Add option to print information about configured SSRCs from RTC event logs.

webrtc/logging/rtc_event_log/rtc_event_log2text.cc

 /*
  *  Copyright (c) 2017 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.
  */
 
@@ skipping 19 matching lines @@
 #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/sli.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"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 namespace {
 
+DEFINE_bool(noconfig, true, "Excludes stream configurations.");
 DEFINE_bool(noincoming, false, "Excludes incoming packets.");
 DEFINE_bool(nooutgoing, false, "Excludes outgoing packets.");
 // TODO(terelius): Note that the media type doesn't work with outgoing packets.
 DEFINE_bool(noaudio, false, "Excludes audio packets.");
 // TODO(terelius): Note that the media type doesn't work with outgoing packets.
 DEFINE_bool(novideo, false, "Excludes video packets.");
 // TODO(terelius): Note that the media type doesn't work with outgoing packets.
 DEFINE_bool(nodata, false, "Excludes data packets.");
 DEFINE_bool(nortp, false, "Excludes RTP packets.");
 DEFINE_bool(nortcp, false, "Excludes RTCP packets.");
@@ skipping 71 matching lines @@
                        uint64_t log_timestamp,
                        webrtc::PacketDirection direction,
                        webrtc::MediaType media_type) {
   webrtc::rtcp::SenderReport sr;
   if (!sr.Parse(rtcp_block))
     return;
   if (ExcludePacket(direction, media_type, sr.sender_ssrc()))
     return;
   std::cout << log_timestamp << "\t"
             << "RTCP_SR" << StreamInfo(direction, media_type)
-            << "\tSSRC=" << sr.sender_ssrc()
+            << "\tssrc=" << sr.sender_ssrc()
             << "\ttimestamp=" << sr.rtp_timestamp() << std::endl;
 }
 
 void PrintReceiverReport(const webrtc::rtcp::CommonHeader& rtcp_block,
                          uint64_t log_timestamp,
                          webrtc::PacketDirection direction,
                          webrtc::MediaType media_type) {
   webrtc::rtcp::ReceiverReport rr;
   if (!rr.Parse(rtcp_block))
     return;
   if (ExcludePacket(direction, media_type, rr.sender_ssrc()))
     return;
   std::cout << log_timestamp << "\t"
             << "RTCP_RR" << StreamInfo(direction, media_type)
-            << "\tSSRC=" << rr.sender_ssrc() << std::endl;
+            << "\tssrc=" << rr.sender_ssrc() << std::endl;
 }
 
 void PrintXr(const webrtc::rtcp::CommonHeader& rtcp_block,
              uint64_t log_timestamp,
              webrtc::PacketDirection direction,
              webrtc::MediaType media_type) {
   webrtc::rtcp::ExtendedReports xr;
   if (!xr.Parse(rtcp_block))
     return;
   if (ExcludePacket(direction, media_type, xr.sender_ssrc()))
     return;
   std::cout << log_timestamp << "\t"
             << "RTCP_XR" << StreamInfo(direction, media_type)
-            << "\tSSRC=" << xr.sender_ssrc() << std::endl;
+            << "\tssrc=" << xr.sender_ssrc() << std::endl;
 }
 
 void PrintSdes(const webrtc::rtcp::CommonHeader& rtcp_block,
                uint64_t log_timestamp,
                webrtc::PacketDirection direction,
                webrtc::MediaType media_type) {
   std::cout << log_timestamp << "\t"
             << "RTCP_SDES" << StreamInfo(direction, media_type) << std::endl;
   RTC_NOTREACHED() << "SDES should have been redacted when writing the log";
 }
 
 void PrintBye(const webrtc::rtcp::CommonHeader& rtcp_block,
               uint64_t log_timestamp,
               webrtc::PacketDirection direction,
               webrtc::MediaType media_type) {
   webrtc::rtcp::Bye bye;
   if (!bye.Parse(rtcp_block))
     return;
   if (ExcludePacket(direction, media_type, bye.sender_ssrc()))
     return;
   std::cout << log_timestamp << "\t"
             << "RTCP_BYE" << StreamInfo(direction, media_type)
-            << "\tSSRC=" << bye.sender_ssrc() << std::endl;
+            << "\tssrc=" << bye.sender_ssrc() << std::endl;
 }
 
