mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-19 00:27:51 +01:00
8a3eaddc95
Added a new sub-module 'GainApplier'. The build target is 'modules/audio_processing/agc2:gain_applier'. A small refactoring makes the GainApplier used in adaptive-digital AGC2. The AGC2 now multiplies samples with a gain in 3 places. It's the GainApplier, the GainCurveApplier, and the FixedGainController. The GainApplier is used in AdaptiveDigitalGainApplier and will be used as a pre-amplifier. Bug: webrtc:9138 Change-Id: Ibc4c0ea109c6757f159d4adb6e3d8614179c9bc6 Reviewed-on: https://webrtc-review.googlesource.com/69321 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22849}
101 lines
3.3 KiB
C++
101 lines
3.3 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/agc2/gain_applier.h"
|
|
|
|
#include "modules/audio_processing/agc2/agc2_common.h"
|
|
#include "rtc_base/numerics/safe_minmax.h"
|
|
|
|
namespace webrtc {
|
|
namespace {
|
|
|
|
// Returns true when the gain factor is so close to 1 that it would
|
|
// not affect int16 samples.
|
|
bool GainCloseToOne(float gain_factor) {
|
|
return 1.f - 1.f / kMaxFloatS16Value <= gain_factor &&
|
|
gain_factor <= 1.f + 1.f / kMaxFloatS16Value;
|
|
}
|
|
|
|
void ClipSignal(AudioFrameView<float> signal) {
|
|
for (size_t k = 0; k < signal.num_channels(); ++k) {
|
|
rtc::ArrayView<float> channel_view = signal.channel(k);
|
|
for (auto& sample : channel_view) {
|
|
sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
|
|
}
|
|
}
|
|
}
|
|
|
|
void ApplyGainWithRamping(float last_gain_linear,
|
|
float gain_at_end_of_frame_linear,
|
|
float inverse_samples_per_channel,
|
|
AudioFrameView<float> float_frame) {
|
|
// Do not modify the signal.
|
|
if (last_gain_linear == gain_at_end_of_frame_linear &&
|
|
GainCloseToOne(gain_at_end_of_frame_linear)) {
|
|
return;
|
|
}
|
|
|
|
// Gain is constant and different from 1.
|
|
if (last_gain_linear == gain_at_end_of_frame_linear) {
|
|
for (size_t k = 0; k < float_frame.num_channels(); ++k) {
|
|
rtc::ArrayView<float> channel_view = float_frame.channel(k);
|
|
for (auto& sample : channel_view) {
|
|
sample *= gain_at_end_of_frame_linear;
|
|
}
|
|
}
|
|
return;
|
|
}
|
|
|
|
// The gain changes. We have to change slowly to avoid discontinuities.
|
|
const float increment = (gain_at_end_of_frame_linear - last_gain_linear) *
|
|
inverse_samples_per_channel;
|
|
float gain = last_gain_linear;
|
|
for (size_t i = 0; i < float_frame.samples_per_channel(); ++i) {
|
|
for (size_t ch = 0; ch < float_frame.num_channels(); ++ch) {
|
|
float_frame.channel(ch)[i] *= gain;
|
|
}
|
|
gain += increment;
|
|
}
|
|
}
|
|
|
|
} // namespace
|
|
|
|
GainApplier::GainApplier(bool hard_clip_samples, float initial_gain_factor)
|
|
: hard_clip_samples_(hard_clip_samples),
|
|
last_gain_factor_(initial_gain_factor),
|
|
current_gain_factor_(initial_gain_factor) {}
|
|
|
|
void GainApplier::ApplyGain(AudioFrameView<float> signal) {
|
|
if (static_cast<int>(signal.samples_per_channel()) != samples_per_channel_) {
|
|
Initialize(signal.samples_per_channel());
|
|
}
|
|
|
|
ApplyGainWithRamping(last_gain_factor_, current_gain_factor_,
|
|
inverse_samples_per_channel_, signal);
|
|
|
|
last_gain_factor_ = current_gain_factor_;
|
|
|
|
if (hard_clip_samples_) {
|
|
ClipSignal(signal);
|
|
}
|
|
}
|
|
|
|
void GainApplier::SetGainFactor(float gain_factor) {
|
|
RTC_DCHECK_GT(gain_factor, 0.f);
|
|
current_gain_factor_ = gain_factor;
|
|
}
|
|
|
|
void GainApplier::Initialize(size_t samples_per_channel) {
|
|
RTC_DCHECK_GT(samples_per_channel, 0);
|
|
samples_per_channel_ = static_cast<int>(samples_per_channel);
|
|
inverse_samples_per_channel_ = 1.f / samples_per_channel_;
|
|
}
|
|
|
|
} // namespace webrtc
|