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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_RENDERER_H_ | 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_ |
6 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_ | 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_ |
7 | 7 |
8 #include <stdint.h> | 8 #include <stdint.h> |
9 | 9 |
10 #include <map> | 10 #include <map> |
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153 SourcePlayingStates; | 153 SourcePlayingStates; |
154 | 154 |
155 // Used to DCHECK that we are called on the correct thread. | 155 // Used to DCHECK that we are called on the correct thread. |
156 base::ThreadChecker thread_checker_; | 156 base::ThreadChecker thread_checker_; |
157 | 157 |
158 // Flag to keep track the state of the renderer. | 158 // Flag to keep track the state of the renderer. |
159 State state_; | 159 State state_; |
160 | 160 |
161 // media::AudioRendererSink::RenderCallback implementation. | 161 // media::AudioRendererSink::RenderCallback implementation. |
162 // These two methods are called on the AudioOutputDevice worker thread. | 162 // These two methods are called on the AudioOutputDevice worker thread. |
163 int Render(media::AudioBus* audio_bus, | 163 int Render(base::TimeDelta delay, |
164 uint32_t frames_delayed, | 164 base::TimeTicks delay_timestamp, |
165 uint32_t frames_skipped) override; | 165 int prior_frames_skipped, |
| 166 media::AudioBus* audio_bus) override; |
166 void OnRenderError() override; | 167 void OnRenderError() override; |
167 | 168 |
168 // Called by AudioPullFifo when more data is necessary. | 169 // Called by AudioPullFifo when more data is necessary. |
169 // This method is called on the AudioOutputDevice worker thread. | 170 // This method is called on the AudioOutputDevice worker thread. |
170 void SourceCallback(int fifo_frame_delay, media::AudioBus* audio_bus); | 171 void SourceCallback(int fifo_frame_delay, media::AudioBus* audio_bus); |
171 | 172 |
172 // Goes through all renderers for the |source| and applies the proper | 173 // Goes through all renderers for the |source| and applies the proper |
173 // volume scaling for the source based on the volume(s) of the renderer(s). | 174 // volume scaling for the source based on the volume(s) of the renderer(s). |
174 void UpdateSourceVolume(webrtc::AudioSourceInterface* source); | 175 void UpdateSourceVolume(webrtc::AudioSourceInterface* source); |
175 | 176 |
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221 | 222 |
222 // Ref count for the MediaPlayers which have called Start() but not Stop(). | 223 // Ref count for the MediaPlayers which have called Start() but not Stop(). |
223 int start_ref_count_; | 224 int start_ref_count_; |
224 | 225 |
225 // Used to buffer data between the client and the output device in cases where | 226 // Used to buffer data between the client and the output device in cases where |
226 // the client buffer size is not the same as the output device buffer size. | 227 // the client buffer size is not the same as the output device buffer size. |
227 std::unique_ptr<media::AudioPullFifo> audio_fifo_; | 228 std::unique_ptr<media::AudioPullFifo> audio_fifo_; |
228 | 229 |
229 // Contains the accumulated delay estimate which is provided to the WebRTC | 230 // Contains the accumulated delay estimate which is provided to the WebRTC |
230 // AEC. | 231 // AEC. |
231 int audio_delay_milliseconds_; | 232 base::TimeDelta audio_delay_; |
232 | 233 |
233 base::TimeDelta current_time_; | 234 base::TimeDelta current_time_; |
234 | 235 |
235 // Saved volume and playing state of the root renderer. | 236 // Saved volume and playing state of the root renderer. |
236 PlayingState playing_state_; | 237 PlayingState playing_state_; |
237 | 238 |
238 // Audio params used by the sink of the renderer. | 239 // Audio params used by the sink of the renderer. |
239 media::AudioParameters sink_params_; | 240 media::AudioParameters sink_params_; |
240 | 241 |
241 // The preferred device id of the output device or empty for the default | 242 // The preferred device id of the output device or empty for the default |
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253 // Stores the maximum time spent waiting for render data from the source. Used | 254 // Stores the maximum time spent waiting for render data from the source. Used |
254 // for logging UMA data. Logged and reset when Stop() is called. | 255 // for logging UMA data. Logged and reset when Stop() is called. |
255 base::TimeDelta max_render_time_; | 256 base::TimeDelta max_render_time_; |
256 | 257 |
257 DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer); | 258 DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer); |
258 }; | 259 }; |
259 | 260 |
260 } // namespace content | 261 } // namespace content |
261 | 262 |
262 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_ | 263 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_ |
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