 void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
                       uint64_t log_timestamp,
                       webrtc::PacketDirection direction,
                       webrtc::MediaType media_type) {
-  std::cout << "Rtp feedback found";
   switch (rtcp_block.fmt()) {
     case webrtc::rtcp::Nack::kFeedbackMessageType: {
       webrtc::rtcp::Nack nack;
       if (!nack.Parse(rtcp_block))
         return;
       if (ExcludePacket(direction, media_type, nack.sender_ssrc()))
         return;
       std::cout << log_timestamp << "\t"
                 << "RTCP_NACK" << StreamInfo(direction, media_type)
-                << "\tSSRC=" << nack.sender_ssrc() << std::endl;
+                << "\tssrc=" << nack.sender_ssrc() << std::endl;
       break;
     }
     case webrtc::rtcp::Tmmbr::kFeedbackMessageType: {
       webrtc::rtcp::Tmmbr tmmbr;
       if (!tmmbr.Parse(rtcp_block))
         return;
       if (ExcludePacket(direction, media_type, tmmbr.sender_ssrc()))
         return;
       std::cout << log_timestamp << "\t"
                 << "RTCP_TMMBR" << StreamInfo(direction, media_type)
-                << "\tSSRC=" << tmmbr.sender_ssrc() << std::endl;
+                << "\tssrc=" << tmmbr.sender_ssrc() << std::endl;
       break;
     }
     case webrtc::rtcp::Tmmbn::kFeedbackMessageType: {
       webrtc::rtcp::Tmmbn tmmbn;
       if (!tmmbn.Parse(rtcp_block))
         return;
       if (ExcludePacket(direction, media_type, tmmbn.sender_ssrc()))
         return;
       std::cout << log_timestamp << "\t"
                 << "RTCP_TMMBN" << StreamInfo(direction, media_type)
-                << "\tSSRC=" << tmmbn.sender_ssrc() << std::endl;
+                << "\tssrc=" << tmmbn.sender_ssrc() << std::endl;
       break;
     }
     case webrtc::rtcp::RapidResyncRequest::kFeedbackMessageType: {
       webrtc::rtcp::RapidResyncRequest sr_req;
       if (!sr_req.Parse(rtcp_block))
         return;
       if (ExcludePacket(direction, media_type, sr_req.sender_ssrc()))
         return;
       std::cout << log_timestamp << "\t"
                 << "RTCP_SRREQ" << StreamInfo(direction, media_type)
-                << "\tSSRC=" << sr_req.sender_ssrc() << std::endl;
+                << "\tssrc=" << sr_req.sender_ssrc() << std::endl;
       break;
     }
     case webrtc::rtcp::TransportFeedback::kFeedbackMessageType: {
       webrtc::rtcp::TransportFeedback transport_feedback;
       if (!transport_feedback.Parse(rtcp_block))
         return;
       if (ExcludePacket(direction, media_type,
                         transport_feedback.sender_ssrc()))
         return;
       std::cout << log_timestamp << "\t"
                 << "RTCP_NEWFB" << StreamInfo(direction, media_type)
-                << "\tSSRC=" << transport_feedback.sender_ssrc() << std::endl;
+                << "\tssrc=" << transport_feedback.sender_ssrc() << std::endl;
       break;
     }
     default:
-      RTC_DCHECK(false);
       break;
   }
 }
 
 void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
                      uint64_t log_timestamp,
                      webrtc::PacketDirection direction,
                      webrtc::MediaType media_type) {
   switch (rtcp_block.fmt()) {
     case webrtc::rtcp::Pli::kFeedbackMessageType: {
       webrtc::rtcp::Pli pli;
       if (!pli.Parse(rtcp_block))
         return;
       if (ExcludePacket(direction, media_type, pli.sender_ssrc()))
         return;
       std::cout << log_timestamp << "\t"
                 << "RTCP_PLI" << StreamInfo(direction, media_type)
-                << "\tSSRC=" << pli.sender_ssrc() << std::endl;
+                << "\tssrc=" << pli.sender_ssrc() << std::endl;
       break;
     }
     case webrtc::rtcp::Sli::kFeedbackMessageType: {
       webrtc::rtcp::Sli sli;
       if (!sli.Parse(rtcp_block))
         return;
       if (ExcludePacket(direction, media_type, sli.sender_ssrc()))
         return;
       std::cout << log_timestamp << "\t"
                 << "RTCP_SLI" << StreamInfo(direction, media_type)
-                << "\tSSRC=" << sli.sender_ssrc() << std::endl;
+                << "\tssrc=" << sli.sender_ssrc() << std::endl;
       break;
     }
     case webrtc::rtcp::Rpsi::kFeedbackMessageType: {
       webrtc::rtcp::Rpsi rpsi;
       if (!rpsi.Parse(rtcp_block))
         return;
       if (ExcludePacket(direction, media_type, rpsi.sender_ssrc()))
         return;
       std::cout << log_timestamp << "\t"
                 << "RTCP_RPSI" << StreamInfo(direction, media_type)
-                << "\tSSRC=" << rpsi.sender_ssrc() << std::endl;
+                << "\tssrc=" << rpsi.sender_ssrc() << std::endl;
       break;
     }
     case webrtc::rtcp::Fir::kFeedbackMessageType: {
       webrtc::rtcp::Fir fir;
       if (!fir.Parse(rtcp_block))
         return;
       if (ExcludePacket(direction, media_type, fir.sender_ssrc()))
         return;
       std::cout << log_timestamp << "\t"
                 << "RTCP_FIR" << StreamInfo(direction, media_type)
-                << "\tSSRC=" << fir.sender_ssrc() << std::endl;
+                << "\tssrc=" << fir.sender_ssrc() << std::endl;
       break;
     }
     case webrtc::rtcp::Remb::kFeedbackMessageType: {
       webrtc::rtcp::Remb remb;
       if (!remb.Parse(rtcp_block))
         return;
       if (ExcludePacket(direction, media_type, remb.sender_ssrc()))
         return;
       std::cout << log_timestamp << "\t"
                 << "RTCP_REMB" << StreamInfo(direction, media_type)
-                << "\tSSRC=" << remb.sender_ssrc() << std::endl;
+                << "\tssrc=" << remb.sender_ssrc() << std::endl;
       break;
     }
     default:
       break;
   }
 }
 
