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; |