Change rtp_event_log2text to ignore webrtc::MediaType from proto.

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 55 matching lines @@
   if (str.size() > 2 &&
       (str.substr(0, 2) == "0x" || str.substr(0, 2) == "0X")) {
     read_mode = std::hex;
     str = str.substr(2);
   }
   std::stringstream ss(str);
   ss >> read_mode >> filtered_ssrc;
   return str.empty() || (!ss.fail() && ss.eof());
 }
 
+// Struct used for storing all SSRCs used in a Stream.

[terelius, 2017/05/19 11:55:59]

Why "all"? The struct only holds one SSRC.

[perkj_webrtc, 2017/05/19 12:02:52]

Done.

+struct Stream {
+  Stream(uint32_t ssrc,
+         webrtc::MediaType media_type,
+         webrtc::PacketDirection direction)
+      : ssrc(ssrc), media_type(media_type), direction(direction) {}
+  uint32_t ssrc;
+  webrtc::MediaType media_type;
+  webrtc::PacketDirection direction;
+};
+
+// All configured streams found in the event log.
+std::vector<Stream> global_streams;
+
+// Returns the MediaType for registered SSRCs. Search from the end to use last
+// registered types first.
+webrtc::MediaType GetMediaType(uint32_t ssrc,
+                               webrtc::PacketDirection direction) {
+  for (auto rit = global_streams.rbegin(); rit != global_streams.rend();
+       ++rit) {
+    if (rit->ssrc == ssrc && rit->direction == direction)
+      return rit->media_type;
+  }
+  return webrtc::MediaType::ANY;
+}
+
 bool ExcludePacket(webrtc::PacketDirection direction,
                    webrtc::MediaType media_type,
                    uint32_t packet_ssrc) {
   if (FLAGS_nooutgoing && direction == webrtc::kOutgoingPacket)
     return true;
   if (FLAGS_noincoming && direction == webrtc::kIncomingPacket)
     return true;
   if (FLAGS_noaudio && media_type == webrtc::MediaType::AUDIO)
     return true;
   if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO)
@@ skipping 25 matching lines @@
     else if (media_type == webrtc::MediaType::DATA)
       return "(in,data)";
     else
       return "(in)";
   }
   return "(unknown)";
 }
 
 void PrintSenderReport(const webrtc::rtcp::CommonHeader& rtcp_block,
                        uint64_t log_timestamp,
-                       webrtc::PacketDirection direction,
-                       webrtc::MediaType media_type) {
+                       webrtc::PacketDirection direction) {
   webrtc::rtcp::SenderReport sr;
   if (!sr.Parse(rtcp_block))
     return;
+  webrtc::MediaType media_type = GetMediaType(sr.sender_ssrc(), direction);
   if (ExcludePacket(direction, media_type, sr.sender_ssrc()))
     return;
   std::cout << log_timestamp << "\t"
             << "RTCP_SR" << StreamInfo(direction, media_type)
             << "\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::PacketDirection direction) {
   webrtc::rtcp::ReceiverReport rr;
   if (!rr.Parse(rtcp_block))
     return;
+  webrtc::MediaType media_type = GetMediaType(rr.sender_ssrc(), direction);
   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;
 }
 
 void PrintXr(const webrtc::rtcp::CommonHeader& rtcp_block,
              uint64_t log_timestamp,
-             webrtc::PacketDirection direction,
-             webrtc::MediaType media_type) {
+             webrtc::PacketDirection direction) {
   webrtc::rtcp::ExtendedReports xr;
   if (!xr.Parse(rtcp_block))
     return;
+  webrtc::MediaType media_type = GetMediaType(xr.sender_ssrc(), direction);
   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;
 }
 
 void PrintSdes(const webrtc::rtcp::CommonHeader& rtcp_block,
                uint64_t log_timestamp,
-               webrtc::PacketDirection direction,
-               webrtc::MediaType media_type) {
+               webrtc::PacketDirection direction) {
   std::cout << log_timestamp << "\t"
-            << "RTCP_SDES" << StreamInfo(direction, media_type) << std::endl;
+            << "RTCP_SDES" << StreamInfo(direction, webrtc::MediaType::ANY)
+            << 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::PacketDirection direction) {
   webrtc::rtcp::Bye bye;
   if (!bye.Parse(rtcp_block))
     return;
+  webrtc::MediaType media_type = GetMediaType(bye.sender_ssrc(), direction);
   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;
 }
 
