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_local_audio_renderer.h" | 5 #include "content/renderer/media/track_audio_renderer.h" |
6 | |
7 #include <utility> | |
8 | 6 |
9 #include "base/location.h" | 7 #include "base/location.h" |
10 #include "base/logging.h" | 8 #include "base/logging.h" |
11 #include "base/metrics/histogram.h" | 9 #include "base/metrics/histogram.h" |
12 #include "base/synchronization/lock.h" | 10 #include "base/synchronization/lock.h" |
13 #include "base/thread_task_runner_handle.h" | 11 #include "base/thread_task_runner_handle.h" |
14 #include "base/trace_event/trace_event.h" | 12 #include "base/trace_event/trace_event.h" |
15 #include "content/renderer/media/audio_device_factory.h" | 13 #include "content/renderer/media/audio_device_factory.h" |
16 #include "content/renderer/media/media_stream_dispatcher.h" | 14 #include "content/renderer/media/media_stream_audio_track.h" |
17 #include "content/renderer/media/webrtc_audio_capturer.h" | |
18 #include "content/renderer/media/webrtc_audio_renderer.h" | |
19 #include "content/renderer/render_frame_impl.h" | |
20 #include "media/audio/audio_output_device.h" | 15 #include "media/audio/audio_output_device.h" |
21 #include "media/base/audio_bus.h" | 16 #include "media/base/audio_bus.h" |
22 #include "media/base/audio_shifter.h" | 17 #include "media/base/audio_shifter.h" |
23 | 18 |
24 namespace content { | 19 namespace content { |
25 | 20 |
26 namespace { | 21 namespace { |
27 | 22 |
28 enum LocalRendererSinkStates { | 23 enum LocalRendererSinkStates { |
29 kSinkStarted = 0, | 24 kSinkStarted = 0, |
30 kSinkNeverStarted, | 25 kSinkNeverStarted, |
31 kSinkStatesMax // Must always be last! | 26 kSinkStatesMax // Must always be last! |
32 }; | 27 }; |
33 | 28 |
| 29 // Translates |num_samples_rendered| into a TimeDelta duration and adds it to |
| 30 // |prior_elapsed_render_time|. |
| 31 base::TimeDelta ComputeTotalElapsedRenderTime( |
| 32 base::TimeDelta prior_elapsed_render_time, |
| 33 int64_t num_samples_rendered, |
| 34 int sample_rate) { |
| 35 return prior_elapsed_render_time + base::TimeDelta::FromMicroseconds( |
| 36 num_samples_rendered * base::Time::kMicrosecondsPerSecond / sample_rate); |
| 37 } |
| 38 |
34 } // namespace | 39 } // namespace |
35 | 40 |
36 // media::AudioRendererSink::RenderCallback implementation | 41 // media::AudioRendererSink::RenderCallback implementation |
37 int WebRtcLocalAudioRenderer::Render(media::AudioBus* audio_bus, | 42 int TrackAudioRenderer::Render(media::AudioBus* audio_bus, |
38 uint32_t audio_delay_milliseconds, | 43 uint32_t audio_delay_milliseconds, |
39 uint32_t frames_skipped) { | 44 uint32_t frames_skipped) { |
40 TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::Render"); | 45 TRACE_EVENT0("audio", "TrackAudioRenderer::Render"); |
41 base::AutoLock auto_lock(thread_lock_); | 46 base::AutoLock auto_lock(thread_lock_); |
42 | 47 |
43 if (!playing_ || !volume_ || !audio_shifter_) { | 48 if (!audio_shifter_) { |
44 audio_bus->Zero(); | 49 audio_bus->Zero(); |
45 return 0; | 50 return 0; |
46 } | 51 } |
47 | 52 |
48 audio_shifter_->Pull( | 53 // TODO(miu): Plumbing is needed to determine the actual playout timestamp |
49 audio_bus, | 54 // of the audio, instead of just snapshotting TimeTicks::Now(), for proper |
50 base::TimeTicks::Now() - | 55 // audio/video sync. http://crbug.com/335335 |
