| /* |
| * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| #include "modules/audio_processing/aec3/echo_canceller3.h" |
| |
| #include <algorithm> |
| #include <utility> |
| |
| #include "modules/audio_processing/aec3/aec3_common.h" |
| #include "modules/audio_processing/high_pass_filter.h" |
| #include "modules/audio_processing/logging/apm_data_dumper.h" |
| #include "rtc_base/atomic_ops.h" |
| #include "rtc_base/experiments/field_trial_parser.h" |
| #include "rtc_base/logging.h" |
| #include "system_wrappers/include/field_trial.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| enum class EchoCanceller3ApiCall { kCapture, kRender }; |
| |
| bool DetectSaturation(rtc::ArrayView<const float> y) { |
| for (size_t k = 0; k < y.size(); ++k) { |
| if (y[k] >= 32700.0f || y[k] <= -32700.0f) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| // Retrieves a value from a field trial if it is available. If no value is |
| // present, the default value is returned. If the retrieved value is beyond the |
| // specified limits, the default value is returned instead. |
| void RetrieveFieldTrialValue(const char* trial_name, |
| float min, |
| float max, |
| float* value_to_update) { |
| const std::string field_trial_str = field_trial::FindFullName(trial_name); |
| |
| FieldTrialParameter<double> field_trial_param(/*key=*/"", *value_to_update); |
| |
| ParseFieldTrial({&field_trial_param}, field_trial_str); |
| float field_trial_value = static_cast<float>(field_trial_param.Get()); |
| |
| if (field_trial_value >= min && field_trial_value <= max && |
| field_trial_value != *value_to_update) { |
| RTC_LOG(LS_INFO) << "Key " << trial_name |
| << " changing AEC3 parameter value from " |
| << *value_to_update << " to " << field_trial_value; |
| *value_to_update = field_trial_value; |
| } |
| } |
| |
| void RetrieveFieldTrialValue(const char* trial_name, |
| int min, |
| int max, |
| int* value_to_update) { |
| const std::string field_trial_str = field_trial::FindFullName(trial_name); |
| |
| FieldTrialParameter<int> field_trial_param(/*key=*/"", *value_to_update); |
| |
| ParseFieldTrial({&field_trial_param}, field_trial_str); |
| float field_trial_value = field_trial_param.Get(); |
| |
| if (field_trial_value >= min && field_trial_value <= max && |
| field_trial_value != *value_to_update) { |
| RTC_LOG(LS_INFO) << "Key " << trial_name |
| << " changing AEC3 parameter value from " |
| << *value_to_update << " to " << field_trial_value; |
| *value_to_update = field_trial_value; |
| } |
| } |
| |
| void FillSubFrameView( |
| AudioBuffer* frame, |
| size_t sub_frame_index, |
| std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame_view) { |
| RTC_DCHECK_GE(1, sub_frame_index); |
| RTC_DCHECK_LE(0, sub_frame_index); |
| RTC_DCHECK_EQ(frame->num_bands(), sub_frame_view->size()); |
| RTC_DCHECK_EQ(frame->num_channels(), (*sub_frame_view)[0].size()); |
| for (size_t band = 0; band < sub_frame_view->size(); ++band) { |
| for (size_t channel = 0; channel < (*sub_frame_view)[0].size(); ++channel) { |
| (*sub_frame_view)[band][channel] = rtc::ArrayView<float>( |
| &frame->split_bands(channel)[band][sub_frame_index * kSubFrameLength], |
| kSubFrameLength); |
| } |
| } |
| } |
| |
| void FillSubFrameView( |
| bool proper_downmix_needed, |
| std::vector<std::vector<std::vector<float>>>* frame, |
| size_t sub_frame_index, |
| std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame_view) { |
| RTC_DCHECK_GE(1, sub_frame_index); |
| RTC_DCHECK_EQ(frame->size(), sub_frame_view->size()); |
| const size_t frame_num_channels = (*frame)[0].size(); |
| const size_t sub_frame_num_channels = (*sub_frame_view)[0].size(); |
| if (frame_num_channels > sub_frame_num_channels) { |
| RTC_DCHECK_EQ(sub_frame_num_channels, 1u); |
| if (proper_downmix_needed) { |
| // When a proper downmix is needed (which is the case when proper stereo |
| // is present in the echo reference signal but the echo canceller does the |
| // processing in mono) downmix the echo reference by averaging the channel |
| // content (otherwise downmixing is done by selecting channel 0). |
| for (size_t band = 0; band < frame->size(); ++band) { |
| for (size_t ch = 1; ch < frame_num_channels; ++ch) { |
| for (size_t k = 0; k < kSubFrameLength; ++k) { |
| (*frame)[band][/*channel=*/0] |
| [sub_frame_index * kSubFrameLength + k] += |
| (*frame)[band][ch][sub_frame_index * kSubFrameLength + k]; |
| } |
| } |
| const float one_by_num_channels = 1.0f / frame_num_channels; |
| for (size_t k = 0; k < kSubFrameLength; ++k) { |
| (*frame)[band][/*channel=*/0][sub_frame_index * kSubFrameLength + |
| k] *= one_by_num_channels; |
| } |
| } |
| } |
| for (size_t band = 0; band < frame->size(); ++band) { |
| (*sub_frame_view)[band][/*channel=*/0] = rtc::ArrayView<float>( |
| &(*frame)[band][/*channel=*/0][sub_frame_index * kSubFrameLength], |
| kSubFrameLength); |
| } |
| } else { |
| RTC_DCHECK_EQ(frame_num_channels, sub_frame_num_channels); |
| for (size_t band = 0; band < frame->size(); ++band) { |
| for (size_t channel = 0; channel < (*frame)[band].size(); ++channel) { |
