| Index: webrtc/audio/audio_receive_stream.h
|
| diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
|
| index 6e7da09d1ddf42d3b3916f312175af7f5db60bf3..3bba54ebb85a5d5b879ebbc8995a23327ea41e75 100644
|
| --- a/webrtc/audio/audio_receive_stream.h
|
| +++ b/webrtc/audio/audio_receive_stream.h
|
| @@ -22,9 +22,9 @@
|
| #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
|
|
|
| namespace webrtc {
|
| -class CongestionController;
|
| class RemoteBitrateEstimator;
|
| class RtcEventLog;
|
| +class PacketRouter;
|
|
|
| namespace voe {
|
| class ChannelProxy;
|
| @@ -36,7 +36,8 @@ class AudioSendStream;
|
| class AudioReceiveStream final : public webrtc::AudioReceiveStream,
|
| public AudioMixer::Source {
|
| public:
|
| - AudioReceiveStream(CongestionController* congestion_controller,
|
| + AudioReceiveStream(PacketRouter* packet_router,
|
| + RemoteBitrateEstimator* remote_bitrate_estimator,
|
| const webrtc::AudioReceiveStream::Config& config,
|
| const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
|
| webrtc::RtcEventLog* event_log);
|
| @@ -69,7 +70,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
|
| int SetVoiceEnginePlayout(bool playout);
|
|
|
| rtc::ThreadChecker thread_checker_;
|
| - RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr;
|
| + RemoteBitrateEstimator* const remote_bitrate_estimator_;
|
| const webrtc::AudioReceiveStream::Config config_;
|
| rtc::scoped_refptr<webrtc::AudioState> audio_state_;
|
| std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
|
|
|