This change enables the use of two different adaptation speeds of the matched filter of the delay estimator of AEC3. One speed is used when no delay has been found, and one is used after a reliable delay has been found. The purpose is to use a slower adaptation speed to reduce the risk of divergence during double-talk without slowing down the search for the initial delay. The CL prepares for experimentation by adding field trials for controlling the two adaptation speeds. Bug: webrtc:12775 Change-Id: I817a1ab5ded0f78d20de45edcf04c708290173fc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219083 Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34055}
894 lines
38 KiB
C++
894 lines
38 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/aec3/echo_canceller3.h"
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#include <algorithm>
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#include <utility>
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#include "modules/audio_processing/aec3/aec3_common.h"
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#include "modules/audio_processing/high_pass_filter.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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#include "rtc_base/atomic_ops.h"
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#include "rtc_base/experiments/field_trial_parser.h"
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#include "rtc_base/logging.h"
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#include "system_wrappers/include/field_trial.h"
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namespace webrtc {
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namespace {
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enum class EchoCanceller3ApiCall { kCapture, kRender };
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bool DetectSaturation(rtc::ArrayView<const float> y) {
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for (auto y_k : y) {
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if (y_k >= 32700.0f || y_k <= -32700.0f) {
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return true;
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}
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}
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return false;
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}
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// Retrieves a value from a field trial if it is available. If no value is
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// present, the default value is returned. If the retrieved value is beyond the
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// specified limits, the default value is returned instead.
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void RetrieveFieldTrialValue(const char* trial_name,
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float min,
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float max,
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float* value_to_update) {
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const std::string field_trial_str = field_trial::FindFullName(trial_name);
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FieldTrialParameter<double> field_trial_param(/*key=*/"", *value_to_update);
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ParseFieldTrial({&field_trial_param}, field_trial_str);
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float field_trial_value = static_cast<float>(field_trial_param.Get());
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if (field_trial_value >= min && field_trial_value <= max) {
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*value_to_update = field_trial_value;
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}
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}
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void RetrieveFieldTrialValue(const char* trial_name,
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int min,
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int max,
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int* value_to_update) {
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const std::string field_trial_str = field_trial::FindFullName(trial_name);
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FieldTrialParameter<int> field_trial_param(/*key=*/"", *value_to_update);
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ParseFieldTrial({&field_trial_param}, field_trial_str);
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float field_trial_value = field_trial_param.Get();
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if (field_trial_value >= min && field_trial_value <= max) {
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*value_to_update = field_trial_value;
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}
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}
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void FillSubFrameView(
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AudioBuffer* frame,
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size_t sub_frame_index,
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std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame_view) {
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RTC_DCHECK_GE(1, sub_frame_index);
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RTC_DCHECK_LE(0, sub_frame_index);
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RTC_DCHECK_EQ(frame->num_bands(), sub_frame_view->size());
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RTC_DCHECK_EQ(frame->num_channels(), (*sub_frame_view)[0].size());
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for (size_t band = 0; band < sub_frame_view->size(); ++band) {
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for (size_t channel = 0; channel < (*sub_frame_view)[0].size(); ++channel) {
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(*sub_frame_view)[band][channel] = rtc::ArrayView<float>(
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&frame->split_bands(channel)[band][sub_frame_index * kSubFrameLength],
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kSubFrameLength);
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}
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}
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}
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void FillSubFrameView(
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std::vector<std::vector<std::vector<float>>>* frame,
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size_t sub_frame_index,
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std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame_view) {
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RTC_DCHECK_GE(1, sub_frame_index);
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RTC_DCHECK_EQ(frame->size(), sub_frame_view->size());
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RTC_DCHECK_EQ((*frame)[0].size(), (*sub_frame_view)[0].size());
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for (size_t band = 0; band < frame->size(); ++band) {
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for (size_t channel = 0; channel < (*frame)[band].size(); ++channel) {
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(*sub_frame_view)[band][channel] = rtc::ArrayView<float>(
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&(*frame)[band][channel][sub_frame_index * kSubFrameLength],
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kSubFrameLength);
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}
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}
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}
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void ProcessCaptureFrameContent(
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AudioBuffer* linear_output,
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AudioBuffer* capture,
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bool level_change,
