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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.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_
#include <stddef.h>
#include "webrtc/base/constructormagic.h"
namespace webrtc {
class BufferLevelFilter {
public:
BufferLevelFilter();
virtual ~BufferLevelFilter() {}
virtual void Reset();
// Updates the filter. Current buffer size is |buffer_size_packets| (Q0).
// If |time_stretched_samples| is non-zero, the value is converted to the
// corresponding number of packets, and is subtracted from the filtered
// value (thus bypassing the filter operation). |packet_len_samples| is the
// number of audio samples carried in each incoming packet.
virtual void Update(size_t buffer_size_packets, int time_stretched_samples,
size_t packet_len_samples);
// Set the current target buffer level (obtained from
// DelayManager::base_target_level()). Used to select the appropriate
// filter coefficient.
virtual void SetTargetBufferLevel(int target_buffer_level);
virtual int filtered_current_level() const;
private:
int level_factor_; // Filter factor for the buffer level filter in Q8.
int filtered_current_level_; // Filtered current buffer level in Q8.
RTC_DISALLOW_COPY_AND_ASSIGN(BufferLevelFilter);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_
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