51 base::TimeDelta::FromMilliseconds(audio_delay_milliseconds)); | 56 const base::TimeTicks playout_time = |
52 | 57 base::TimeTicks::Now() + |
| 58 base::TimeDelta::FromMilliseconds(audio_delay_milliseconds); |
| 59 DVLOG(2) << "Pulling audio out of shifter to be played " |
| 60 << audio_delay_milliseconds << " ms from now."; |
| 61 audio_shifter_->Pull(audio_bus, playout_time); |
| 62 num_samples_rendered_ += audio_bus->frames(); |
53 return audio_bus->frames(); | 63 return audio_bus->frames(); |
54 } | 64 } |
55 | 65 |
56 void WebRtcLocalAudioRenderer::OnRenderError() { | 66 void TrackAudioRenderer::OnRenderError() { |
57 NOTIMPLEMENTED(); | 67 NOTIMPLEMENTED(); |
58 } | 68 } |
59 | 69 |
60 // content::MediaStreamAudioSink implementation | 70 // content::MediaStreamAudioSink implementation |
61 void WebRtcLocalAudioRenderer::OnData(const media::AudioBus& audio_bus, | 71 void TrackAudioRenderer::OnData(const media::AudioBus& audio_bus, |
62 base::TimeTicks estimated_capture_time) { | 72 base::TimeTicks reference_time) { |
63 DCHECK(capture_thread_checker_.CalledOnValidThread()); | 73 DCHECK(audio_thread_checker_.CalledOnValidThread()); |
64 DCHECK(!estimated_capture_time.is_null()); | 74 DCHECK(!reference_time.is_null()); |
65 | 75 |
66 TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::CaptureData"); | 76 TRACE_EVENT0("audio", "TrackAudioRenderer::CaptureData"); |
67 | 77 |
68 base::AutoLock auto_lock(thread_lock_); | 78 base::AutoLock auto_lock(thread_lock_); |
69 if (!playing_ || !volume_ || !audio_shifter_) | 79 if (!audio_shifter_) |
70 return; | 80 return; |
71 | 81 |
72 scoped_ptr<media::AudioBus> audio_data( | 82 scoped_ptr<media::AudioBus> audio_data( |
73 media::AudioBus::Create(audio_bus.channels(), audio_bus.frames())); | 83 media::AudioBus::Create(audio_bus.channels(), audio_bus.frames())); |
74 audio_bus.CopyTo(audio_data.get()); | 84 audio_bus.CopyTo(audio_data.get()); |
75 audio_shifter_->Push(std::move(audio_data), estimated_capture_time); | 85 // Note: For remote audio sources, |reference_time| is the local playout time, |
76 const base::TimeTicks now = base::TimeTicks::Now(); | 86 // the ideal point-in-time at which the first audio sample should be played |
77 total_render_time_ += now - last_render_time_; | 87 // out in the future. For local sources, |reference_time| is the |
78 last_render_time_ = now; | 88 // point-in-time at which the first audio sample was captured in the past. In |
| 89 // either case, AudioShifter will auto-detect and do the right thing when |
| 90 // audio is pulled from it. |
| 91 audio_shifter_->Push(std::move(audio_data), reference_time); |
79 } | 92 } |
80 | 93 |
81 void WebRtcLocalAudioRenderer::OnSetFormat( | 94 void TrackAudioRenderer::OnSetFormat(const media::AudioParameters& params) { |
82 const media::AudioParameters& params) { | 95 DVLOG(1) << "TrackAudioRenderer::OnSetFormat()"; |
83 DVLOG(1) << "WebRtcLocalAudioRenderer::OnSetFormat()"; | |
84 // If the source is restarted, we might have changed to another capture | 96 // If the source is restarted, we might have changed to another capture |
85 // thread. | 97 // thread. |
86 capture_thread_checker_.DetachFromThread(); | 98 audio_thread_checker_.DetachFromThread(); |
87 DCHECK(capture_thread_checker_.CalledOnValidThread()); | 99 DCHECK(audio_thread_checker_.CalledOnValidThread()); |
