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/*
* Copyright (c) 2014 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_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
#include <cstddef>
#include <cstdint>
#include <list>
#include "api/array_view.h"
#include "api/rtp_headers.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "rtc_base/copy_on_write_buffer.h"
namespace webrtc {
namespace test {
// Class for handling RTP packets in test applications.
class Packet {
public:
// Creates a packet, with the packet payload (including header bytes) in
// `packet`. The `time_ms` is an extra time associated with this packet,
// typically used to denote arrival time.
// `virtual_packet_length_bytes` is typically used when reading RTP dump files
// that only contain the RTP headers, and no payload (a.k.a RTP dummy files or
// RTP light). The `virtual_packet_length_bytes` tells what size the packet
// had on wire, including the now discarded payload.
Packet(CopyOnWriteBuffer packet,
size_t virtual_packet_length_bytes,
double time_ms,
const RtpHeaderExtensionMap* extension_map = nullptr);
Packet(CopyOnWriteBuffer packet,
double time_ms,
const RtpHeaderExtensionMap* extension_map = nullptr)
: Packet(packet, packet.size(), time_ms, extension_map) {}
// Same as above, but creates the packet from an already parsed RTPHeader.
// This is typically used when reading RTP dump files that only contain the
// RTP headers, and no payload. The `virtual_packet_length_bytes` tells what
// size the packet had on wire, including the now discarded payload,
// The `virtual_payload_length_bytes` tells the size of the payload.
Packet(const RTPHeader& header,
size_t virtual_packet_length_bytes,
size_t virtual_payload_length_bytes,
double time_ms);
virtual ~Packet();
Packet(const Packet&) = delete;
Packet& operator=(const Packet&) = delete;
// Parses the first bytes of the RTP payload, interpreting them as RED headers
// according to RFC 2198. The headers will be inserted into `headers`. The
// caller of the method assumes ownership of the objects in the list, and
// must delete them properly.
bool ExtractRedHeaders(std::list<RTPHeader*>* headers) const;
// Deletes all RTPHeader objects in `headers`, but does not delete `headers`
// itself.
static void DeleteRedHeaders(std::list<RTPHeader*>* headers);
const uint8_t* payload() const { return rtp_payload_.data(); }
size_t packet_length_bytes() const { return packet_.size(); }
size_t payload_length_bytes() const { return rtp_payload_.size(); }
size_t virtual_packet_length_bytes() const {
return virtual_packet_length_bytes_;
}
size_t virtual_payload_length_bytes() const {
return virtual_payload_length_bytes_;
}
const RTPHeader& header() const { return header_; }
double time_ms() const { return time_ms_; }
bool valid_header() const { return valid_header_; }
private:
bool ParseHeader(const RtpHeaderExtensionMap* extension_map);
void CopyToHeader(RTPHeader* destination) const;
RTPHeader header_;
const CopyOnWriteBuffer packet_;
ArrayView<const uint8_t> rtp_payload_; // Empty for dummy RTP packets.
// Virtual lengths are used when parsing RTP header files (dummy RTP files).
const size_t virtual_packet_length_bytes_;
size_t virtual_payload_length_bytes_ = 0;
const double time_ms_; // Used to denote a packet's arrival time.
const bool valid_header_;
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
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