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604fd2f1ab
Bug: webrtc:13555, webrtc:13082 Change-Id: I2c2cbcbd918f0cfa970c1a964893220ba11d4b41 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247960 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com> Cr-Commit-Position: refs/heads/main@{#35771}
104 lines
3.9 KiB
C++
104 lines
3.9 KiB
C++
/*
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* Copyright (c) 2014 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 MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
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#include <list>
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#include "api/array_view.h"
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#include "api/rtp_headers.h"
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#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
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#include "rtc_base/copy_on_write_buffer.h"
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namespace webrtc {
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namespace test {
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// Class for handling RTP packets in test applications.
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class Packet {
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public:
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// Creates a packet, with the packet payload (including header bytes) in
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// `packet`. The `time_ms` is an extra time associated with this packet,
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// typically used to denote arrival time.
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// `virtual_packet_length_bytes` is typically used when reading RTP dump files
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// that only contain the RTP headers, and no payload (a.k.a RTP dummy files or
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// RTP light). The `virtual_packet_length_bytes` tells what size the packet
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// had on wire, including the now discarded payload.
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Packet(rtc::CopyOnWriteBuffer packet,
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size_t virtual_packet_length_bytes,
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double time_ms,
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const RtpHeaderExtensionMap* extension_map = nullptr);
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Packet(rtc::CopyOnWriteBuffer packet,
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double time_ms,
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const RtpHeaderExtensionMap* extension_map = nullptr)
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: Packet(packet, packet.size(), time_ms, extension_map) {}
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// Same as above, but creates the packet from an already parsed RTPHeader.
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// This is typically used when reading RTP dump files that only contain the
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// RTP headers, and no payload. The `virtual_packet_length_bytes` tells what
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// size the packet had on wire, including the now discarded payload,
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// The `virtual_payload_length_bytes` tells the size of the payload.
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Packet(const RTPHeader& header,
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size_t virtual_packet_length_bytes,
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size_t virtual_payload_length_bytes,
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double time_ms);
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virtual ~Packet();
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Packet(const Packet&) = delete;
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Packet& operator=(const Packet&) = delete;
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// Parses the first bytes of the RTP payload, interpreting them as RED headers
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// according to RFC 2198. The headers will be inserted into `headers`. The
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// caller of the method assumes ownership of the objects in the list, and
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// must delete them properly.
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bool ExtractRedHeaders(std::list<RTPHeader*>* headers) const;
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// Deletes all RTPHeader objects in `headers`, but does not delete `headers`
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// itself.
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static void DeleteRedHeaders(std::list<RTPHeader*>* headers);
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const uint8_t* payload() const { return rtp_payload_.data(); }
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size_t packet_length_bytes() const { return packet_.size(); }
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size_t payload_length_bytes() const { return rtp_payload_.size(); }
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size_t virtual_packet_length_bytes() const {
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return virtual_packet_length_bytes_;
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}
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size_t virtual_payload_length_bytes() const {
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return virtual_payload_length_bytes_;
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}
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const RTPHeader& header() const { return header_; }
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double time_ms() const { return time_ms_; }
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bool valid_header() const { return valid_header_; }
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private:
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bool ParseHeader(const RtpHeaderExtensionMap* extension_map);
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void CopyToHeader(RTPHeader* destination) const;
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RTPHeader header_;
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const rtc::CopyOnWriteBuffer packet_;
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rtc::ArrayView<const uint8_t> rtp_payload_; // Empty for dummy RTP packets.
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// Virtual lengths are used when parsing RTP header files (dummy RTP files).
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const size_t virtual_packet_length_bytes_;
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size_t virtual_payload_length_bytes_ = 0;
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const double time_ms_; // Used to denote a packet's arrival time.
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const bool valid_header_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
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