| (*sub_frame_view)[band][channel] = rtc::ArrayView<float>( |
| &(*frame)[band][channel][sub_frame_index * kSubFrameLength], |
| kSubFrameLength); |
| } |
| } |
| } |
| } |
| |
| void ProcessCaptureFrameContent( |
| AudioBuffer* linear_output, |
| AudioBuffer* capture, |
| bool level_change, |
| bool aec_reference_is_downmixed_stereo, |
| bool saturated_microphone_signal, |
| size_t sub_frame_index, |
| FrameBlocker* capture_blocker, |
| BlockFramer* linear_output_framer, |
| BlockFramer* output_framer, |
| BlockProcessor* block_processor, |
| Block* linear_output_block, |
| std::vector<std::vector<rtc::ArrayView<float>>>* |
| linear_output_sub_frame_view, |
| Block* capture_block, |
| std::vector<std::vector<rtc::ArrayView<float>>>* capture_sub_frame_view) { |
| FillSubFrameView(capture, sub_frame_index, capture_sub_frame_view); |
| |
| if (linear_output) { |
| RTC_DCHECK(linear_output_framer); |
| RTC_DCHECK(linear_output_block); |
| RTC_DCHECK(linear_output_sub_frame_view); |
| FillSubFrameView(linear_output, sub_frame_index, |
| linear_output_sub_frame_view); |
| } |
| |
| capture_blocker->InsertSubFrameAndExtractBlock(*capture_sub_frame_view, |
| capture_block); |
| block_processor->ProcessCapture( |
| /*echo_path_gain_change=*/level_change || |
| aec_reference_is_downmixed_stereo, |
| saturated_microphone_signal, linear_output_block, capture_block); |
| output_framer->InsertBlockAndExtractSubFrame(*capture_block, |
| capture_sub_frame_view); |
| |
| if (linear_output) { |
| RTC_DCHECK(linear_output_framer); |
| linear_output_framer->InsertBlockAndExtractSubFrame( |
| *linear_output_block, linear_output_sub_frame_view); |
| } |
| } |
| |
| void ProcessRemainingCaptureFrameContent(bool level_change, |
| bool aec_reference_is_downmixed_stereo, |
| bool saturated_microphone_signal, |
| FrameBlocker* capture_blocker, |
| BlockFramer* linear_output_framer, |
| BlockFramer* output_framer, |
| BlockProcessor* block_processor, |
| Block* linear_output_block, |
| Block* block) { |
| if (!capture_blocker->IsBlockAvailable()) { |
| return; |
| } |
| |
| capture_blocker->ExtractBlock(block); |
| block_processor->ProcessCapture( |
| /*echo_path_gain_change=*/level_change || |
| aec_reference_is_downmixed_stereo, |
| saturated_microphone_signal, linear_output_block, block); |
| output_framer->InsertBlock(*block); |
| |
| if (linear_output_framer) { |
| RTC_DCHECK(linear_output_block); |
| linear_output_framer->InsertBlock(*linear_output_block); |
| } |
| } |
| |
| void BufferRenderFrameContent( |
| bool proper_downmix_needed, |
| std::vector<std::vector<std::vector<float>>>* render_frame, |
| size_t sub_frame_index, |
| FrameBlocker* render_blocker, |
| BlockProcessor* block_processor, |
| Block* block, |
| std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame_view) { |
| FillSubFrameView(proper_downmix_needed, render_frame, sub_frame_index, |
| sub_frame_view); |
| render_blocker->InsertSubFrameAndExtractBlock(*sub_frame_view, block); |
| block_processor->BufferRender(*block); |
| } |
| |
| void BufferRemainingRenderFrameContent(FrameBlocker* render_blocker, |
| BlockProcessor* block_processor, |
| Block* block) { |
| if (!render_blocker->IsBlockAvailable()) { |
| return; |
| } |
| render_blocker->ExtractBlock(block); |
| block_processor->BufferRender(*block); |
| } |
| |
| void CopyBufferIntoFrame(const AudioBuffer& buffer, |
| size_t num_bands, |
| size_t num_channels, |
| std::vector<std::vector<std::vector<float>>>* frame) { |
| RTC_DCHECK_EQ(num_bands, frame->size()); |
| RTC_DCHECK_EQ(num_channels, (*frame)[0].size()); |
| RTC_DCHECK_EQ(AudioBuffer::kSplitBandSize, (*frame)[0][0].size()); |
| for (size_t band = 0; band < num_bands; ++band) { |
| for (size_t channel = 0; channel < num_channels; ++channel) { |
| rtc::ArrayView<const float> buffer_view( |
| &buffer.split_bands_const(channel)[band][0], |
| AudioBuffer::kSplitBandSize); |
| std::copy(buffer_view.begin(), buffer_view.end(), |
| (*frame)[band][channel].begin()); |
| } |
| } |
| } |
| |
| } // namespace |
| |
| // TODO(webrtc:5298): Move this to a separate file. |
| EchoCanceller3Config AdjustConfig(const EchoCanceller3Config& config) { |
| EchoCanceller3Config adjusted_cfg = config; |
| |
| if (field_trial::IsEnabled("WebRTC-Aec3StereoContentDetectionKillSwitch")) { |
| adjusted_cfg.multi_channel.detect_stereo_content = false; |
| } |
| |
| if (field_trial::IsEnabled("WebRTC-Aec3AntiHowlingMinimizationKillSwitch")) { |
| adjusted_cfg.suppressor.high_bands_suppression |
| .anti_howling_activation_threshold = 25.f; |
| adjusted_cfg.suppressor.high_bands_suppression.anti_howling_gain = 0.01f; |
| } |
| |
| if (field_trial::IsEnabled("WebRTC-Aec3UseShortConfigChangeDuration")) { |
| adjusted_cfg.filter.config_change_duration_blocks = 10; |
| } |
| |
| if (field_trial::IsEnabled("WebRTC-Aec3UseZeroInitialStateDuration")) { |
| adjusted_cfg.filter.initial_state_seconds = 0.f; |
| } else if (field_trial::IsEnabled( |
| "WebRTC-Aec3UseDot1SecondsInitialStateDuration")) { |
| adjusted_cfg.filter.initial_state_seconds = .1f; |
| } else if (field_trial::IsEnabled( |
