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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_audio_capturer.h" | |
| 6 | |
| 7 #include "base/bind.h" | |
| 8 #include "base/logging.h" | |
| 9 #include "base/macros.h" | |
| 10 #include "base/metrics/histogram.h" | |
| 11 #include "base/strings/string_util.h" | |
| 12 #include "base/strings/stringprintf.h" | |
| 13 #include "build/build_config.h" | |
| 14 #include "content/child/child_process.h" | |
| 15 #include "content/renderer/media/audio_device_factory.h" | |
| 16 #include "content/renderer/media/media_stream_audio_processor.h" | |
| 17 #include "content/renderer/media/media_stream_audio_processor_options.h" | |
| 18 #include "content/renderer/media/media_stream_audio_source.h" | |
| 19 #include "content/renderer/media/media_stream_constraints_util.h" | |
| 20 #include "content/renderer/media/webrtc_audio_device_impl.h" | |
| 21 #include "content/renderer/media/webrtc_local_audio_track.h" | |
| 22 #include "content/renderer/media/webrtc_logging.h" | |
| 23 #include "media/audio/sample_rates.h" | |
| 24 | |
| 25 namespace content { | |
| 26 | |
| 27 // Reference counted container of WebRtcLocalAudioTrack delegate. | |
| 28 // TODO(xians): Switch to MediaStreamAudioSinkOwner. | |
| 29 class WebRtcAudioCapturer::TrackOwner | |
| 30 : public base::RefCountedThreadSafe<WebRtcAudioCapturer::TrackOwner> { | |
| 31 public: | |
| 32 explicit TrackOwner(WebRtcLocalAudioTrack* track) | |
| 33 : delegate_(track) {} | |
| 34 | |
| 35 void Capture(const media::AudioBus& audio_bus, | |
| 36 base::TimeTicks estimated_capture_time) { | |
| 37 base::AutoLock lock(lock_); | |
| 38 if (delegate_) { | |
| 39 delegate_->Capture(audio_bus, estimated_capture_time); | |
| 40 } | |
| 41 } | |
| 42 | |
| 43 void OnSetFormat(const media::AudioParameters& params) { | |
| 44 base::AutoLock lock(lock_); | |
| 45 if (delegate_) | |
| 46 delegate_->OnSetFormat(params); | |
| 47 } | |
| 48 | |
| 49 void Reset() { | |
| 50 base::AutoLock lock(lock_); | |
| 51 delegate_ = NULL; | |
| 52 } | |
| 53 | |
| 54 void Stop() { | |
| 55 base::AutoLock lock(lock_); | |
| 56 DCHECK(delegate_); | |
| 57 | |
| 58 // This can be reentrant so reset |delegate_| before calling out. | |
| 59 WebRtcLocalAudioTrack* temp = delegate_; | |
| 60 delegate_ = NULL; | |
| 61 temp->Stop(); | |
| 62 } | |
| 63 | |
| 64 // Wrapper which allows to use std::find_if() when adding and removing | |
| 65 // sinks to/from the list. | |
| 66 struct TrackWrapper { | |
| 67 explicit TrackWrapper(WebRtcLocalAudioTrack* track) : track_(track) {} | |
| 68 bool operator()( | |
| 69 const scoped_refptr<WebRtcAudioCapturer::TrackOwner>& owner) const { | |
| 70 return owner->IsEqual(track_); | |
| 71 } | |
| 72 WebRtcLocalAudioTrack* track_; | |
| 73 }; | |
| 74 | |
| 75 protected: | |
| 76 virtual ~TrackOwner() {} | |
| 77 | |
| 78 private: | |
| 79 friend class base::RefCountedThreadSafe<WebRtcAudioCapturer::TrackOwner>; | |
| 80 | |
| 81 bool IsEqual(const WebRtcLocalAudioTrack* other) const { | |
| 82 base::AutoLock lock(lock_); | |
| 83 return (other == delegate_); | |
| 84 } | |
| 85 | |
| 86 // Do NOT reference count the |delegate_| to avoid cyclic reference counting. | |
| 87 WebRtcLocalAudioTrack* delegate_; | |
| 88 mutable base::Lock lock_; | |
| 89 | |
| 90 DISALLOW_COPY_AND_ASSIGN(TrackOwner); | |
| 91 }; | |
| 92 | |
| 93 // static | |
| 94 std::unique_ptr<WebRtcAudioCapturer> WebRtcAudioCapturer::CreateCapturer( | |
