Chromium Code Reviews| 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..62fc0e6cbc5f5cac6917da868bc07190584446c6 | 
| --- /dev/null | 
| +++ b/webrtc/modules/audio_processing/aec3/block_framer.cc | 
| @@ -0,0 +1,58 @@ | 
| +/* | 
| + * 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)) {} | 
| 
 
aleloi
2016/12/23 14:28:38
Suggestion: initialize the buffer with 16 zeroes a
 
peah-webrtc
2017/01/02 08:45:10
That does not seem to work.
I think the reason is
 
aleloi
2017/01/09 13:49:28
Ok, I see. Sorry for the delay.
 
 | 
| + | 
| +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].resize(0); | 
| + buffer_[i].insert(buffer_[i].begin(), block[i].begin() + samples_to_frame, | 
| + block[i].end()); | 
| + } | 
| +} | 
| + | 
| +} // namespace webrtc |