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| 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 | |
| 3 // found in the LICENSE file. | |
| 4 | |
| 5 #include "content/renderer/media/webrtc_local_audio_renderer.h" | |
| 6 | |
| 7 #include <utility> | |
| 8 | |
| 9 #include "base/location.h" | |
| 10 #include "base/logging.h" | |
| 11 #include "base/metrics/histogram.h" | |
| 12 #include "base/synchronization/lock.h" | |
| 13 #include "base/thread_task_runner_handle.h" | |
| 14 #include "base/trace_event/trace_event.h" | |
| 15 #include "content/renderer/media/audio_device_factory.h" | |
| 16 #include "content/renderer/media/media_stream_dispatcher.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" | |
| 21 #include "media/base/audio_bus.h" | |
| 22 #include "media/base/audio_shifter.h" | |
| 23 | |
| 24 namespace content { | |
| 25 | |
| 26 namespace { | |
| 27 | |
| 28 enum LocalRendererSinkStates { | |
| 29 kSinkStarted = 0, | |
| 30 kSinkNeverStarted, | |
| 31 kSinkStatesMax // Must always be last! | |
| 32 }; | |
| 33 | |
| 34 } // namespace | |
| 35 | |
| 36 // media::AudioRendererSink::RenderCallback implementation | |
| 37 int WebRtcLocalAudioRenderer::Render(media::AudioBus* audio_bus, | |
| 38 uint32_t audio_delay_milliseconds, | |
| 39 uint32_t frames_skipped) { | |
| 40 TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::Render"); | |
| 41 base::AutoLock auto_lock(thread_lock_); | |
| 42 | |
| 43 if (!playing_ || !volume_ || !audio_shifter_) { | |
| 44 audio_bus->Zero(); | |
| 45 return 0; | |
| 46 } | |
| 47 | |
| 48 audio_shifter_->Pull( | |
| 49 audio_bus, | |
| 50 base::TimeTicks::Now() - | |
| 51 base::TimeDelta::FromMilliseconds(audio_delay_milliseconds)); | |
| 52 | |
| 53 return audio_bus->frames(); | |
| 54 } | |
| 55 | |
| 56 void WebRtcLocalAudioRenderer::OnRenderError() { | |
| 57 NOTIMPLEMENTED(); | |
| 58 } | |
| 59 | |
| 60 // content::MediaStreamAudioSink implementation | |
| 61 void WebRtcLocalAudioRenderer::OnData(const media::AudioBus& audio_bus, | |
| 62 base::TimeTicks estimated_capture_time) { | |
| 63 DCHECK(capture_thread_checker_.CalledOnValidThread()); | |
| 64 DCHECK(!estimated_capture_time.is_null()); | |
| 65 | |
| 66 TRACE_EVENT0("audio", "WebRtcLocalAudioRenderer::CaptureData"); | |
| 67 | |
| 68 base::AutoLock auto_lock(thread_lock_); | |
| 69 if (!playing_ || !volume_ || !audio_shifter_) | |
| 70 return; | |
| 71 | |
| 72 scoped_ptr<media::AudioBus> audio_data( | |
| 73 media::AudioBus::Create(audio_bus.channels(), audio_bus.frames())); | |
| 74 audio_bus.CopyTo(audio_data.get()); | |
| 75 audio_shifter_->Push(std::move(audio_data), estimated_capture_time); | |
| 76 const base::TimeTicks now = base::TimeTicks::Now(); | |
| 77 total_render_time_ += now - last_render_time_; | |
| 78 last_render_time_ = now; | |
| 79 } | |
| 80 | |
| 81 void WebRtcLocalAudioRenderer::OnSetFormat( | |
| 82 const media::AudioParameters& params) { | |
| 83 DVLOG(1) << "WebRtcLocalAudioRenderer::OnSetFormat()"; | |
| 84 // If the source is restarted, we might have changed to another capture | |
| 85 // thread. | |
| 86 capture_thread_checker_.DetachFromThread(); | |
| 87 DCHECK(capture_thread_checker_.CalledOnValidThread()); | |
| 88 | |
| 89 // Post a task on the main render thread to reconfigure the |sink_| with the | |
| 90 // new format. | |
| 91 task_runner_->PostTask( | |
| 92 FROM_HERE, | |
| 93 base::Bind(&WebRtcLocalAudioRenderer::ReconfigureSink, this, params)); | |
| 94 } | |
| 95 | |
| 96 // WebRtcLocalAudioRenderer::WebRtcLocalAudioRenderer implementation. | |
| 97 WebRtcLocalAudioRenderer::WebRtcLocalAudioRenderer( | |
| 98 const blink::WebMediaStreamTrack& audio_track, | |
| 99 int source_render_frame_id, | |
