Print configured header extensions and codecs in rtc_event_log2text.

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 16 matching lines @@
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h"
 #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/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(noconfig, false, "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 301 matching lines @@
   }
 
   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);
       std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
                 << "\tssrc=" << config.remote_ssrc
-                << "\tfeedback_ssrc=" << config.local_ssrc << std::endl;
+                << "\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 (size_t j = 0; j < configs.size(); j++) {
+      for (const auto& config : configs) {
         std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
-        std::cout << "\tssrcs=" << configs[j].local_ssrc;
-        std::cout << "\trtx_ssrcs=" << configs[j].rtx_ssrc;
-        std::cout << std::endl;
+        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) {
+          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_noaudio && !FLAGS_noincoming &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
       webrtc::rtclog::StreamConfig config =
           parsed_stream.GetAudioReceiveConfig(i);
       std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
                 << "\tssrc=" << config.remote_ssrc
-                << "\tfeedback_ssrc=" << config.local_ssrc << std::endl;
+                << "\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_noaudio && !FLAGS_nooutgoing &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
       webrtc::rtclog::StreamConfig config = parsed_stream.GetAudioSendConfig(i);
       std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
-                << "\tssrc=" << config.local_ssrc << std::endl;
+                << "\tssrc=" << 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_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);
@@ skipping 59 matching lines @@
                             direction);
             break;
           default:
             break;
         }
       }
     }
   }
   return 0;
 }