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https://github.com/danog/libtgvoip.git
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45 lines
1.5 KiB
C++
45 lines
1.5 KiB
C++
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/*
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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/render_reverb_model.h"
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#include <algorithm>
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#include "api/array_view.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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RenderReverbModel::RenderReverbModel() {
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Reset();
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}
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RenderReverbModel::~RenderReverbModel() = default;
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void RenderReverbModel::Reset() {
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render_reverb_.Reset();
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}
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void RenderReverbModel::Apply(const VectorBuffer& spectrum_buffer,
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int delay_blocks,
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float reverb_decay,
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rtc::ArrayView<float> reverb_power_spectrum) {
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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 idx_past = spectrum_buffer.IncIndex(idx_at_delay);
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const auto& X2 = spectrum_buffer.buffer[idx_at_delay];
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RTC_DCHECK_EQ(X2.size(), reverb_power_spectrum.size());
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std::copy(X2.begin(), X2.end(), reverb_power_spectrum.begin());
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render_reverb_.AddReverbNoFreqShaping(spectrum_buffer.buffer[idx_past], 1.0f,
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reverb_decay, reverb_power_spectrum);
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}
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} // namespace webrtc
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