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Side by Side Diff: webrtc/modules/audio_processing/audio_processing_impl.cc

Issue 2457783003: Added offline data logpoints and logging functionality to the gain controller (Closed)
Patch Set: Created 4 years, 1 month ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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428 render_.render_audio.reset(nullptr); 428 render_.render_audio.reset(nullptr);
429 render_.render_converter.reset(nullptr); 429 render_.render_converter.reset(nullptr);
430 } 430 }
431 capture_.capture_audio.reset( 431 capture_.capture_audio.reset(
432 new AudioBuffer(formats_.api_format.input_stream().num_frames(), 432 new AudioBuffer(formats_.api_format.input_stream().num_frames(),
433 formats_.api_format.input_stream().num_channels(), 433 formats_.api_format.input_stream().num_channels(),
434 capture_nonlocked_.capture_processing_format.num_frames(), 434 capture_nonlocked_.capture_processing_format.num_frames(),
435 capture_audiobuffer_num_channels, 435 capture_audiobuffer_num_channels,
436 formats_.api_format.output_stream().num_frames())); 436 formats_.api_format.output_stream().num_frames()));
437 437
438 public_submodules_->gain_control->Initialize(num_proc_channels(),
439 proc_sample_rate_hz());
440
441 public_submodules_->echo_cancellation->Initialize( 438 public_submodules_->echo_cancellation->Initialize(
442 proc_sample_rate_hz(), num_reverse_channels(), num_output_channels(), 439 proc_sample_rate_hz(), num_reverse_channels(), num_output_channels(),
443 num_proc_channels()); 440 num_proc_channels());
444 AllocateRenderQueue(); 441 AllocateRenderQueue();
445 442
446 public_submodules_->echo_control_mobile->Initialize( 443 public_submodules_->echo_control_mobile->Initialize(
447 proc_split_sample_rate_hz(), num_reverse_channels(), 444 proc_split_sample_rate_hz(), num_reverse_channels(),
448 num_output_channels()); 445 num_output_channels());
446
peah-webrtc 2016/10/28 08:37:49 I moved the initialization of the gain_control to
447 public_submodules_->gain_control->Initialize(num_proc_channels(),
448 proc_sample_rate_hz());
449 if (constants_.use_experimental_agc) { 449 if (constants_.use_experimental_agc) {
450 if (!private_submodules_->agc_manager.get()) { 450 if (!private_submodules_->agc_manager.get()) {
451 private_submodules_->agc_manager.reset(new AgcManagerDirect( 451 private_submodules_->agc_manager.reset(new AgcManagerDirect(
452 public_submodules_->gain_control.get(), 452 public_submodules_->gain_control.get(),
453 public_submodules_->gain_control_for_experimental_agc.get(), 453 public_submodules_->gain_control_for_experimental_agc.get(),
454 constants_.agc_startup_min_volume)); 454 constants_.agc_startup_min_volume));
455 } 455 }
456 private_submodules_->agc_manager->Initialize(); 456 private_submodules_->agc_manager->Initialize();
457 private_submodules_->agc_manager->SetCaptureMuted( 457 private_submodules_->agc_manager->SetCaptureMuted(
458 capture_.output_will_be_muted); 458 capture_.output_will_be_muted);
459 public_submodules_->gain_control_for_experimental_agc->Initialize();
peah-webrtc 2016/10/28 08:37:50 This is needed to ensure that a new set of data du
459 } 460 }
460 InitializeTransient(); 461 InitializeTransient();
461 InitializeBeamformer(); 462 InitializeBeamformer();
462 #if WEBRTC_INTELLIGIBILITY_ENHANCER 463 #if WEBRTC_INTELLIGIBILITY_ENHANCER
463 InitializeIntelligibility(); 464 InitializeIntelligibility();
464 #endif 465 #endif
465 public_submodules_->high_pass_filter->Initialize(num_proc_channels(), 466 public_submodules_->high_pass_filter->Initialize(num_proc_channels(),
466 proc_sample_rate_hz()); 467 proc_sample_rate_hz());
467 public_submodules_->noise_suppression->Initialize(num_proc_channels(), 468 public_submodules_->noise_suppression->Initialize(num_proc_channels(),
468 proc_sample_rate_hz()); 469 proc_sample_rate_hz());
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1733 capture_processing_format(kSampleRate16kHz), 1734 capture_processing_format(kSampleRate16kHz),
1734 split_rate(kSampleRate16kHz) {} 1735 split_rate(kSampleRate16kHz) {}
1735 1736
1736 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default; 1737 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
1737 1738
1738 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default; 1739 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
1739 1740
1740 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default; 1741 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
1741 1742
1742 } // namespace webrtc 1743 } // namespace webrtc
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