| "WebRTC-Aec3UseDot2SecondsInitialStateDuration")) { |
| adjusted_cfg.filter.initial_state_seconds = .2f; |
| } else if (field_trial::IsEnabled( |
| "WebRTC-Aec3UseDot3SecondsInitialStateDuration")) { |
| adjusted_cfg.filter.initial_state_seconds = .3f; |
| } else if (field_trial::IsEnabled( |
| "WebRTC-Aec3UseDot6SecondsInitialStateDuration")) { |
| adjusted_cfg.filter.initial_state_seconds = .6f; |
| } else if (field_trial::IsEnabled( |
| "WebRTC-Aec3UseDot9SecondsInitialStateDuration")) { |
| adjusted_cfg.filter.initial_state_seconds = .9f; |
| } else if (field_trial::IsEnabled( |
| "WebRTC-Aec3Use1Dot2SecondsInitialStateDuration")) { |
| adjusted_cfg.filter.initial_state_seconds = 1.2f; |
| } else if (field_trial::IsEnabled( |
| "WebRTC-Aec3Use1Dot6SecondsInitialStateDuration")) { |
| adjusted_cfg.filter.initial_state_seconds = 1.6f; |
| } else if (field_trial::IsEnabled( |
| "WebRTC-Aec3Use2Dot0SecondsInitialStateDuration")) { |
| adjusted_cfg.filter.initial_state_seconds = 2.0f; |
| } |
| |
| if (field_trial::IsEnabled("WebRTC-Aec3HighPassFilterEchoReference")) { |
| adjusted_cfg.filter.high_pass_filter_echo_reference = true; |
| } |
| |
| if (field_trial::IsEnabled("WebRTC-Aec3EchoSaturationDetectionKillSwitch")) { |
| adjusted_cfg.ep_strength.echo_can_saturate = false; |
| } |
| |
| const std::string use_nearend_reverb_len_tunings = |
| field_trial::FindFullName("WebRTC-Aec3UseNearendReverbLen"); |
| FieldTrialParameter<double> nearend_reverb_default_len( |
| "default_len", adjusted_cfg.ep_strength.default_len); |
| FieldTrialParameter<double> nearend_reverb_nearend_len( |
| "nearend_len", adjusted_cfg.ep_strength.nearend_len); |
| |
| ParseFieldTrial({&nearend_reverb_default_len, &nearend_reverb_nearend_len}, |
| use_nearend_reverb_len_tunings); |
| float default_len = static_cast<float>(nearend_reverb_default_len.Get()); |
| float nearend_len = static_cast<float>(nearend_reverb_nearend_len.Get()); |
| if (default_len > -1 && default_len < 1 && nearend_len > -1 && |
| nearend_len < 1) { |
| adjusted_cfg.ep_strength.default_len = |
| static_cast<float>(nearend_reverb_default_len.Get()); |
| adjusted_cfg.ep_strength.nearend_len = |
| static_cast<float>(nearend_reverb_nearend_len.Get()); |
| } |
| |
| if (field_trial::IsEnabled("WebRTC-Aec3ConservativeTailFreqResponse")) { |
| adjusted_cfg.ep_strength.use_conservative_tail_frequency_response = true; |
| } |
| |
| if (field_trial::IsDisabled("WebRTC-Aec3ConservativeTailFreqResponse")) { |
| adjusted_cfg.ep_strength.use_conservative_tail_frequency_response = false; |
| } |
| |
| if (field_trial::IsEnabled("WebRTC-Aec3ShortHeadroomKillSwitch")) { |
| // Two blocks headroom. |
| adjusted_cfg.delay.delay_headroom_samples = kBlockSize * 2; |
| } |
| |
| if (field_trial::IsEnabled("WebRTC-Aec3ClampInstQualityToZeroKillSwitch")) { |
| adjusted_cfg.erle.clamp_quality_estimate_to_zero = false; |
| } |
| |
| if (field_trial::IsEnabled("WebRTC-Aec3ClampInstQualityToOneKillSwitch")) { |
| adjusted_cfg.erle.clamp_quality_estimate_to_one = false; |
| } |
| |
| if (field_trial::IsEnabled("WebRTC-Aec3OnsetDetectionKillSwitch")) { |
| adjusted_cfg.erle.onset_detection = false; |
| } |
| |
| if (field_trial::IsEnabled( |
| "WebRTC-Aec3EnforceRenderDelayEstimationDownmixing")) { |
| adjusted_cfg.delay.render_alignment_mixing.downmix = true; |
| adjusted_cfg.delay.render_alignment_mixing.adaptive_selection = false; |
| } |
| |
| if (field_trial::IsEnabled( |
| "WebRTC-Aec3EnforceCaptureDelayEstimationDownmixing")) { |
| adjusted_cfg.delay.capture_alignment_mixing.downmix = true; |
| adjusted_cfg.delay.capture_alignment_mixing.adaptive_selection = false; |
| } |
| |
| if (field_trial::IsEnabled( |
| "WebRTC-Aec3EnforceCaptureDelayEstimationLeftRightPrioritization")) { |
| adjusted_cfg.delay.capture_alignment_mixing.prefer_first_two_channels = |
| true; |
| } |
| |
| if (field_trial::IsEnabled( |
| "WebRTC-" |
| "Aec3RenderDelayEstimationLeftRightPrioritizationKillSwitch")) { |
| adjusted_cfg.delay.capture_alignment_mixing.prefer_first_two_channels = |
| false; |
| } |
| |
| if (field_trial::IsEnabled("WebRTC-Aec3SensitiveDominantNearendActivation")) { |
| adjusted_cfg.suppressor.dominant_nearend_detection.enr_threshold = 0.5f; |
| } else if (field_trial::IsEnabled( |
| "WebRTC-Aec3VerySensitiveDominantNearendActivation")) { |
| adjusted_cfg.suppressor.dominant_nearend_detection.enr_threshold = 0.75f; |
| } |
| |
| if (field_trial::IsEnabled("WebRTC-Aec3TransparentAntiHowlingGain")) { |
| adjusted_cfg.suppressor.high_bands_suppression.anti_howling_gain = 1.f; |
| } |
| |
| if (field_trial::IsEnabled( |
| "WebRTC-Aec3EnforceMoreTransparentNormalSuppressorTuning")) { |
| adjusted_cfg.suppressor.normal_tuning.mask_lf.enr_transparent = 0.4f; |
| adjusted_cfg.suppressor.normal_tuning.mask_lf.enr_suppress = 0.5f; |
| } |
| |
| if (field_trial::IsEnabled( |
| "WebRTC-Aec3EnforceMoreTransparentNearendSuppressorTuning")) { |
| adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_transparent = 1.29f; |
| adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_suppress = 1.3f; |
| } |
| |
| if (field_trial::IsEnabled( |
