| Index: webrtc/audio/audio_receive_stream.h
|
| diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
|
| index 6721c7ee6531e4af9d1299074983e65ce697ef6b..13869c453e1e2f9a1c095e8b5825f4cdf947cac1 100644
|
| --- a/webrtc/audio/audio_receive_stream.h
|
| +++ b/webrtc/audio/audio_receive_stream.h
|
| @@ -19,7 +19,6 @@
|
| #include "webrtc/base/thread_checker.h"
|
| #include "webrtc/call/audio_receive_stream.h"
|
| #include "webrtc/call/syncable.h"
|
| -#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
|
|
|
| namespace webrtc {
|
| class PacketRouter;
|
| @@ -81,7 +80,6 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
|
| RemoteBitrateEstimator* const remote_bitrate_estimator_;
|
| const webrtc::AudioReceiveStream::Config config_;
|
| rtc::scoped_refptr<webrtc::AudioState> audio_state_;
|
| - std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
|
| std::unique_ptr<voe::ChannelProxy> channel_proxy_;
|
|
|
| bool playing_ ACCESS_ON(worker_thread_checker_) = false;
|
|
|