OLD | NEW |
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 #include "content/renderer/media/webrtc_audio_renderer.h" | 5 #include "content/renderer/media/webrtc_audio_renderer.h" |
6 | 6 |
7 #include "base/logging.h" | 7 #include "base/logging.h" |
8 #include "base/metrics/histogram.h" | 8 #include "base/metrics/histogram.h" |
9 #include "base/strings/string_util.h" | 9 #include "base/strings/string_util.h" |
10 #include "base/strings/stringprintf.h" | 10 #include "base/strings/stringprintf.h" |
11 #include "content/renderer/media/audio_device_factory.h" | 11 #include "content/renderer/media/audio_device_factory.h" |
| 12 #include "content/renderer/media/media_stream_dispatcher.h" |
12 #include "content/renderer/media/webrtc_audio_device_impl.h" | 13 #include "content/renderer/media/webrtc_audio_device_impl.h" |
13 #include "content/renderer/media/webrtc_logging.h" | 14 #include "content/renderer/media/webrtc_logging.h" |
| 15 #include "content/renderer/render_view_impl.h" |
14 #include "media/audio/audio_output_device.h" | 16 #include "media/audio/audio_output_device.h" |
15 #include "media/audio/audio_parameters.h" | 17 #include "media/audio/audio_parameters.h" |
16 #include "media/audio/sample_rates.h" | 18 #include "media/audio/sample_rates.h" |
17 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" | 19 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" |
18 #include "third_party/libjingle/source/talk/media/base/audiorenderer.h" | 20 #include "third_party/libjingle/source/talk/media/base/audiorenderer.h" |
19 | 21 |
20 | 22 |
21 #if defined(OS_WIN) | 23 #if defined(OS_WIN) |
22 #include "base/win/windows_version.h" | 24 #include "base/win/windows_version.h" |
23 #include "media/audio/win/core_audio_util_win.h" | 25 #include "media/audio/win/core_audio_util_win.h" |
(...skipping 150 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
174 | 176 |
175 private: | 177 private: |
176 base::ThreadChecker thread_checker_; | 178 base::ThreadChecker thread_checker_; |
177 const scoped_refptr<MediaStreamAudioRenderer> delegate_; | 179 const scoped_refptr<MediaStreamAudioRenderer> delegate_; |
178 const scoped_refptr<webrtc::MediaStreamInterface> media_stream_; | 180 const scoped_refptr<webrtc::MediaStreamInterface> media_stream_; |
179 bool started_; | 181 bool started_; |
180 WebRtcAudioRenderer::PlayingState playing_state_; | 182 WebRtcAudioRenderer::PlayingState playing_state_; |
181 OnPlayStateChanged on_play_state_changed_; | 183 OnPlayStateChanged on_play_state_changed_; |
182 }; | 184 }; |
183 | 185 |
| 186 // Returns either AudioParameters::NO_EFFECTS or AudioParameters::DUCKING |
| 187 // depending on whether or not an input element is currently open with |
| 188 // ducking enabled. |
| 189 int GetCurrentDuckingFlag(int render_view_id) { |
| 190 RenderViewImpl* render_view = RenderViewImpl::FromRoutingID(render_view_id); |
| 191 if (render_view && render_view->media_stream_dispatcher() && |
| 192 render_view->media_stream_dispatcher()->IsAudioDuckingActive()) { |
| 193 return media::AudioParameters::DUCKING; |
| 194 } |
| 195 |
| 196 return media::AudioParameters::NO_EFFECTS; |
| 197 } |
| 198 |
184 } // namespace | 199 } // namespace |
185 | 200 |
186 WebRtcAudioRenderer::WebRtcAudioRenderer( | 201 WebRtcAudioRenderer::WebRtcAudioRenderer( |
187 const scoped_refptr<webrtc::MediaStreamInterface>& media_stream, | 202 const scoped_refptr<webrtc::MediaStreamInterface>& media_stream, |
188 int source_render_view_id, | 203 int source_render_view_id, |
189 int source_render_frame_id, | 204 int source_render_frame_id, |
