mirror of
https://github.com/oxen-io/session-android.git
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d83a3d71bc
Merge in RedPhone // FREEBIE
69 lines
2.2 KiB
C++
69 lines
2.2 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/modules/audio_coding/neteq/packet.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// Forward declaration.
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class DecoderDatabase;
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// This class scales timestamps for codecs that need timestamp scaling.
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// This is done for codecs where one RTP timestamp does not correspond to
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// one sample.
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class TimestampScaler {
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public:
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explicit TimestampScaler(const DecoderDatabase& decoder_database)
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: first_packet_received_(false),
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numerator_(1),
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denominator_(1),
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external_ref_(0),
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internal_ref_(0),
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decoder_database_(decoder_database) {}
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virtual ~TimestampScaler() {}
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// Start over.
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virtual void Reset() { first_packet_received_ = false; }
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// Scale the timestamp in |packet| from external to internal.
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virtual void ToInternal(Packet* packet);
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// Scale the timestamp for all packets in |packet_list| from external to
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// internal.
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virtual void ToInternal(PacketList* packet_list);
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// Returns the internal equivalent of |external_timestamp|, given the
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// RTP payload type |rtp_payload_type|.
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virtual uint32_t ToInternal(uint32_t external_timestamp,
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uint8_t rtp_payload_type);
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// Scales back to external timestamp. This is the inverse of ToInternal().
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virtual uint32_t ToExternal(uint32_t internal_timestamp) const;
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private:
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bool first_packet_received_;
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int numerator_;
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int denominator_;
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uint32_t external_ref_;
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uint32_t internal_ref_;
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const DecoderDatabase& decoder_database_;
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DISALLOW_COPY_AND_ASSIGN(TimestampScaler);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_
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