| "WebRTC-Aec3EnforceMoreTransparentNormalSuppressorHfTuning")) { |
| adjusted_cfg.suppressor.normal_tuning.mask_hf.enr_transparent = 0.3f; |
| adjusted_cfg.suppressor.normal_tuning.mask_hf.enr_suppress = 0.4f; |
| } |
| |
| if (field_trial::IsEnabled( |
| "WebRTC-Aec3EnforceMoreTransparentNearendSuppressorHfTuning")) { |
| adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_transparent = 1.09f; |
| adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_suppress = 1.1f; |
| } |
| |
| if (field_trial::IsEnabled( |
| "WebRTC-Aec3EnforceRapidlyAdjustingNormalSuppressorTunings")) { |
| adjusted_cfg.suppressor.normal_tuning.max_inc_factor = 2.5f; |
| } |
| |
| if (field_trial::IsEnabled( |
| "WebRTC-Aec3EnforceRapidlyAdjustingNearendSuppressorTunings")) { |
| adjusted_cfg.suppressor.nearend_tuning.max_inc_factor = 2.5f; |
| } |
| |
| if (field_trial::IsEnabled( |
| "WebRTC-Aec3EnforceSlowlyAdjustingNormalSuppressorTunings")) { |
| adjusted_cfg.suppressor.normal_tuning.max_dec_factor_lf = .2f; |
| } |
| |
| if (field_trial::IsEnabled( |
| "WebRTC-Aec3EnforceSlowlyAdjustingNearendSuppressorTunings")) { |
| adjusted_cfg.suppressor.nearend_tuning.max_dec_factor_lf = .2f; |
| } |
| |
| if (field_trial::IsEnabled("WebRTC-Aec3EnforceConservativeHfSuppression")) { |
| adjusted_cfg.suppressor.conservative_hf_suppression = true; |
| } |
| |
| if (field_trial::IsEnabled("WebRTC-Aec3EnforceStationarityProperties")) { |
| adjusted_cfg.echo_audibility.use_stationarity_properties = true; |
| } |
| |
| if (field_trial::IsEnabled( |
| "WebRTC-Aec3EnforceStationarityPropertiesAtInit")) { |
| adjusted_cfg.echo_audibility.use_stationarity_properties_at_init = true; |
| } |
| |
| if (field_trial::IsEnabled("WebRTC-Aec3EnforceLowActiveRenderLimit")) { |
| adjusted_cfg.render_levels.active_render_limit = 50.f; |
| } else if (field_trial::IsEnabled( |
| "WebRTC-Aec3EnforceVeryLowActiveRenderLimit")) { |
| adjusted_cfg.render_levels.active_render_limit = 30.f; |
| } |
| |
| if (field_trial::IsEnabled("WebRTC-Aec3NonlinearModeReverbKillSwitch")) { |
| adjusted_cfg.echo_model.model_reverb_in_nonlinear_mode = false; |
| } |
| |
| // Field-trial based override for the whole suppressor tuning. |
| const std::string suppressor_tuning_override_trial_name = |
| field_trial::FindFullName("WebRTC-Aec3SuppressorTuningOverride"); |
| |
| FieldTrialParameter<double> nearend_tuning_mask_lf_enr_transparent( |
| "nearend_tuning_mask_lf_enr_transparent", |
| adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_transparent); |
| FieldTrialParameter<double> nearend_tuning_mask_lf_enr_suppress( |
| "nearend_tuning_mask_lf_enr_suppress", |
| adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_suppress); |
| FieldTrialParameter<double> nearend_tuning_mask_hf_enr_transparent( |
| "nearend_tuning_mask_hf_enr_transparent", |
| adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_transparent); |
| FieldTrialParameter<double> nearend_tuning_mask_hf_enr_suppress( |
| "nearend_tuning_mask_hf_enr_suppress", |
| adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_suppress); |
| FieldTrialParameter<double> nearend_tuning_max_inc_factor( |
| "nearend_tuning_max_inc_factor", |
| adjusted_cfg.suppressor.nearend_tuning.max_inc_factor); |
| FieldTrialParameter<double> nearend_tuning_max_dec_factor_lf( |
| "nearend_tuning_max_dec_factor_lf", |
| adjusted_cfg.suppressor.nearend_tuning.max_dec_factor_lf); |
| FieldTrialParameter<double> normal_tuning_mask_lf_enr_transparent( |
| "normal_tuning_mask_lf_enr_transparent", |
| adjusted_cfg.suppressor.normal_tuning.mask_lf.enr_transparent); |
| FieldTrialParameter<double> normal_tuning_mask_lf_enr_suppress( |
| "normal_tuning_mask_lf_enr_suppress", |
| adjusted_cfg.suppressor.normal_tuning.mask_lf.enr_suppress); |
| FieldTrialParameter<double> normal_tuning_mask_hf_enr_transparent( |
| "normal_tuning_mask_hf_enr_transparent", |
| adjusted_cfg.suppressor.normal_tuning.mask_hf.enr_transparent); |
| FieldTrialParameter<double> normal_tuning_mask_hf_enr_suppress( |
| "normal_tuning_mask_hf_enr_suppress", |
| adjusted_cfg.suppressor.normal_tuning.mask_hf.enr_suppress); |
| FieldTrialParameter<double> normal_tuning_max_inc_factor( |
| "normal_tuning_max_inc_factor", |
| adjusted_cfg.suppressor.normal_tuning.max_inc_factor); |
| FieldTrialParameter<double> normal_tuning_max_dec_factor_lf( |
| "normal_tuning_max_dec_factor_lf", |
| adjusted_cfg.suppressor.normal_tuning.max_dec_factor_lf); |
| FieldTrialParameter<double> dominant_nearend_detection_enr_threshold( |
| "dominant_nearend_detection_enr_threshold", |
| adjusted_cfg.suppressor.dominant_nearend_detection.enr_threshold); |
| FieldTrialParameter<double> dominant_nearend_detection_enr_exit_threshold( |
| "dominant_nearend_detection_enr_exit_threshold", |
| adjusted_cfg.suppressor.dominant_nearend_detection.enr_exit_threshold); |
| FieldTrialParameter<double> dominant_nearend_detection_snr_threshold( |
| "dominant_nearend_detection_snr_threshold", |
| adjusted_cfg.suppressor.dominant_nearend_detection.snr_threshold); |
| FieldTrialParameter<int> dominant_nearend_detection_hold_duration( |