| 95 int render_frame_id, | |
| 96 const StreamDeviceInfo& device_info, | |
| 97 const blink::WebMediaConstraints& constraints, | |
| 98 WebRtcAudioDeviceImpl* audio_device, | |
| 99 MediaStreamAudioSource* audio_source) { | |
| 100 std::unique_ptr<WebRtcAudioCapturer> capturer(new WebRtcAudioCapturer( | |
| 101 render_frame_id, device_info, constraints, audio_device, audio_source)); | |
| 102 if (capturer->Initialize()) | |
| 103 return capturer; | |
| 104 | |
| 105 return NULL; | |
| 106 } | |
| 107 | |
| 108 bool WebRtcAudioCapturer::Initialize() { | |
| 109 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 110 DVLOG(1) << "WebRtcAudioCapturer::Initialize()"; | |
| 111 WebRtcLogMessage(base::StringPrintf( | |
| 112 "WAC::Initialize. render_frame_id=%d" | |
| 113 ", channel_layout=%d, sample_rate=%d, buffer_size=%d" | |
| 114 ", session_id=%d, paired_output_sample_rate=%d" | |
| 115 ", paired_output_frames_per_buffer=%d, effects=%d. ", | |
| 116 render_frame_id_, device_info_.device.input.channel_layout, | |
| 117 device_info_.device.input.sample_rate, | |
| 118 device_info_.device.input.frames_per_buffer, device_info_.session_id, | |
| 119 device_info_.device.matched_output.sample_rate, | |
| 120 device_info_.device.matched_output.frames_per_buffer, | |
| 121 device_info_.device.input.effects)); | |
| 122 | |
| 123 if (render_frame_id_ == -1) { | |
| 124 // Return true here to allow injecting a new source via | |
| 125 // SetCapturerSourceForTesting() at a later state. | |
| 126 return true; | |
| 127 } | |
| 128 | |
| 129 MediaAudioConstraints audio_constraints(constraints_, | |
| 130 device_info_.device.input.effects); | |
| 131 if (!audio_constraints.IsValid()) | |
| 132 return false; | |
| 133 | |
| 134 media::ChannelLayout channel_layout = static_cast<media::ChannelLayout>( | |
| 135 device_info_.device.input.channel_layout); | |
| 136 | |
| 137 // If KEYBOARD_MIC effect is set, change the layout to the corresponding | |
| 138 // layout that includes the keyboard mic. | |
| 139 if ((device_info_.device.input.effects & | |
| 140 media::AudioParameters::KEYBOARD_MIC) && | |
| 141 audio_constraints.GetGoogExperimentalNoiseSuppression()) { | |
| 142 if (channel_layout == media::CHANNEL_LAYOUT_STEREO) { | |
| 143 channel_layout = media::CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC; | |
| 144 DVLOG(1) << "Changed stereo layout to stereo + keyboard mic layout due " | |
| 145 << "to KEYBOARD_MIC effect."; | |
| 146 } else { | |
| 147 DVLOG(1) << "KEYBOARD_MIC effect ignored, not compatible with layout " | |
| 148 << channel_layout; | |
| 149 } | |
| 150 } | |
| 151 | |
| 152 DVLOG(1) << "Audio input hardware channel layout: " << channel_layout; | |
| 153 UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioInputChannelLayout", | |
| 154 channel_layout, media::CHANNEL_LAYOUT_MAX + 1); | |
| 155 | |
| 156 // Verify that the reported input channel configuration is supported. | |
| 157 if (channel_layout != media::CHANNEL_LAYOUT_MONO && | |
| 158 channel_layout != media::CHANNEL_LAYOUT_STEREO && | |
| 159 channel_layout != media::CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC) { | |
| 160 DLOG(ERROR) << channel_layout | |
| 161 << " is not a supported input channel configuration."; | |
| 162 return false; | |
| 163 } | |
| 164 | |
| 165 DVLOG(1) << "Audio input hardware sample rate: " | |
| 166 << device_info_.device.input.sample_rate; | |
