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 292 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

[danilchap, 2017/02/08 14:30:50]

may be label them remote/local rather than ssrc/rtcp ssrc

[terelius, 2017/02/08 14:41:05]

Do you find local/remote to be more clear? The local ssrc is only used for RTCP,
right?

[danilchap, 2017/02/08 14:44:44]

Is it? as I understand
rtp sender care about local ssrc only
rtp receiver care about remote ssrc only
rtcp (both sender and receiver) use both.

[terelius, 2017/02/08 14:49:11]

I agree, but since this is a receive stream there should not be any RTP sender,
right? So that only leaves RTCP using local ssrc.

[danilchap, 2017/02/08 15:41:45]

Right,
I'm still not sure about 'rtcp' label - it is id of the receiver endpoint, but
just happen to appear in rtcp messages only.
How about label 'Feedback ssrc'?

[terelius, 2017/02/08 15:55:42]

I'm fine with any label. Do you prefer "ssrc"/"feedback_ssrc" over
"local_ssrc"/"remote_ssrc"?

[danilchap, 2017/02/08 16:00:41]

Yes, I think ssrc/feedback_ssrc is better.

+                << "\tRTCP_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

[danilchap, 2017/02/08 14:30:50]

ditto

+                << "\tRTCP_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);
 
@@ skipping 54 matching lines @@
             PrintPsFeedback(rtcp_block, log_timestamp, direction, media_type);
             break;
           default:
             break;
         }
       }
     }
   }
   return 0;
 }