From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001
From: Daniel Baumann <daniel.baumann@progress-linux.org>
Date: Fri, 19 Apr 2024 02:47:55 +0200
Subject: Adding upstream version 124.0.1.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
---
 third_party/libwebrtc/test/rtcp_packet_parser.h | 130 ++++++++++++++++++++++++
 1 file changed, 130 insertions(+)
 create mode 100644 third_party/libwebrtc/test/rtcp_packet_parser.h

(limited to 'third_party/libwebrtc/test/rtcp_packet_parser.h')

diff --git a/third_party/libwebrtc/test/rtcp_packet_parser.h b/third_party/libwebrtc/test/rtcp_packet_parser.h
new file mode 100644
index 0000000000..0aa1cbf499
--- /dev/null
+++ b/third_party/libwebrtc/test/rtcp_packet_parser.h
@@ -0,0 +1,130 @@
+/*
+ *  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 TEST_RTCP_PACKET_PARSER_H_
+#define TEST_RTCP_PACKET_PARSER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/app.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/fir.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/loss_notification.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/pli.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sdes.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "rtc_base/checks.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_;
+    }
+    bool Parse(const rtcp::CommonHeader& header, uint32_t* sender_ssrc) {
+      const bool result = TypedRtcpPacket::Parse(header);
+      if (result) {
+        ++num_packets_;
+        if (*sender_ssrc == 0)  // Use first sender ssrc in compound packet.
+          *sender_ssrc = TypedRtcpPacket::sender_ssrc();
+      }
+      return result;
+    }
+
+   private:
+    int num_packets_ = 0;
+  };
+
+  RtcpPacketParser();
+  ~RtcpPacketParser();
+
+  bool Parse(rtc::ArrayView<const uint8_t> packet);
+
+  PacketCounter<rtcp::App>* app() { return &app_; }
+  PacketCounter<rtcp::Bye>* bye() { return &bye_; }
+  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::LossNotification>* loss_notification() {
+    return &loss_notification_;
+  }
+  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_; }
+  size_t processed_rtcp_packets() const { return processed_rtcp_packets_; }
+
+ private:
+  PacketCounter<rtcp::App> app_;
+  PacketCounter<rtcp::Bye> bye_;
+  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::LossNotification> loss_notification_;
+  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;
+  size_t processed_rtcp_packets_ = 0;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // TEST_RTCP_PACKET_PARSER_H_
-- 
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