| 167 media::AudioSampleRate asr; | |
| 168 if (media::ToAudioSampleRate(device_info_.device.input.sample_rate, &asr)) { | |
| 169 UMA_HISTOGRAM_ENUMERATION( | |
| 170 "WebRTC.AudioInputSampleRate", asr, media::kAudioSampleRateMax + 1); | |
| 171 } else { | |
| 172 UMA_HISTOGRAM_COUNTS("WebRTC.AudioInputSampleRateUnexpected", | |
| 173 device_info_.device.input.sample_rate); | |
| 174 } | |
| 175 | |
| 176 // Create and configure the default audio capturing source. | |
| 177 SetCapturerSourceInternal( | |
| 178 AudioDeviceFactory::NewAudioCapturerSource(render_frame_id_), | |
| 179 channel_layout, device_info_.device.input.sample_rate); | |
| 180 | |
| 181 // Add the capturer to the WebRtcAudioDeviceImpl since it needs some hardware | |
| 182 // information from the capturer. | |
| 183 if (audio_device_) | |
| 184 audio_device_->AddAudioCapturer(this); | |
| 185 | |
| 186 return true; | |
| 187 } | |
| 188 | |
| 189 WebRtcAudioCapturer::WebRtcAudioCapturer( | |
| 190 int render_frame_id, | |
| 191 const StreamDeviceInfo& device_info, | |
| 192 const blink::WebMediaConstraints& constraints, | |
| 193 WebRtcAudioDeviceImpl* audio_device, | |
| 194 MediaStreamAudioSource* audio_source) | |
| 195 : constraints_(constraints), | |
| 196 audio_processor_(new rtc::RefCountedObject<MediaStreamAudioProcessor>( | |
| 197 constraints, | |
| 198 device_info.device.input, | |
| 199 audio_device)), | |
| 200 running_(false), | |
| 201 render_frame_id_(render_frame_id), | |
| 202 device_info_(device_info), | |
| 203 volume_(0), | |
| 204 peer_connection_mode_(false), | |
| 205 audio_device_(audio_device), | |
| 206 audio_source_(audio_source) { | |
| 207 DVLOG(1) << "WebRtcAudioCapturer::WebRtcAudioCapturer()"; | |
| 208 } | |
| 209 | |
| 210 WebRtcAudioCapturer::~WebRtcAudioCapturer() { | |
| 211 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 212 DCHECK(tracks_.IsEmpty()); | |
| 213 DVLOG(1) << "WebRtcAudioCapturer::~WebRtcAudioCapturer()"; | |
| 214 Stop(); | |
| 215 } | |
| 216 | |
| 217 void WebRtcAudioCapturer::AddTrack(WebRtcLocalAudioTrack* track) { | |
| 218 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 219 DCHECK(track); | |
| 220 DVLOG(1) << "WebRtcAudioCapturer::AddTrack()"; | |
| 221 | |
| 222 track->SetLevel(level_calculator_.level()); | |
| 223 | |
| 224 // The track only grabs stats from the audio processor. Stats are only | |
| 225 // available if audio processing is turned on. Therefore, only provide the | |
| 226 // track a reference if audio processing is turned on. | |
| 227 if (audio_processor_->has_audio_processing()) | |
| 228 track->SetAudioProcessor(audio_processor_); | |
| 229 | |
| 230 { | |
| 231 base::AutoLock auto_lock(lock_); | |
| 232 // Verify that |track| is not already added to the list. | |
| 233 DCHECK(!tracks_.Contains(TrackOwner::TrackWrapper(track))); | |
| 234 | |
| 235 // Add with a tag, so we remember to call OnSetFormat() on the new | |
| 236 // track. | |
| 237 scoped_refptr<TrackOwner> track_owner(new TrackOwner(track)); | |
| 238 tracks_.AddAndTag(track_owner.get()); | |
| 239 } | |
| 240 } | |
| 241 | |
| 242 void WebRtcAudioCapturer::RemoveTrack(WebRtcLocalAudioTrack* track) { | |
| 243 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 244 DVLOG(1) << "WebRtcAudioCapturer::RemoveTrack()"; | |
| 245 bool stop_source = false; | |
| 246 { | |
| 247 base::AutoLock auto_lock(lock_); | |
| 248 | |
| 249 scoped_refptr<TrackOwner> removed_item = | |
