| Index: webrtc/audio/audio_send_stream.h
|
| diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
|
| index 567799c336f7bbaba1436a323148fce71f0f1329..2072b5a38f5546639f06b0d9d0e1cbd18f170568 100644
|
| --- a/webrtc/audio/audio_send_stream.h
|
| +++ b/webrtc/audio/audio_send_stream.h
|
| @@ -47,6 +47,8 @@ class AudioSendStream final : public webrtc::AudioSendStream,
|
| RtcpRttStats* rtcp_rtt_stats);
|
| ~AudioSendStream() override;
|
|
|
| + void Reconfigure(const webrtc::AudioSendStream::Config& config) override;
|
| +
|
| // webrtc::AudioSendStream implementation.
|
| void Start() override;
|
| void Stop() override;
|
| @@ -75,14 +77,22 @@ class AudioSendStream final : public webrtc::AudioSendStream,
|
| private:
|
| VoiceEngine* voice_engine() const;
|
|
|
| - bool SetupSendCodec();
|
| + bool SetupSendCodec(const Config& new_config);
|
| + bool ReconfigureSendCodec(const Config& new_config);
|
| + void ReconfigureANA(const Config& new_config);
|
| + void ReconfigureCNG(const Config& new_config);
|
| + void ReconfigureBitrateObserver(const Config& new_config);
|
| +
|
| + void ConfigureBitrateObserver(int min_bitrate_bps, int max_bitrate_bps);
|
| + void RemoveBitrateObserver();
|
|
|
| rtc::ThreadChecker worker_thread_checker_;
|
| rtc::ThreadChecker pacer_thread_checker_;
|
| rtc::TaskQueue* worker_queue_;
|
| - const webrtc::AudioSendStream::Config config_;
|
| + webrtc::AudioSendStream::Config config_;
|
| rtc::scoped_refptr<webrtc::AudioState> audio_state_;
|
| std::unique_ptr<voe::ChannelProxy> channel_proxy_;
|
| + RtcEventLog* const event_log_;
|
|
|
| BitrateAllocator* const bitrate_allocator_;
|
| RtpTransportControllerSendInterface* const transport_;
|
|
|