| "dominant_nearend_detection_hold_duration", |
| adjusted_cfg.suppressor.dominant_nearend_detection.hold_duration); |
| FieldTrialParameter<int> dominant_nearend_detection_trigger_threshold( |
| "dominant_nearend_detection_trigger_threshold", |
| adjusted_cfg.suppressor.dominant_nearend_detection.trigger_threshold); |
| |
| ParseFieldTrial( |
| {&nearend_tuning_mask_lf_enr_transparent, |
| &nearend_tuning_mask_lf_enr_suppress, |
| &nearend_tuning_mask_hf_enr_transparent, |
| &nearend_tuning_mask_hf_enr_suppress, &nearend_tuning_max_inc_factor, |
| &nearend_tuning_max_dec_factor_lf, |
| &normal_tuning_mask_lf_enr_transparent, |
| &normal_tuning_mask_lf_enr_suppress, |
| &normal_tuning_mask_hf_enr_transparent, |
| &normal_tuning_mask_hf_enr_suppress, &normal_tuning_max_inc_factor, |
| &normal_tuning_max_dec_factor_lf, |
| &dominant_nearend_detection_enr_threshold, |
| &dominant_nearend_detection_enr_exit_threshold, |
| &dominant_nearend_detection_snr_threshold, |
| &dominant_nearend_detection_hold_duration, |
| &dominant_nearend_detection_trigger_threshold}, |
| suppressor_tuning_override_trial_name); |
| |
| adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_transparent = |
| static_cast<float>(nearend_tuning_mask_lf_enr_transparent.Get()); |
| adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_suppress = |
| static_cast<float>(nearend_tuning_mask_lf_enr_suppress.Get()); |
| adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_transparent = |
| static_cast<float>(nearend_tuning_mask_hf_enr_transparent.Get()); |
| adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_suppress = |
| static_cast<float>(nearend_tuning_mask_hf_enr_suppress.Get()); |
| adjusted_cfg.suppressor.nearend_tuning.max_inc_factor = |
| static_cast<float>(nearend_tuning_max_inc_factor.Get()); |
| adjusted_cfg.suppressor.nearend_tuning.max_dec_factor_lf = |
| static_cast<float>(nearend_tuning_max_dec_factor_lf.Get()); |
| adjusted_cfg.suppressor.normal_tuning.mask_lf.enr_transparent = |
| static_cast<float>(normal_tuning_mask_lf_enr_transparent.Get()); |
| adjusted_cfg.suppressor.normal_tuning.mask_lf.enr_suppress = |
| static_cast<float>(normal_tuning_mask_lf_enr_suppress.Get()); |
| adjusted_cfg.suppressor.normal_tuning.mask_hf.enr_transparent = |
| static_cast<float>(normal_tuning_mask_hf_enr_transparent.Get()); |
| adjusted_cfg.suppressor.normal_tuning.mask_hf.enr_suppress = |
| static_cast<float>(normal_tuning_mask_hf_enr_suppress.Get()); |
| adjusted_cfg.suppressor.normal_tuning.max_inc_factor = |
| static_cast<float>(normal_tuning_max_inc_factor.Get()); |
| adjusted_cfg.suppressor.normal_tuning.max_dec_factor_lf = |
| static_cast<float>(normal_tuning_max_dec_factor_lf.Get()); |
| adjusted_cfg.suppressor.dominant_nearend_detection.enr_threshold = |
| static_cast<float>(dominant_nearend_detection_enr_threshold.Get()); |
| adjusted_cfg.suppressor.dominant_nearend_detection.enr_exit_threshold = |
| static_cast<float>(dominant_nearend_detection_enr_exit_threshold.Get()); |
| adjusted_cfg.suppressor.dominant_nearend_detection.snr_threshold = |
| static_cast<float>(dominant_nearend_detection_snr_threshold.Get()); |
| adjusted_cfg.suppressor.dominant_nearend_detection.hold_duration = |
| dominant_nearend_detection_hold_duration.Get(); |
| adjusted_cfg.suppressor.dominant_nearend_detection.trigger_threshold = |
| dominant_nearend_detection_trigger_threshold.Get(); |
| |
| // Field trial-based overrides of individual suppressor parameters. |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3SuppressorNearendLfMaskTransparentOverride", 0.f, 10.f, |
| &adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_transparent); |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3SuppressorNearendLfMaskSuppressOverride", 0.f, 10.f, |
| &adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_suppress); |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3SuppressorNearendHfMaskTransparentOverride", 0.f, 10.f, |
| &adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_transparent); |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3SuppressorNearendHfMaskSuppressOverride", 0.f, 10.f, |
| &adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_suppress); |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3SuppressorNearendMaxIncFactorOverride", 0.f, 10.f, |
| &adjusted_cfg.suppressor.nearend_tuning.max_inc_factor); |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3SuppressorNearendMaxDecFactorLfOverride", 0.f, 10.f, |
| &adjusted_cfg.suppressor.nearend_tuning.max_dec_factor_lf); |
| |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3SuppressorNormalLfMaskTransparentOverride", 0.f, 10.f, |
| &adjusted_cfg.suppressor.normal_tuning.mask_lf.enr_transparent); |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3SuppressorNormalLfMaskSuppressOverride", 0.f, 10.f, |
| &adjusted_cfg.suppressor.normal_tuning.mask_lf.enr_suppress); |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3SuppressorNormalHfMaskTransparentOverride", 0.f, 10.f, |