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bool saturated_microphone_signal,
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size_t sub_frame_index,
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FrameBlocker* capture_blocker,
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BlockFramer* linear_output_framer,
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BlockFramer* output_framer,
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BlockProcessor* block_processor,
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std::vector<std::vector<std::vector<float>>>* linear_output_block,
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std::vector<std::vector<rtc::ArrayView<float>>>*
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linear_output_sub_frame_view,
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std::vector<std::vector<std::vector<float>>>* capture_block,
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std::vector<std::vector<rtc::ArrayView<float>>>* capture_sub_frame_view) {
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FillSubFrameView(capture, sub_frame_index, capture_sub_frame_view);
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if (linear_output) {
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RTC_DCHECK(linear_output_framer);
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RTC_DCHECK(linear_output_block);
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RTC_DCHECK(linear_output_sub_frame_view);
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FillSubFrameView(linear_output, sub_frame_index,
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linear_output_sub_frame_view);
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}
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capture_blocker->InsertSubFrameAndExtractBlock(*capture_sub_frame_view,
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capture_block);
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block_processor->ProcessCapture(level_change, saturated_microphone_signal,
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linear_output_block, capture_block);
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output_framer->InsertBlockAndExtractSubFrame(*capture_block,
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capture_sub_frame_view);
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if (linear_output) {
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RTC_DCHECK(linear_output_framer);
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linear_output_framer->InsertBlockAndExtractSubFrame(
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*linear_output_block, linear_output_sub_frame_view);
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}
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}
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void ProcessRemainingCaptureFrameContent(
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bool level_change,
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bool saturated_microphone_signal,
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FrameBlocker* capture_blocker,
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BlockFramer* linear_output_framer,
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BlockFramer* output_framer,
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BlockProcessor* block_processor,
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std::vector<std::vector<std::vector<float>>>* linear_output_block,
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std::vector<std::vector<std::vector<float>>>* block) {
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if (!capture_blocker->IsBlockAvailable()) {
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return;
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}
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capture_blocker->ExtractBlock(block);
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block_processor->ProcessCapture(level_change, saturated_microphone_signal,
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linear_output_block, block);
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output_framer->InsertBlock(*block);
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if (linear_output_framer) {
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RTC_DCHECK(linear_output_block);
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linear_output_framer->InsertBlock(*linear_output_block);
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}
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}
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void BufferRenderFrameContent(
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std::vector<std::vector<std::vector<float>>>* render_frame,
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size_t sub_frame_index,
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FrameBlocker* render_blocker,
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BlockProcessor* block_processor,
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std::vector<std::vector<std::vector<float>>>* block,
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std::vector<std::vector<rtc::ArrayView<float>>>* sub_frame_view) {
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FillSubFrameView(render_frame, sub_frame_index, sub_frame_view);
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render_blocker->InsertSubFrameAndExtractBlock(*sub_frame_view, block);
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block_processor->BufferRender(*block);
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}
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void BufferRemainingRenderFrameContent(
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FrameBlocker* render_blocker,
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BlockProcessor* block_processor,
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std::vector<std::vector<std::vector<float>>>* block) {
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if (!render_blocker->IsBlockAvailable()) {
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return;
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}
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render_blocker->ExtractBlock(block);
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block_processor->BufferRender(*block);
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}
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void CopyBufferIntoFrame(const AudioBuffer& buffer,
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size_t num_bands,
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size_t num_channels,
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std::vector<std::vector<std::vector<float>>>* frame) {
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RTC_DCHECK_EQ(num_bands, frame->size());
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RTC_DCHECK_EQ(num_channels, (*frame)[0].size());
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RTC_DCHECK_EQ(AudioBuffer::kSplitBandSize, (*frame)[0][0].size());
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for (size_t band = 0; band < num_bands; ++band) {
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for (size_t channel = 0; channel < num_channels; ++channel) {
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rtc::ArrayView<const float> buffer_view(
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&buffer.split_bands_const(channel)[band][0],
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AudioBuffer::kSplitBandSize);
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std::copy(buffer_view.begin(), buffer_view.end(),
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(*frame)[band][channel].begin());
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}
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}
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}
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} // namespace
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// TODO(webrtc:5298): Move this to a separate file.
