Index: webrtc/audio/audio_receive_stream.cc |
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc |
index 05d6edfa4c1fccd33b953ec2e38c3c456eefcb83..c0893b9044f6239c60edab4c30f6c80e86f6e384 100644 |
--- a/webrtc/audio/audio_receive_stream.cc |
+++ b/webrtc/audio/audio_receive_stream.cc |
@@ -302,14 +302,12 @@ bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) { |
return channel_proxy_->ReceivedRTCPPacket(packet, length); |
} |
-bool AudioReceiveStream::DeliverRtp(const uint8_t* packet, |
- size_t length, |
- const PacketTime& packet_time) { |
+void AudioReceiveStream::OnRtpPacket(const RtpPacketReceived& packet) { |
// TODO(solenberg): Tests call this function on a network thread, libjingle |
// 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 channel_proxy_->ReceivedRTPPacket(packet, length, packet_time); |
+ channel_proxy_->OnRtpPacket(packet); |
} |
const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const { |