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), |
the sun
2016/04/22 12:40:31
Looks like this cast is not needed anymore?
mflodman
2016/04/27 13:42:17
Doh!
|
+ 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, |
the sun
2016/04/22 12:40:31
stefan: is it still appropriate to fork packets of
mflodman
2016/04/27 13:42:17
For video this is done in ViEReciever, that we're
the sun
2016/04/28 09:15:56
ok, just wanted to make sure we won't magically se
stefan-webrtc
2016/04/28 09:27:48
Should be fine, and I also prefer it being done th
|
header, false); |
} |
- return true; |
+ |
+ return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time); |
the sun
2016/04/22 12:40:31
VoE::ReceivedRTPPacket() will:
if ((length < 12)
mflodman
2016/04/27 13:42:17
Channel::ReceivedRTPPacket is calling rtp_header_p
the sun
2016/04/28 09:15:56
Yes, please add the check in Channel::ReceivedRTPP
|
} |
webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const { |