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/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_processing/aec3/block_delay_buffer.h"
#include "rtc_base/checks.h"
namespace webrtc {
BlockDelayBuffer::BlockDelayBuffer(size_t num_bands,
size_t frame_length,
size_t delay_samples)
: frame_length_(frame_length),
delay_(delay_samples),
buf_(num_bands, std::vector<float>(delay_, 0.f)) {}
BlockDelayBuffer::~BlockDelayBuffer() = default;
void BlockDelayBuffer::DelaySignal(AudioBuffer* frame) {
RTC_DCHECK_EQ(1, frame->num_channels());
RTC_DCHECK_EQ(buf_.size(), frame->num_bands());
if (delay_ == 0) {
return;
}
const size_t i_start = last_insert_;
size_t i = 0;
for (size_t j = 0; j < buf_.size(); ++j) {
i = i_start;
for (size_t k = 0; k < frame_length_; ++k) {
const float tmp = buf_[j][i];
buf_[j][i] = frame->split_bands_f(0)[j][k];
frame->split_bands_f(0)[j][k] = tmp;
i = i < buf_[0].size() - 1 ? i + 1 : 0;
}
}
last_insert_ = i;
}
} // namespace webrtc
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