/* * 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 #include #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* headers) const; // Deletes all RTPHeader objects in |headers|, but does not delete |headers| // itself. static void DeleteRedHeaders(std::list* 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 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_