 void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
                       uint64_t log_timestamp,
-                      webrtc::PacketDirection direction,
-                      webrtc::MediaType media_type) {
+                      webrtc::PacketDirection direction) {
   switch (rtcp_block.fmt()) {
     case webrtc::rtcp::Nack::kFeedbackMessageType: {
       webrtc::rtcp::Nack nack;
       if (!nack.Parse(rtcp_block))
         return;
+      webrtc::MediaType media_type =
+          GetMediaType(nack.sender_ssrc(), direction);
       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;
       break;
     }
     case webrtc::rtcp::Tmmbr::kFeedbackMessageType: {
       webrtc::rtcp::Tmmbr tmmbr;
       if (!tmmbr.Parse(rtcp_block))
         return;
+      webrtc::MediaType media_type =
+          GetMediaType(tmmbr.sender_ssrc(), direction);
       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;
       break;
     }
     case webrtc::rtcp::Tmmbn::kFeedbackMessageType: {
       webrtc::rtcp::Tmmbn tmmbn;
       if (!tmmbn.Parse(rtcp_block))
         return;
+      webrtc::MediaType media_type =
+          GetMediaType(tmmbn.sender_ssrc(), direction);
       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;
       break;
     }
     case webrtc::rtcp::RapidResyncRequest::kFeedbackMessageType: {
       webrtc::rtcp::RapidResyncRequest sr_req;
       if (!sr_req.Parse(rtcp_block))
         return;
+      webrtc::MediaType media_type =
+          GetMediaType(sr_req.sender_ssrc(), direction);
       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;
       break;
     }
     case webrtc::rtcp::TransportFeedback::kFeedbackMessageType: {
       webrtc::rtcp::TransportFeedback transport_feedback;
       if (!transport_feedback.Parse(rtcp_block))
         return;
+      webrtc::MediaType media_type =
+          GetMediaType(transport_feedback.sender_ssrc(), direction);
       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;
       break;
     }
     default:
       break;
   }
 }
 
 void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
                      uint64_t log_timestamp,
-                     webrtc::PacketDirection direction,
-                     webrtc::MediaType media_type) {
+                     webrtc::PacketDirection direction) {
   switch (rtcp_block.fmt()) {
     case webrtc::rtcp::Pli::kFeedbackMessageType: {
       webrtc::rtcp::Pli pli;
       if (!pli.Parse(rtcp_block))
         return;
+      webrtc::MediaType media_type = GetMediaType(pli.sender_ssrc(), direction);
       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;
       break;
     }
     case webrtc::rtcp::Fir::kFeedbackMessageType: {
       webrtc::rtcp::Fir fir;
       if (!fir.Parse(rtcp_block))
         return;
+      webrtc::MediaType media_type = GetMediaType(fir.sender_ssrc(), direction);
       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;
       break;
     }
     case webrtc::rtcp::Remb::kFeedbackMessageType: {
       webrtc::rtcp::Remb remb;
       if (!remb.Parse(rtcp_block))
         return;
+      webrtc::MediaType media_type =
+          GetMediaType(remb.sender_ssrc(), direction);
       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;
       break;
     }
     default:
       break;
   }
@@ skipping 25 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) {
+    if (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;
+
+      global_streams.emplace_back(config.rtp.remote_ssrc,
+                                  webrtc::MediaType::VIDEO,
+                                  webrtc::kIncomingPacket);
+      global_streams.emplace_back(config.rtp.local_ssrc,
+                                  webrtc::MediaType::VIDEO,
+                                  webrtc::kOutgoingPacket);
+
+      if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_noincoming) {
+        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) {
+    if (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;
+
+      for (uint32_t ssrc : config.rtp.ssrcs) {
+        global_streams.emplace_back(ssrc, webrtc::MediaType::VIDEO,
+                                    webrtc::kOutgoingPacket);
+      }
+      for (uint32_t ssrc : config.rtp.rtx.ssrcs) {
+        global_streams.emplace_back(ssrc, webrtc::MediaType::VIDEO,
+                                    webrtc::kOutgoingPacket);
+      }
+      if (config.rtp.flexfec.ssrc > 0) {

[terelius, 2017/05/19 11:55:59]

We don't log FlexFEC SSRCs, so this is always going to be empty. It also does
not exist in your upcoming StreamConfig struct.

