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libtgvoip/webrtc_dsp/modules/audio_processing/aec3/reverb_model_fallback.cc
Grishka 5caaaafa42 Updated WebRTC APM
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.
2018-11-23 04:02:53 +03:00

71 lines
2.2 KiB
C++

/*
* Copyright (c) 2018 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/reverb_model_fallback.h"
#include <algorithm>
#include <functional>
#include "modules/audio_processing/aec3/aec3_common.h"
#include "rtc_base/checks.h"
namespace webrtc {
ReverbModelFallback::ReverbModelFallback(size_t length_blocks)
: S2_old_(length_blocks) {
Reset();
}
ReverbModelFallback::~ReverbModelFallback() = default;
void ReverbModelFallback::Reset() {
R2_reverb_.fill(0.f);
for (auto& S2_k : S2_old_) {
S2_k.fill(0.f);
}
}
void ReverbModelFallback::AddEchoReverb(
const std::array<float, kFftLengthBy2Plus1>& S2,
size_t delay,
float reverb_decay_factor,
std::array<float, kFftLengthBy2Plus1>* R2) {
// Compute the decay factor for how much the echo has decayed before leaving
// the region covered by the linear model.
auto integer_power = [](float base, int exp) {
float result = 1.f;
for (int k = 0; k < exp; ++k) {
result *= base;
}
return result;
};
RTC_DCHECK_LE(delay, S2_old_.size());
const float reverb_decay_for_delay =
integer_power(reverb_decay_factor, S2_old_.size() - delay);
// Update the estimate of the reverberant residual echo power.
S2_old_index_ = S2_old_index_ > 0 ? S2_old_index_ - 1 : S2_old_.size() - 1;
const auto& S2_end = S2_old_[S2_old_index_];
std::transform(
S2_end.begin(), S2_end.end(), R2_reverb_.begin(), R2_reverb_.begin(),
[reverb_decay_for_delay, reverb_decay_factor](float a, float b) {
return (b + a * reverb_decay_for_delay) * reverb_decay_factor;
});
// Update the buffer of old echo powers.
std::copy(S2.begin(), S2.end(), S2_old_[S2_old_index_].begin());
// Add the power of the echo reverb to the residual echo power.
std::transform(R2->begin(), R2->end(), R2_reverb_.begin(), R2->begin(),
std::plus<float>());
}
} // namespace webrtc