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/*
* Copyright (c) 2016 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 MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
#include <stdint.h>
#include <utility>
#include "absl/base/attributes.h"
#include "api/array_view.h"
#include "api/ref_counted_base.h"
#include "api/rtp_headers.h"
#include "api/scoped_refptr.h"
#include "api/units/timestamp.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
namespace webrtc {
// Class to hold rtp packet with metadata for receiver side.
// The metadata is not parsed from the rtp packet, but may be derived from the
// data that is parsed from the rtp packet.
class RtpPacketReceived : public RtpPacket {
public:
RtpPacketReceived();
explicit RtpPacketReceived(
const ExtensionManager* extensions,
webrtc::Timestamp arrival_time = webrtc::Timestamp::MinusInfinity());
RtpPacketReceived(const RtpPacketReceived& packet);
RtpPacketReceived(RtpPacketReceived&& packet);
RtpPacketReceived& operator=(const RtpPacketReceived& packet);
RtpPacketReceived& operator=(RtpPacketReceived&& packet);
~RtpPacketReceived();
// TODO(danilchap): Remove this function when all code update to use RtpPacket
// directly. Function is there just for easier backward compatibilty.
void GetHeader(RTPHeader* header) const;
// Time in local time base as close as it can to packet arrived on the
// network.
webrtc::Timestamp arrival_time() const { return arrival_time_; }
void set_arrival_time(webrtc::Timestamp time) { arrival_time_ = time; }
ABSL_DEPRECATED("Use arrival_time() instead")
int64_t arrival_time_ms() const {
return arrival_time_.IsMinusInfinity() ? -1 : arrival_time_.ms();
}
ABSL_DEPRECATED("Use set_arrival_time() instead")
void set_arrival_time_ms(int64_t time) {
arrival_time_ = webrtc::Timestamp::Millis(time);
}
// Flag if packet was recovered via RTX or FEC.
bool recovered() const { return recovered_; }
void set_recovered(bool value) { recovered_ = value; }
int payload_type_frequency() const { return payload_type_frequency_; }
void set_payload_type_frequency(int value) {
payload_type_frequency_ = value;
}
// An application can attach arbitrary data to an RTP packet using
// `additional_data`. The additional data does not affect WebRTC processing.
rtc::scoped_refptr<rtc::RefCountedBase> additional_data() const {
return additional_data_;
}
void set_additional_data(rtc::scoped_refptr<rtc::RefCountedBase> data) {
additional_data_ = std::move(data);
}
private:
webrtc::Timestamp arrival_time_ = Timestamp::MinusInfinity();
int payload_type_frequency_ = 0;
bool recovered_ = false;
rtc::scoped_refptr<rtc::RefCountedBase> additional_data_;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTP_PACKET_RECEIVED_H_
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