1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include <iostream>
12 #include <sstream>
13 #include <string>
14
15 #include "gflags/gflags.h"
16 #include "webrtc/base/checks.h"
17 #include "webrtc/base/scoped_ptr.h"
18 #include "webrtc/call/rtc_event_log.h"
19 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
20 #include "webrtc/test/rtp_file_writer.h"
21
22 // Files generated at build-time by the protobuf compiler.
23 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
24 #include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
25 #else
26 #include "webrtc/call/rtc_event_log.pb.h"
27 #endif
28
29 namespace {
30
31 DEFINE_bool(noaudio,
32 false,
33 "Excludes audio packets from the converted RTPdump file.");
34 DEFINE_bool(novideo,
35 false,
36 "Excludes video packets from the converted RTPdump file.");
37 DEFINE_bool(nodata,
38 false,
39 "Excludes data packets from the converted RTPdump file.");
40 DEFINE_bool(nortp,
41 false,
42 "Excludes RTP packets from the converted RTPdump file.");
43 DEFINE_bool(nortcp,
44 false,
45 "Excludes RTCP packets from the converted RTPdump file.");
46 DEFINE_string(ssrc,
47 "",
48 "Store only packets with this SSRC (decimal or hex, the latter "
49 "starting with 0x).");
50
51 // Parses the input string for a valid SSRC. If a valid SSRC is found, it is
52 // written to the output variable |ssrc|, and true is returned. Otherwise,
53 // false is returned.
54 // The empty string must be validated as true, because it is the default value
55 // of the command-line flag. In this case, no value is written to the output
56 // variable.
ParseSsrc(std::string str,uint32_t * ssrc)57 bool ParseSsrc(std::string str, uint32_t* ssrc) {
58 // If the input string starts with 0x or 0X it indicates a hexadecimal number.
59 auto read_mode = std::dec;
60 if (str.size() > 2 &&
61 (str.substr(0, 2) == "0x" || str.substr(0, 2) == "0X")) {
62 read_mode = std::hex;
63 str = str.substr(2);
64 }
65 std::stringstream ss(str);
66 ss >> read_mode >> *ssrc;
67 return str.empty() || (!ss.fail() && ss.eof());
68 }
69
70 } // namespace
71
72 // This utility will convert a stored event log to the rtpdump format.
main(int argc,char * argv[])73 int main(int argc, char* argv[]) {
74 std::string program_name = argv[0];
75 std::string usage =
76 "Tool for converting an RtcEventLog file to an RTP dump file.\n"
77 "Run " +
78 program_name +
79 " --helpshort for usage.\n"
80 "Example usage:\n" +
81 program_name + " input.rel output.rtp\n";
82 google::SetUsageMessage(usage);
83 google::ParseCommandLineFlags(&argc, &argv, true);
84
85 if (argc != 3) {
86 std::cout << google::ProgramUsage();
87 return 0;
88 }
89 std::string input_file = argv[1];
90 std::string output_file = argv[2];
91
92 uint32_t ssrc_filter = 0;
93 if (!FLAGS_ssrc.empty())
94 RTC_CHECK(ParseSsrc(FLAGS_ssrc, &ssrc_filter))
95 << "Flag verification has failed.";
96
97 webrtc::rtclog::EventStream event_stream;
98 if (!webrtc::RtcEventLog::ParseRtcEventLog(input_file, &event_stream)) {
99 std::cerr << "Error while parsing input file: " << input_file << std::endl;
100 return -1;
101 }
102
103 rtc::scoped_ptr<webrtc::test::RtpFileWriter> rtp_writer(
104 webrtc::test::RtpFileWriter::Create(
105 webrtc::test::RtpFileWriter::FileFormat::kRtpDump, output_file));
106
107 if (!rtp_writer.get()) {
108 std::cerr << "Error while opening output file: " << output_file
109 << std::endl;
110 return -1;
111 }
112
113 std::cout << "Found " << event_stream.stream_size()
114 << " events in the input file." << std::endl;
115 int rtp_counter = 0, rtcp_counter = 0;
116 bool header_only = false;
117 // TODO(ivoc): This can be refactored once the packet interpretation
118 // functions are finished.
