| Index: webrtc/audio/audio_send_stream.h
|
| diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
|
| index 183ad88189d12104aaeec0e143ce946226955ba2..61dd7f24b45f2662e897df12936f72bd86252c29 100644
|
| --- a/webrtc/audio/audio_send_stream.h
|
| +++ b/webrtc/audio/audio_send_stream.h
|
| @@ -34,17 +34,15 @@ class AudioSendStream final : public webrtc::AudioSendStream {
|
| CongestionController* congestion_controller);
|
| ~AudioSendStream() override;
|
|
|
| - // webrtc::SendStream implementation.
|
| + // webrtc::AudioSendStream implementation.
|
| void Start() override;
|
| void Stop() override;
|
| - void SignalNetworkState(NetworkState state) override;
|
| - bool DeliverRtcp(const uint8_t* packet, size_t length) override;
|
| -
|
| - // webrtc::AudioSendStream implementation.
|
| bool SendTelephoneEvent(int payload_type, int event,
|
| int duration_ms) override;
|
| webrtc::AudioSendStream::Stats GetStats() const override;
|
|
|
| + void SignalNetworkState(NetworkState state);
|
| + bool DeliverRtcp(const uint8_t* packet, size_t length);
|
| const webrtc::AudioSendStream::Config& config() const;
|
|
|
| private:
|
|
|