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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 <list>
#include <memory>
#include "api/rtp_headers.h" // NOLINT(build/include)
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
namespace RtpUtility {
class RtpHeaderParser;
} // namespace RtpUtility
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_memory|. The length of |packet_memory| is |allocated_bytes|.
// The new object assumes ownership of |packet_memory| and will delete it
// when the Packet object is deleted. The |time_ms| is an extra time
// associated with this packet, typically used to denote arrival time.
// The first bytes in |packet_memory| will be parsed using |parser|.
// |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, whereas |allocated_bytes|
// is the length of the remaining payload (typically only the RTP header).
Packet(uint8_t* packet_memory,
size_t allocated_bytes,
size_t virtual_packet_length_bytes,
double time_ms,
const RtpUtility::RtpHeaderParser& parser,
const RtpHeaderExtensionMap* extension_map = nullptr);
// 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);
// The following constructors are the same as the first two, but without a
// parser. Note that when the object is constructed using any of these
// methods, the header will be parsed using a default RtpHeaderParser object.
// In particular, RTP header extensions won't be parsed.
Packet(uint8_t* packet_memory, size_t allocated_bytes, double time_ms);
Packet(uint8_t* packet_memory,
size_t allocated_bytes,
size_t virtual_packet_length_bytes,
double time_ms);
virtual ~Packet();
// 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 payload_; }
size_t packet_length_bytes() const { return packet_length_bytes_; }
size_t payload_length_bytes() const { return payload_length_bytes_; }
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 webrtc::RtpUtility::RtpHeaderParser& parser,
const RtpHeaderExtensionMap* extension_map);
void CopyToHeader(RTPHeader* destination) const;
RTPHeader header_;
const std::unique_ptr<uint8_t[]> payload_memory_;
const uint8_t* payload_ = nullptr; // First byte after header.
const size_t packet_length_bytes_ = 0; // Total length of packet.
size_t payload_length_bytes_ = 0; // Length of the payload, after RTP header.
// Zero 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_; // Set by the RtpHeaderParser.
RTC_DISALLOW_COPY_AND_ASSIGN(Packet);
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
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