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/*
* Copyright (c) 2021 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/saturation_protector_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_compare.h"
namespace webrtc {
SaturationProtectorBuffer::SaturationProtectorBuffer() = default;
SaturationProtectorBuffer::~SaturationProtectorBuffer() = default;
bool SaturationProtectorBuffer::operator==(
const SaturationProtectorBuffer& b) const {
RTC_DCHECK_LE(size_, buffer_.size());
RTC_DCHECK_LE(b.size_, b.buffer_.size());
if (size_ != b.size_) {
return false;
}
for (int i = 0, i0 = FrontIndex(), i1 = b.FrontIndex(); i < size_;
++i, ++i0, ++i1) {
if (buffer_[i0 % buffer_.size()] != b.buffer_[i1 % b.buffer_.size()]) {
return false;
}
}
return true;
}
int SaturationProtectorBuffer::Capacity() const {
return buffer_.size();
}
int SaturationProtectorBuffer::Size() const {
return size_;
}
void SaturationProtectorBuffer::Reset() {
next_ = 0;
size_ = 0;
}
void SaturationProtectorBuffer::PushBack(float v) {
RTC_DCHECK_GE(next_, 0);
RTC_DCHECK_GE(size_, 0);
RTC_DCHECK_LT(next_, buffer_.size());
RTC_DCHECK_LE(size_, buffer_.size());
buffer_[next_++] = v;
if (rtc::SafeEq(next_, buffer_.size())) {
next_ = 0;
}
if (rtc::SafeLt(size_, buffer_.size())) {
size_++;
}
}
absl::optional<float> SaturationProtectorBuffer::Front() const {
if (size_ == 0) {
return absl::nullopt;
}
RTC_DCHECK_LT(FrontIndex(), buffer_.size());
return buffer_[FrontIndex()];
}
int SaturationProtectorBuffer::FrontIndex() const {
return rtc::SafeEq(size_, buffer_.size()) ? next_ : 0;
}
} // namespace webrtc
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