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Side by Side Diff: webrtc/audio/audio_receive_stream.h

Issue 2436033002: Replace AudioConferenceMixer with AudioMixer. (Closed)
Patch Set: Added errors and logs to AudioTransport. Created 4 years, 1 month ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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62 private: 62 private:
63 VoiceEngine* voice_engine() const; 63 VoiceEngine* voice_engine() const;
64 64
65 rtc::ThreadChecker thread_checker_; 65 rtc::ThreadChecker thread_checker_;
66 RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr; 66 RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr;
67 const webrtc::AudioReceiveStream::Config config_; 67 const webrtc::AudioReceiveStream::Config config_;
68 rtc::scoped_refptr<webrtc::AudioState> audio_state_; 68 rtc::scoped_refptr<webrtc::AudioState> audio_state_;
69 std::unique_ptr<RtpHeaderParser> rtp_header_parser_; 69 std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
70 std::unique_ptr<voe::ChannelProxy> channel_proxy_; 70 std::unique_ptr<voe::ChannelProxy> channel_proxy_;
71 71
72 bool playing_ ACCESS_ON(thread_checker_) = false;
73
72 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); 74 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
73 }; 75 };
74 } // namespace internal 76 } // namespace internal
75 } // namespace webrtc 77 } // namespace webrtc
76 78
77 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ 79 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
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