| &adjusted_cfg.suppressor.normal_tuning.mask_hf.enr_transparent); |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3SuppressorNormalHfMaskSuppressOverride", 0.f, 10.f, |
| &adjusted_cfg.suppressor.normal_tuning.mask_hf.enr_suppress); |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3SuppressorNormalMaxIncFactorOverride", 0.f, 10.f, |
| &adjusted_cfg.suppressor.normal_tuning.max_inc_factor); |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3SuppressorNormalMaxDecFactorLfOverride", 0.f, 10.f, |
| &adjusted_cfg.suppressor.normal_tuning.max_dec_factor_lf); |
| |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3SuppressorDominantNearendEnrThresholdOverride", 0.f, 100.f, |
| &adjusted_cfg.suppressor.dominant_nearend_detection.enr_threshold); |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3SuppressorDominantNearendEnrExitThresholdOverride", 0.f, |
| 100.f, |
| &adjusted_cfg.suppressor.dominant_nearend_detection.enr_exit_threshold); |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3SuppressorDominantNearendSnrThresholdOverride", 0.f, 100.f, |
| &adjusted_cfg.suppressor.dominant_nearend_detection.snr_threshold); |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3SuppressorDominantNearendHoldDurationOverride", 0, 1000, |
| &adjusted_cfg.suppressor.dominant_nearend_detection.hold_duration); |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3SuppressorDominantNearendTriggerThresholdOverride", 0, 1000, |
| &adjusted_cfg.suppressor.dominant_nearend_detection.trigger_threshold); |
| |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3SuppressorAntiHowlingGainOverride", 0.f, 10.f, |
| &adjusted_cfg.suppressor.high_bands_suppression.anti_howling_gain); |
| |
| // Field trial-based overrides of individual delay estimator parameters. |
| RetrieveFieldTrialValue("WebRTC-Aec3DelayEstimateSmoothingOverride", 0.f, 1.f, |
| &adjusted_cfg.delay.delay_estimate_smoothing); |
| RetrieveFieldTrialValue( |
| "WebRTC-Aec3DelayEstimateSmoothingDelayFoundOverride", 0.f, 1.f, |
| &adjusted_cfg.delay.delay_estimate_smoothing_delay_found); |
| |
| return adjusted_cfg; |
| } |
| |
| class EchoCanceller3::RenderWriter { |
| public: |
| RenderWriter(ApmDataDumper* data_dumper, |
| const EchoCanceller3Config& config, |
| SwapQueue<std::vector<std::vector<std::vector<float>>>, |
| Aec3RenderQueueItemVerifier>* render_transfer_queue, |
| size_t num_bands, |
| size_t num_channels); |
| |
| RenderWriter() = delete; |
| RenderWriter(const RenderWriter&) = delete; |
| RenderWriter& operator=(const RenderWriter&) = delete; |
| |
| ~RenderWriter(); |
| void Insert(const AudioBuffer& input); |
| |
| private: |
| ApmDataDumper* data_dumper_; |
| const size_t num_bands_; |
| const size_t num_channels_; |
| std::unique_ptr<HighPassFilter> high_pass_filter_; |
| std::vector<std::vector<std::vector<float>>> render_queue_input_frame_; |
| SwapQueue<std::vector<std::vector<std::vector<float>>>, |
| Aec3RenderQueueItemVerifier>* render_transfer_queue_; |
| }; |
| |
| EchoCanceller3::RenderWriter::RenderWriter( |
| ApmDataDumper* data_dumper, |
| const EchoCanceller3Config& config, |
| SwapQueue<std::vector<std::vector<std::vector<float>>>, |
| Aec3RenderQueueItemVerifier>* render_transfer_queue, |
| size_t num_bands, |
| size_t num_channels) |
| : data_dumper_(data_dumper), |
| num_bands_(num_bands), |
| num_channels_(num_channels), |
| render_queue_input_frame_( |
| num_bands_, |
| std::vector<std::vector<float>>( |
| num_channels_, |
| std::vector<float>(AudioBuffer::kSplitBandSize, 0.f))), |
| render_transfer_queue_(render_transfer_queue) { |
| RTC_DCHECK(data_dumper); |
| if (config.filter.high_pass_filter_echo_reference) { |
| high_pass_filter_ = std::make_unique<HighPassFilter>(16000, num_channels); |
| } |
| } |
| |
| EchoCanceller3::RenderWriter::~RenderWriter() = default; |
| |
| void EchoCanceller3::RenderWriter::Insert(const AudioBuffer& input) { |
| RTC_DCHECK_EQ(AudioBuffer::kSplitBandSize, input.num_frames_per_band()); |
| RTC_DCHECK_EQ(num_bands_, input.num_bands()); |
| RTC_DCHECK_EQ(num_channels_, input.num_channels()); |
| |
| // TODO(bugs.webrtc.org/8759) Temporary work-around. |
| if (num_bands_ != input.num_bands()) |
| return; |
| |
| data_dumper_->DumpWav("aec3_render_input", AudioBuffer::kSplitBandSize, |
| &input.split_bands_const(0)[0][0], 16000, 1); |
| |
| CopyBufferIntoFrame(input, num_bands_, num_channels_, |
| &render_queue_input_frame_); |
| if (high_pass_filter_) { |
| high_pass_filter_->Process(&render_queue_input_frame_[0]); |
| } |
| |
| static_cast<void>(render_transfer_queue_->Insert(&render_queue_input_frame_)); |
| } |
| |
| int EchoCanceller3::instance_count_ = 0; |
| |
| EchoCanceller3::EchoCanceller3( |
| const EchoCanceller3Config& config, |
| const absl::optional<EchoCanceller3Config>& multichannel_config, |
| int sample_rate_hz, |
| size_t num_render_channels, |
| size_t num_capture_channels) |
| : data_dumper_( |
| new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))), |
| config_(AdjustConfig(config)), |
| sample_rate_hz_(sample_rate_hz), |
| num_bands_(NumBandsForRate(sample_rate_hz_)), |
| num_render_input_channels_(num_render_channels), |
