| Index: webrtc/modules/audio_processing/aec3/block_framer.cc
|
| diff --git a/webrtc/modules/audio_processing/aec3/block_framer.cc b/webrtc/modules/audio_processing/aec3/block_framer.cc
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..6425dae8c87551419fc30089155c65bdf5bbbedc
|
| --- /dev/null
|
| +++ b/webrtc/modules/audio_processing/aec3/block_framer.cc
|
| @@ -0,0 +1,59 @@
|
| +/*
|
| + * 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/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(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].clear();
|
| + buffer_[i].insert(buffer_[i].begin(), block[i].begin() + samples_to_frame,
|
| + block[i].end());
|
| + }
|
| +}
|
| +
|
| +} // namespace webrtc
|
|
|