mirror of
https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
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];
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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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