Make rtc_event_log2text output header extensions

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.
  */
 
 #include <iostream>
+#include <map>
 #include <sstream>
 #include <string>
+#include <utility>  // pair
 
 #include "gflags/gflags.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
@@ skipping 317 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;
   }
 
+  typedef std::pair<uint32_t, webrtc::PacketDirection> StreamId;
+  // For each separate stream identified by StreamId we need a
+  // separate extension_map, because negotiated extensions may differ.
+  std::map<StreamId, webrtc::RtpHeaderExtensionMap> extension_maps;
+
   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::rtclog::StreamConfig config =
           parsed_stream.GetVideoReceiveConfig(i);
+      extension_maps[StreamId(config.remote_ssrc, webrtc::kIncomingPacket)] =
+          webrtc::RtpHeaderExtensionMap(config.rtp_extensions);
+      extension_maps[StreamId(config.rtx_ssrc, webrtc::kIncomingPacket)] =
+          webrtc::RtpHeaderExtensionMap(config.rtp_extensions);
       std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
                 << "\tssrc=" << config.remote_ssrc
                 << "\tfeedback_ssrc=" << config.local_ssrc;
       std::cout << "\textensions={";
       for (const auto& extension : config.rtp_extensions) {
         std::cout << extension.ToString() << ",";
       }
       std::cout << "}";
       std::cout << "\tcodecs={";
       for (const auto& codec : config.codecs) {
         std::cout << "{name: " << codec.payload_name
                   << ", payload_type: " << codec.payload_type
                   << ", rtx_payload_type: " << codec.rtx_payload_type << "}";
       }
       std::cout << "}" << std::endl;
     }
     if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_nooutgoing &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
       std::vector<webrtc::rtclog::StreamConfig> configs =
           parsed_stream.GetVideoSendConfig(i);
       for (const auto& config : configs) {
+        extension_maps[StreamId(config.local_ssrc, webrtc::kOutgoingPacket)] =
+            webrtc::RtpHeaderExtensionMap(config.rtp_extensions);
+        extension_maps[StreamId(config.rtx_ssrc, webrtc::kOutgoingPacket)] =
+            webrtc::RtpHeaderExtensionMap(config.rtp_extensions);

[terelius, 2017/06/02 10:55:03]

Maybe this should be parsed from audio configs too even though it's not yet
written to the log? WDYT?

[ilnik, 2017/06/02 11:25:59]

Done.

         std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
         std::cout << "\tssrcs=" << config.local_ssrc;
         std::cout << "\trtx_ssrcs=" << config.rtx_ssrc;
         std::cout << "\textensions={";
         for (const auto& extension : config.rtp_extensions) {
           std::cout << extension.ToString() << ",";
         }
         std::cout << "}";
         std::cout << "\tcodecs={";
         for (const auto& codec : config.codecs) {
@@ skipping 43 matching lines @@
                   << ", rtx_payload_type: " << codec.rtx_payload_type << "}";
       }
       std::cout << "}" << 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;
-
       parsed_stream.GetRtpHeader(i, &direction, 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);
+      StreamId stream_id(parsed_header.ssrc, direction);
+      if (extension_maps.count(stream_id) == 1) {
+        webrtc::RtpHeaderExtensionMap* extension_map =
+            &extension_maps[stream_id];
+        rtp_parser.Parse(&parsed_header, extension_map);
+      }
       MediaType media_type =
           parsed_stream.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;
+                << "\ttimestamp=" << parsed_header.timestamp;
+      if (parsed_header.extension.hasAbsoluteSendTime) {
+        std::cout << "\tAbsSendTime="

[terelius, 2017/06/02 10:55:03]

Is this the format that perkj suggested?

[ilnik, 2017/06/02 11:25:59]

No. It was something quick I used for myself. Do we even need special format for
this text tool?
+perkj, WDYT about this reporting format?

[ilnik, 2017/06/02 11:43:35]

We had an offline discussion with perkj@. He doesn't have any opinion on report
format, but he asked me to refactor things a little, so extension map is
extracted and returned by ParsedRtcEventLog. That way there's no need to copy
here the code from tools/event_log_visualizer/analyzer.cc. I am working on it.

+                  << parsed_header.extension.absoluteSendTime;
+      }
+      if (parsed_header.extension.hasVideoContentType) {
+        std::cout << "\tContentType="
+                  << static_cast<int>(parsed_header.extension.videoContentType);
+      }
+      if (parsed_header.extension.hasVideoRotation) {
+        std::cout << "\tRotation="
+                  << static_cast<int>(parsed_header.extension.videoRotation);
+      }
+      if (parsed_header.extension.hasTransportSequenceNumber) {
+        std::cout << "\tTransportSeq="
+                  << parsed_header.extension.transportSequenceNumber;
+      }
+      if (parsed_header.extension.hasTransmissionTimeOffset) {
+        std::cout << "\tTransmTimeOffset="
+                  << parsed_header.extension.transmissionTimeOffset;
+      }
+      if (parsed_header.extension.hasAudioLevel) {
+        std::cout << "\tAudioLevel=" << parsed_header.extension.audioLevel;
+      }
+      std::cout << 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;
       parsed_stream.GetRtcpPacket(i, &direction, packet, &length);
 
       webrtc::rtcp::CommonHeader rtcp_block;
@@ skipping 34 matching lines @@
                             direction);
             break;
           default:
             break;
         }
       }
     }
   }
   return 0;
 }