blob: 80ac83645e8b0600feef1811647022071097a011 [file] [log] [blame]
* Copyright (c) 2016 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 "webrtc/modules/audio_processing/aec3/block_framer.h"
#include <algorithm>
#include "webrtc/rtc_base/checks.h"
namespace webrtc {
BlockFramer::BlockFramer(size_t num_bands)
: num_bands_(num_bands),
buffer_(num_bands_, std::vector<float>(kBlockSize, 0.f)) {}
BlockFramer::~BlockFramer() = default;
// All the constants are chosen so that the buffer is either empty or has enough
// samples for InsertBlockAndExtractSubFrame to produce a frame. In order to
// achieve this, the InsertBlockAndExtractSubFrame and InsertBlock methods need
// to be called in the correct order.
void BlockFramer::InsertBlock(const std::vector<std::vector<float>>& block) {
RTC_DCHECK_EQ(num_bands_, block.size());
for (size_t i = 0; i < num_bands_; ++i) {
RTC_DCHECK_EQ(kBlockSize, block[i].size());
RTC_DCHECK_EQ(0, buffer_[i].size());
buffer_[i].insert(buffer_[i].begin(), block[i].begin(), block[i].end());
void BlockFramer::InsertBlockAndExtractSubFrame(
const std::vector<std::vector<float>>& block,
std::vector<rtc::ArrayView<float>>* sub_frame) {
RTC_DCHECK_EQ(num_bands_, block.size());
RTC_DCHECK_EQ(num_bands_, sub_frame->size());
for (size_t i = 0; i < num_bands_; ++i) {
RTC_DCHECK_LE(kSubFrameLength, buffer_[i].size() + kBlockSize);
RTC_DCHECK_EQ(kBlockSize, block[i].size());
RTC_DCHECK_GE(kBlockSize, buffer_[i].size());
RTC_DCHECK_EQ(kSubFrameLength, (*sub_frame)[i].size());
const int samples_to_frame = kSubFrameLength - buffer_[i].size();
std::copy(buffer_[i].begin(), buffer_[i].end(), (*sub_frame)[i].begin());
std::copy(block[i].begin(), block[i].begin() + samples_to_frame,
(*sub_frame)[i].begin() + buffer_[i].size());
buffer_[i].insert(buffer_[i].begin(), block[i].begin() + samples_to_frame,
} // namespace webrtc