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

Issue 396263004: Switch the input code to use AudioBlockFifo (Closed) Base URL: svn://svn.chromium.org/chrome/trunk/src
Patch Set: addressed the comments. Created 6 years, 5 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_LOCAL_AUDIO_RENDERER_H_ 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_
6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_ 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_
7 7
8 #include <vector> 8 #include <vector>
9 9
10 #include "base/callback.h" 10 #include "base/callback.h"
11 #include "base/memory/ref_counted.h" 11 #include "base/memory/ref_counted.h"
12 #include "base/message_loop/message_loop_proxy.h" 12 #include "base/message_loop/message_loop_proxy.h"
13 #include "base/synchronization/lock.h" 13 #include "base/synchronization/lock.h"
14 #include "base/threading/thread_checker.h" 14 #include "base/threading/thread_checker.h"
15 #include "content/common/content_export.h" 15 #include "content/common/content_export.h"
16 #include "content/public/renderer/media_stream_audio_sink.h" 16 #include "content/public/renderer/media_stream_audio_sink.h"
17 #include "content/renderer/media/media_stream_audio_renderer.h" 17 #include "content/renderer/media/media_stream_audio_renderer.h"
18 #include "content/renderer/media/webrtc_audio_device_impl.h" 18 #include "content/renderer/media/webrtc_audio_device_impl.h"
19 #include "content/renderer/media/webrtc_local_audio_track.h" 19 #include "content/renderer/media/webrtc_local_audio_track.h"
20 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" 20 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
21 21
22 namespace media { 22 namespace media {
23 class AudioBus; 23 class AudioBus;
24 class AudioFifo; 24 class AudioBlockFifo;
25 class AudioOutputDevice; 25 class AudioOutputDevice;
26 class AudioParameters; 26 class AudioParameters;
27 } 27 }
28 28
29 namespace content { 29 namespace content {
30 30
31 class WebRtcAudioCapturer; 31 class WebRtcAudioCapturer;
32 32
33 // WebRtcLocalAudioRenderer is a MediaStreamAudioRenderer designed for rendering 33 // WebRtcLocalAudioRenderer is a MediaStreamAudioRenderer designed for rendering
34 // local audio media stream tracks, 34 // local audio media stream tracks,
(...skipping 82 matching lines...) Expand 10 before | Expand all | Expand 10 after
117 const int session_id_; 117 const int session_id_;
118 118
119 // MessageLoop associated with the single thread that performs all control 119 // MessageLoop associated with the single thread that performs all control
120 // tasks. Set to the MessageLoop that invoked the ctor. 120 // tasks. Set to the MessageLoop that invoked the ctor.
121 const scoped_refptr<base::MessageLoopProxy> message_loop_; 121 const scoped_refptr<base::MessageLoopProxy> message_loop_;
122 122
123 // The sink (destination) for rendered audio. 123 // The sink (destination) for rendered audio.
124 scoped_refptr<media::AudioOutputDevice> sink_; 124 scoped_refptr<media::AudioOutputDevice> sink_;
125 125
126 // Contains copies of captured audio frames. 126 // Contains copies of captured audio frames.
127 scoped_ptr<media::AudioFifo> loopback_fifo_; 127 scoped_ptr<media::AudioBlockFifo> loopback_fifo_;
128 128
129 // Stores last time a render callback was received. The time difference 129 // Stores last time a render callback was received. The time difference
130 // between a new time stamp and this value can be used to derive the 130 // between a new time stamp and this value can be used to derive the
131 // total render time. 131 // total render time.
132 base::TimeTicks last_render_time_; 132 base::TimeTicks last_render_time_;
133 133
134 // Keeps track of total time audio has been rendered. 134 // Keeps track of total time audio has been rendered.
135 base::TimeDelta total_render_time_; 135 base::TimeDelta total_render_time_;
136 136
137 // The audio parameters of the capture source. 137 // The audio parameters of the capture source.
(...skipping 25 matching lines...) Expand all
163 163
164 // Used to DCHECK that some methods are called on the capture audio thread. 164 // Used to DCHECK that some methods are called on the capture audio thread.
165 base::ThreadChecker capture_thread_checker_; 165 base::ThreadChecker capture_thread_checker_;
166 166
167 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer); 167 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer);
168 }; 168 };
169 169
170 } // namespace content 170 } // namespace content
171 171
172 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_ 172 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_
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