tg2sip/webrtc_dsp/modules/audio_processing/aec3/frame_blocker.cc

71 lines
2.4 KiB
C++

/*
* 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 "modules/audio_processing/aec3/frame_blocker.h"
#include "modules/audio_processing/aec3/aec3_common.h"
#include "rtc_base/checks.h"
namespace webrtc {
FrameBlocker::FrameBlocker(size_t num_bands)
: num_bands_(num_bands), buffer_(num_bands_) {
for (auto& b : buffer_) {
b.reserve(kBlockSize);
RTC_DCHECK(b.empty());
}
}
FrameBlocker::~FrameBlocker() = default;
void FrameBlocker::InsertSubFrameAndExtractBlock(
const std::vector<rtc::ArrayView<float>>& sub_frame,
std::vector<std::vector<float>>* block) {
RTC_DCHECK(block);
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_GE(kBlockSize - 16, buffer_[i].size());
RTC_DCHECK_EQ(kBlockSize, (*block)[i].size());
RTC_DCHECK_EQ(kSubFrameLength, sub_frame[i].size());
const int samples_to_block = kBlockSize - buffer_[i].size();
(*block)[i].clear();
(*block)[i].insert((*block)[i].begin(), buffer_[i].begin(),
buffer_[i].end());
(*block)[i].insert((*block)[i].begin() + buffer_[i].size(),
sub_frame[i].begin(),
sub_frame[i].begin() + samples_to_block);
buffer_[i].clear();
buffer_[i].insert(buffer_[i].begin(),
sub_frame[i].begin() + samples_to_block,
sub_frame[i].end());
}
}
bool FrameBlocker::IsBlockAvailable() const {
return kBlockSize == buffer_[0].size();
}
void FrameBlocker::ExtractBlock(std::vector<std::vector<float>>* block) {
RTC_DCHECK(block);
RTC_DCHECK_EQ(num_bands_, block->size());
RTC_DCHECK(IsBlockAvailable());
for (size_t i = 0; i < num_bands_; ++i) {
RTC_DCHECK_EQ(kBlockSize, buffer_[i].size());
RTC_DCHECK_EQ(kBlockSize, (*block)[i].size());
(*block)[i].clear();
(*block)[i].insert((*block)[i].begin(), buffer_[i].begin(),
buffer_[i].end());
buffer_[i].clear();
}
}
} // namespace webrtc