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Unified Diff: webrtc/audio/audio_receive_stream.cc

Issue 1909333002: Switch voice transport to use Call and Stream instead of VoENetwork. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 8 months ago
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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 {
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