| 100 |
| 101 // If the parameters changed, the audio in the AudioShifter is invalid and |
| 102 // should be dropped. |
| 103 { |
| 104 base::AutoLock auto_lock(thread_lock_); |
| 105 if (audio_shifter_ && |
| 106 (audio_shifter_->sample_rate() != params.sample_rate() || |
| 107 audio_shifter_->channels() != params.channels())) { |
| 108 HaltAudioFlowWhileLockHeld(); |
| 109 } |
| 110 } |
88 | 111 |
89 // Post a task on the main render thread to reconfigure the |sink_| with the | 112 // Post a task on the main render thread to reconfigure the |sink_| with the |
90 // new format. | 113 // new format. |
91 task_runner_->PostTask( | 114 task_runner_->PostTask( |
92 FROM_HERE, | 115 FROM_HERE, |
93 base::Bind(&WebRtcLocalAudioRenderer::ReconfigureSink, this, params)); | 116 base::Bind(&TrackAudioRenderer::ReconfigureSink, this, params)); |
94 } | 117 } |
95 | 118 |
96 // WebRtcLocalAudioRenderer::WebRtcLocalAudioRenderer implementation. | 119 TrackAudioRenderer::TrackAudioRenderer( |
97 WebRtcLocalAudioRenderer::WebRtcLocalAudioRenderer( | |
98 const blink::WebMediaStreamTrack& audio_track, | 120 const blink::WebMediaStreamTrack& audio_track, |
99 int source_render_frame_id, | 121 int playout_render_frame_id, |
100 int session_id, | 122 int session_id, |
101 const std::string& device_id, | 123 const std::string& device_id, |
102 const url::Origin& security_origin) | 124 const url::Origin& security_origin) |
103 : audio_track_(audio_track), | 125 : audio_track_(audio_track), |
104 source_render_frame_id_(source_render_frame_id), | 126 playout_render_frame_id_(playout_render_frame_id), |
105 session_id_(session_id), | 127 session_id_(session_id), |
106 task_runner_(base::ThreadTaskRunnerHandle::Get()), | 128 task_runner_(base::ThreadTaskRunnerHandle::Get()), |
| 129 num_samples_rendered_(0), |
107 playing_(false), | 130 playing_(false), |
108 output_device_id_(device_id), | 131 output_device_id_(device_id), |
109 security_origin_(security_origin), | 132 security_origin_(security_origin), |
110 volume_(0.0), | 133 volume_(0.0), |
111 sink_started_(false) { | 134 sink_started_(false) { |
112 DVLOG(1) << "WebRtcLocalAudioRenderer::WebRtcLocalAudioRenderer()"; | 135 DCHECK(MediaStreamAudioTrack::GetTrack(audio_track_)); |
| 136 DVLOG(1) << "TrackAudioRenderer::TrackAudioRenderer()"; |
113 } | 137 } |
114 | 138 |
115 WebRtcLocalAudioRenderer::~WebRtcLocalAudioRenderer() { | 139 TrackAudioRenderer::~TrackAudioRenderer() { |
116 DCHECK(task_runner_->BelongsToCurrentThread()); | 140 DCHECK(task_runner_->BelongsToCurrentThread()); |
117 DCHECK(!sink_.get()); | 141 DCHECK(!sink_.get()); |
118 DVLOG(1) << "WebRtcLocalAudioRenderer::~WebRtcLocalAudioRenderer()"; | 142 DVLOG(1) << "TrackAudioRenderer::~TrackAudioRenderer()"; |
119 } | 143 } |
120 | 144 |
121 void WebRtcLocalAudioRenderer::Start() { | 145 void TrackAudioRenderer::Start() { |
122 DVLOG(1) << "WebRtcLocalAudioRenderer::Start()"; | 146 DVLOG(1) << "TrackAudioRenderer::Start()"; |
123 DCHECK(task_runner_->BelongsToCurrentThread()); | 147 DCHECK(task_runner_->BelongsToCurrentThread()); |
| 148 DCHECK_EQ(playing_, false); |
124 | 149 |
125 // We get audio data from |audio_track_|... | 150 // We get audio data from |audio_track_|... |
126 MediaStreamAudioSink::AddToAudioTrack(this, audio_track_); | 151 MediaStreamAudioSink::AddToAudioTrack(this, audio_track_); |
