Chromium Code Reviews| Index: webrtc/audio/audio_send_stream.h |
| diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h |
| index 42a04aee0932b3d06ea0508d1b40373934c02c73..79501a1c93386a1bc58ae9b71f6e394e49271ce4 100644 |
| --- a/webrtc/audio/audio_send_stream.h |
| +++ b/webrtc/audio/audio_send_stream.h |
| @@ -51,6 +51,9 @@ class AudioSendStream final : public webrtc::AudioSendStream, |
| // webrtc::AudioSendStream implementation. |
| void Reconfigure(const webrtc::AudioSendStream::Config& config) override; |
| + const webrtc::AudioSendStream::Config& GetConfig() const override; |
|
eladalon
2017/07/25 13:35:59
I'm not going to move this in the .cc file, though
|
| + RtpState GetRtpState() const override; |
| + const TimeInterval& GetActiveLifetime() const override; |
| void Start() override; |
| void Stop() override; |
| @@ -73,12 +76,8 @@ class AudioSendStream final : public webrtc::AudioSendStream, |
| void OnPacketFeedbackVector( |
| const std::vector<PacketFeedback>& packet_feedback_vector) override; |
| - const webrtc::AudioSendStream::Config& config() const; |
| void SetTransportOverhead(int transport_overhead_per_packet); |
| - RtpState GetRtpState() const; |
| - const TimeInterval& GetActiveLifetime() const; |
| - |
| private: |
| class TimedTransport; |