| num_capture_channels_(num_capture_channels), |
| config_selector_(AdjustConfig(config), |
| multichannel_config, |
| num_render_input_channels_), |
| multichannel_content_detector_( |
| config_selector_.active_config().multi_channel.detect_stereo_content, |
| num_render_input_channels_, |
| config_selector_.active_config() |
| .multi_channel.stereo_detection_threshold, |
| config_selector_.active_config() |
| .multi_channel.stereo_detection_timeout_threshold_seconds, |
| config_selector_.active_config() |
| .multi_channel.stereo_detection_hysteresis_seconds), |
| output_framer_(num_bands_, num_capture_channels_), |
| capture_blocker_(num_bands_, num_capture_channels_), |
| render_transfer_queue_( |
| kRenderTransferQueueSizeFrames, |
| std::vector<std::vector<std::vector<float>>>( |
| num_bands_, |
| std::vector<std::vector<float>>( |
| num_render_input_channels_, |
| std::vector<float>(AudioBuffer::kSplitBandSize, 0.f))), |
| Aec3RenderQueueItemVerifier(num_bands_, |
| num_render_input_channels_, |
| AudioBuffer::kSplitBandSize)), |
| render_queue_output_frame_( |
| num_bands_, |
| std::vector<std::vector<float>>( |
| num_render_input_channels_, |
| std::vector<float>(AudioBuffer::kSplitBandSize, 0.f))), |
| render_block_(num_bands_, num_render_input_channels_), |
| capture_block_(num_bands_, num_capture_channels_), |
| capture_sub_frame_view_( |
| num_bands_, |
| std::vector<rtc::ArrayView<float>>(num_capture_channels_)) { |
| RTC_DCHECK(ValidFullBandRate(sample_rate_hz_)); |
| |
| if (config_selector_.active_config().delay.fixed_capture_delay_samples > 0) { |
| block_delay_buffer_.reset(new BlockDelayBuffer( |
| num_capture_channels_, num_bands_, AudioBuffer::kSplitBandSize, |
| config_.delay.fixed_capture_delay_samples)); |
| } |
| |
| render_writer_.reset(new RenderWriter( |
| data_dumper_.get(), config_selector_.active_config(), |
| &render_transfer_queue_, num_bands_, num_render_input_channels_)); |
| |
| RTC_DCHECK_EQ(num_bands_, std::max(sample_rate_hz_, 16000) / 16000); |
| RTC_DCHECK_GE(kMaxNumBands, num_bands_); |
| |
| if (config_selector_.active_config().filter.export_linear_aec_output) { |
| linear_output_framer_.reset( |
| new BlockFramer(/*num_bands=*/1, num_capture_channels_)); |
| linear_output_block_ = |
| std::make_unique<Block>(/*num_bands=*/1, num_capture_channels_), |
| linear_output_sub_frame_view_ = |
| std::vector<std::vector<rtc::ArrayView<float>>>( |
| 1, std::vector<rtc::ArrayView<float>>(num_capture_channels_)); |
| } |
| |
| Initialize(); |
| |
| RTC_LOG(LS_INFO) << "AEC3 created with sample rate: " << sample_rate_hz_ |
| << " Hz, num render channels: " << num_render_input_channels_ |
| << ", num capture channels: " << num_capture_channels_; |
| } |
| |
| EchoCanceller3::~EchoCanceller3() = default; |
| |
| void EchoCanceller3::Initialize() { |
| RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_); |
| |
| num_render_channels_to_aec_ = |
| multichannel_content_detector_.IsProperMultiChannelContentDetected() |
| ? num_render_input_channels_ |
| : 1; |
| |
| config_selector_.Update( |
| multichannel_content_detector_.IsProperMultiChannelContentDetected()); |
| |
| render_block_.SetNumChannels(num_render_channels_to_aec_); |
| |
| render_blocker_.reset( |
| new FrameBlocker(num_bands_, num_render_channels_to_aec_)); |
| |
| block_processor_.reset(BlockProcessor::Create( |
| config_selector_.active_config(), sample_rate_hz_, |
| num_render_channels_to_aec_, num_capture_channels_)); |
| |
| render_sub_frame_view_ = std::vector<std::vector<rtc::ArrayView<float>>>( |
| num_bands_, |
| std::vector<rtc::ArrayView<float>>(num_render_channels_to_aec_)); |
| } |
| |
| void EchoCanceller3::AnalyzeRender(const AudioBuffer& render) { |
| RTC_DCHECK_RUNS_SERIALIZED(&render_race_checker_); |
| |
| RTC_DCHECK_EQ(render.num_channels(), num_render_input_channels_); |
| data_dumper_->DumpRaw("aec3_call_order", |
| static_cast<int>(EchoCanceller3ApiCall::kRender)); |
| |
| return render_writer_->Insert(render); |
| } |
| |
| void EchoCanceller3::AnalyzeCapture(const AudioBuffer& capture) { |
| RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_); |
| data_dumper_->DumpWav("aec3_capture_analyze_input", capture.num_frames(), |
| capture.channels_const()[0], sample_rate_hz_, 1); |
| saturated_microphone_signal_ = false; |
| for (size_t channel = 0; channel < capture.num_channels(); ++channel) { |
| saturated_microphone_signal_ |= |
| DetectSaturation(rtc::ArrayView<const float>( |
| capture.channels_const()[channel], capture.num_frames())); |
| if (saturated_microphone_signal_) { |
| break; |
| } |
| } |
| } |
| |
| void EchoCanceller3::ProcessCapture(AudioBuffer* capture, bool level_change) { |
| ProcessCapture(capture, nullptr, level_change); |
| } |
| |
| void EchoCanceller3::ProcessCapture(AudioBuffer* capture, |
| AudioBuffer* linear_output, |
| bool level_change) { |
| RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_); |
| RTC_DCHECK(capture); |
| RTC_DCHECK_EQ(num_bands_, capture->num_bands()); |
| RTC_DCHECK_EQ(AudioBuffer::kSplitBandSize, capture->num_frames_per_band()); |
