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
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 46b95ed..5cb2ac3 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -625,9 +625,21 @@
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 @@
_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);
}