[perkj_webrtc, 2017/05/19 12:02:52]

removed.

+        global_streams.emplace_back(config.rtp.flexfec.ssrc,
+                                    webrtc::MediaType::VIDEO,
+                                    webrtc::kOutgoingPacket);
+      }
+
+      if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_nooutgoing) {
+        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) {
+    if (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;
+      global_streams.emplace_back(config.rtp.remote_ssrc,
+                                  webrtc::MediaType::AUDIO,
+                                  webrtc::kIncomingPacket);
+      global_streams.emplace_back(config.rtp.local_ssrc,
+                                  webrtc::MediaType::AUDIO,
+                                  webrtc::kOutgoingPacket);
+      if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_noincoming) {
+        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) {
+    if (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;
+      global_streams.emplace_back(config.rtp.ssrc, webrtc::MediaType::AUDIO,
+                                  webrtc::kOutgoingPacket);
+      if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_nooutgoing) {
+        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);
+      media_type = GetMediaType(parsed_header.ssrc, direction);
 
       if (ExcludePacket(direction, media_type, parsed_header.ssrc))
         continue;
 
       std::cout << parsed_stream.GetTimestamp(i) << "\tRTP"
                 << StreamInfo(direction, media_type)
                 << "\tssrc=" << parsed_header.ssrc
                 << "\ttimestamp=" << parsed_header.timestamp << std::endl;
     }
     if (!FLAGS_nortcp &&
@@ skipping 11 matching lines @@
            next_block = rtcp_block.NextPacket()) {
         ptrdiff_t remaining_blocks_size = packet_end - next_block;
         RTC_DCHECK_GT(remaining_blocks_size, 0);
         if (!rtcp_block.Parse(next_block, remaining_blocks_size)) {
           break;
         }
 
         uint64_t log_timestamp = parsed_stream.GetTimestamp(i);
         switch (rtcp_block.type()) {
           case webrtc::rtcp::SenderReport::kPacketType:
-            PrintSenderReport(rtcp_block, log_timestamp, direction, media_type);
+            PrintSenderReport(rtcp_block, log_timestamp, direction);
             break;
           case webrtc::rtcp::ReceiverReport::kPacketType:
-            PrintReceiverReport(rtcp_block, log_timestamp, direction,
-                                media_type);
+            PrintReceiverReport(rtcp_block, log_timestamp, direction);
             break;
           case webrtc::rtcp::Sdes::kPacketType:
-            PrintSdes(rtcp_block, log_timestamp, direction, media_type);
+            PrintSdes(rtcp_block, log_timestamp, direction);
             break;
           case webrtc::rtcp::ExtendedReports::kPacketType:
-            PrintXr(rtcp_block, log_timestamp, direction, media_type);
+            PrintXr(rtcp_block, log_timestamp, direction);
             break;
           case webrtc::rtcp::Bye::kPacketType:
-            PrintBye(rtcp_block, log_timestamp, direction, media_type);
+            PrintBye(rtcp_block, log_timestamp, direction);
             break;
           case webrtc::rtcp::Rtpfb::kPacketType:
-            PrintRtpFeedback(rtcp_block, log_timestamp, direction, media_type);
+            PrintRtpFeedback(rtcp_block, log_timestamp, direction);
             break;
           case webrtc::rtcp::Psfb::kPacketType:
-            PrintPsFeedback(rtcp_block, log_timestamp, direction, media_type);
+            PrintPsFeedback(rtcp_block, log_timestamp, direction);
             break;
           default:
             break;
         }
       }
     }
   }
   return 0;
 }