127 // ...and |sink_| will get audio data from us. | 152 // ...and |sink_| will get audio data from us. |
128 DCHECK(!sink_.get()); | 153 DCHECK(!sink_.get()); |
129 sink_ = | 154 sink_ = |
130 AudioDeviceFactory::NewOutputDevice(source_render_frame_id_, session_id_, | 155 AudioDeviceFactory::NewOutputDevice(playout_render_frame_id_, session_id_, |
131 output_device_id_, security_origin_); | 156 output_device_id_, security_origin_); |
132 | 157 |
133 base::AutoLock auto_lock(thread_lock_); | 158 base::AutoLock auto_lock(thread_lock_); |
134 last_render_time_ = base::TimeTicks::Now(); | 159 prior_elapsed_render_time_ = base::TimeDelta(); |
135 playing_ = false; | 160 num_samples_rendered_ = 0; |
136 } | 161 } |
137 | 162 |
138 void WebRtcLocalAudioRenderer::Stop() { | 163 void TrackAudioRenderer::Stop() { |
139 DVLOG(1) << "WebRtcLocalAudioRenderer::Stop()"; | 164 DVLOG(1) << "TrackAudioRenderer::Stop()"; |
140 DCHECK(task_runner_->BelongsToCurrentThread()); | 165 DCHECK(task_runner_->BelongsToCurrentThread()); |
141 | 166 |
142 { | 167 Pause(); |
143 base::AutoLock auto_lock(thread_lock_); | |
144 playing_ = false; | |
145 audio_shifter_.reset(); | |
146 } | |
147 | 168 |
148 // Stop the output audio stream, i.e, stop asking for data to render. | 169 // Stop the output audio stream, i.e, stop asking for data to render. |
149 // It is safer to call Stop() on the |sink_| to clean up the resources even | 170 // It is safer to call Stop() on the |sink_| to clean up the resources even |
150 // when the |sink_| is never started. | 171 // when the |sink_| is never started. |
151 if (sink_.get()) { | 172 if (sink_.get()) { |
152 sink_->Stop(); | 173 sink_->Stop(); |
153 sink_ = NULL; | 174 sink_ = NULL; |
154 } | 175 } |
155 | 176 |
156 if (!sink_started_) { | 177 if (!sink_started_ && IsLocalRenderer()) { |
157 UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates", | 178 UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates", |
158 kSinkNeverStarted, kSinkStatesMax); | 179 kSinkNeverStarted, kSinkStatesMax); |
159 } | 180 } |
160 sink_started_ = false; | 181 sink_started_ = false; |
161 | 182 |
162 // Ensure that the capturer stops feeding us with captured audio. | 183 // Ensure that the capturer stops feeding us with captured audio. |
163 MediaStreamAudioSink::RemoveFromAudioTrack(this, audio_track_); | 184 MediaStreamAudioSink::RemoveFromAudioTrack(this, audio_track_); |
164 } | 185 } |
165 | 186 |
166 void WebRtcLocalAudioRenderer::Play() { | 187 void TrackAudioRenderer::Play() { |
167 DVLOG(1) << "WebRtcLocalAudioRenderer::Play()"; | 188 DVLOG(1) << "TrackAudioRenderer::Play()"; |
168 DCHECK(task_runner_->BelongsToCurrentThread()); | 189 DCHECK(task_runner_->BelongsToCurrentThread()); |
169 | 190 |
170 if (!sink_.get()) | 191 if (!sink_.get()) |
171 return; | 192 return; |
172 | 193 |
173 { | 194 playing_ = true; |
174 base::AutoLock auto_lock(thread_lock_); | |
175 // Resumes rendering by ensuring that WebRtcLocalAudioRenderer::Render() | |
176 // now reads data from the local FIFO. | |
177 playing_ = true; | |
178 last_render_time_ = base::TimeTicks::Now(); | |
179 } | |
180 | 195 |
181 // Note: If volume_ is currently muted, the |sink_| will not be started yet. | |
182 MaybeStartSink(); | 196 MaybeStartSink(); |
183 } | 197 } |
184 | 198 |
185 void WebRtcLocalAudioRenderer::Pause() { | 199 void TrackAudioRenderer::Pause() { |
