ce202a0f98
This is a reland of a66395e72f9fc86873bf443579ec73c3d78af240 Original change's description: > Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3." > > This is a reland of f3a197e55323aee974a932c52dd19fa88e5d4e38 > > Original change's description: > > Add core multi-channel pipeline in AEC3 > > This CL adds basic the basic pipeline to support multi-channel > > processing in AEC3. > > > > Apart from that, it removes the 8 kHz processing support in several > > places of the AEC3 code. > > > > Bug: webrtc:10913 > > Change-Id: If5b75fa325ed0071deea94a7546cb4a7adf22137 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150332 > > Commit-Queue: Per Åhgren <peah@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#29017} > > Bug: webrtc:10913 > Change-Id: Ifc4b13bd994cfd22dca8f8755fa5700617cc379d > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151124 > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > Commit-Queue: Per Åhgren <peah@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29034} Bug: webrtc:10913 Change-Id: Id8da5666df8c86f290c73ad5dc9958199f1a7ebe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151127 Commit-Queue: Sam Zackrisson <saza@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29042}
114 lines
3.8 KiB
C++
114 lines
3.8 KiB
C++
/*
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* Copyright (c) 2018 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_audibility.h"
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#include <algorithm>
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#include <cmath>
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#include <utility>
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#include <vector>
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#include "api/array_view.h"
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#include "modules/audio_processing/aec3/matrix_buffer.h"
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#include "modules/audio_processing/aec3/stationarity_estimator.h"
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#include "modules/audio_processing/aec3/vector_buffer.h"
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namespace webrtc {
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EchoAudibility::EchoAudibility(bool use_render_stationarity_at_init)
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: use_render_stationarity_at_init_(use_render_stationarity_at_init) {
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Reset();
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}
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EchoAudibility::~EchoAudibility() = default;
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void EchoAudibility::Update(
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const RenderBuffer& render_buffer,
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rtc::ArrayView<const float> render_reverb_contribution_spectrum,
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int delay_blocks,
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bool external_delay_seen) {
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UpdateRenderNoiseEstimator(render_buffer.GetSpectrumBuffer(),
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render_buffer.GetBlockBuffer(),
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external_delay_seen);
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if (external_delay_seen || use_render_stationarity_at_init_) {
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UpdateRenderStationarityFlags(
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render_buffer, render_reverb_contribution_spectrum, delay_blocks);
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}
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}
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void EchoAudibility::Reset() {
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render_stationarity_.Reset();
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non_zero_render_seen_ = false;
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render_spectrum_write_prev_ = absl::nullopt;
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}
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void EchoAudibility::UpdateRenderStationarityFlags(
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const RenderBuffer& render_buffer,
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rtc::ArrayView<const float> render_reverb_contribution_spectrum,
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int delay_blocks) {
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const VectorBuffer& spectrum_buffer = render_buffer.GetSpectrumBuffer();
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int idx_at_delay =
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spectrum_buffer.OffsetIndex(spectrum_buffer.read, delay_blocks);
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int num_lookahead = render_buffer.Headroom() - delay_blocks + 1;
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num_lookahead = std::max(0, num_lookahead);
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render_stationarity_.UpdateStationarityFlags(
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spectrum_buffer, render_reverb_contribution_spectrum, idx_at_delay,
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num_lookahead);
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}
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void EchoAudibility::UpdateRenderNoiseEstimator(
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const VectorBuffer& spectrum_buffer,
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const MatrixBuffer& block_buffer,
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bool external_delay_seen) {
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if (!render_spectrum_write_prev_) {
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render_spectrum_write_prev_ = spectrum_buffer.write;
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render_block_write_prev_ = block_buffer.write;
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return;
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}
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int render_spectrum_write_current = spectrum_buffer.write;
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if (!non_zero_render_seen_ && !external_delay_seen) {
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non_zero_render_seen_ = !IsRenderTooLow(block_buffer);
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}
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if (non_zero_render_seen_) {
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for (int idx = render_spectrum_write_prev_.value();
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idx != render_spectrum_write_current;
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idx = spectrum_buffer.DecIndex(idx)) {
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render_stationarity_.UpdateNoiseEstimator(spectrum_buffer.buffer[idx]);
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}
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}
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render_spectrum_write_prev_ = render_spectrum_write_current;
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}
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bool EchoAudibility::IsRenderTooLow(const MatrixBuffer& block_buffer) {
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bool too_low = false;
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const int render_block_write_current = block_buffer.write;
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if (render_block_write_current == render_block_write_prev_) {
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too_low = true;
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} else {
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for (int idx = render_block_write_prev_; idx != render_block_write_current;
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idx = block_buffer.IncIndex(idx)) {
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auto block = block_buffer.buffer[idx][0][0];
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auto r = std::minmax_element(block.cbegin(), block.cend());
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float max_abs = std::max(std::fabs(*r.first), std::fabs(*r.second));
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if (max_abs < 10) {
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too_low = true; // Discards all blocks if one of them is too low.
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break;
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}
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}
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}
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render_block_write_prev_ = render_block_write_current;
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return too_low;
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}
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} // namespace webrtc
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