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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.
48 lines
1.4 KiB
C++
48 lines
1.4 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/block_delay_buffer.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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BlockDelayBuffer::BlockDelayBuffer(size_t num_bands,
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size_t frame_length,
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size_t delay_samples)
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: frame_length_(frame_length),
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delay_(delay_samples),
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buf_(num_bands, std::vector<float>(delay_, 0.f)) {}
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BlockDelayBuffer::~BlockDelayBuffer() = default;
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void BlockDelayBuffer::DelaySignal(AudioBuffer* frame) {
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RTC_DCHECK_EQ(1, frame->num_channels());
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RTC_DCHECK_EQ(buf_.size(), frame->num_bands());
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if (delay_ == 0) {
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return;
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}
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const size_t i_start = last_insert_;
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size_t i = 0;
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for (size_t j = 0; j < buf_.size(); ++j) {
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i = i_start;
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for (size_t k = 0; k < frame_length_; ++k) {
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const float tmp = buf_[j][i];
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buf_[j][i] = frame->split_bands_f(0)[j][k];
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frame->split_bands_f(0)[j][k] = tmp;
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i = i < buf_[0].size() - 1 ? i + 1 : 0;
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}
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}
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last_insert_ = i;
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}
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} // namespace webrtc
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