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d47941e018
This is a reland of 81c0cf287c
Original change's description:
> Simplification and refactoring of the AudioBuffer code
>
> This CL performs a major refactoring and simplification
> of the AudioBuffer code that.
> -Removes 7 of the 9 internal buffers of the AudioBuffer.
> -Avoids the implicit copying required to keep the
> internal buffers in sync.
> -Removes all code relating to handling of fixed-point
> sample data in the AudioBuffer.
> -Changes the naming of the class methods to reflect
> that only floating point is handled.
> -Corrects some bugs in the code.
> -Extends the handling of internal downmixing to be
> more generic.
>
> Bug: webrtc:10882
> Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28928}
Bug: webrtc:10882
Change-Id: I2ddf327e80a03468c41662ae63c619ff34f2363a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150101
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28938}
47 lines
1.4 KiB
C++
47 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(0)[j][k];
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frame->split_bands(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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