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EchoCanceller3Config AdjustConfig(const EchoCanceller3Config& config) {
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EchoCanceller3Config adjusted_cfg = config;
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if (field_trial::IsEnabled("WebRTC-Aec3AntiHowlingMinimizationKillSwitch")) {
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adjusted_cfg.suppressor.high_bands_suppression
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.anti_howling_activation_threshold = 25.f;
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adjusted_cfg.suppressor.high_bands_suppression.anti_howling_gain = 0.01f;
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}
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if (field_trial::IsEnabled("WebRTC-Aec3UseShortConfigChangeDuration")) {
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adjusted_cfg.filter.config_change_duration_blocks = 10;
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}
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if (field_trial::IsEnabled("WebRTC-Aec3UseZeroInitialStateDuration")) {
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adjusted_cfg.filter.initial_state_seconds = 0.f;
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} else if (field_trial::IsEnabled(
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"WebRTC-Aec3UseDot1SecondsInitialStateDuration")) {
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adjusted_cfg.filter.initial_state_seconds = .1f;
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} else if (field_trial::IsEnabled(
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"WebRTC-Aec3UseDot2SecondsInitialStateDuration")) {
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adjusted_cfg.filter.initial_state_seconds = .2f;
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} else if (field_trial::IsEnabled(
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"WebRTC-Aec3UseDot3SecondsInitialStateDuration")) {
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adjusted_cfg.filter.initial_state_seconds = .3f;
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} else if (field_trial::IsEnabled(
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"WebRTC-Aec3UseDot6SecondsInitialStateDuration")) {
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adjusted_cfg.filter.initial_state_seconds = .6f;
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} else if (field_trial::IsEnabled(
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"WebRTC-Aec3UseDot9SecondsInitialStateDuration")) {
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adjusted_cfg.filter.initial_state_seconds = .9f;
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} else if (field_trial::IsEnabled(
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"WebRTC-Aec3Use1Dot2SecondsInitialStateDuration")) {
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adjusted_cfg.filter.initial_state_seconds = 1.2f;
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} else if (field_trial::IsEnabled(
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"WebRTC-Aec3Use1Dot6SecondsInitialStateDuration")) {
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adjusted_cfg.filter.initial_state_seconds = 1.6f;
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} else if (field_trial::IsEnabled(
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"WebRTC-Aec3Use2Dot0SecondsInitialStateDuration")) {
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adjusted_cfg.filter.initial_state_seconds = 2.0f;
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}
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if (field_trial::IsEnabled("WebRTC-Aec3HighPassFilterEchoReference")) {
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adjusted_cfg.filter.high_pass_filter_echo_reference = true;
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}
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if (field_trial::IsEnabled("WebRTC-Aec3EchoSaturationDetectionKillSwitch")) {
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adjusted_cfg.ep_strength.echo_can_saturate = false;
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}
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if (field_trial::IsEnabled("WebRTC-Aec3UseDot2ReverbDefaultLen")) {
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adjusted_cfg.ep_strength.default_len = 0.2f;
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} else if (field_trial::IsEnabled("WebRTC-Aec3UseDot3ReverbDefaultLen")) {
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adjusted_cfg.ep_strength.default_len = 0.3f;
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} else if (field_trial::IsEnabled("WebRTC-Aec3UseDot4ReverbDefaultLen")) {
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adjusted_cfg.ep_strength.default_len = 0.4f;
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} else if (field_trial::IsEnabled("WebRTC-Aec3UseDot5ReverbDefaultLen")) {
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adjusted_cfg.ep_strength.default_len = 0.5f;
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} else if (field_trial::IsEnabled("WebRTC-Aec3UseDot6ReverbDefaultLen")) {
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adjusted_cfg.ep_strength.default_len = 0.6f;
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} else if (field_trial::IsEnabled("WebRTC-Aec3UseDot7ReverbDefaultLen")) {
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adjusted_cfg.ep_strength.default_len = 0.7f;
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} else if (field_trial::IsEnabled("WebRTC-Aec3UseDot8ReverbDefaultLen")) {
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adjusted_cfg.ep_strength.default_len = 0.8f;
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}
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if (field_trial::IsEnabled("WebRTC-Aec3ShortHeadroomKillSwitch")) {
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// Two blocks headroom.
