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https://github.com/danog/libtgvoip.git
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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
1.7 KiB
C++
61 lines
1.7 KiB
C++
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/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/aec3/moving_average.h"
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#include <algorithm>
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#include <functional>
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace aec3 {
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MovingAverage::MovingAverage(size_t num_elem, size_t mem_len)
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: num_elem_(num_elem),
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mem_len_(mem_len - 1),
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scaling_(1.0f / static_cast<float>(mem_len)),
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memory_(num_elem * mem_len_, 0.f),
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mem_index_(0) {
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RTC_DCHECK(num_elem_ > 0);
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RTC_DCHECK(mem_len > 0);
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}
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MovingAverage::~MovingAverage() = default;
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void MovingAverage::Average(rtc::ArrayView<const float> input,
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rtc::ArrayView<float> output) {
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RTC_DCHECK(input.size() == num_elem_);
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RTC_DCHECK(output.size() == num_elem_);
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// Sum all contributions.
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std::copy(input.begin(), input.end(), output.begin());
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for (auto i = memory_.begin(); i < memory_.end(); i += num_elem_) {
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std::transform(i, i + num_elem_, output.begin(), output.begin(),
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std::plus<float>());
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}
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// Divide by mem_len_.
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for (float& o : output) {
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o *= scaling_;
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}
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// Update memory.
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if (mem_len_ > 0) {
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std::copy(input.begin(), input.end(),
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memory_.begin() + mem_index_ * num_elem_);
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mem_index_ = (mem_index_ + 1) % mem_len_;
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}
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}
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} // namespace aec3
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} // namespace webrtc
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