 }  // namespace
 
 // This utility will print basic information about each packet to stdout.
@@ skipping 20 matching lines @@
   if (!FLAGS_ssrc.empty())
     RTC_CHECK(ParseSsrc(FLAGS_ssrc)) << "Flag verification has failed.";
 
   webrtc::ParsedRtcEventLog parsed_stream;
   if (!parsed_stream.ParseFile(input_file)) {
     std::cerr << "Error while parsing input file: " << input_file << std::endl;
     return -1;
   }
 
   for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
+    if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_noincoming &&
+        parsed_stream.GetEventType(i) ==
+            webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
+      webrtc::VideoReceiveStream::Config config(nullptr);
+      parsed_stream.GetVideoReceiveConfig(i, &config);
+      std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
+                << "\tssrc=" << config.rtp.remote_ssrc
+                << "\tfeedback_ssrc=" << config.rtp.local_ssrc << std::endl;
+    }
+    if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_nooutgoing &&
+        parsed_stream.GetEventType(i) ==
+            webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
+      webrtc::VideoSendStream::Config config(nullptr);
+      parsed_stream.GetVideoSendConfig(i, &config);
+      std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
+      std::cout << "\tssrcs=";
+      for (const auto& ssrc : config.rtp.ssrcs)
+        std::cout << ssrc << ',';
+      std::cout << "\trtx_ssrcs=";
+      for (const auto& ssrc : config.rtp.rtx.ssrcs)
+        std::cout << ssrc << ',';
+      std::cout << std::endl;
+    }
+    if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_noincoming &&
+        parsed_stream.GetEventType(i) ==
+            webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
+      webrtc::AudioReceiveStream::Config config;
+      parsed_stream.GetAudioReceiveConfig(i, &config);
+      std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
+                << "\tssrc=" << config.rtp.remote_ssrc
+                << "\tfeedback_ssrc=" << config.rtp.local_ssrc << std::endl;
+    }
+    if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_nooutgoing &&
+        parsed_stream.GetEventType(i) ==
+            webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
+      webrtc::AudioSendStream::Config config(nullptr);
+      parsed_stream.GetAudioSendConfig(i, &config);
+      std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
+                << "\tssrc=" << config.rtp.ssrc << std::endl;
+    }
     if (!FLAGS_nortp &&
         parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
       size_t header_length;
       size_t total_length;
       uint8_t header[IP_PACKET_SIZE];
       webrtc::PacketDirection direction;
       webrtc::MediaType media_type;
       parsed_stream.GetRtpHeader(i, &direction, &media_type, header,
                                  &header_length, &total_length);
 
       // Parse header to get SSRC and RTP time.
       webrtc::RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
       webrtc::RTPHeader parsed_header;
       rtp_parser.Parse(&parsed_header);
 
       if (ExcludePacket(direction, media_type, parsed_header.ssrc))
         continue;
 
       std::cout << parsed_stream.GetTimestamp(i) << "\tRTP"
                 << StreamInfo(direction, media_type)
-                << "\tSSRC=" << parsed_header.ssrc
+                << "\tssrc=" << parsed_header.ssrc
                 << "\ttimestamp=" << parsed_header.timestamp << std::endl;
     }
     if (!FLAGS_nortcp &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::RTCP_EVENT) {
       size_t length;
       uint8_t packet[IP_PACKET_SIZE];
       webrtc::PacketDirection direction;
       webrtc::MediaType media_type;
       parsed_stream.GetRtcpPacket(i, &direction, &media_type, packet, &length);
@@ skipping 33 matching lines @@
             PrintPsFeedback(rtcp_block, log_timestamp, direction, media_type);
             break;
           default:
             break;
         }
       }
     }
   }
   return 0;
 }