AmAudioRtpFormat::AmAudioRtpFormat(const vector<SdpPayload *>& payloads)
: AmAudioFormat(), m_payloads(payloads), m_currentPayload(-1)
{
for (vector<SdpPayload *>::iterator it = m_payloads.begin();
it != m_payloads.end(); ++it) {
DBG("(*it)->payload_type = %i\n",(*it)->payload_type);
m_sdpPayloadByPayload[(*it)->payload_type] = *it;
}
setCurrentPayload(m_payloads[0]->payload_type);
}
/*
@param wallclock_ts [in] the current ts in the audio buffer
*/
int AmRtpAudio::receive(unsigned long long system_ts)
{
int size;
unsigned int rtp_ts;
int new_payload = -1;
if(!fmt.get() || (!playout_buffer.get())) {
DBG("audio format not initialized\n");
return RTP_ERROR;
}
unsigned int wallclock_ts = scaleSystemTS(system_ts);
while(true) {
size = AmRtpStream::receive((unsigned char*)samples,
(unsigned int)AUDIO_BUFFER_SIZE, rtp_ts,
new_payload);
if(size <= 0) {
switch(size){
case 0: break;
case RTP_DTMF:
case RTP_UNKNOWN_PL:
case RTP_PARSE_ERROR:
continue;
case RTP_TIMEOUT:
//FIXME: postRequest(new SchedRequest(AmMediaProcessor::RemoveSession,s));
// post to the session (FIXME: is session always set? seems to be...)
session->postEvent(new AmRtpTimeoutEvent());
return -1;
case RTP_BUFFER_SIZE:
default:
ERROR("AmRtpStream::receive() returned %i\n",size);
//FIXME: postRequest(new SchedRequest(AmMediaProcessor::ClearSession,s));
// or AmMediaProcessor::instance()->clearSession(session);
return -1;
break;
}
break;
}
if (// don't process if we don't need to
// ignore CN
COMFORT_NOISE_PAYLOAD_TYPE == new_payload ||
// ignore packet if payload not found
setCurrentPayload(new_payload)
){
playout_buffer->clearLastTs();
continue;
}
size = decode(size);
if(size <= 0){
ERROR("decode() returned %i\n",size);
return -1;
}
// This only works because the possible ratio (Rate/TSRate)
// is 2. Rate and TSRate are only different in case of g722.
// For g722, TSRate=8000 and Rate=16000
//
AmAudioRtpFormat* rtp_fmt = (AmAudioRtpFormat*)fmt.get();
unsigned long long adjusted_rtp_ts = rtp_ts;
if(rtp_fmt->getRate() != rtp_fmt->getTSRate()) {
adjusted_rtp_ts =
(unsigned long long)rtp_ts *
(unsigned long long)rtp_fmt->getRate()
/ (unsigned long long)rtp_fmt->getTSRate();
}
playout_buffer->write(wallclock_ts, adjusted_rtp_ts,
(ShortSample*)((unsigned char *)samples),
PCM16_B2S(size), begin_talk);
if(!active) {
DBG("switching to active-mode\t(ts=%u;stream=%p)\n",
rtp_ts,this);
active = true;
}
}
return size;
}
