Connect ACM with RTP module for audio NACK.

Depends on http://review.webrtc.org/1507004/

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1613007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4189 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pwestin@webrtc.org
2013-06-06 21:09:01 +00:00
parent a305e9612a
commit d30859e58e

View File

@ -625,9 +625,21 @@ Channel::OnReceivedPayloadData(const uint8_t* payloadData,
return -1;
}
// Update the packet delay
// Update the packet delay.
UpdatePacketDelay(rtpHeader->header.timestamp,
rtpHeader->header.sequenceNumber);
if (kNackOff != _rtpRtcpModule->NACK()) { // Is NACK on?
uint16_t round_trip_time = 0;
_rtpRtcpModule->RTT(_rtpRtcpModule->RemoteSSRC(), &round_trip_time,
NULL, NULL, NULL);
std::vector<uint16_t> nack_list = _audioCodingModule.GetNackList(
round_trip_time);
if (!nack_list.empty()) {
ResendPackets(nack_list.data(), nack_list.size());
}
}
return 0;
}
@ -4235,11 +4247,14 @@ void Channel::SetNACKStatus(bool enable, int maxNumberOfPackets) {
_rtpRtcpModule->SetStorePacketsStatus(enable, maxNumberOfPackets);
_rtpRtcpModule->SetNACKStatus(enable ? kNackRtcp : kNackOff,
maxNumberOfPackets);
if (enable)
_audioCodingModule.EnableNack(maxNumberOfPackets);
else
_audioCodingModule.DisableNack();
}
// Called by the ACM when it's missing one or more packets.
int Channel::ResendPackets(const uint16_t* sequence_numbers,
int length) {
// Called when we are missing one or more packets.
int Channel::ResendPackets(const uint16_t* sequence_numbers, int length) {
return _rtpRtcpModule->SendNACK(sequence_numbers, length);
}