diff options
Diffstat (limited to 'webrtc/modules/audio_processing/aec3/block_framer.cc')
-rw-r--r-- | webrtc/modules/audio_processing/aec3/block_framer.cc | 86 |
1 files changed, 86 insertions, 0 deletions
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 0000000..8241ce6 --- /dev/null +++ b/webrtc/modules/audio_processing/aec3/block_framer.cc @@ -0,0 +1,86 @@ +/* + * 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/block_framer.h" + +#include <algorithm> + +#include "modules/audio_processing/aec3/aec3_common.h" +#include "rtc_base/checks.h" + +namespace webrtc { + +BlockFramer::BlockFramer(size_t num_bands, size_t num_channels) + : num_bands_(num_bands), + num_channels_(num_channels), + buffer_(num_bands_, + std::vector<std::vector<float>>( + num_channels, + std::vector<float>(kBlockSize, 0.f))) { + RTC_DCHECK_LT(0, num_bands); + RTC_DCHECK_LT(0, num_channels); +} + +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<std::vector<float>>>& block) { + RTC_DCHECK_EQ(num_bands_, block.size()); + for (size_t band = 0; band < num_bands_; ++band) { + RTC_DCHECK_EQ(num_channels_, block[band].size()); + for (size_t channel = 0; channel < num_channels_; ++channel) { + RTC_DCHECK_EQ(kBlockSize, block[band][channel].size()); + RTC_DCHECK_EQ(0, buffer_[band][channel].size()); + + buffer_[band][channel].insert(buffer_[band][channel].begin(), + block[band][channel].begin(), + block[band][channel].end()); + } + } +} + +void BlockFramer::InsertBlockAndExtractSubFrame( + const std::vector<std::vector<std::vector<float>>>& block, + std::vector<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 band = 0; band < num_bands_; ++band) { + RTC_DCHECK_EQ(num_channels_, block[band].size()); + RTC_DCHECK_EQ(num_channels_, (*sub_frame)[0].size()); + for (size_t channel = 0; channel < num_channels_; ++channel) { + RTC_DCHECK_LE(kSubFrameLength, + buffer_[band][channel].size() + kBlockSize); + RTC_DCHECK_EQ(kBlockSize, block[band][channel].size()); + RTC_DCHECK_GE(kBlockSize, buffer_[band][channel].size()); + RTC_DCHECK_EQ(kSubFrameLength, (*sub_frame)[band][channel].size()); + + const int samples_to_frame = + kSubFrameLength - buffer_[band][channel].size(); + std::copy(buffer_[band][channel].begin(), buffer_[band][channel].end(), + (*sub_frame)[band][channel].begin()); + std::copy( + block[band][channel].begin(), + block[band][channel].begin() + samples_to_frame, + (*sub_frame)[band][channel].begin() + buffer_[band][channel].size()); + buffer_[band][channel].clear(); + buffer_[band][channel].insert( + buffer_[band][channel].begin(), + block[band][channel].begin() + samples_to_frame, + block[band][channel].end()); + } + } +} + +} // namespace webrtc |