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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_coding/neteq/underrun_optimizer.h"
#include <algorithm>
namespace webrtc {
namespace {
constexpr int kDelayBuckets = 100;
constexpr int kBucketSizeMs = 20;
} // namespace
UnderrunOptimizer::UnderrunOptimizer(const TickTimer* tick_timer,
int histogram_quantile,
int forget_factor,
absl::optional<int> start_forget_weight,
absl::optional<int> resample_interval_ms)
: tick_timer_(tick_timer),
histogram_(kDelayBuckets, forget_factor, start_forget_weight),
histogram_quantile_(histogram_quantile),
resample_interval_ms_(resample_interval_ms) {}
void UnderrunOptimizer::Update(int relative_delay_ms) {
absl::optional<int> histogram_update;
if (resample_interval_ms_) {
if (!resample_stopwatch_) {
resample_stopwatch_ = tick_timer_->GetNewStopwatch();
}
if (static_cast<int>(resample_stopwatch_->ElapsedMs()) >
*resample_interval_ms_) {
histogram_update = max_delay_in_interval_ms_;
resample_stopwatch_ = tick_timer_->GetNewStopwatch();
max_delay_in_interval_ms_ = 0;
}
max_delay_in_interval_ms_ =
std::max(max_delay_in_interval_ms_, relative_delay_ms);
} else {
histogram_update = relative_delay_ms;
}
if (!histogram_update) {
return;
}
const int index = *histogram_update / kBucketSizeMs;
if (index < histogram_.NumBuckets()) {
// Maximum delay to register is 2000 ms.
histogram_.Add(index);
}
int bucket_index = histogram_.Quantile(histogram_quantile_);
optimal_delay_ms_ = (1 + bucket_index) * kBucketSizeMs;
}
void UnderrunOptimizer::Reset() {
histogram_.Reset();
resample_stopwatch_.reset();
max_delay_in_interval_ms_ = 0;
optimal_delay_ms_.reset();
}
} // namespace webrtc
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