Index: webrtc/audio/audio_receive_stream.cc |
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc |
index 9c253894719278a48b354763b4aef0a3235d0817..8d04f978d7d62fd19fc9f58f158cff9f2070cdf5 100644 |
--- a/webrtc/audio/audio_receive_stream.cc |
+++ b/webrtc/audio/audio_receive_stream.cc |
@@ -150,7 +150,8 @@ bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) { |
// calls on the worker thread. We should move towards always using a network |
// thread. Then this check can be enabled. |
// RTC_DCHECK(!thread_checker_.CalledOnValidThread()); |
- return false; |
+ return channel_proxy_->ReceivedRTCPPacket(static_cast<const uint8_t*>(packet), |
+ length) == 0; |
} |
bool AudioReceiveStream::DeliverRtp(const uint8_t* packet, |
@@ -177,7 +178,8 @@ bool AudioReceiveStream::DeliverRtp(const uint8_t* packet, |
remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size, |
header, false); |
} |
- return true; |
+ |
+ return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time); |
} |
webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const { |