session-android/jni/webrtc/modules/audio_coding/neteq/timestamp_scaler.h

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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_TIMESTAMP_SCALER_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/neteq/packet.h"
#include "webrtc/typedefs.h"
namespace webrtc {
// Forward declaration.
class DecoderDatabase;
// This class scales timestamps for codecs that need timestamp scaling.
// This is done for codecs where one RTP timestamp does not correspond to
// one sample.
class TimestampScaler {
public:
explicit TimestampScaler(const DecoderDatabase& decoder_database)
: first_packet_received_(false),
numerator_(1),
denominator_(1),
external_ref_(0),
internal_ref_(0),
decoder_database_(decoder_database) {}
virtual ~TimestampScaler() {}
// Start over.
virtual void Reset() { first_packet_received_ = false; }
// Scale the timestamp in |packet| from external to internal.
virtual void ToInternal(Packet* packet);
// Scale the timestamp for all packets in |packet_list| from external to
// internal.
virtual void ToInternal(PacketList* packet_list);
// Returns the internal equivalent of |external_timestamp|, given the
// RTP payload type |rtp_payload_type|.
virtual uint32_t ToInternal(uint32_t external_timestamp,
uint8_t rtp_payload_type);
// Scales back to external timestamp. This is the inverse of ToInternal().
virtual uint32_t ToExternal(uint32_t internal_timestamp) const;
private:
bool first_packet_received_;
int numerator_;
int denominator_;
uint32_t external_ref_;
uint32_t internal_ref_;
const DecoderDatabase& decoder_database_;
DISALLOW_COPY_AND_ASSIGN(TimestampScaler);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_