mirror of
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.
103 lines
3.3 KiB
C++
103 lines
3.3 KiB
C++
/*
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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/agc2/gain_applier.h"
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#include "api/array_view.h"
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#include "modules/audio_processing/agc2/agc2_common.h"
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#include "rtc_base/numerics/safe_minmax.h"
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namespace webrtc {
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namespace {
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// Returns true when the gain factor is so close to 1 that it would
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// not affect int16 samples.
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bool GainCloseToOne(float gain_factor) {
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return 1.f - 1.f / kMaxFloatS16Value <= gain_factor &&
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gain_factor <= 1.f + 1.f / kMaxFloatS16Value;
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}
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void ClipSignal(AudioFrameView<float> signal) {
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for (size_t k = 0; k < signal.num_channels(); ++k) {
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rtc::ArrayView<float> channel_view = signal.channel(k);
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for (auto& sample : channel_view) {
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sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
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}
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}
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}
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void ApplyGainWithRamping(float last_gain_linear,
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float gain_at_end_of_frame_linear,
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float inverse_samples_per_channel,
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AudioFrameView<float> float_frame) {
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// Do not modify the signal.
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if (last_gain_linear == gain_at_end_of_frame_linear &&
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GainCloseToOne(gain_at_end_of_frame_linear)) {
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return;
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}
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// Gain is constant and different from 1.
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if (last_gain_linear == gain_at_end_of_frame_linear) {
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for (size_t k = 0; k < float_frame.num_channels(); ++k) {
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rtc::ArrayView<float> channel_view = float_frame.channel(k);
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for (auto& sample : channel_view) {
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sample *= gain_at_end_of_frame_linear;
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}
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}
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return;
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}
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// The gain changes. We have to change slowly to avoid discontinuities.
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const float increment = (gain_at_end_of_frame_linear - last_gain_linear) *
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inverse_samples_per_channel;
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float gain = last_gain_linear;
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for (size_t i = 0; i < float_frame.samples_per_channel(); ++i) {
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for (size_t ch = 0; ch < float_frame.num_channels(); ++ch) {
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float_frame.channel(ch)[i] *= gain;
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}
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gain += increment;
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}
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}
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} // namespace
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GainApplier::GainApplier(bool hard_clip_samples, float initial_gain_factor)
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: hard_clip_samples_(hard_clip_samples),
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last_gain_factor_(initial_gain_factor),
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current_gain_factor_(initial_gain_factor) {}
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void GainApplier::ApplyGain(AudioFrameView<float> signal) {
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if (static_cast<int>(signal.samples_per_channel()) != samples_per_channel_) {
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Initialize(signal.samples_per_channel());
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}
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ApplyGainWithRamping(last_gain_factor_, current_gain_factor_,
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inverse_samples_per_channel_, signal);
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last_gain_factor_ = current_gain_factor_;
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if (hard_clip_samples_) {
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ClipSignal(signal);
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}
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}
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void GainApplier::SetGainFactor(float gain_factor) {
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RTC_DCHECK_GT(gain_factor, 0.f);
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current_gain_factor_ = gain_factor;
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}
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void GainApplier::Initialize(size_t samples_per_channel) {
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RTC_DCHECK_GT(samples_per_channel, 0);
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samples_per_channel_ = static_cast<int>(samples_per_channel);
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inverse_samples_per_channel_ = 1.f / samples_per_channel_;
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}
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} // namespace webrtc
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