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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/modules/rtp_rtcp/source/byte_io.h" | |
19 #include "webrtc/test/rtp_file_writer.h" | |
20 #include "webrtc/video/rtc_event_log.h" | |
21 | |
22 // Files generated at build-time by the protobuf compiler. | |
23 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD | |
24 #include "external/webrtc/webrtc/video/rtc_event_log.pb.h" | |
25 #else | |
26 #include "webrtc/video/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. | |
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. | |
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 } | |
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