| 100 int session_id, | |
| 101 const std::string& device_id, | |
| 102 const url::Origin& security_origin) | |
| 103 : audio_track_(audio_track), | |
| 104 source_render_frame_id_(source_render_frame_id), | |
| 105 session_id_(session_id), | |
| 106 task_runner_(base::ThreadTaskRunnerHandle::Get()), | |
| 107 playing_(false), | |
| 108 output_device_id_(device_id), | |
| 109 security_origin_(security_origin), | |
| 110 volume_(0.0), | |
| 111 sink_started_(false) { | |
| 112 DVLOG(1) << "WebRtcLocalAudioRenderer::WebRtcLocalAudioRenderer()"; | |
| 113 } | |
| 114 | |
| 115 WebRtcLocalAudioRenderer::~WebRtcLocalAudioRenderer() { | |
| 116 DCHECK(task_runner_->BelongsToCurrentThread()); | |
| 117 DCHECK(!sink_.get()); | |
| 118 DVLOG(1) << "WebRtcLocalAudioRenderer::~WebRtcLocalAudioRenderer()"; | |
| 119 } | |
| 120 | |
| 121 void WebRtcLocalAudioRenderer::Start() { | |
| 122 DVLOG(1) << "WebRtcLocalAudioRenderer::Start()"; | |
| 123 DCHECK(task_runner_->BelongsToCurrentThread()); | |
| 124 | |
| 125 // We get audio data from |audio_track_|... | |
| 126 MediaStreamAudioSink::AddToAudioTrack(this, audio_track_); | |
| 127 // ...and |sink_| will get audio data from us. | |
| 128 DCHECK(!sink_.get()); | |
| 129 sink_ = | |
| 130 AudioDeviceFactory::NewOutputDevice(source_render_frame_id_, session_id_, | |
| 131 output_device_id_, security_origin_); | |
| 132 | |
| 133 base::AutoLock auto_lock(thread_lock_); | |
| 134 last_render_time_ = base::TimeTicks::Now(); | |
| 135 playing_ = false; | |
| 136 } | |
| 137 | |
| 138 void WebRtcLocalAudioRenderer::Stop() { | |
| 139 DVLOG(1) << "WebRtcLocalAudioRenderer::Stop()"; | |
| 140 DCHECK(task_runner_->BelongsToCurrentThread()); | |
| 141 | |
| 142 { | |
| 143 base::AutoLock auto_lock(thread_lock_); | |
| 144 playing_ = false; | |
| 145 audio_shifter_.reset(); | |
| 146 } | |
| 147 | |
| 148 // 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 | |
| 150 // when the |sink_| is never started. | |
| 151 if (sink_.get()) { | |
| 152 sink_->Stop(); | |
| 153 sink_ = NULL; | |
| 154 } | |
| 155 | |
| 156 if (!sink_started_) { | |
| 157 UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates", | |
| 158 kSinkNeverStarted, kSinkStatesMax); | |
| 159 } | |
| 160 sink_started_ = false; | |
| 161 | |
| 162 // Ensure that the capturer stops feeding us with captured audio. | |
| 163 MediaStreamAudioSink::RemoveFromAudioTrack(this, audio_track_); | |
| 164 } | |
| 165 | |
| 166 void WebRtcLocalAudioRenderer::Play() { | |
| 167 DVLOG(1) << "WebRtcLocalAudioRenderer::Play()"; | |
| 168 DCHECK(task_runner_->BelongsToCurrentThread()); | |
| 169 | |
| 170 if (!sink_.get()) | |
| 171 return; | |
| 172 | |
| 173 { | |
| 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 | |
| 181 // Note: If volume_ is currently muted, the |sink_| will not be started yet. | |
| 182 MaybeStartSink(); | |
| 183 } | |
| 184 | |
| 185 void WebRtcLocalAudioRenderer::Pause() { | |
| 186 DVLOG(1) << "WebRtcLocalAudioRenderer::Pause()"; | |
| 187 DCHECK(task_runner_->BelongsToCurrentThread()); | |
| 188 | |
| 189 if (!sink_.get()) | |
| 190 return; | |
| 191 | |
| 192 base::AutoLock auto_lock(thread_lock_); | |
| 193 // Temporarily suspends rendering audio. | |
| 194 // WebRtcLocalAudioRenderer::Render() will return early during this state | |
| 195 // and only zeros will be provided to the active sink. | |
| 196 playing_ = false; | |
| 197 } | |
| 198 | |
| 199 void WebRtcLocalAudioRenderer::SetVolume(float volume) { | |
| 200 DVLOG(1) << "WebRtcLocalAudioRenderer::SetVolume(" << volume << ")"; | |
| 201 DCHECK(task_runner_->BelongsToCurrentThread()); | |
| 202 | |
| 203 { | |
| 204 base::AutoLock auto_lock(thread_lock_); | |