186 DVLOG(1) << "WebRtcLocalAudioRenderer::Pause()"; | 200 DVLOG(1) << "TrackAudioRenderer::Pause()"; |
187 DCHECK(task_runner_->BelongsToCurrentThread()); | 201 DCHECK(task_runner_->BelongsToCurrentThread()); |
188 | 202 |
189 if (!sink_.get()) | 203 if (!sink_.get()) |
190 return; | 204 return; |
191 | 205 |
| 206 playing_ = false; |
| 207 |
192 base::AutoLock auto_lock(thread_lock_); | 208 base::AutoLock auto_lock(thread_lock_); |
193 // Temporarily suspends rendering audio. | 209 HaltAudioFlowWhileLockHeld(); |
194 // WebRtcLocalAudioRenderer::Render() will return early during this state | |
195 // and only zeros will be provided to the active sink. | |
196 playing_ = false; | |
197 } | 210 } |
198 | 211 |
199 void WebRtcLocalAudioRenderer::SetVolume(float volume) { | 212 void TrackAudioRenderer::SetVolume(float volume) { |
200 DVLOG(1) << "WebRtcLocalAudioRenderer::SetVolume(" << volume << ")"; | 213 DVLOG(1) << "TrackAudioRenderer::SetVolume(" << volume << ")"; |
| 214 DCHECK(task_runner_->BelongsToCurrentThread()); |
| 215 |
| 216 // Cache the volume. Whenever |sink_| is re-created, call SetVolume() with |
| 217 // this cached volume. |
| 218 volume_ = volume; |
| 219 if (sink_.get()) |
| 220 sink_->SetVolume(volume); |
| 221 } |
| 222 |
| 223 media::OutputDevice* TrackAudioRenderer::GetOutputDevice() { |
| 224 DCHECK(task_runner_->BelongsToCurrentThread()); |
| 225 return this; |
| 226 } |
| 227 |
| 228 base::TimeDelta TrackAudioRenderer::GetCurrentRenderTime() const { |
| 229 DCHECK(task_runner_->BelongsToCurrentThread()); |
| 230 base::AutoLock auto_lock(thread_lock_); |
| 231 if (source_params_.IsValid()) { |
| 232 return ComputeTotalElapsedRenderTime(prior_elapsed_render_time_, |
| 233 num_samples_rendered_, |
| 234 source_params_.sample_rate()); |
| 235 } |
| 236 return prior_elapsed_render_time_; |
| 237 } |
| 238 |
| 239 bool TrackAudioRenderer::IsLocalRenderer() const { |
| 240 DCHECK(task_runner_->BelongsToCurrentThread()); |
| 241 return MediaStreamAudioTrack::GetTrack(audio_track_)->is_local_track(); |
| 242 } |
| 243 |
| 244 void TrackAudioRenderer::SwitchOutputDevice( |
| 245 const std::string& device_id, |
| 246 const url::Origin& security_origin, |
| 247 const media::SwitchOutputDeviceCB& callback) { |
| 248 DVLOG(1) << "TrackAudioRenderer::SwitchOutputDevice()"; |
201 DCHECK(task_runner_->BelongsToCurrentThread()); | 249 DCHECK(task_runner_->BelongsToCurrentThread()); |
202 | 250 |
203 { | 251 { |
204 base::AutoLock auto_lock(thread_lock_); | 252 base::AutoLock auto_lock(thread_lock_); |
205 // Cache the volume. | 253 HaltAudioFlowWhileLockHeld(); |
206 volume_ = volume; | |
207 } | 254 } |
208 | 255 |
209 // Lazily start the |sink_| when the local renderer is unmuted during | |
210 // playing. | |
211 MaybeStartSink(); | |
212 | |
213 if (sink_.get()) | |
214 sink_->SetVolume(volume); | |
215 } | |
216 | |
217 media::OutputDevice* WebRtcLocalAudioRenderer::GetOutputDevice() { | |
218 DCHECK(task_runner_->BelongsToCurrentThread()); | |
219 return this; | |
220 } | |
221 | |
222 base::TimeDelta WebRtcLocalAudioRenderer::GetCurrentRenderTime() const { | |
223 DCHECK(task_runner_->BelongsToCurrentThread()); | |
224 base::AutoLock auto_lock(thread_lock_); | |
225 if (!sink_.get()) | |
226 return base::TimeDelta(); | |
227 return total_render_time(); | |