| 250 tracks_.Remove(TrackOwner::TrackWrapper(track)); | |
| 251 | |
| 252 // Clear the delegate to ensure that no more capture callbacks will | |
| 253 // be sent to this sink. Also avoids a possible crash which can happen | |
| 254 // if this method is called while capturing is active. | |
| 255 if (removed_item.get()) { | |
| 256 removed_item->Reset(); | |
| 257 stop_source = tracks_.IsEmpty(); | |
| 258 } | |
| 259 } | |
| 260 if (stop_source) { | |
| 261 // Since WebRtcAudioCapturer does not inherit MediaStreamAudioSource, | |
| 262 // and instead MediaStreamAudioSource is composed of a WebRtcAudioCapturer, | |
| 263 // we have to call StopSource on the MediaStreamSource. This will call | |
| 264 // MediaStreamAudioSource::DoStopSource which in turn call | |
| 265 // WebRtcAudioCapturerer::Stop(); | |
| 266 audio_source_->StopSource(); | |
| 267 } | |
| 268 } | |
| 269 | |
| 270 void WebRtcAudioCapturer::SetCapturerSourceInternal( | |
| 271 const scoped_refptr<media::AudioCapturerSource>& source, | |
| 272 media::ChannelLayout channel_layout, | |
| 273 int sample_rate) { | |
| 274 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 275 DVLOG(1) << "SetCapturerSource(channel_layout=" << channel_layout << "," | |
| 276 << "sample_rate=" << sample_rate << ")"; | |
| 277 scoped_refptr<media::AudioCapturerSource> old_source; | |
| 278 { | |
| 279 base::AutoLock auto_lock(lock_); | |
| 280 if (source_.get() == source.get()) | |
| 281 return; | |
| 282 | |
| 283 source_.swap(old_source); | |
| 284 source_ = source; | |
| 285 | |
| 286 // Reset the flag to allow starting the new source. | |
| 287 running_ = false; | |
| 288 } | |
| 289 | |
| 290 DVLOG(1) << "Switching to a new capture source."; | |
| 291 if (old_source.get()) | |
| 292 old_source->Stop(); | |
| 293 | |
| 294 // Dispatch the new parameters both to the sink(s) and to the new source, | |
| 295 // also apply the new |constraints|. | |
| 296 // The idea is to get rid of any dependency of the microphone parameters | |
| 297 // which would normally be used by default. | |
| 298 // bits_per_sample is always 16 for now. | |
| 299 media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | |
| 300 channel_layout, sample_rate, 16, | |
| 301 GetBufferSize(sample_rate)); | |
| 302 params.set_effects(device_info_.device.input.effects); | |
| 303 DCHECK(params.IsValid()); | |
| 304 | |
| 305 { | |
| 306 base::AutoLock auto_lock(lock_); | |
| 307 | |
| 308 // Notify the |audio_processor_| of the new format. We're doing this while | |
| 309 // the lock is held only because the signaling thread might be calling | |
| 310 // GetInputFormat(). Simultaneous reads from the audio thread are NOT the | |
| 311 // concern here since the source is currently stopped (i.e., no audio | |
| 312 // capture calls can be executing). | |
| 313 audio_processor_->OnCaptureFormatChanged(params); | |
| 314 | |
| 315 // Notify all tracks about the new format. | |
| 316 tracks_.TagAll(); | |
| 317 } | |
| 318 | |
| 319 if (source.get()) | |
| 320 source->Initialize(params, this, device_info_.session_id); | |
| 321 | |
| 322 Start(); | |
| 323 } | |
| 324 | |
| 325 void WebRtcAudioCapturer::EnablePeerConnectionMode() { | |
| 326 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 327 DVLOG(1) << "EnablePeerConnectionMode"; | |
| 328 // Do nothing if the peer connection mode has been enabled. | |
| 329 if (peer_connection_mode_) | |
