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

Issue 21183002: Adding key press detection in the browser process. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: add linux impl Created 7 years, 4 months ago
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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. 1 // Copyright (c) 2012 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_AUDIO_CAPTURER_H_ 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
6 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_ 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
7 7
8 #include <list> 8 #include <list>
9 #include <string> 9 #include <string>
10 10
(...skipping 93 matching lines...) Expand 10 before | Expand all | Expand 10 after
104 WebRtcAudioCapturer(); 104 WebRtcAudioCapturer();
105 virtual ~WebRtcAudioCapturer(); 105 virtual ~WebRtcAudioCapturer();
106 106
107 private: 107 private:
108 typedef std::list<scoped_refptr<WebRtcAudioCapturerSinkOwner> > TrackList; 108 typedef std::list<scoped_refptr<WebRtcAudioCapturerSinkOwner> > TrackList;
109 109
110 // AudioCapturerSource::CaptureCallback implementation. 110 // AudioCapturerSource::CaptureCallback implementation.
111 // Called on the AudioInputDevice audio thread. 111 // Called on the AudioInputDevice audio thread.
112 virtual void Capture(media::AudioBus* audio_source, 112 virtual void Capture(media::AudioBus* audio_source,
113 int audio_delay_milliseconds, 113 int audio_delay_milliseconds,
114 double volume) OVERRIDE; 114 double volume,
115 bool key_pressed) OVERRIDE;
115 virtual void OnCaptureError() OVERRIDE; 116 virtual void OnCaptureError() OVERRIDE;
116 117
117 // Reconfigures the capturer with a new buffer size and capture parameters. 118 // Reconfigures the capturer with a new buffer size and capture parameters.
118 // Must be called without holding the lock. Returns true on success. 119 // Must be called without holding the lock. Returns true on success.
119 bool Reconfigure(int sample_rate, media::ChannelLayout channel_layout); 120 bool Reconfigure(int sample_rate, media::ChannelLayout channel_layout);
120 121
121 // Starts recording audio. 122 // Starts recording audio.
122 // Triggered by AddSink() on the main render thread or a Libjingle working 123 // Triggered by AddSink() on the main render thread or a Libjingle working
123 // thread. It should NOT be called under |lock_|. 124 // thread. It should NOT be called under |lock_|.
124 void Start(); 125 void Start();
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157 158
158 // The media session ID used to identify which input device to be started. 159 // The media session ID used to identify which input device to be started.
159 int session_id_; 160 int session_id_;
160 161
161 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer); 162 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
162 }; 163 };
163 164
164 } // namespace content 165 } // namespace content
165 166
166 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_ 167 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
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