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: |