1 /*
2  *  Copyright (c) 2012 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 <assert.h>
12 #include <stdio.h>
13 #include <vector>
14 
15 #include "google/gflags.h"
16 #include "webrtc/base/scoped_ptr.h"
17 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
18 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
19 
20 // Flag validator.
ValidatePayloadType(const char * flagname,int32_t value)21 static bool ValidatePayloadType(const char* flagname, int32_t value) {
22   if (value >= 0 && value <= 127)  // Value is ok.
23     return true;
24   printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
25   return false;
26 }
ValidateExtensionId(const char * flagname,int32_t value)27 static bool ValidateExtensionId(const char* flagname, int32_t value) {
28   if (value > 0 && value <= 255)  // Value is ok.
29     return true;
30   printf("Invalid value for --%s: %d\n", flagname, static_cast<int>(value));
31   return false;
32 }
33 
34 // Define command line flags.
35 DEFINE_int32(red, 117, "RTP payload type for RED");
36 static const bool red_dummy =
37     google::RegisterFlagValidator(&FLAGS_red, &ValidatePayloadType);
38 DEFINE_int32(audio_level, 1, "Extension ID for audio level (RFC 6464)");
39 static const bool audio_level_dummy =
40     google::RegisterFlagValidator(&FLAGS_audio_level, &ValidateExtensionId);
41 DEFINE_int32(abs_send_time, 3, "Extension ID for absolute sender time");
42 static const bool abs_send_time_dummy =
43     google::RegisterFlagValidator(&FLAGS_abs_send_time, &ValidateExtensionId);
44 
main(int argc,char * argv[])45 int main(int argc, char* argv[]) {
46   std::string program_name = argv[0];
47   std::string usage =
48       "Tool for parsing an RTP dump file to text output.\n"
49       "Run " +
50       program_name +
51       " --helpshort for usage.\n"
52       "Example usage:\n" +
53       program_name + " input.rtp output.txt\n\n" +
54       "Output is sent to stdout if no output file is given." +
55       "Note that this tool can read files with our without payloads.";
56   google::SetUsageMessage(usage);
57   google::ParseCommandLineFlags(&argc, &argv, true);
58 
59   if (argc != 2 && argc != 3) {
60     // Print usage information.
61     printf("%s", google::ProgramUsage());
62     return 0;
63   }
64 
65   printf("Input file: %s\n", argv[1]);
66   rtc::scoped_ptr<webrtc::test::RtpFileSource> file_source(
67       webrtc::test::RtpFileSource::Create(argv[1]));
68   assert(file_source.get());
69   // Set RTP extension IDs.
70   bool print_audio_level = false;
71   if (!google::GetCommandLineFlagInfoOrDie("audio_level").is_default) {
72     print_audio_level = true;
73     file_source->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
74                                             FLAGS_audio_level);
75   }
76   bool print_abs_send_time = false;
77   if (!google::GetCommandLineFlagInfoOrDie("abs_send_time").is_default) {
78     print_abs_send_time = true;
79     file_source->RegisterRtpHeaderExtension(
80         webrtc::kRtpExtensionAbsoluteSendTime, FLAGS_abs_send_time);
81   }
82 
83   FILE* out_file;
84   if (argc == 3) {
85     out_file = fopen(argv[2], "wt");
86     if (!out_file) {
87       printf("Cannot open output file %s\n", argv[2]);
88       return -1;
89     }
90     printf("Output file: %s\n\n", argv[2]);
91   } else {
92     out_file = stdout;
93   }
94 
95   // Print file header.
96   fprintf(out_file, "SeqNo  TimeStamp   SendTime  Size    PT  M       SSRC");
97   if (print_audio_level) {
98     fprintf(out_file, " AuLvl (V)");
99   }
100   if (print_abs_send_time) {
101     fprintf(out_file, " AbsSendTime");
102   }
103   fprintf(out_file, "\n");
104 
105   uint32_t max_abs_send_time = 0;
106   int cycles = -1;
107   rtc::scoped_ptr<webrtc::test::Packet> packet;
108   while (true) {
109     packet.reset(file_source->NextPacket());
110     if (!packet.get()) {
111       // End of file reached.
112       break;
113     }
114     // Write packet data to file. Use virtual_packet_length_bytes so that the
115     // correct packet sizes are printed also for RTP header-only dumps.
116     fprintf(out_file,
117             "%5u %10u %10u %5i %5i %2i %#08X",
118             packet->header().sequenceNumber,
119             packet->header().timestamp,
120             static_cast<unsigned int>(packet->time_ms()),
121             static_cast<int>(packet->virtual_packet_length_bytes()),
122             packet->header().payloadType,
123             packet->header().markerBit,
124             packet->header().ssrc);
125     if (print_audio_level && packet->header().extension.hasAudioLevel) {
126       fprintf(out_file,
127               " %5u (%1i)",
128               packet->header().extension.audioLevel,
129               packet->header().extension.voiceActivity);
130     }
131     if (print_abs_send_time && packet->header().extension.hasAbsoluteSendTime) {
132       if (cycles == -1) {
133         // Initialize.
134         max_abs_send_time = packet->header().extension.absoluteSendTime;
135         cycles = 0;
136       }
137       // Abs sender time is 24 bit 6.18 fixed point. Shift by 8 to normalize to
138       // 32 bits (unsigned). Calculate the difference between this packet's
139       // send time and the maximum observed. Cast to signed 32-bit to get the
140       // desired wrap-around behavior.
141       if (static_cast<int32_t>(
142               (packet->header().extension.absoluteSendTime << 8) -
143               (max_abs_send_time << 8)) >= 0) {
144         // The difference is non-negative, meaning that this packet is newer
145         // than the previously observed maximum absolute send time.
146         if (packet->header().extension.absoluteSendTime < max_abs_send_time) {
147           // Wrap detected.
148           cycles++;
149         }
150         max_abs_send_time = packet->header().extension.absoluteSendTime;
151       }
152       // Abs sender time is 24 bit 6.18 fixed point. Divide by 2^18 to convert
153       // to floating point representation.
154       double send_time_seconds =
155           static_cast<double>(packet->header().extension.absoluteSendTime) /
156               262144 +
157           64.0 * cycles;
158       fprintf(out_file, " %11f", send_time_seconds);
159     }
160     fprintf(out_file, "\n");
161 
162     if (packet->header().payloadType == FLAGS_red) {
163       std::list<webrtc::RTPHeader*> red_headers;
164       packet->ExtractRedHeaders(&red_headers);
165       while (!red_headers.empty()) {
166         webrtc::RTPHeader* red = red_headers.front();
167         assert(red);
168         fprintf(out_file,
169                 "* %5u %10u %10u %5i\n",
170                 red->sequenceNumber,
171                 red->timestamp,
172                 static_cast<unsigned int>(packet->time_ms()),
173                 red->payloadType);
174         red_headers.pop_front();
175         delete red;
176       }
177     }
178   }
179 
180   fclose(out_file);
181 
182   return 0;
183 }
184