| 205 // Cache the volume. | |
| 206 volume_ = volume; | |
| 207 } | |
| 208 | |
| 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 = | |
| 242 AudioDeviceFactory::NewOutputDevice(source_render_frame_id_, session_id_, | |
| 243 device_id, security_origin); | |
| 244 if (new_sink->GetDeviceStatus() != media::OUTPUT_DEVICE_STATUS_OK) { | |
| 245 callback.Run(new_sink->GetDeviceStatus()); | |
| 246 return; | |
| 247 } | |
| 248 | |
| 249 output_device_id_ = device_id; | |
| 250 security_origin_ = security_origin; | |
| 251 bool was_sink_started = sink_started_; | |
| 252 | |
| 253 if (sink_.get()) | |
| 254 sink_->Stop(); | |
| 255 | |
| 256 sink_started_ = false; | |
| 257 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) | |
| 264 MaybeStartSink(); | |
| 265 | |
| 266 callback.Run(media::OUTPUT_DEVICE_STATUS_OK); | |
| 267 } | |
| 268 | |
| 269 media::AudioParameters WebRtcLocalAudioRenderer::GetOutputParameters() { | |
| 270 DCHECK(task_runner_->BelongsToCurrentThread()); | |
| 271 if (!sink_.get()) | |
| 272 return media::AudioParameters(); | |
| 273 | |
| 274 return sink_->GetOutputParameters(); | |
| 275 } | |
| 276 | |
| 277 media::OutputDeviceStatus WebRtcLocalAudioRenderer::GetDeviceStatus() { | |
| 278 DCHECK(task_runner_->BelongsToCurrentThread()); | |
| 279 if (!sink_.get()) | |
| 280 return media::OUTPUT_DEVICE_STATUS_ERROR_INTERNAL; | |
| 281 | |
| 282 return sink_->GetDeviceStatus(); | |
| 283 } | |
| 284 | |
| 285 void WebRtcLocalAudioRenderer::MaybeStartSink() { | |
| 286 DCHECK(task_runner_->BelongsToCurrentThread()); | |
| 287 DVLOG(1) << "WebRtcLocalAudioRenderer::MaybeStartSink()"; | |
| 288 | |
| 289 if (!sink_.get() || !source_params_.IsValid()) | |
| 290 return; | |
| 291 | |
| 292 { | |
| 293 // Clear up the old data in the FIFO. | |
| 294 base::AutoLock auto_lock(thread_lock_); | |
| 295 audio_shifter_->Flush(); | |
| 296 } | |
| 297 | |
| 298 if (!sink_params_.IsValid() || !playing_ || !volume_ || sink_started_ || | |
| 299 sink_->GetDeviceStatus() != media::OUTPUT_DEVICE_STATUS_OK) | |
| 300 return; | |
| 301 | |
| 302 DVLOG(1) << "WebRtcLocalAudioRenderer::MaybeStartSink() -- Starting sink_."; | |
| 303 sink_->Initialize(sink_params_, this); | |
| 304 sink_->Start(); | |
| 305 sink_started_ = true; | |
| 306 UMA_HISTOGRAM_ENUMERATION("Media.LocalRendererSinkStates", | |
| 307 kSinkStarted, kSinkStatesMax); | |
| 308 } | |
| 309 | |
| 310 void WebRtcLocalAudioRenderer::ReconfigureSink( | |
| 311 const media::AudioParameters& params) { | |
| 312 DCHECK(task_runner_->BelongsToCurrentThread()); | |
| 313 | |
| 314 DVLOG(1) << "WebRtcLocalAudioRenderer::ReconfigureSink()"; | |
| 315 | |
| 316 if (source_params_.Equals(params)) | |
| 317 return; | |
| 318 | |
| 319 // Reset the |source_params_|, |sink_params_| and |loopback_fifo_| to match | |
| 320 // the new format. | |
| 321 | |
| 322 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 | |
| 339 if (!sink_.get()) | |
| 340 return; // WebRtcLocalAudioRenderer has not yet been started. | |
| 341 | |
| 342 // Stop |sink_| and re-create a new one to be initialized with different audio | |
| 343 // parameters. Then, invoke MaybeStartSink() to restart everything again. | |
| 344 sink_->Stop(); | |
| 345 sink_started_ = false; | |
| 346 sink_ = | |
| 347 AudioDeviceFactory::NewOutputDevice(source_render_frame_id_, session_id_, | |
| 348 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(); | |
| 354 } | |
| 355 | |
| 356 } // namespace content | |
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Chromium Code Reviews
Issue 1633423002:
MediaStream audio rendering: Bypass audio processing for non-WebRTC cases. (Closed)
Base URL: https://chromium.googlesource.com/chromium/src.git@master