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/*
* 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 "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
#include <algorithm> // Provide access to std::max.
namespace webrtc {
BufferLevelFilter::BufferLevelFilter() {
Reset();
}
void BufferLevelFilter::Reset() {
filtered_current_level_ = 0;
level_factor_ = 253;
}
void BufferLevelFilter::Update(int buffer_size_packets,
int time_stretched_samples,
int packet_len_samples) {
// Filter:
// |filtered_current_level_| = |level_factor_| * |filtered_current_level_| +
// (1 - |level_factor_|) * |buffer_size_packets|
// |level_factor_| and |filtered_current_level_| are in Q8.
// |buffer_size_packets| is in Q0.
filtered_current_level_ = ((level_factor_ * filtered_current_level_) >> 8) +
((256 - level_factor_) * buffer_size_packets);
// Account for time-scale operations (accelerate and pre-emptive expand).
if (time_stretched_samples && packet_len_samples > 0) {
// Time-scaling has been performed since last filter update. Subtract the
// value of |time_stretched_samples| from |filtered_current_level_| after
// converting |time_stretched_samples| from samples to packets in Q8.
// Make sure that the filtered value remains non-negative.
filtered_current_level_ = std::max(0,
filtered_current_level_ -
(time_stretched_samples << 8) / packet_len_samples);
}
}
void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level) {
if (target_buffer_level <= 1) {
level_factor_ = 251;
} else if (target_buffer_level <= 3) {
level_factor_ = 252;
} else if (target_buffer_level <= 7) {
level_factor_ = 253;
} else {
level_factor_ = 254;
}
}
} // namespace webrtc
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