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 5ee49da91a7a1469285d1fcf675c39c53a43ac5c..efd0a6e960335f15ccdbd638522f01c9d7c962e0 100644 | 
| --- a/webrtc/audio/audio_send_stream.h | 
| +++ b/webrtc/audio/audio_send_stream.h | 
| @@ -45,6 +45,8 @@ class AudioSendStream final : public webrtc::AudioSendStream, | 
| RtcpRttStats* rtcp_rtt_stats); | 
| ~AudioSendStream() override; | 
| + void Reconfigure(const webrtc::AudioSendStream::Config& config) override; | 
| 
 
the sun
2017/04/04 23:02:54
nit: looks like it should go under the below comme
 
 | 
| + | 
| // webrtc::AudioSendStream implementation. | 
| void Start() override; | 
| void Stop() override; | 
| @@ -68,14 +70,23 @@ 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 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_; | 
| + PacketRouter* const packet_router_; | 
| BitrateAllocator* const bitrate_allocator_; | 
| CongestionController* const congestion_controller_; | 
| std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_; |