webrtc_m130/webrtc/test/rtcp_packet_parser.h
nisse 25d0bdc1bc Delete support for receiving RTCP RPSI and SLI message.
This code has been unused for years, and at least the RTCP RSPI sending
logic appears broken.

This cl is part 3, following

  https://codereview.webrtc.org/2746413003 (delete sending)
  https://codereview.webrtc.org/2753783002 (delete vp8 feedback mode)

BUG=webrtc:7338

Review-Url: https://codereview.webrtc.org/2742383004
Cr-Commit-Position: refs/heads/master@{#17342}
2017-03-22 14:15:09 +00:00

122 lines
4.8 KiB
C++

/*
* 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 WEBRTC_TEST_RTCP_PACKET_PARSER_H_
#define WEBRTC_TEST_RTCP_PACKET_PARSER_H_
#include "webrtc/base/array_view.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
namespace webrtc {
namespace test {
// Parse RTCP packet of given type. Assumes RTCP header is valid and that there
// is excatly one packet of correct type in the buffer.
template <typename Packet>
bool ParseSinglePacket(const uint8_t* buffer, size_t size, Packet* packet) {
rtcp::CommonHeader header;
RTC_CHECK(header.Parse(buffer, size));
RTC_CHECK_EQ(size, header.NextPacket() - buffer);
return packet->Parse(header);
}
// Same function, but takes raw buffer as single argument instead of pair.
template <typename Packet>
bool ParseSinglePacket(rtc::ArrayView<const uint8_t> buffer, Packet* packet) {
return ParseSinglePacket(buffer.data(), buffer.size(), packet);
}
class RtcpPacketParser {
public:
// Keeps last parsed packet, count number of parsed packets of given type.
template <typename TypedRtcpPacket>
class PacketCounter : public TypedRtcpPacket {
public:
int num_packets() const { return num_packets_; }
void Parse(const rtcp::CommonHeader& header) {
if (TypedRtcpPacket::Parse(header))
++num_packets_;
}
void Parse(const rtcp::CommonHeader& header, uint32_t* sender_ssrc) {
if (TypedRtcpPacket::Parse(header)) {
++num_packets_;
if (*sender_ssrc == 0) // Use first sender ssrc in compound packet.
*sender_ssrc = TypedRtcpPacket::sender_ssrc();
}
}
private:
int num_packets_ = 0;
};
RtcpPacketParser();
~RtcpPacketParser();
bool Parse(const void* packet, size_t packet_len);
PacketCounter<rtcp::App>* app() { return &app_; }
PacketCounter<rtcp::Bye>* bye() { return &bye_; }
PacketCounter<rtcp::ExtendedJitterReport>* ij() { return &ij_; }
PacketCounter<rtcp::ExtendedReports>* xr() { return &xr_; }
PacketCounter<rtcp::Fir>* fir() { return &fir_; }
PacketCounter<rtcp::Nack>* nack() { return &nack_; }
PacketCounter<rtcp::Pli>* pli() { return &pli_; }
PacketCounter<rtcp::RapidResyncRequest>* rrr() { return &rrr_; }
PacketCounter<rtcp::ReceiverReport>* receiver_report() {
return &receiver_report_;
}
PacketCounter<rtcp::Remb>* remb() { return &remb_; }
PacketCounter<rtcp::Sdes>* sdes() { return &sdes_; }
PacketCounter<rtcp::SenderReport>* sender_report() { return &sender_report_; }
PacketCounter<rtcp::Tmmbn>* tmmbn() { return &tmmbn_; }
PacketCounter<rtcp::Tmmbr>* tmmbr() { return &tmmbr_; }
PacketCounter<rtcp::TransportFeedback>* transport_feedback() {
return &transport_feedback_;
}
uint32_t sender_ssrc() const { return sender_ssrc_; }
private:
PacketCounter<rtcp::App> app_;
PacketCounter<rtcp::Bye> bye_;
PacketCounter<rtcp::ExtendedJitterReport> ij_;
PacketCounter<rtcp::ExtendedReports> xr_;
PacketCounter<rtcp::Fir> fir_;
PacketCounter<rtcp::Nack> nack_;
PacketCounter<rtcp::Pli> pli_;
PacketCounter<rtcp::RapidResyncRequest> rrr_;
PacketCounter<rtcp::ReceiverReport> receiver_report_;
PacketCounter<rtcp::Remb> remb_;
PacketCounter<rtcp::Sdes> sdes_;
PacketCounter<rtcp::SenderReport> sender_report_;
PacketCounter<rtcp::Tmmbn> tmmbn_;
PacketCounter<rtcp::Tmmbr> tmmbr_;
PacketCounter<rtcp::TransportFeedback> transport_feedback_;
uint32_t sender_ssrc_ = 0;
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_TEST_RTCP_PACKET_PARSER_H_