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Side by Side Diff: content/renderer/media/webrtc_local_audio_track.h

Issue 670683003: Standardize usage of virtual/override/final in content/renderer/ (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 6 years, 2 months ago
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1 // Copyright 2013 The Chromium Authors. All rights reserved. 1 // Copyright 2013 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_LOCAL_AUDIO_TRACK_H_ 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
7 7
8 #include <list> 8 #include <list>
9 #include <string> 9 #include <string>
10 10
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55 // same sink interface as MediaStreamAudioSink. 55 // same sink interface as MediaStreamAudioSink.
56 void AddSink(PeerConnectionAudioSink* sink); 56 void AddSink(PeerConnectionAudioSink* sink);
57 void RemoveSink(PeerConnectionAudioSink* sink); 57 void RemoveSink(PeerConnectionAudioSink* sink);
58 58
59 // Starts the local audio track. Called on the main render thread and 59 // Starts the local audio track. Called on the main render thread and
60 // should be called only once when audio track is created. 60 // should be called only once when audio track is created.
61 void Start(); 61 void Start();
62 62
63 // Stops the local audio track. Called on the main render thread and 63 // Stops the local audio track. Called on the main render thread and
64 // should be called only once when audio track going away. 64 // should be called only once when audio track going away.
65 virtual void Stop() override; 65 void Stop() override;
66 66
67 // Method called by the capturer to deliver the capture data. 67 // Method called by the capturer to deliver the capture data.
68 // Called on the capture audio thread. 68 // Called on the capture audio thread.
69 void Capture(const int16* audio_data, 69 void Capture(const int16* audio_data,
70 base::TimeDelta delay, 70 base::TimeDelta delay,
71 int volume, 71 int volume,
72 bool key_pressed, 72 bool key_pressed,
73 bool need_audio_processing, 73 bool need_audio_processing,
74 bool force_report_nonzero_energy); 74 bool force_report_nonzero_energy);
75 75
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119 // Used to calculate the signal level that shows in the UI. 119 // Used to calculate the signal level that shows in the UI.
120 // Accessed on only the audio thread. 120 // Accessed on only the audio thread.
121 scoped_ptr<MediaStreamAudioLevelCalculator> level_calculator_; 121 scoped_ptr<MediaStreamAudioLevelCalculator> level_calculator_;
122 122
123 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack); 123 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack);
124 }; 124 };
125 125
126 } // namespace content 126 } // namespace content
127 127
128 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_ 128 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
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