTP timestamp increment: %d", _timestampIncrement);
//TODO: figure out why this is necessary.
// Even if specified as a 16 kHz codec, G722 requires rtp sending rate to be 8 kHz
if (_audiocodec->getPayload() == 9) {
_debug ("RTP: Setting payload format to G722");
_debug ("RTP: Setting G722 payload format");
static_cast<D*>(this)->setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
} else if (_audiocodec->hasDynamicPayload()) {
_debug ("RTP: Setting a dynamic payload format");
_debug ("RTP: Setting dynamic payload format");
static_cast<D*>(this)->setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) _audiocodec->getPayload(), _audiocodec->getClockRate()));
} else if (!_audiocodec->hasDynamicPayload() && _audiocodec->getPayload() != 9) {
_debug ("RTP: Setting a static payload format");
_debug ("RTP: Setting static payload format");
static_cast<D*>(this)->setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) _audiocodec->getPayload()));
}
}
......@@ -412,44 +436,44 @@ namespace sfl {
template<typename D>
void AudioRtpSession<D>::sendDtmfEvent(sfl::DtmfEvent *dtmf)
{
_debug("RTP: Send Dtmf %d", _eventQueue.size());
_timestamp += 160;
// discard equivalent size of audio
processDataEncode();
// change Payload type for DTMF payload
static_cast<D*>(this)->setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) 101, 8000));
_debug("RTP: Send Dtmf %d", _eventQueue.size());
// Set marker in case this is a new Event
if(dtmf->newevent)
static_cast<D*>(this)->setMark (true);
_timestamp += 160;
static_cast<D*>(this)->putData (_timestamp, (const unsigned char*)(&(dtmf->payload)), sizeof(ost::RTPPacket::RFC2833Payload));
// discard equivalent size of audio
processDataEncode();
// This is no more a new event
if(dtmf->newevent) {
dtmf->newevent = false;
static_cast<D*>(this)->setMark (false);
}
// get back the payload to audio
static_cast<D*>(this)->setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) _audiocodec->getPayload()));
// decrease length remaining to process for this event
dtmf->length -= 160;
// change Payload type for DTMF payload
static_cast<D*>(this)->setPayloadFormat (ost::DynamicPayloadFormat ( (ost::PayloadType) 101, 8000));
// Set marker in case this is a new Event
if(dtmf->newevent)
static_cast<D*>(this)->setMark (true);
static_cast<D*>(this)->putData (_timestamp, (const unsigned char*)(&(dtmf->payload)), sizeof(ost::RTPPacket::RFC2833Payload));
dtmf->payload.duration += 1;
// This is no more a new event
if(dtmf->newevent) {
dtmf->newevent = false;
static_cast<D*>(this)->setMark (false);
}
// next packet is going to be the last one
if((dtmf->length - 160) < 160)
dtmf->payload.ebit = true;
// get back the payload to audio
static_cast<D*>(this)->setPayloadFormat (ost::StaticPayloadFormat ( (ost::StaticPayloadType) _audiocodec->getPayload()));
if(dtmf->length < 160) {
delete dtmf;
_eventQueue.pop_front();
}
// decrease length remaining to process for this event
dtmf->length -= 160;
dtmf->payload.duration += 1;
// next packet is going to be the last one
if((dtmf->length - 160) < 160)
dtmf->payload.ebit = true;
if(dtmf->length < 160) {
delete dtmf;
_eventQueue.pop_front();
}
}
template <typename D>
......@@ -567,7 +591,7 @@ namespace sfl {
// 4. send it
// Increment timestamp for outgoing packet
_timestamp += _codecFrameSize;
_timestamp += _timestampIncrement;
if (!_audiolayer) {
_debug ("No audiolayer available for MIC");
......@@ -643,11 +667,11 @@ namespace sfl {
int sessionWaiting;
int threadSleep = 0;
if (_codecSampleRate != 0){
threadSleep = (_codecFrameSize * 1000) / _codecSampleRate;
}
if (_codecSampleRate != 0){
threadSleep = (_codecFrameSize * 1000) / _codecSampleRate;
}
else {
threadSleep = _layerFrameSize;
threadSleep = _layerFrameSize;
}
TimerPort::setTimer (threadSleep);
......@@ -668,15 +692,20 @@ namespace sfl {
while (!testCancel()) {
// ost::MutexLock lock(*(_manager->getAudioLayerMutex()));
_manager->getAudioLayerMutex()->enter();
// Reset timestamp to make sure the timing information are up to date
if(_timestampCount > RTP_TIMESTAMP_RESET_FREQ) {
_timestamp = static_cast<D*>(this)->getCurrentTimestamp();
_timestampCount = 0;
}
_timestampCount++;
// converterSamplingRate = _audiolayer->getMainBuffer()->getInternalSamplingRate();
_converterSamplingRate = _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
_manager->getAudioLayerMutex()->enter();
// converterSamplingRate = _audiolayer->getMainBuffer()->getInternalSamplingRate();
_converterSamplingRate = _manager->getAudioDriver()->getMainBuffer()->getInternalSamplingRate();
sessionWaiting = static_cast<D*>(this)->isWaiting();
sessionWaiting = static_cast<D*>(this)->isWaiting();
// Send session
if(_eventQueue.size() > 0) {
......
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