modules/audio_coding/neteq/buffer_level_filter.cc

/*
 *  Copyright (c) 2012 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/buffer_level_filter.h"

#include <stdint.h>

#include <algorithm>

#include "rtc_base/numerics/safe_conversions.h"

namespace webrtc {

BufferLevelFilter::BufferLevelFilter() {
  Reset();
}

void BufferLevelFilter::Reset() {
  filtered_current_level_ = 0;
  level_factor_ = 253;
}

void BufferLevelFilter::Update(size_t buffer_size_samples,
                               int time_stretched_samples) {
  // Filter:
  // `filtered_current_level_` = `level_factor_` * `filtered_current_level_` +
  //                            (1 - `level_factor_`) * `buffer_size_samples`
  // `level_factor_` and `filtered_current_level_` are in Q8.
  // `buffer_size_samples` is in Q0.
  const int64_t filtered_current_level =
      (level_factor_* int64_t{filtered_current_level_} >> 8) +
      (256 - level_factor_) * rtc::dchecked_cast<int64_t>(buffer_size_samples);

  // Account for time-scale operations (accelerate and pre-emptive expand) and
  // make sure that the filtered value remains non-negative.
  filtered_current_level_ = rtc::saturated_cast<int>(std::max<int64_t>(
      0, filtered_current_level - int64_t{time_stretched_samples} * (1 << 8)));
}

void BufferLevelFilter::SetFilteredBufferLevel(int buffer_size_samples) {
  filtered_current_level_ =
      rtc::saturated_cast<int>(int64_t{buffer_size_samples} * 256);
}

void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level_ms) {
  if (target_buffer_level_ms <= 20) {
    level_factor_ = 251;
  } else if (target_buffer_level_ms <= 60) {
    level_factor_ = 252;
  } else if (target_buffer_level_ms <= 140) {
    level_factor_ = 253;
  } else {
    level_factor_ = 254;
  }
}

}  // namespace webrtc