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

Issue 2436033002: Replace AudioConferenceMixer with AudioMixer. (Closed)
Patch Set: Added mock mixer, merged audio state tests. Created 4 years 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
11 #ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ 11 #ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
12 #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ 12 #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
13 13
14 #include <memory> 14 #include <memory>
15 15
16 #include "webrtc/api/audio/audio_mixer.h" 16 #include "webrtc/api/audio/audio_mixer.h"
17 #include "webrtc/api/call/audio_receive_stream.h" 17 #include "webrtc/api/call/audio_receive_stream.h"
18 #include "webrtc/api/call/audio_state.h" 18 #include "webrtc/api/call/audio_state.h"
19 #include "webrtc/audio/audio_state.h"
19 #include "webrtc/base/constructormagic.h" 20 #include "webrtc/base/constructormagic.h"
20 #include "webrtc/base/thread_checker.h" 21 #include "webrtc/base/thread_checker.h"
21 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" 22 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
22 23
23 namespace webrtc { 24 namespace webrtc {
24 class CongestionController; 25 class CongestionController;
25 class RemoteBitrateEstimator; 26 class RemoteBitrateEstimator;
26 class RtcEventLog; 27 class RtcEventLog;
27 28
28 namespace voe { 29 namespace voe {
(...skipping 28 matching lines...) Expand all
57 const webrtc::AudioReceiveStream::Config& config() const; 58 const webrtc::AudioReceiveStream::Config& config() const;
58 59
59 // AudioMixer::Source 60 // AudioMixer::Source
60 AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz, 61 AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
61 AudioFrame* audio_frame) override; 62 AudioFrame* audio_frame) override;
62 int PreferredSampleRate() const override; 63 int PreferredSampleRate() const override;
63 int Ssrc() const override; 64 int Ssrc() const override;
64 65
65 private: 66 private:
66 VoiceEngine* voice_engine() const; 67 VoiceEngine* voice_engine() const;
68 AudioState* audio_state() const;
69 int SetVoiceEnginePlayout(bool playout);
67 70
68 rtc::ThreadChecker thread_checker_; 71 rtc::ThreadChecker thread_checker_;
69 RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr; 72 RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr;
70 const webrtc::AudioReceiveStream::Config config_; 73 const webrtc::AudioReceiveStream::Config config_;
71 rtc::scoped_refptr<webrtc::AudioState> audio_state_; 74 rtc::scoped_refptr<webrtc::AudioState> audio_state_;
72 std::unique_ptr<RtpHeaderParser> rtp_header_parser_; 75 std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
73 std::unique_ptr<voe::ChannelProxy> channel_proxy_; 76 std::unique_ptr<voe::ChannelProxy> channel_proxy_;
74 77
78 bool playing_ ACCESS_ON(thread_checker_) = false;
79
75 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream); 80 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
76 }; 81 };
77 } // namespace internal 82 } // namespace internal
78 } // namespace webrtc 83 } // namespace webrtc
79 84
80 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ 85 #endif // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
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