| 330 return; | |
| 331 | |
| 332 peer_connection_mode_ = true; | |
| 333 int render_frame_id = -1; | |
| 334 media::AudioParameters input_params; | |
| 335 { | |
| 336 base::AutoLock auto_lock(lock_); | |
| 337 // Simply return if there is no existing source or the |render_frame_id_| is | |
| 338 // not valid. | |
| 339 if (!source_.get() || render_frame_id_ == -1) | |
| 340 return; | |
| 341 | |
| 342 render_frame_id = render_frame_id_; | |
| 343 input_params = audio_processor_->InputFormat(); | |
| 344 } | |
| 345 | |
| 346 // Do nothing if the current buffer size is the WebRtc native buffer size. | |
| 347 if (GetBufferSize(input_params.sample_rate()) == | |
| 348 input_params.frames_per_buffer()) { | |
| 349 return; | |
| 350 } | |
| 351 | |
| 352 // Create a new audio stream as source which will open the hardware using | |
| 353 // WebRtc native buffer size. | |
| 354 SetCapturerSourceInternal( | |
| 355 AudioDeviceFactory::NewAudioCapturerSource(render_frame_id), | |
| 356 input_params.channel_layout(), input_params.sample_rate()); | |
| 357 } | |
| 358 | |
| 359 void WebRtcAudioCapturer::Start() { | |
| 360 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 361 DVLOG(1) << "WebRtcAudioCapturer::Start()"; | |
| 362 base::AutoLock auto_lock(lock_); | |
| 363 if (running_ || !source_.get()) | |
| 364 return; | |
| 365 | |
| 366 // Start the data source, i.e., start capturing data from the current source. | |
| 367 // We need to set the AGC control before starting the stream. | |
| 368 source_->SetAutomaticGainControl(true); | |
| 369 source_->Start(); | |
| 370 running_ = true; | |
| 371 } | |
| 372 | |
| 373 void WebRtcAudioCapturer::Stop() { | |
| 374 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 375 DVLOG(1) << "WebRtcAudioCapturer::Stop()"; | |
| 376 scoped_refptr<media::AudioCapturerSource> source; | |
| 377 TrackList::ItemList tracks; | |
| 378 { | |
| 379 base::AutoLock auto_lock(lock_); | |
| 380 if (!running_) | |
| 381 return; | |
| 382 | |
| 383 source = source_; | |
| 384 tracks = tracks_.Items(); | |
| 385 tracks_.Clear(); | |
| 386 running_ = false; | |
| 387 } | |
| 388 | |
| 389 // Remove the capturer object from the WebRtcAudioDeviceImpl. | |
| 390 if (audio_device_) | |
| 391 audio_device_->RemoveAudioCapturer(this); | |
| 392 | |
| 393 for (TrackList::ItemList::const_iterator it = tracks.begin(); | |
| 394 it != tracks.end(); | |
| 395 ++it) { | |
| 396 (*it)->Stop(); | |
| 397 } | |
| 398 | |
| 399 if (source.get()) | |
| 400 source->Stop(); | |
| 401 | |
| 402 // Stop the audio processor to avoid feeding render data into the processor. | |
| 403 audio_processor_->Stop(); | |
| 404 } | |
| 405 | |
| 406 void WebRtcAudioCapturer::SetVolume(int volume) { | |
| 407 DVLOG(1) << "WebRtcAudioCapturer::SetVolume()"; | |
| 408 DCHECK_LE(volume, MaxVolume()); | |
| 409 double normalized_volume = static_cast<double>(volume) / MaxVolume(); | |
| 410 base::AutoLock auto_lock(lock_); | |
| 411 if (source_.get()) | |
| 412 source_->SetVolume(normalized_volume); | |
| 413 } | |
| 414 | |
| 415 int WebRtcAudioCapturer::Volume() const { | |
| 416 base::AutoLock auto_lock(lock_); | |
| 417 return volume_; | |
| 418 } | |
| 419 | |
| 420 int WebRtcAudioCapturer::MaxVolume() const { | |
| 421 return WebRtcAudioDeviceImpl::kMaxVolumeLevel; | |
| 422 } | |
| 423 | |
| 424 media::AudioParameters WebRtcAudioCapturer::GetOutputFormat() const { | |
