Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(267)

Side by Side Diff: content/renderer/media/webrtc/webrtc_local_audio_track_adapter.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
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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 24 matching lines...) Expand all
35 public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) { 35 public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) {
36 public: 36 public:
37 static scoped_refptr<WebRtcLocalAudioTrackAdapter> Create( 37 static scoped_refptr<WebRtcLocalAudioTrackAdapter> Create(
38 const std::string& label, 38 const std::string& label,
39 webrtc::AudioSourceInterface* track_source); 39 webrtc::AudioSourceInterface* track_source);
40 40
41 WebRtcLocalAudioTrackAdapter( 41 WebRtcLocalAudioTrackAdapter(
42 const std::string& label, 42 const std::string& label,
43 webrtc::AudioSourceInterface* track_source); 43 webrtc::AudioSourceInterface* track_source);
44 44
45 virtual ~WebRtcLocalAudioTrackAdapter(); 45 ~WebRtcLocalAudioTrackAdapter() override;
46 46
47 void Initialize(WebRtcLocalAudioTrack* owner); 47 void Initialize(WebRtcLocalAudioTrack* owner);
48 48
49 std::vector<int> VoeChannels() const; 49 std::vector<int> VoeChannels() const;
50 50
51 // Called on the audio thread by the WebRtcLocalAudioTrack to set the signal 51 // Called on the audio thread by the WebRtcLocalAudioTrack to set the signal
52 // level of the audio data. 52 // level of the audio data.
53 void SetSignalLevel(int signal_level); 53 void SetSignalLevel(int signal_level);
54 54
55 // Method called by the WebRtcLocalAudioTrack to set the processor that 55 // Method called by the WebRtcLocalAudioTrack to set the processor that
56 // applies signal processing on the data of the track. 56 // applies signal processing on the data of the track.
57 // This class will keep a reference of the |processor|. 57 // This class will keep a reference of the |processor|.
58 // Called on the main render thread. 58 // Called on the main render thread.
59 void SetAudioProcessor( 59 void SetAudioProcessor(
60 const scoped_refptr<MediaStreamAudioProcessor>& processor); 60 const scoped_refptr<MediaStreamAudioProcessor>& processor);
61 61
62 private: 62 private:
63 // webrtc::MediaStreamTrack implementation. 63 // webrtc::MediaStreamTrack implementation.
64 virtual std::string kind() const override; 64 std::string kind() const override;
65 65
66 // webrtc::AudioTrackInterface implementation. 66 // webrtc::AudioTrackInterface implementation.
67 virtual void AddSink(webrtc::AudioTrackSinkInterface* sink) override; 67 void AddSink(webrtc::AudioTrackSinkInterface* sink) override;
68 virtual void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override; 68 void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override;
69 virtual bool GetSignalLevel(int* level) override; 69 bool GetSignalLevel(int* level) override;
70 virtual rtc::scoped_refptr<webrtc::AudioProcessorInterface> 70 rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor()
71 GetAudioProcessor() override; 71 override;
72 72
73 // cricket::AudioCapturer implementation. 73 // cricket::AudioCapturer implementation.
74 virtual void AddChannel(int channel_id) override; 74 void AddChannel(int channel_id) override;
75 virtual void RemoveChannel(int channel_id) override; 75 void RemoveChannel(int channel_id) override;
76 76
77 // webrtc::AudioTrackInterface implementation. 77 // webrtc::AudioTrackInterface implementation.
78 virtual webrtc::AudioSourceInterface* GetSource() const override; 78 webrtc::AudioSourceInterface* GetSource() const override;
79 virtual cricket::AudioRenderer* GetRenderer() override; 79 cricket::AudioRenderer* GetRenderer() override;
80 80
81 // Weak reference. 81 // Weak reference.
82 WebRtcLocalAudioTrack* owner_; 82 WebRtcLocalAudioTrack* owner_;
83 83
84 // The source of the audio track which handles the audio constraints. 84 // The source of the audio track which handles the audio constraints.
85 // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack. 85 // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack.
86 rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_; 86 rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
87 87
88 // The audio processsor that applies audio processing on the data of audio 88 // The audio processsor that applies audio processing on the data of audio
89 // track. 89 // track.
90 scoped_refptr<MediaStreamAudioProcessor> audio_processor_; 90 scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
91 91
92 // A vector of WebRtc VoE channels that the capturer sends data to. 92 // A vector of WebRtc VoE channels that the capturer sends data to.
93 std::vector<int> voe_channels_; 93 std::vector<int> voe_channels_;
94 94
95 // A vector of the peer connection sink adapters which receive the audio data 95 // A vector of the peer connection sink adapters which receive the audio data
96 // from the audio track. 96 // from the audio track.
97 ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_; 97 ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_;
98 98
99 // The amplitude of the signal. 99 // The amplitude of the signal.
100 int signal_level_; 100 int signal_level_;
101 101
102 // Protects |voe_channels_|, |audio_processor_| and |signal_level_|. 102 // Protects |voe_channels_|, |audio_processor_| and |signal_level_|.
103 mutable base::Lock lock_; 103 mutable base::Lock lock_;
104 }; 104 };
105 105
106 } // namespace content 106 } // namespace content
107 107
108 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_ 108 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
OLDNEW
« no previous file with comments | « content/renderer/media/webrtc/webrtc_audio_sink_adapter.h ('k') | content/renderer/media/webrtc/webrtc_media_stream_adapter.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698