Chromium Code Reviews| Index: webrtc/audio/audio_receive_stream.h |
| diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h |
| index 37117427a172d11ad48a95cd414b2d985d4f79fb..75143814462121456add2dbe3ea159e9cc8857ef 100644 |
| --- a/webrtc/audio/audio_receive_stream.h |
| +++ b/webrtc/audio/audio_receive_stream.h |
| @@ -36,16 +36,16 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream { |
| const rtc::scoped_refptr<webrtc::AudioState>& audio_state); |
| ~AudioReceiveStream() override; |
| - // webrtc::ReceiveStream implementation. |
| + void SignalNetworkState(NetworkState state); |
|
The Sun (google.com)
2016/04/27 19:39:46
nit: move this to sit together with config()
anoth
pbos-webrtc
2016/04/28 07:10:13
Done.
|
| + |
| + // webrtc::AudioReceiveStream implementation. |
| void Start() override; |
| void Stop() override; |
| - void SignalNetworkState(NetworkState state) override; |
| bool DeliverRtcp(const uint8_t* packet, size_t length) override; |
| bool DeliverRtp(const uint8_t* packet, |
| size_t length, |
| const PacketTime& packet_time) override; |
| - // webrtc::AudioReceiveStream implementation. |
| webrtc::AudioReceiveStream::Stats GetStats() const override; |
| void SetSink(std::unique_ptr<AudioSinkInterface> sink) override; |