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adjusted_cfg.delay.delay_headroom_samples = kBlockSize * 2;
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}
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if (field_trial::IsEnabled("WebRTC-Aec3ClampInstQualityToZeroKillSwitch")) {
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adjusted_cfg.erle.clamp_quality_estimate_to_zero = false;
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}
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if (field_trial::IsEnabled("WebRTC-Aec3ClampInstQualityToOneKillSwitch")) {
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adjusted_cfg.erle.clamp_quality_estimate_to_one = false;
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}
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if (field_trial::IsEnabled("WebRTC-Aec3OnsetDetectionKillSwitch")) {
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adjusted_cfg.erle.onset_detection = false;
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}
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if (field_trial::IsEnabled(
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"WebRTC-Aec3EnforceRenderDelayEstimationDownmixing")) {
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adjusted_cfg.delay.render_alignment_mixing.downmix = true;
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adjusted_cfg.delay.render_alignment_mixing.adaptive_selection = false;
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}
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if (field_trial::IsEnabled(
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"WebRTC-Aec3EnforceCaptureDelayEstimationDownmixing")) {
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adjusted_cfg.delay.capture_alignment_mixing.downmix = true;
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adjusted_cfg.delay.capture_alignment_mixing.adaptive_selection = false;
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}
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if (field_trial::IsEnabled(
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"WebRTC-Aec3EnforceCaptureDelayEstimationLeftRightPrioritization")) {
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adjusted_cfg.delay.capture_alignment_mixing.prefer_first_two_channels =
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true;
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}
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if (field_trial::IsEnabled(
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"WebRTC-"
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"Aec3RenderDelayEstimationLeftRightPrioritizationKillSwitch")) {
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adjusted_cfg.delay.capture_alignment_mixing.prefer_first_two_channels =
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false;
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}
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if (field_trial::IsEnabled("WebRTC-Aec3SensitiveDominantNearendActivation")) {
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adjusted_cfg.suppressor.dominant_nearend_detection.enr_threshold = 0.5f;
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} else if (field_trial::IsEnabled(
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"WebRTC-Aec3VerySensitiveDominantNearendActivation")) {
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adjusted_cfg.suppressor.dominant_nearend_detection.enr_threshold = 0.75f;
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}
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if (field_trial::IsEnabled("WebRTC-Aec3TransparentAntiHowlingGain")) {
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adjusted_cfg.suppressor.high_bands_suppression.anti_howling_gain = 1.f;
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}
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if (field_trial::IsEnabled(
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"WebRTC-Aec3EnforceMoreTransparentNormalSuppressorTuning")) {
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adjusted_cfg.suppressor.normal_tuning.mask_lf.enr_transparent = 0.4f;
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adjusted_cfg.suppressor.normal_tuning.mask_lf.enr_suppress = 0.5f;
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}
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if (field_trial::IsEnabled(
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"WebRTC-Aec3EnforceMoreTransparentNearendSuppressorTuning")) {
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adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_transparent = 1.29f;
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adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_suppress = 1.3f;
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}
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if (field_trial::IsEnabled(
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"WebRTC-Aec3EnforceMoreTransparentNormalSuppressorHfTuning")) {
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adjusted_cfg.suppressor.normal_tuning.mask_hf.enr_transparent = 0.3f;
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adjusted_cfg.suppressor.normal_tuning.mask_hf.enr_suppress = 0.4f;