228 } | |
229 | |
230 bool WebRtcLocalAudioRenderer::IsLocalRenderer() const { | |
231 return true; | |
232 } | |
233 | |
234 void WebRtcLocalAudioRenderer::SwitchOutputDevice( | |
235 const std::string& device_id, | |
236 const url::Origin& security_origin, | |
237 const media::SwitchOutputDeviceCB& callback) { | |
238 DVLOG(1) << "WebRtcLocalAudioRenderer::SwitchOutputDevice()"; | |
239 DCHECK(task_runner_->BelongsToCurrentThread()); | |
240 | |
241 scoped_refptr<media::AudioOutputDevice> new_sink = | 256 scoped_refptr<media::AudioOutputDevice> new_sink = |
242 AudioDeviceFactory::NewOutputDevice(source_render_frame_id_, session_id_, | 257 AudioDeviceFactory::NewOutputDevice(playout_render_frame_id_, session_id_, |
243 device_id, security_origin); | 258 device_id, security_origin); |
244 if (new_sink->GetDeviceStatus() != media::OUTPUT_DEVICE_STATUS_OK) { | 259 if (new_sink->GetDeviceStatus() != media::OUTPUT_DEVICE_STATUS_OK) { |
245 callback.Run(new_sink->GetDeviceStatus()); | 260 callback.Run(new_sink->GetDeviceStatus()); |
246 return; | 261 return; |
247 } | 262 } |
248 | 263 |
249 output_device_id_ = device_id; | 264 output_device_id_ = device_id; |
250 security_origin_ = security_origin; | 265 security_origin_ = security_origin; |
251 bool was_sink_started = sink_started_; | 266 bool was_sink_started = sink_started_; |
252 | 267 |
253 if (sink_.get()) | 268 if (sink_.get()) |
254 sink_->Stop(); | 269 sink_->Stop(); |
255 | 270 |
256 sink_started_ = false; | 271 sink_started_ = false; |
257 sink_ = new_sink; | 272 sink_ = new_sink; |
258 int frames_per_buffer = sink_->GetOutputParameters().frames_per_buffer(); | |
259 sink_params_ = source_params_; | |
260 sink_params_.set_frames_per_buffer(WebRtcAudioRenderer::GetOptimalBufferSize( | |
261 source_params_.sample_rate(), frames_per_buffer)); | |
262 | |
263 if (was_sink_started) | 273 if (was_sink_started) |
264 MaybeStartSink(); | 274 MaybeStartSink(); |
265 | 275 |
266 callback.Run(media::OUTPUT_DEVICE_STATUS_OK); | 276 callback.Run(media::OUTPUT_DEVICE_STATUS_OK); |
267 } | 277 } |
268 | 278 |
269 media::AudioParameters WebRtcLocalAudioRenderer::GetOutputParameters() { | 279 media::AudioParameters TrackAudioRenderer::GetOutputParameters() { |
270 DCHECK(task_runner_->BelongsToCurrentThread()); | 280 DCHECK(task_runner_->BelongsToCurrentThread()); |
271 if (!sink_.get()) | 281 if (!sink_ || !source_params_.IsValid()) |
272 return media::AudioParameters(); | 282 return media::AudioParameters(); |
273 | 283 |
274 return sink_->GetOutputParameters(); | 284 // Output parameters consist of the same channel layout and sample rate as the |
| 285 // source, but having the buffer duration preferred by the hardware. |
| 286 const media::AudioParameters& preferred_params = sink_->GetOutputParameters(); |
| 287 return media::AudioParameters( |
| 288 preferred_params.format(), source_params_.channel_layout(), |
| 289 source_params_.sample_rate(), source_params_.bits_per_sample(), |
| 290 preferred_params.frames_per_buffer() * source_params_.sample_rate() / |
| 291 preferred_params.sample_rate()); |
275 } | 292 } |
276 | 293 |
277 media::OutputDeviceStatus WebRtcLocalAudioRenderer::GetDeviceStatus() { | 294 media::OutputDeviceStatus TrackAudioRenderer::GetDeviceStatus() { |
278 DCHECK(task_runner_->BelongsToCurrentThread()); | 295 DCHECK(task_runner_->BelongsToCurrentThread()); |
