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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/clipping_predictor_level_buffer.h"
#include <algorithm>
#include <cmath>
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
bool ClippingPredictorLevelBuffer::Level::operator==(const Level& level) const {
constexpr float kEpsilon = 1e-6f;
return std::fabs(average - level.average) < kEpsilon &&
std::fabs(max - level.max) < kEpsilon;
}
ClippingPredictorLevelBuffer::ClippingPredictorLevelBuffer(int capacity)
: tail_(-1), size_(0), data_(std::max(1, capacity)) {
if (capacity > kMaxCapacity) {
RTC_LOG(LS_WARNING) << "[agc]: ClippingPredictorLevelBuffer exceeds the "
<< "maximum allowed capacity. Capacity: " << capacity;
}
RTC_DCHECK(!data_.empty());
}
void ClippingPredictorLevelBuffer::Reset() {
tail_ = -1;
size_ = 0;
}
void ClippingPredictorLevelBuffer::Push(Level level) {
++tail_;
if (tail_ == Capacity()) {
tail_ = 0;
}
if (size_ < Capacity()) {
size_++;
}
data_[tail_] = level;
}
// TODO(bugs.webrtc.org/12774): Optimize partial computation for long buffers.
absl::optional<ClippingPredictorLevelBuffer::Level>
ClippingPredictorLevelBuffer::ComputePartialMetrics(int delay,
int num_items) const {
RTC_DCHECK_GE(delay, 0);
RTC_DCHECK_LT(delay, Capacity());
RTC_DCHECK_GT(num_items, 0);
RTC_DCHECK_LE(num_items, Capacity());
RTC_DCHECK_LE(delay + num_items, Capacity());
if (delay + num_items > Size()) {
return absl::nullopt;
}
float sum = 0.0f;
float max = 0.0f;
for (int i = 0; i < num_items && i < Size(); ++i) {
int idx = tail_ - delay - i;
if (idx < 0) {
idx += Capacity();
}
sum += data_[idx].average;
max = std::fmax(data_[idx].max, max);
}
return absl::optional<Level>({sum / static_cast<float>(num_items), max});
}
} // namespace webrtc