mirror of
https://github.com/mollyim/webrtc.git
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980c4601e1
- Bug fix: the desired initial gain quickly dropped to 0 dB hence starting a call with a too low level - New tuning to make AGC2 more robust to VAD mistakes - Smarter max gain increase speed: to deal with an increased threshold of adjacent speech frames, the gain applier temporarily allows a faster gain increase to deal with a longer time spent waiting for enough speech frames in a row to be observed - Saturation protector isolated from `AdaptiveModeLevelEstimator` to simplify the unit tests for the latter (non bit-exact change) - AGC2 adaptive digital config: unnecessary params deprecated - Code readability improvements - Data dumps clean-up and better naming Bug: webrtc:7494 Change-Id: I4e36059bdf2566cc2a7e1a7e95b7430ba9ae9844 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215140 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33736}
77 lines
2 KiB
C++
77 lines
2 KiB
C++
/*
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* Copyright (c) 2021 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/saturation_protector_buffer.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/numerics/safe_compare.h"
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namespace webrtc {
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SaturationProtectorBuffer::SaturationProtectorBuffer() = default;
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SaturationProtectorBuffer::~SaturationProtectorBuffer() = default;
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bool SaturationProtectorBuffer::operator==(
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const SaturationProtectorBuffer& b) const {
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RTC_DCHECK_LE(size_, buffer_.size());
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RTC_DCHECK_LE(b.size_, b.buffer_.size());
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if (size_ != b.size_) {
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return false;
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}
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for (int i = 0, i0 = FrontIndex(), i1 = b.FrontIndex(); i < size_;
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++i, ++i0, ++i1) {
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if (buffer_[i0 % buffer_.size()] != b.buffer_[i1 % b.buffer_.size()]) {
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return false;
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}
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}
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return true;
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}
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int SaturationProtectorBuffer::Capacity() const {
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return buffer_.size();
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}
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int SaturationProtectorBuffer::Size() const {
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return size_;
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}
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void SaturationProtectorBuffer::Reset() {
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next_ = 0;
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size_ = 0;
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}
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void SaturationProtectorBuffer::PushBack(float v) {
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RTC_DCHECK_GE(next_, 0);
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RTC_DCHECK_GE(size_, 0);
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RTC_DCHECK_LT(next_, buffer_.size());
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RTC_DCHECK_LE(size_, buffer_.size());
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buffer_[next_++] = v;
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if (rtc::SafeEq(next_, buffer_.size())) {
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next_ = 0;
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}
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if (rtc::SafeLt(size_, buffer_.size())) {
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size_++;
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}
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}
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absl::optional<float> SaturationProtectorBuffer::Front() const {
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if (size_ == 0) {
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return absl::nullopt;
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}
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RTC_DCHECK_LT(FrontIndex(), buffer_.size());
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return buffer_[FrontIndex()];
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}
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int SaturationProtectorBuffer::FrontIndex() const {
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return rtc::SafeEq(size_, buffer_.size()) ? next_ : 0;
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}
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} // namespace webrtc
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