279 if (!sink_.get()) | 296 if (!sink_.get()) |
280 return media::OUTPUT_DEVICE_STATUS_ERROR_INTERNAL; | 297 return media::OUTPUT_DEVICE_STATUS_ERROR_INTERNAL; |
281 | 298 |
282 return sink_->GetDeviceStatus(); | 299 return sink_->GetDeviceStatus(); |
283 } | 300 } |
284 | 301 |
285 void WebRtcLocalAudioRenderer::MaybeStartSink() { | 302 void TrackAudioRenderer::MaybeStartSink() { |
286 DCHECK(task_runner_->BelongsToCurrentThread()); | 303 DCHECK(task_runner_->BelongsToCurrentThread()); |
287 DVLOG(1) << "WebRtcLocalAudioRenderer::MaybeStartSink()"; | 304 DVLOG(1) << "TrackAudioRenderer::MaybeStartSink()"; |
288 | 305 |
289 if (!sink_.get() || !source_params_.IsValid()) | 306 if (!sink_.get() || !source_params_.IsValid() || !playing_) |
290 return; | 307 return; |
291 | 308 |
292 { | 309 // Re-create the AudioShifter to drop old audio data and reset to a starting |
293 // Clear up the old data in the FIFO. | 310 // state. MaybeStartSink() is always called in a situation where either the |
294 base::AutoLock auto_lock(thread_lock_); | 311 // source or sink has changed somehow and so all of AudioShifter's internal |
295 audio_shifter_->Flush(); | 312 // time-sync state is invalid. |
| 313 CreateAudioShifter(); |
| 314 |
| 315 if (sink_started_ || |
| 316 sink_->GetDeviceStatus() != media::OUTPUT_DEVICE_STATUS_OK) { |
| 317 return; |
296 } | 318 } |
297 | 319 |
298 if (!sink_params_.IsValid() || !playing_ || !volume_ || sink_started_ || | 320 DVLOG(1) << ("TrackAudioRenderer::MaybeStartSink() -- Starting sink. " |
299 sink_->GetDeviceStatus() != media::OUTPUT_DEVICE_STATUS_OK) | 321 "source_params_={") |
300 return; | 322 << source_params_.AsHumanReadableString() << "}, sink parameters={" |
301 | 323 << GetOutputParameters().AsHumanReadableString() << '}'; |
302 DVLOG(1) << "WebRtcLocalAudioRenderer::MaybeStartSink() -- Starting sink_."; | 324 sink_->Initialize(GetOutputParameters(), this); |
303 sink_->Initialize(sink_params_, this); | |
304 sink_->Start(); | 325 sink_->Start(); |
| 326 sink_->SetVolume(volume_); |
305 sink_started_ = true; | 327 sink_started_ = true; |
306 UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates", | 328 if (IsLocalRenderer()) { |
307 kSinkStarted, kSinkStatesMax); | 329 UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates", kSinkStarted, |
| 330 kSinkStatesMax); |
| 331 } |
308 } | 332 } |
309 | 333 |
310 void WebRtcLocalAudioRenderer::ReconfigureSink( | 334 void TrackAudioRenderer::ReconfigureSink(const media::AudioParameters& params) { |
311 const media::AudioParameters& params) { | |
312 DCHECK(task_runner_->BelongsToCurrentThread()); | 335 DCHECK(task_runner_->BelongsToCurrentThread()); |
313 | 336 |
314 DVLOG(1) << "WebRtcLocalAudioRenderer::ReconfigureSink()"; | 337 DVLOG(1) << "TrackAudioRenderer::ReconfigureSink()"; |
315 | 338 |
316 if (source_params_.Equals(params)) | 339 if (source_params_.Equals(params)) |
317 return; | 340 return; |
318 | |
319 // Reset the |source_params_|, |sink_params_| and |loopback_fifo_| to match | |
320 // the new format. | |
321 | |
322 source_params_ = params; | 341 source_params_ = params; |
323 { | |
324 // Note: The max buffer is fairly large, but will rarely be used. | |
325 // Cast needs the buffer to hold at least one second of audio. | |
326 // The clock accuracy is set to 20ms because clock accuracy is | |