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}
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if (field_trial::IsEnabled(
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"WebRTC-Aec3EnforceMoreTransparentNearendSuppressorHfTuning")) {
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adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_transparent = 1.09f;
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adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_suppress = 1.1f;
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}
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if (field_trial::IsEnabled(
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"WebRTC-Aec3EnforceRapidlyAdjustingNormalSuppressorTunings")) {
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adjusted_cfg.suppressor.normal_tuning.max_inc_factor = 2.5f;
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}
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if (field_trial::IsEnabled(
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"WebRTC-Aec3EnforceRapidlyAdjustingNearendSuppressorTunings")) {
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adjusted_cfg.suppressor.nearend_tuning.max_inc_factor = 2.5f;
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}
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if (field_trial::IsEnabled(
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"WebRTC-Aec3EnforceSlowlyAdjustingNormalSuppressorTunings")) {
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adjusted_cfg.suppressor.normal_tuning.max_dec_factor_lf = .2f;
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}
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if (field_trial::IsEnabled(
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"WebRTC-Aec3EnforceSlowlyAdjustingNearendSuppressorTunings")) {
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adjusted_cfg.suppressor.nearend_tuning.max_dec_factor_lf = .2f;
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}
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if (field_trial::IsEnabled("WebRTC-Aec3EnforceConservativeHfSuppression")) {
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adjusted_cfg.suppressor.conservative_hf_suppression = true;
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}
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if (field_trial::IsEnabled("WebRTC-Aec3EnforceStationarityProperties")) {
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adjusted_cfg.echo_audibility.use_stationarity_properties = true;
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}
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if (field_trial::IsEnabled(
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"WebRTC-Aec3EnforceStationarityPropertiesAtInit")) {
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adjusted_cfg.echo_audibility.use_stationarity_properties_at_init = true;
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}
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if (field_trial::IsEnabled("WebRTC-Aec3EnforceLowActiveRenderLimit")) {
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adjusted_cfg.render_levels.active_render_limit = 50.f;
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} else if (field_trial::IsEnabled(
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"WebRTC-Aec3EnforceVeryLowActiveRenderLimit")) {
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adjusted_cfg.render_levels.active_render_limit = 30.f;
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}
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if (field_trial::IsEnabled("WebRTC-Aec3NonlinearModeReverbKillSwitch")) {
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adjusted_cfg.echo_model.model_reverb_in_nonlinear_mode = false;
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}
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// Field-trial based override for the whole suppressor tuning.
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const std::string suppressor_tuning_override_trial_name =
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field_trial::FindFullName("WebRTC-Aec3SuppressorTuningOverride");
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FieldTrialParameter<double> nearend_tuning_mask_lf_enr_transparent(
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"nearend_tuning_mask_lf_enr_transparent",
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adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_transparent);
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FieldTrialParameter<double> nearend_tuning_mask_lf_enr_suppress(
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"nearend_tuning_mask_lf_enr_suppress",
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adjusted_cfg.suppressor.nearend_tuning.mask_lf.enr_suppress);
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FieldTrialParameter<double> nearend_tuning_mask_hf_enr_transparent(
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"nearend_tuning_mask_hf_enr_transparent",
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adjusted_cfg.suppressor.nearend_tuning.mask_hf.enr_transparent);
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FieldTrialParameter<double> nearend_tuning_mask_hf_enr_suppress(
