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Side by Side Diff: content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h

Issue 1618653004: Delete WebRtcLocalAudioTrackAdapter::GetRenderer. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 4 years, 11 months ago
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1 // Copyright 2014 The Chromium Authors. All rights reserved. 1 // Copyright 2014 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
6 #define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
7 7
8 #include <vector> 8 #include <vector>
9 9
10 #include "base/memory/ref_counted.h" 10 #include "base/memory/ref_counted.h"
(...skipping 53 matching lines...) Expand 10 before | Expand all | Expand 10 after
64 bool set_enabled(bool enable) override; 64 bool set_enabled(bool enable) override;
65 65
66 private: 66 private:
67 // webrtc::AudioTrackInterface implementation. 67 // webrtc::AudioTrackInterface implementation.
68 void AddSink(webrtc::AudioTrackSinkInterface* sink) override; 68 void AddSink(webrtc::AudioTrackSinkInterface* sink) override;
69 void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override; 69 void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override;
70 bool GetSignalLevel(int* level) override; 70 bool GetSignalLevel(int* level) override;
71 rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor() 71 rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor()
72 override; 72 override;
73 webrtc::AudioSourceInterface* GetSource() const override; 73 webrtc::AudioSourceInterface* GetSource() const override;
74 cricket::AudioRenderer* GetRenderer() override;
75 74
76 // Weak reference. 75 // Weak reference.
77 WebRtcLocalAudioTrack* owner_; 76 WebRtcLocalAudioTrack* owner_;
78 77
79 // The source of the audio track which handles the audio constraints. 78 // The source of the audio track which handles the audio constraints.
80 // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack. 79 // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack.
81 rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_; 80 rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
82 81
83 // Libjingle's signaling thread. 82 // Libjingle's signaling thread.
84 const scoped_refptr<base::SingleThreadTaskRunner> signaling_thread_; 83 const scoped_refptr<base::SingleThreadTaskRunner> signaling_thread_;
(...skipping 16 matching lines...) Expand all
101 base::ThreadChecker signaling_thread_checker_; 100 base::ThreadChecker signaling_thread_checker_;
102 base::ThreadChecker capture_thread_; 101 base::ThreadChecker capture_thread_;
103 102
104 // Protects |voe_channels_|, |audio_processor_| and |signal_level_|. 103 // Protects |voe_channels_|, |audio_processor_| and |signal_level_|.
105 mutable base::Lock lock_; 104 mutable base::Lock lock_;
106 }; 105 };
107 106
108 } // namespace content 107 } // namespace content
109 108
110 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ 109 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
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