| RTC_DCHECK_EQ(capture->num_channels(), num_capture_channels_); |
| data_dumper_->DumpRaw("aec3_call_order", |
| static_cast<int>(EchoCanceller3ApiCall::kCapture)); |
| |
| if (linear_output && !linear_output_framer_) { |
| RTC_LOG(LS_ERROR) << "Trying to retrieve the linear AEC output without " |
| "properly configuring AEC3."; |
| RTC_DCHECK_NOTREACHED(); |
| } |
| |
| // Report capture call in the metrics and periodically update API call |
| // metrics. |
| api_call_metrics_.ReportCaptureCall(); |
| |
| // Optionally delay the capture signal. |
| if (config_selector_.active_config().delay.fixed_capture_delay_samples > 0) { |
| RTC_DCHECK(block_delay_buffer_); |
| block_delay_buffer_->DelaySignal(capture); |
| } |
| |
| rtc::ArrayView<float> capture_lower_band = rtc::ArrayView<float>( |
| &capture->split_bands(0)[0][0], AudioBuffer::kSplitBandSize); |
| |
| data_dumper_->DumpWav("aec3_capture_input", capture_lower_band, 16000, 1); |
| |
| EmptyRenderQueue(); |
| |
| ProcessCaptureFrameContent( |
| linear_output, capture, level_change, |
| multichannel_content_detector_.IsTemporaryMultiChannelContentDetected(), |
| saturated_microphone_signal_, 0, &capture_blocker_, |
| linear_output_framer_.get(), &output_framer_, block_processor_.get(), |
| linear_output_block_.get(), &linear_output_sub_frame_view_, |
| &capture_block_, &capture_sub_frame_view_); |
| |
| ProcessCaptureFrameContent( |
| linear_output, capture, level_change, |
| multichannel_content_detector_.IsTemporaryMultiChannelContentDetected(), |
| saturated_microphone_signal_, 1, &capture_blocker_, |
| linear_output_framer_.get(), &output_framer_, block_processor_.get(), |
| linear_output_block_.get(), &linear_output_sub_frame_view_, |
| &capture_block_, &capture_sub_frame_view_); |
| |
| ProcessRemainingCaptureFrameContent( |
| level_change, |
| multichannel_content_detector_.IsTemporaryMultiChannelContentDetected(), |
| saturated_microphone_signal_, &capture_blocker_, |
| linear_output_framer_.get(), &output_framer_, block_processor_.get(), |
| linear_output_block_.get(), &capture_block_); |
| |
| data_dumper_->DumpWav("aec3_capture_output", AudioBuffer::kSplitBandSize, |
| &capture->split_bands(0)[0][0], 16000, 1); |
| } |
| |
| EchoControl::Metrics EchoCanceller3::GetMetrics() const { |
| RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_); |
| Metrics metrics; |
| block_processor_->GetMetrics(&metrics); |
| return metrics; |
| } |
| |
| void EchoCanceller3::SetAudioBufferDelay(int delay_ms) { |
| RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_); |
| block_processor_->SetAudioBufferDelay(delay_ms); |
| } |
| |
| void EchoCanceller3::SetCaptureOutputUsage(bool capture_output_used) { |
| RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_); |
| block_processor_->SetCaptureOutputUsage(capture_output_used); |
| } |
| |
| bool EchoCanceller3::ActiveProcessing() const { |
| return true; |
| } |
| |
| EchoCanceller3Config EchoCanceller3::CreateDefaultMultichannelConfig() { |
| EchoCanceller3Config cfg; |
| // Use shorter and more rapidly adapting coarse filter to compensate for |
| // thge increased number of total filter parameters to adapt. |
| cfg.filter.coarse.length_blocks = 11; |
| cfg.filter.coarse.rate = 0.95f; |
| cfg.filter.coarse_initial.length_blocks = 11; |
| cfg.filter.coarse_initial.rate = 0.95f; |
| |
| // Use more concervative suppressor behavior for non-nearend speech. |
| cfg.suppressor.normal_tuning.max_dec_factor_lf = 0.35f; |
| cfg.suppressor.normal_tuning.max_inc_factor = 1.5f; |
| return cfg; |
| } |
| |
| void EchoCanceller3::SetBlockProcessorForTesting( |
| std::unique_ptr<BlockProcessor> block_processor) { |
| RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_); |
| RTC_DCHECK(block_processor); |
| block_processor_ = std::move(block_processor); |
| } |
| |
| void EchoCanceller3::EmptyRenderQueue() { |
| RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_); |
| bool frame_to_buffer = |
| render_transfer_queue_.Remove(&render_queue_output_frame_); |
| while (frame_to_buffer) { |
| // Report render call in the metrics. |
| api_call_metrics_.ReportRenderCall(); |
| |
| if (multichannel_content_detector_.UpdateDetection( |
| render_queue_output_frame_)) { |
| // Reinitialize the AEC when proper stereo is detected. |
| Initialize(); |
| } |
| |
| // Buffer frame content. |
| BufferRenderFrameContent( |
| /*proper_downmix_needed=*/multichannel_content_detector_ |
| .IsTemporaryMultiChannelContentDetected(), |
| &render_queue_output_frame_, 0, render_blocker_.get(), |
| block_processor_.get(), &render_block_, &render_sub_frame_view_); |
| |
| BufferRenderFrameContent( |
| /*proper_downmix_needed=*/multichannel_content_detector_ |
| .IsTemporaryMultiChannelContentDetected(), |
| &render_queue_output_frame_, 1, render_blocker_.get(), |
| block_processor_.get(), &render_block_, &render_sub_frame_view_); |
| |
| BufferRemainingRenderFrameContent(render_blocker_.get(), |
| block_processor_.get(), &render_block_); |
| |
| frame_to_buffer = |
| render_transfer_queue_.Remove(&render_queue_output_frame_); |
| } |
| } |
| } // namespace webrtc |