190 int session_id, | 205 int session_id, |
191 int sample_rate, | 206 int sample_rate, |
192 int frames_per_buffer) | 207 int frames_per_buffer) |
193 : state_(UNINITIALIZED), | 208 : state_(UNINITIALIZED), |
194 source_render_view_id_(source_render_view_id), | 209 source_render_view_id_(source_render_view_id), |
195 source_render_frame_id_(source_render_frame_id), | 210 source_render_frame_id_(source_render_frame_id), |
196 session_id_(session_id), | 211 session_id_(session_id), |
197 media_stream_(media_stream), | 212 media_stream_(media_stream), |
198 source_(NULL), | 213 source_(NULL), |
199 play_ref_count_(0), | 214 play_ref_count_(0), |
200 start_ref_count_(0), | 215 start_ref_count_(0), |
201 audio_delay_milliseconds_(0), | 216 audio_delay_milliseconds_(0), |
202 fifo_delay_milliseconds_(0), | 217 fifo_delay_milliseconds_(0), |
203 // TODO(tommi): Ducking is currently not set on sink_params due to an | |
204 // issue on Windows that causes the ducked state to be pinned if an output | |
205 // stream is opened before an input stream (both in communication mode). | |
206 // Furthermore the input stream may not be associated with the output | |
207 // stream, which results in the output stream getting incorrectly ducked. | |
208 // What should happen here is that the ducking flag should be raised | |
209 // iff an input device is currently open with ducking set. | |
210 // Bugs: crbug/391414, crbug/391247. | |
211 sink_params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | 218 sink_params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
212 media::CHANNEL_LAYOUT_STEREO, 0, sample_rate, 16, | 219 media::CHANNEL_LAYOUT_STEREO, 0, sample_rate, 16, |
213 frames_per_buffer, media::AudioParameters::NO_EFFECTS) { | 220 frames_per_buffer, |
| 221 GetCurrentDuckingFlag(source_render_view_id)) { |
214 WebRtcLogMessage(base::StringPrintf( | 222 WebRtcLogMessage(base::StringPrintf( |
215 "WAR::WAR. source_render_view_id=%d" | 223 "WAR::WAR. source_render_view_id=%d" |
216 ", session_id=%d, sample_rate=%d, frames_per_buffer=%d", | 224 ", session_id=%d, sample_rate=%d, frames_per_buffer=%d, effects=%i", |
217 source_render_view_id, | 225 source_render_view_id, |
218 session_id, | 226 session_id, |
219 sample_rate, | 227 sample_rate, |
220 frames_per_buffer)); | 228 frames_per_buffer, |
| 229 sink_params_.effects())); |
221 } | 230 } |
222 | 231 |
223 WebRtcAudioRenderer::~WebRtcAudioRenderer() { | 232 WebRtcAudioRenderer::~WebRtcAudioRenderer() { |
224 DCHECK(thread_checker_.CalledOnValidThread()); | 233 DCHECK(thread_checker_.CalledOnValidThread()); |
225 DCHECK_EQ(state_, UNINITIALIZED); | 234 DCHECK_EQ(state_, UNINITIALIZED); |
226 } | 235 } |
227 | 236 |
228 bool WebRtcAudioRenderer::Initialize(WebRtcAudioRendererSource* source) { | 237 bool WebRtcAudioRenderer::Initialize(WebRtcAudioRendererSource* source) { |
229 DVLOG(1) << "WebRtcAudioRenderer::Initialize()"; | 238 DVLOG(1) << "WebRtcAudioRenderer::Initialize()"; |
230 DCHECK(thread_checker_.CalledOnValidThread()); | 239 DCHECK(thread_checker_.CalledOnValidThread()); |
(...skipping 361 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
592 if (RemovePlayingState(source, state)) | 601 if (RemovePlayingState(source, state)) |
593 EnterPauseState(); | 602 EnterPauseState(); |
594 } else if (AddPlayingState(source, state)) { | 603 } else if (AddPlayingState(source, state)) { |
595 EnterPlayState(); | 604 EnterPlayState(); |
596 } | 605 } |
597 UpdateSourceVolume(source); | 606 UpdateSourceVolume(source); |
598 } | 607 } |
599 } | 608 } |
600 | 609 |
601 } // namespace content | 610 } // namespace content |
OLD | NEW |