Chromium Code Reviews| Index: webrtc/video/audio_receive_stream.h |
| diff --git a/webrtc/video/audio_receive_stream.h b/webrtc/video/audio_receive_stream.h |
| index 9935117d0f565dc1a7698a9d587146765ba6ae98..08d929adb32620904cc8eafc054f402399a4a510 100644 |
| --- a/webrtc/video/audio_receive_stream.h |
| +++ b/webrtc/video/audio_receive_stream.h |
| @@ -26,11 +26,16 @@ class AudioReceiveStream : public webrtc::AudioReceiveStream { |
| const webrtc::AudioReceiveStream::Config& config); |
| ~AudioReceiveStream() override {} |
| + // webrtc::ReceiveStream implementation. |
|
pbos-webrtc
2015/07/14 14:50:14
Same w/r/t comments, imo. remove.
Jelena
2015/07/15 11:12:03
Acknowledged.
|
| + 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) override; |
| + |
| + // webrtc::AudioReceiveStream implementation. |
| webrtc::AudioReceiveStream::Stats GetStats() const override; |
| - bool DeliverRtcp(const uint8_t* packet, size_t length); |
| - bool DeliverRtp(const uint8_t* packet, size_t length); |
| - |
| const webrtc::AudioReceiveStream::Config& config() const { |
| return config_; |
| } |