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_; |