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: |