327 // ~15ms on windows. | |
328 media::AudioShifter* const new_shifter = new media::AudioShifter( | |
329 base::TimeDelta::FromSeconds(2), | |
330 base::TimeDelta::FromMilliseconds(20), | |
331 base::TimeDelta::FromSeconds(20), | |
332 source_params_.sample_rate(), | |
333 params.channels()); | |
334 | |
335 base::AutoLock auto_lock(thread_lock_); | |
336 audio_shifter_.reset(new_shifter); | |
337 } | |
338 | 342 |
339 if (!sink_.get()) | 343 if (!sink_.get()) |
340 return; // WebRtcLocalAudioRenderer has not yet been started. | 344 return; // TrackAudioRenderer has not yet been started. |
341 | 345 |
342 // Stop |sink_| and re-create a new one to be initialized with different audio | 346 // Stop |sink_| and re-create a new one to be initialized with different audio |
343 // parameters. Then, invoke MaybeStartSink() to restart everything again. | 347 // parameters. Then, invoke MaybeStartSink() to restart everything again. |
344 sink_->Stop(); | 348 sink_->Stop(); |
345 sink_started_ = false; | 349 sink_started_ = false; |
346 sink_ = | 350 sink_ = |
347 AudioDeviceFactory::NewOutputDevice(source_render_frame_id_, session_id_, | 351 AudioDeviceFactory::NewOutputDevice(playout_render_frame_id_, session_id_, |
348 output_device_id_, security_origin_); | 352 output_device_id_, security_origin_); |
349 int frames_per_buffer = sink_->GetOutputParameters().frames_per_buffer(); | |
350 sink_params_ = source_params_; | |
351 sink_params_.set_frames_per_buffer(WebRtcAudioRenderer::GetOptimalBufferSize( | |
352 source_params_.sample_rate(), frames_per_buffer)); | |
353 MaybeStartSink(); | 353 MaybeStartSink(); |
354 } | 354 } |
355 | 355 |
| 356 void TrackAudioRenderer::CreateAudioShifter() { |
| 357 DCHECK(task_runner_->BelongsToCurrentThread()); |
| 358 |
| 359 // Note 1: The max buffer is fairly large to cover the case where |
| 360 // remotely-sourced audio is delivered well ahead of its scheduled playout |
| 361 // time (e.g., content streaming with a very large end-to-end |
| 362 // latency). However, there is no penalty for making it large in the |
| 363 // low-latency use cases since AudioShifter will discard data as soon as it is |
| 364 // no longer needed. |
| 365 // |
| 366 // Note 2: The clock accuracy is set to 20ms because clock accuracy is |
| 367 // ~15ms on Windows machines without a working high-resolution clock. See |
| 368 // comments in base/time/time.h for details. |
| 369 media::AudioShifter* const new_shifter = new media::AudioShifter( |
| 370 base::TimeDelta::FromSeconds(5), base::TimeDelta::FromMilliseconds(20), |
| 371 base::TimeDelta::FromSeconds(20), source_params_.sample_rate(), |
| 372 source_params_.channels()); |
| 373 |
| 374 base::AutoLock auto_lock(thread_lock_); |
| 375 audio_shifter_.reset(new_shifter); |
| 376 } |
| 377 |
| 378 void TrackAudioRenderer::HaltAudioFlowWhileLockHeld() { |
| 379 thread_lock_.AssertAcquired(); |
| 380 |
| 381 audio_shifter_.reset(); |
| 382 |
| 383 if (source_params_.IsValid()) { |
| 384 prior_elapsed_render_time_ = |
| 385 ComputeTotalElapsedRenderTime(prior_elapsed_render_time_, |
| 386 num_samples_rendered_, |
| 387 source_params_.sample_rate()); |
| 388 num_samples_rendered_ = 0; |
| 389 } |
| 390 } |
| 391 |
356 } // namespace content | 392 } // namespace content |
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