d112c75801
This reverts commit a66395e72f9fc86873bf443579ec73c3d78af240. Reason for revert: Breaking downstream tests Original change's description: > Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3." > > This is a reland of f3a197e55323aee974a932c52dd19fa88e5d4e38 > > Original change's description: > > Add core multi-channel pipeline in AEC3 > > This CL adds basic the basic pipeline to support multi-channel > > processing in AEC3. > > > > Apart from that, it removes the 8 kHz processing support in several > > places of the AEC3 code. > > > > Bug: webrtc:10913 > > Change-Id: If5b75fa325ed0071deea94a7546cb4a7adf22137 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150332 > > Commit-Queue: Per Åhgren <peah@webrtc.org> > > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#29017} > > Bug: webrtc:10913 > Change-Id: Ifc4b13bd994cfd22dca8f8755fa5700617cc379d > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151124 > Reviewed-by: Sam Zackrisson <saza@webrtc.org> > Commit-Queue: Per Åhgren <peah@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29034} TBR=saza@webrtc.org,peah@webrtc.org Change-Id: I0e9fd154da5910d73b7a4c82e4e588f3220fd39d No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10913 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151126 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29038}
61 lines
2.3 KiB
C++
61 lines
2.3 KiB
C++
/*
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* Copyright (c) 2016 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_framer.h"
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#include <algorithm>
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#include "modules/audio_processing/aec3/aec3_common.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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BlockFramer::BlockFramer(size_t num_bands)
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: num_bands_(num_bands),
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buffer_(num_bands_, std::vector<float>(kBlockSize, 0.f)) {}
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BlockFramer::~BlockFramer() = default;
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// All the constants are chosen so that the buffer is either empty or has enough
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// samples for InsertBlockAndExtractSubFrame to produce a frame. In order to
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// achieve this, the InsertBlockAndExtractSubFrame and InsertBlock methods need
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// to be called in the correct order.
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void BlockFramer::InsertBlock(const std::vector<std::vector<float>>& block) {
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RTC_DCHECK_EQ(num_bands_, block.size());
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for (size_t i = 0; i < num_bands_; ++i) {
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RTC_DCHECK_EQ(kBlockSize, block[i].size());
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RTC_DCHECK_EQ(0, buffer_[i].size());
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buffer_[i].insert(buffer_[i].begin(), block[i].begin(), block[i].end());
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}
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}
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void BlockFramer::InsertBlockAndExtractSubFrame(
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const std::vector<std::vector<float>>& block,
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std::vector<rtc::ArrayView<float>>* sub_frame) {
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RTC_DCHECK(sub_frame);
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RTC_DCHECK_EQ(num_bands_, block.size());
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RTC_DCHECK_EQ(num_bands_, sub_frame->size());
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for (size_t i = 0; i < num_bands_; ++i) {
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RTC_DCHECK_LE(kSubFrameLength, buffer_[i].size() + kBlockSize);
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RTC_DCHECK_EQ(kBlockSize, block[i].size());
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RTC_DCHECK_GE(kBlockSize, buffer_[i].size());
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RTC_DCHECK_EQ(kSubFrameLength, (*sub_frame)[i].size());
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const int samples_to_frame = kSubFrameLength - buffer_[i].size();
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std::copy(buffer_[i].begin(), buffer_[i].end(), (*sub_frame)[i].begin());
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std::copy(block[i].begin(), block[i].begin() + samples_to_frame,
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(*sub_frame)[i].begin() + buffer_[i].size());
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buffer_[i].clear();
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buffer_[i].insert(buffer_[i].begin(), block[i].begin() + samples_to_frame,
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block[i].end());
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}
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}
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} // namespace webrtc
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