Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(82)

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

Issue 1036993003: Fix the way locks are acquired in WebRtcAudioDeviceImpl::SetAudioRenderer. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Name variables in comments according to convention, extra header comment. Created 5 years, 8 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 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_AUDIO_DEVICE_IMPL_H_ 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
6 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
7 7
8 #include <string> 8 #include <string>
9 #include <vector> 9 #include <vector>
10 10
(...skipping 149 matching lines...) Expand 10 before | Expand all | Expand 10 after
160 // steady state is that the volume control reaches its max and the new_volume 160 // steady state is that the volume control reaches its max and the new_volume
161 // value from the AGC is zero. A loud voice input is required to break this 161 // value from the AGC is zero. A loud voice input is required to break this
162 // state and start lowering the level again. 162 // state and start lowering the level again.
163 // 163 //
164 // Implementation notes: 164 // Implementation notes:
165 // 165 //
166 // - This class must be created and destroyed on the main render thread and 166 // - This class must be created and destroyed on the main render thread and
167 // most methods are called on the same thread. However, some methods are 167 // most methods are called on the same thread. However, some methods are
168 // also called on a Libjingle worker thread. RenderData is called on the 168 // also called on a Libjingle worker thread. RenderData is called on the
169 // AudioOutputDevice thread and CaptureData on the AudioInputDevice thread. 169 // AudioOutputDevice thread and CaptureData on the AudioInputDevice thread.
170 // To summarize: this class lives on four different threads. 170 // To summarize: this class lives on four different threads, so it is
171 // important to be careful with the order in which locks are acquired in
172 // order to avoid potential deadlocks.
171 // - The webrtc::AudioDeviceModule is reference counted. 173 // - The webrtc::AudioDeviceModule is reference counted.
172 // - AGC is only supported in combination with the WASAPI-based audio layer 174 // - AGC is only supported in combination with the WASAPI-based audio layer
173 // on Windows, i.e., it is not supported on Windows XP. 175 // on Windows, i.e., it is not supported on Windows XP.
174 // - All volume levels required for the AGC scheme are transfered in a 176 // - All volume levels required for the AGC scheme are transfered in a
175 // normalized range [0.0, 1.0]. Scaling takes place in both endpoints 177 // normalized range [0.0, 1.0]. Scaling takes place in both endpoints
176 // (WebRTC client a media layer). This approach ensures that we can avoid 178 // (WebRTC client a media layer). This approach ensures that we can avoid
177 // transferring maximum levels between the renderer and the browser. 179 // transferring maximum levels between the renderer and the browser.
178 // 180 //
179 181
180 namespace content { 182 namespace content {
(...skipping 213 matching lines...) Expand 10 before | Expand all | Expand 10 after
394 // Buffer used for temporary storage during render callback. 396 // Buffer used for temporary storage during render callback.
395 // It is only accessed by the audio render thread. 397 // It is only accessed by the audio render thread.
396 std::vector<int16> render_buffer_; 398 std::vector<int16> render_buffer_;
397 399
398 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl); 400 DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
399 }; 401 };
400 402
401 } // namespace content 403 } // namespace content
402 404
403 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_ 405 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698