mirror of
https://github.com/danog/libtgvoip.git
synced 2024-12-02 17:51:06 +01:00
5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
61 lines
2.3 KiB
C++
61 lines
2.3 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/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)
|
|
: 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
|