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1 // Copyright 2013 The Chromium Authors. All rights reserved. | 1 // Copyright 2013 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/media_stream_audio_processor.h" | 5 #include "content/renderer/media/media_stream_audio_processor.h" |
6 | 6 |
7 #include "base/debug/trace_event.h" | 7 #include "base/debug/trace_event.h" |
8 #if defined(OS_MACOSX) | 8 #if defined(OS_MACOSX) |
9 #include "base/metrics/field_trial.h" | 9 #include "base/metrics/field_trial.h" |
10 #endif | 10 #endif |
11 #include "base/metrics/histogram.h" | 11 #include "base/metrics/histogram.h" |
12 #include "content/renderer/media/media_stream_audio_processor_options.h" | 12 #include "content/renderer/media/media_stream_audio_processor_options.h" |
13 #include "content/renderer/media/rtc_media_constraints.h" | 13 #include "content/renderer/media/rtc_media_constraints.h" |
14 #include "content/renderer/media/webrtc_audio_device_impl.h" | 14 #include "content/renderer/media/webrtc_audio_device_impl.h" |
15 #include "media/audio/audio_parameters.h" | 15 #include "media/audio/audio_parameters.h" |
16 #include "media/base/audio_converter.h" | 16 #include "media/base/audio_converter.h" |
17 #include "media/base/audio_fifo.h" | 17 #include "media/base/audio_fifo.h" |
18 #include "media/base/channel_layout.h" | 18 #include "media/base/channel_layout.h" |
19 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" | 19 #include "third_party/WebKit/public/platform/WebMediaConstraints.h" |
20 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface .h" | 20 #include "third_party/libjingle/source/talk/app/webrtc/mediaconstraintsinterface .h" |
21 #include "third_party/webrtc/modules/audio_processing/typing_detection.h" | 21 #include "third_party/webrtc/modules/audio_processing/typing_detection.h" |
22 | 22 |
23 #if defined(OS_CHROMEOS) | |
24 #include "base/sys_info.h" | |
25 #endif | |
26 | |
23 namespace content { | 27 namespace content { |
24 | 28 |
25 namespace { | 29 namespace { |
26 | 30 |
27 using webrtc::AudioProcessing; | 31 using webrtc::AudioProcessing; |
28 | 32 |
29 #if defined(OS_ANDROID) | 33 #if defined(OS_ANDROID) |
30 const int kAudioProcessingSampleRate = 16000; | 34 const int kAudioProcessingSampleRate = 16000; |
31 #else | 35 #else |
32 const int kAudioProcessingSampleRate = 32000; | 36 const int kAudioProcessingSampleRate = 32000; |
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438 const bool goog_experimental_aec = audio_constraints.GetProperty( | 442 const bool goog_experimental_aec = audio_constraints.GetProperty( |
439 MediaAudioConstraints::kGoogExperimentalEchoCancellation); | 443 MediaAudioConstraints::kGoogExperimentalEchoCancellation); |
440 const bool goog_typing_detection = audio_constraints.GetProperty( | 444 const bool goog_typing_detection = audio_constraints.GetProperty( |
441 MediaAudioConstraints::kGoogTypingNoiseDetection); | 445 MediaAudioConstraints::kGoogTypingNoiseDetection); |
442 #endif | 446 #endif |
443 | 447 |
444 const bool goog_ns = audio_constraints.GetProperty( | 448 const bool goog_ns = audio_constraints.GetProperty( |
445 MediaAudioConstraints::kGoogNoiseSuppression); | 449 MediaAudioConstraints::kGoogNoiseSuppression); |
446 const bool goog_experimental_ns = audio_constraints.GetProperty( | 450 const bool goog_experimental_ns = audio_constraints.GetProperty( |
447 MediaAudioConstraints::kGoogExperimentalNoiseSuppression); | 451 MediaAudioConstraints::kGoogExperimentalNoiseSuppression); |
452 const bool goog_beamforming = audio_constraints.GetProperty( | |
453 MediaAudioConstraints::kGoogBeamforming); | |
448 const bool goog_high_pass_filter = audio_constraints.GetProperty( | 454 const bool goog_high_pass_filter = audio_constraints.GetProperty( |
449 MediaAudioConstraints::kGoogHighpassFilter); | 455 MediaAudioConstraints::kGoogHighpassFilter); |
450 | 456 |
451 // Return immediately if no goog constraint is enabled. | 457 // Return immediately if no goog constraint is enabled. |
452 if (!echo_cancellation && !goog_experimental_aec && !goog_ns && | 458 if (!echo_cancellation && !goog_experimental_aec && !goog_ns && |
453 !goog_high_pass_filter && !goog_typing_detection && | 459 !goog_high_pass_filter && !goog_typing_detection && |
454 !goog_agc && !goog_experimental_ns) { | 460 !goog_agc && !goog_experimental_ns && !goog_beamforming) { |
455 RecordProcessingState(AUDIO_PROCESSING_DISABLED); | 461 RecordProcessingState(AUDIO_PROCESSING_DISABLED); |
456 return; | 462 return; |
457 } | 463 } |
458 | 464 |
459 // Experimental options provided at creation. | 465 // Experimental options provided at creation. |
460 webrtc::Config config; | 466 webrtc::Config config; |
461 if (goog_experimental_aec) | 467 if (goog_experimental_aec) |
462 config.Set<webrtc::DelayCorrection>(new webrtc::DelayCorrection(true)); | 468 config.Set<webrtc::DelayCorrection>(new webrtc::DelayCorrection(true)); |