| 425 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 426 return audio_processor_->OutputFormat(); | |
| 427 } | |
| 428 | |
| 429 void WebRtcAudioCapturer::Capture(const media::AudioBus* audio_source, | |
| 430 int audio_delay_milliseconds, | |
| 431 double volume, | |
| 432 bool key_pressed) { | |
| 433 // This callback is driven by AudioInputDevice::AudioThreadCallback if | |
| 434 // |source_| is AudioInputDevice, otherwise it is driven by client's | |
| 435 // CaptureCallback. | |
| 436 #if defined(OS_WIN) || defined(OS_MACOSX) | |
| 437 DCHECK_LE(volume, 1.0); | |
| 438 #elif (defined(OS_LINUX) && !defined(OS_CHROMEOS)) || defined(OS_OPENBSD) | |
| 439 // We have a special situation on Linux where the microphone volume can be | |
| 440 // "higher than maximum". The input volume slider in the sound preference | |
| 441 // allows the user to set a scaling that is higher than 100%. It means that | |
| 442 // even if the reported maximum levels is N, the actual microphone level can | |
| 443 // go up to 1.5x*N and that corresponds to a normalized |volume| of 1.5x. | |
| 444 DCHECK_LE(volume, 1.6); | |
| 445 #endif | |
| 446 | |
| 447 // TODO(miu): Plumbing is needed to determine the actual capture timestamp | |
| 448 // of the audio, instead of just snapshotting TimeTicks::Now(), for proper | |
| 449 // audio/video sync. http://crbug.com/335335 | |
| 450 const base::TimeTicks reference_clock_snapshot = base::TimeTicks::Now(); | |
| 451 | |
| 452 TrackList::ItemList tracks; | |
| 453 TrackList::ItemList tracks_to_notify_format; | |
| 454 int current_volume = 0; | |
| 455 { | |
| 456 base::AutoLock auto_lock(lock_); | |
| 457 if (!running_) | |
| 458 return; | |
| 459 | |
| 460 // Map internal volume range of [0.0, 1.0] into [0, 255] used by AGC. | |
| 461 // The volume can be higher than 255 on Linux, and it will be cropped to | |
| 462 // 255 since AGC does not allow values out of range. | |
| 463 volume_ = static_cast<int>((volume * MaxVolume()) + 0.5); | |
| 464 current_volume = volume_ > MaxVolume() ? MaxVolume() : volume_; | |
| 465 tracks = tracks_.Items(); | |
| 466 tracks_.RetrieveAndClearTags(&tracks_to_notify_format); | |
| 467 } | |
| 468 | |
| 469 // Sanity-check the input audio format in debug builds. Then, notify the | |
| 470 // tracks if the format has changed. | |
| 471 // | |
| 472 // Locking is not needed here to read the audio input/output parameters | |
| 473 // because the audio processor format changes only occur while audio capture | |
| 474 // is stopped. | |
| 475 DCHECK(audio_processor_->InputFormat().IsValid()); | |
| 476 DCHECK_EQ(audio_source->channels(), | |
| 477 audio_processor_->InputFormat().channels()); | |
| 478 DCHECK_EQ(audio_source->frames(), | |
| 479 audio_processor_->InputFormat().frames_per_buffer()); | |
| 480 if (!tracks_to_notify_format.empty()) { | |
| 481 const media::AudioParameters& output_params = | |
| 482 audio_processor_->OutputFormat(); | |
| 483 for (const auto& track : tracks_to_notify_format) | |
| 484 track->OnSetFormat(output_params); | |
| 485 } | |
| 486 | |
| 487 // Figure out if the pre-processed data has any energy or not. This | |
| 488 // information will be passed to the level calculator to force it to report | |
| 489 // energy in case the post-processed data is zeroed by the audio processing. | |
| 490 const bool force_report_nonzero_energy = !audio_source->AreFramesZero(); | |
| 491 | |
| 492 // Push the data to the processor for processing. | |
