Index: webrtc/audio/audio_receive_stream.h |
diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h |
index 42286af22b9d584f7744292b8050ecd47cafd118..4940c6a64cc8d6ad7e8c1b521a9d2c937d1de52f 100644 |
--- a/webrtc/audio/audio_receive_stream.h |
+++ b/webrtc/audio/audio_receive_stream.h |
@@ -17,6 +17,7 @@ |
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
namespace webrtc { |
+class CongestionController; |
class RemoteBitrateEstimator; |
namespace voe { |
@@ -27,7 +28,7 @@ namespace internal { |
class AudioReceiveStream final : public webrtc::AudioReceiveStream { |
public: |
- AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator, |
+ AudioReceiveStream(CongestionController* congestion_controller, |
const webrtc::AudioReceiveStream::Config& config, |
const rtc::scoped_refptr<webrtc::AudioState>& audio_state); |
~AudioReceiveStream() override; |
@@ -52,7 +53,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream { |
VoiceEngine* voice_engine() const; |
rtc::ThreadChecker thread_checker_; |
- RemoteBitrateEstimator* const remote_bitrate_estimator_; |
+ RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr; |
const webrtc::AudioReceiveStream::Config config_; |
rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_; |