463 if (goog_experimental_ns) | 469 if (goog_experimental_ns) |
464 config.Set<webrtc::ExperimentalNs>(new webrtc::ExperimentalNs(true)); | 470 config.Set<webrtc::ExperimentalNs>(new webrtc::ExperimentalNs(true)); |
465 #if defined(OS_MACOSX) | 471 #if defined(OS_MACOSX) |
466 if (base::FieldTrialList::FindFullName("NoReportedDelayOnMac") == "Enabled") | 472 if (base::FieldTrialList::FindFullName("NoReportedDelayOnMac") == "Enabled") |
467 config.Set<webrtc::ReportedDelay>(new webrtc::ReportedDelay(false)); | 473 config.Set<webrtc::ReportedDelay>(new webrtc::ReportedDelay(false)); |
468 #endif | 474 #endif |
475 if (goog_beamforming) { | |
476 ConfigBeamforming(&config); | |
477 } | |
469 | 478 |
470 // Create and configure the webrtc::AudioProcessing. | 479 // Create and configure the webrtc::AudioProcessing. |
471 audio_processing_.reset(webrtc::AudioProcessing::Create(config)); | 480 audio_processing_.reset(webrtc::AudioProcessing::Create(config)); |
472 | 481 |
473 // Enable the audio processing components. | 482 // Enable the audio processing components. |
474 if (echo_cancellation) { | 483 if (echo_cancellation) { |
475 EnableEchoCancellation(audio_processing_.get()); | 484 EnableEchoCancellation(audio_processing_.get()); |
476 | 485 |
477 if (playout_data_source_) | 486 if (playout_data_source_) |
478 playout_data_source_->AddPlayoutSink(this); | 487 playout_data_source_->AddPlayoutSink(this); |
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494 typing_detector_.reset(new webrtc::TypingDetection()); | 503 typing_detector_.reset(new webrtc::TypingDetection()); |
495 EnableTypingDetection(audio_processing_.get(), typing_detector_.get()); | 504 EnableTypingDetection(audio_processing_.get(), typing_detector_.get()); |
496 } | 505 } |
497 | 506 |
498 if (goog_agc) | 507 if (goog_agc) |
499 EnableAutomaticGainControl(audio_processing_.get()); | 508 EnableAutomaticGainControl(audio_processing_.get()); |
500 | 509 |
501 RecordProcessingState(AUDIO_PROCESSING_ENABLED); | 510 RecordProcessingState(AUDIO_PROCESSING_ENABLED); |
502 } | 511 } |
503 | 512 |
513 void MediaStreamAudioProcessor::ConfigBeamforming(webrtc::Config* config) { | |
514 bool enabled = false; | |
515 std::vector<webrtc::Point> geometry; | |
516 geometry.push_back(webrtc::Point(0.f, 0.f, 0.f)); | |
517 #if defined(OS_CHROMEOS) | |
518 const std::string board = base::SysInfo::GetLsbReleaseBoard(); | |
519 if (board == "peach_pi") { | |
520 enabled = true; | |
521 geometry.push_back(webrtc::Point(0.05f, 0.f, 0.f)); | |
ajm
2015/01/08 00:03:17
nit: perhaps use 0.050f to indicate the measuremen
aluebs-chromium
2015/01/08 00:29:17
Done.
| |
522 } else if (board == "swanky") { | |
523 // TODO(aluebs): Verify beamforming works on Swanky and enable. | |
524 enabled = false; | |
525 geometry.push_back(webrtc::Point(0.052f, 0.f, 0.f)); | |
526 } | |
527 #endif | |
528 config->Set<webrtc::Beamforming>(new webrtc::Beamforming(enabled, geometry)); | |
529 } | |
530 | |
504 void MediaStreamAudioProcessor::InitializeCaptureFifo( | 531 void MediaStreamAudioProcessor::InitializeCaptureFifo( |
505 const media::AudioParameters& input_format) { | 532 const media::AudioParameters& input_format) { |
506 DCHECK(main_thread_checker_.CalledOnValidThread()); | 533 DCHECK(main_thread_checker_.CalledOnValidThread()); |
507 DCHECK(input_format.IsValid()); | 534 DCHECK(input_format.IsValid()); |
508 input_format_ = input_format; | 535 input_format_ = input_format; |
509 | 536 |
510 // TODO(ajm): For now, we assume fixed parameters for the output when audio | 537 // TODO(ajm): For now, we assume fixed parameters for the output when audio |
511 // processing is enabled, to match the previous behavior. We should either | 538 // processing is enabled, to match the previous behavior. We should either |
512 // use the input parameters (in which case, audio processing will convert | 539 // use the input parameters (in which case, audio processing will convert |
513 // at output) or ideally, have a backchannel from the sink to know what | 540 // at output) or ideally, have a backchannel from the sink to know what |
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644 vad->stream_has_voice()); | 671 vad->stream_has_voice()); |
645 base::subtle::Release_Store(&typing_detected_, detected); | 672 base::subtle::Release_Store(&typing_detected_, detected); |
646 } | 673 } |
647 | 674 |
648 // Return 0 if the volume hasn't been changed, and otherwise the new volume. | 675 // Return 0 if the volume hasn't been changed, and otherwise the new volume. |
649 return (agc->stream_analog_level() == volume) ? | 676 return (agc->stream_analog_level() == volume) ? |
650 0 : agc->stream_analog_level(); | 677 0 : agc->stream_analog_level(); |
651 } | 678 } |
652 | 679 |
653 } // namespace content | 680 } // namespace content |
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