119 for (int i = 0; i < event_stream.stream_size(); i++) {
120 const webrtc::rtclog::Event& event = event_stream.stream(i);
121 if (!FLAGS_nortp && event.has_type() && event.type() == event.RTP_EVENT) {
122 if (event.has_timestamp_us() && event.has_rtp_packet() &&
123 event.rtp_packet().has_header() &&
124 event.rtp_packet().header().size() >= 12 &&
125 event.rtp_packet().has_packet_length() &&
126 event.rtp_packet().has_type()) {
127 const webrtc::rtclog::RtpPacket& rtp_packet = event.rtp_packet();
128 if (FLAGS_noaudio && rtp_packet.type() == webrtc::rtclog::AUDIO)
129 continue;
130 if (FLAGS_novideo && rtp_packet.type() == webrtc::rtclog::VIDEO)
131 continue;
132 if (FLAGS_nodata && rtp_packet.type() == webrtc::rtclog::DATA)
133 continue;
134 if (!FLAGS_ssrc.empty()) {
135 const uint32_t packet_ssrc =
136 webrtc::ByteReader<uint32_t>::ReadBigEndian(
137 reinterpret_cast<const uint8_t*>(rtp_packet.header().data() +
138 8));
139 if (packet_ssrc != ssrc_filter)
140 continue;
141 }
142
143 webrtc::test::RtpPacket packet;
144 packet.length = rtp_packet.header().size();
145 if (packet.length > packet.kMaxPacketBufferSize) {
146 std::cout << "Skipping packet with size " << packet.length
147 << ", the maximum supported size is "
148 << packet.kMaxPacketBufferSize << std::endl;
149 continue;
150 }
151 packet.original_length = rtp_packet.packet_length();
152 if (packet.original_length > packet.length)
153 header_only = true;
154 packet.time_ms = event.timestamp_us() / 1000;
155 memcpy(packet.data, rtp_packet.header().data(), packet.length);
156 rtp_writer->WritePacket(&packet);
157 rtp_counter++;
158 } else {
159 std::cout << "Skipping malformed event." << std::endl;
160 }
161 }
162 if (!FLAGS_nortcp && event.has_type() && event.type() == event.RTCP_EVENT) {
163 if (event.has_timestamp_us() && event.has_rtcp_packet() &&
164 event.rtcp_packet().has_type() &&
165 event.rtcp_packet().has_packet_data() &&
166 event.rtcp_packet().packet_data().size() > 0) {
167 const webrtc::rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
168 if (FLAGS_noaudio && rtcp_packet.type() == webrtc::rtclog::AUDIO)
169 continue;
170 if (FLAGS_novideo && rtcp_packet.type() == webrtc::rtclog::VIDEO)
171 continue;
172 if (FLAGS_nodata && rtcp_packet.type() == webrtc::rtclog::DATA)
173 continue;
174 if (!FLAGS_ssrc.empty()) {
175 const uint32_t packet_ssrc =
176 webrtc::ByteReader<uint32_t>::ReadBigEndian(
177 reinterpret_cast<const uint8_t*>(
178 rtcp_packet.packet_data().data() + 4));
179 if (packet_ssrc != ssrc_filter)
180 continue;
181 }
182
183 webrtc::test::RtpPacket packet;
184 packet.length = rtcp_packet.packet_data().size();
185 if (packet.length > packet.kMaxPacketBufferSize) {
186 std::cout << "Skipping packet with size " << packet.length
187 << ", the maximum supported size is "
188 << packet.kMaxPacketBufferSize << std::endl;
189 continue;
190 }
191 // For RTCP packets the original_length should be set to 0 in the
192 // RTPdump format.
193 packet.original_length = 0;
194 packet.time_ms = event.timestamp_us() / 1000;
195 memcpy(packet.data, rtcp_packet.packet_data().data(), packet.length);
196 rtp_writer->WritePacket(&packet);
197 rtcp_counter++;
198 } else {
199 std::cout << "Skipping malformed event." << std::endl;
200 }
201 }
202 }
203 std::cout << "Wrote " << rtp_counter << (header_only ? " header-only" : "")
204 << " RTP packets and " << rtcp_counter << " RTCP packets to the "
205 << "output file." << std::endl;
206 return 0;
207 }
208