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_; |