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"nearend_tuning_mask_hf_enr_suppress",
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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);
|
|
FieldTrialParameter<double> ep_strength_default_len(
|
|
"ep_strength_default_len", adjusted_cfg.ep_strength.default_len);
|
|
|
|
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, &ep_strength_default_len},
|
|
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();
|
|
adjusted_cfg.ep_strength.default_len =
|
|
static_cast<float>(ep_strength_default_len.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);
|
|
|
|
RetrieveFieldTrialValue("WebRTC-Aec3SuppressorEpStrengthDefaultLenOverride",
|
|
-1.f, 1.f, &adjusted_cfg.ep_strength.default_len);
|
|
|
|
// 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,
|
|
int sample_rate_hz,
|
|
size_t num_render_channels,
|
|
size_t num_capture_channels)
|
|
: EchoCanceller3(AdjustConfig(config),
|
|
sample_rate_hz,
|
|
num_render_channels,
|
|
num_capture_channels,
|
|
std::unique_ptr<BlockProcessor>(
|
|
BlockProcessor::Create(AdjustConfig(config),
|
|
sample_rate_hz,
|
|
num_render_channels,
|
|
num_capture_channels))) {}
|
|
EchoCanceller3::EchoCanceller3(const EchoCanceller3Config& config,
|
|
int sample_rate_hz,
|
|
size_t num_render_channels,
|
|
size_t num_capture_channels,
|
|
std::unique_ptr<BlockProcessor> block_processor)
|
|
: data_dumper_(
|
|
new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
|
|
config_(config),
|
|
sample_rate_hz_(sample_rate_hz),
|
|
num_bands_(NumBandsForRate(sample_rate_hz_)),
|
|
num_render_channels_(num_render_channels),
|
|
num_capture_channels_(num_capture_channels),
|
|
output_framer_(num_bands_, num_capture_channels_),
|
|
capture_blocker_(num_bands_, num_capture_channels_),
|
|
render_blocker_(num_bands_, num_render_channels_),
|
|
render_transfer_queue_(
|
|
kRenderTransferQueueSizeFrames,
|
|
std::vector<std::vector<std::vector<float>>>(
|
|
num_bands_,
|
|
std::vector<std::vector<float>>(
|
|
num_render_channels_,
|
|
std::vector<float>(AudioBuffer::kSplitBandSize, 0.f))),
|
|
Aec3RenderQueueItemVerifier(num_bands_,
|
|
num_render_channels_,
|
|
AudioBuffer::kSplitBandSize)),
|
|
block_processor_(std::move(block_processor)),
|
|
render_queue_output_frame_(
|
|
num_bands_,
|
|
std::vector<std::vector<float>>(
|
|
num_render_channels_,
|
|
std::vector<float>(AudioBuffer::kSplitBandSize, 0.f))),
|
|
render_block_(
|
|
num_bands_,
|
|
std::vector<std::vector<float>>(num_render_channels_,
|
|
std::vector<float>(kBlockSize, 0.f))),
|
|
capture_block_(
|
|
num_bands_,
|
|
std::vector<std::vector<float>>(num_capture_channels_,
|
|
std::vector<float>(kBlockSize, 0.f))),
|
|
render_sub_frame_view_(
|
|
num_bands_,
|
|
std::vector<rtc::ArrayView<float>>(num_render_channels_)),
|
|
capture_sub_frame_view_(
|
|
num_bands_,
|
|
std::vector<rtc::ArrayView<float>>(num_capture_channels_)) {
|
|
RTC_DCHECK(ValidFullBandRate(sample_rate_hz_));
|
|
|
|
if (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_,
|
|
&render_transfer_queue_, num_bands_,
|
|
num_render_channels_));
|
|
|
|
RTC_DCHECK_EQ(num_bands_, std::max(sample_rate_hz_, 16000) / 16000);
|
|
RTC_DCHECK_GE(kMaxNumBands, num_bands_);
|
|
|
|
if (config_.filter.export_linear_aec_output) {
|
|
linear_output_framer_.reset(new BlockFramer(1, num_capture_channels_));
|
|
linear_output_block_ =
|
|
std::make_unique<std::vector<std::vector<std::vector<float>>>>(
|
|
1, std::vector<std::vector<float>>(
|
|
num_capture_channels_, std::vector<float>(kBlockSize, 0.f)));
|
|
linear_output_sub_frame_view_ =
|
|
std::vector<std::vector<rtc::ArrayView<float>>>(
|
|
1, std::vector<rtc::ArrayView<float>>(num_capture_channels_));
|
|
}
|
|
}
|
|
|
|
EchoCanceller3::~EchoCanceller3() = default;
|
|
|
|
void EchoCanceller3::AnalyzeRender(const AudioBuffer& render) {
|
|
RTC_DCHECK_RUNS_SERIALIZED(&render_race_checker_);
|
|
|
|
RTC_DCHECK_EQ(render.num_channels(), num_render_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_NOTREACHED();
|
|
}
|
|
|
|
// Report capture call in the metrics and periodically update API call
|
|
// metrics.
|
|
api_call_metrics_.ReportCaptureCall();
|
|
|
|
// Optionally delay the capture signal.
|
|
if (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,
|
|
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,
|
|
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, 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::CreateDefaultConfig(
|
|
size_t num_render_channels,
|
|
size_t num_capture_channels) {
|
|
EchoCanceller3Config cfg;
|
|
if (num_render_channels > 1) {
|
|
// 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::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();
|
|
|
|
BufferRenderFrameContent(&render_queue_output_frame_, 0, &render_blocker_,
|
|
block_processor_.get(), &render_block_,
|
|
&render_sub_frame_view_);
|
|
|
|
BufferRenderFrameContent(&render_queue_output_frame_, 1, &render_blocker_,
|
|
block_processor_.get(), &render_block_,
|
|
&render_sub_frame_view_);
|
|
|
|
BufferRemainingRenderFrameContent(&render_blocker_, block_processor_.get(),
|
|
&render_block_);
|
|
|
|
frame_to_buffer =
|
|
render_transfer_queue_.Remove(&render_queue_output_frame_);
|
|
}
|
|
}
|
|
} // namespace webrtc
|