| 493 audio_processor_->PushCaptureData( | |
| 494 *audio_source, | |
| 495 base::TimeDelta::FromMilliseconds(audio_delay_milliseconds)); | |
| 496 | |
| 497 // Process and consume the data in the processor until there is not enough | |
| 498 // data in the processor. | |
| 499 media::AudioBus* processed_data = nullptr; | |
| 500 base::TimeDelta processed_data_audio_delay; | |
| 501 int new_volume = 0; | |
| 502 while (audio_processor_->ProcessAndConsumeData( | |
| 503 current_volume, key_pressed, | |
| 504 &processed_data, &processed_data_audio_delay, &new_volume)) { | |
| 505 DCHECK(processed_data); | |
| 506 | |
| 507 level_calculator_.Calculate(*processed_data, force_report_nonzero_energy); | |
| 508 | |
| 509 const base::TimeTicks processed_data_capture_time = | |
| 510 reference_clock_snapshot - processed_data_audio_delay; | |
| 511 for (const auto& track : tracks) | |
| 512 track->Capture(*processed_data, processed_data_capture_time); | |
| 513 | |
| 514 if (new_volume) { | |
| 515 SetVolume(new_volume); | |
| 516 | |
| 517 // Update the |current_volume| to avoid passing the old volume to AGC. | |
| 518 current_volume = new_volume; | |
| 519 } | |
| 520 } | |
| 521 } | |
| 522 | |
| 523 void WebRtcAudioCapturer::OnCaptureError(const std::string& message) { | |
| 524 WebRtcLogMessage("WAC::OnCaptureError: " + message); | |
| 525 } | |
| 526 | |
| 527 media::AudioParameters WebRtcAudioCapturer::GetInputFormat() const { | |
| 528 base::AutoLock auto_lock(lock_); | |
| 529 return audio_processor_->InputFormat(); | |
| 530 } | |
| 531 | |
| 532 int WebRtcAudioCapturer::GetBufferSize(int sample_rate) const { | |
| 533 DCHECK(thread_checker_.CalledOnValidThread()); | |
| 534 #if defined(OS_ANDROID) | |
| 535 // TODO(henrika): Tune and adjust buffer size on Android. | |
| 536 return (2 * sample_rate / 100); | |
| 537 #endif | |
| 538 | |
| 539 // PeerConnection is running at a buffer size of 10ms data. A multiple of | |
| 540 // 10ms as the buffer size can give the best performance to PeerConnection. | |
| 541 int peer_connection_buffer_size = sample_rate / 100; | |
| 542 | |
| 543 // Use the native hardware buffer size in non peer connection mode when the | |
| 544 // platform is using a native buffer size smaller than the PeerConnection | |
| 545 // buffer size and audio processing is off. | |
| 546 int hardware_buffer_size = device_info_.device.input.frames_per_buffer; | |
| 547 if (!peer_connection_mode_ && hardware_buffer_size && | |
| 548 hardware_buffer_size <= peer_connection_buffer_size && | |
| 549 !audio_processor_->has_audio_processing()) { | |
| 550 DVLOG(1) << "WebRtcAudioCapturer is using hardware buffer size " | |
| 551 << hardware_buffer_size; | |
| 552 return hardware_buffer_size; | |
| 553 } | |
| 554 | |
| 555 return (sample_rate / 100); | |
| 556 } | |
| 557 | |
| 558 void WebRtcAudioCapturer::SetCapturerSource( | |
| 559 const scoped_refptr<media::AudioCapturerSource>& source, | |
| 560 media::AudioParameters params) { | |
| 561 // Create a new audio stream as source which uses the new source. | |
| 562 SetCapturerSourceInternal(source, params.channel_layout(), | |
| 563 params.sample_rate()); | |
| 564 } | |
| 565 | |
| 566 } // namespace content | |
| OLD | NEW |
|---|
Chromium Code Reviews
Issue 1834323002:
MediaStream audio: Refactor 3 separate "glue" implementations into one. (Closed)
Base URL: https://chromium.googlesource.com/chromium/src.git@master