| 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..d99956c2ddd381b447ce5b7c11679ad7a50d9fce 100644
|
| --- a/webrtc/audio/audio_receive_stream.h
|
| +++ b/webrtc/audio/audio_receive_stream.h
|
| @@ -36,20 +36,17 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream {
|
| const rtc::scoped_refptr<webrtc::AudioState>& audio_state);
|
| ~AudioReceiveStream() override;
|
|
|
| - // webrtc::ReceiveStream implementation.
|
| + // 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;
|
|
|
| + void SignalNetworkState(NetworkState state);
|
| + bool DeliverRtcp(const uint8_t* packet, size_t length);
|
| + bool DeliverRtp(const uint8_t* packet,
|
| + size_t length,
|
| + const PacketTime& packet_time);
|
| const webrtc::AudioReceiveStream::Config& config() const;
|
|
|
| private:
|
|
|