Removed RtcEventLog deps to call:call_interfaces.

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 <sstream>
 #include <string>
 
 #include "gflags/gflags.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/call/call.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"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
@@ skipping 17 matching lines @@
 // 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.");
 // TODO(terelius): Allow a list of SSRCs.
 DEFINE_string(ssrc,
               "",
               "Print only packets with this SSRC (decimal or hex, the latter "
               "starting with 0x).");
 
+enum class MediaType { ANY, AUDIO, VIDEO, DATA };
+
 static uint32_t filtered_ssrc = 0;
 
 // Parses the input string for a valid SSRC. If a valid SSRC is found, it is
 // written to the static global variable |filtered_ssrc|, and true is returned.
 // Otherwise, false is returned.
 // The empty string must be validated as true, because it is the default value
 // of the command-line flag. In this case, no value is written to the output
 // variable.
 bool ParseSsrc(std::string str) {
   // If the input string starts with 0x or 0X it indicates a hexadecimal number.
   auto read_mode = std::dec;
   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 SSRCs used in a Stream.
 struct Stream {
-  Stream(uint32_t ssrc,
-         webrtc::MediaType media_type,
-         webrtc::PacketDirection direction)
+  Stream(uint32_t ssrc, MediaType media_type, webrtc::PacketDirection direction)
       : ssrc(ssrc), media_type(media_type), direction(direction) {}
   uint32_t ssrc;
-  webrtc::MediaType media_type;
+  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) {
+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;
+  return MediaType::ANY;
 }
 
 bool ExcludePacket(webrtc::PacketDirection direction,
-                   webrtc::MediaType media_type,
+                   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)
+  if (FLAGS_noaudio && media_type == MediaType::AUDIO)
     return true;
-  if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO)
+  if (FLAGS_novideo && media_type == MediaType::VIDEO)
     return true;
-  if (FLAGS_nodata && media_type == webrtc::MediaType::DATA)
+  if (FLAGS_nodata && media_type == MediaType::DATA)
     return true;
   if (!FLAGS_ssrc.empty() && packet_ssrc != filtered_ssrc)
     return true;
   return false;
 }
 
 const char* StreamInfo(webrtc::PacketDirection direction,

[terelius, 2017/05/23 11:57:58]

This could be a toString() on the Stream object.

[perkj_webrtc, 2017/05/24 12:32:21]

Prefer not to. I moved the Stream into the parser.

-                       webrtc::MediaType media_type) {
+                       MediaType media_type) {
   if (direction == webrtc::kOutgoingPacket) {
-    if (media_type == webrtc::MediaType::AUDIO)
+    if (media_type == MediaType::AUDIO)
       return "(out,audio)";
-    else if (media_type == webrtc::MediaType::VIDEO)
+    else if (media_type == MediaType::VIDEO)
       return "(out,video)";
-    else if (media_type == webrtc::MediaType::DATA)
+    else if (media_type == MediaType::DATA)
       return "(out,data)";
     else
       return "(out)";
   }
   if (direction == webrtc::kIncomingPacket) {
-    if (media_type == webrtc::MediaType::AUDIO)
+    if (media_type == MediaType::AUDIO)
       return "(in,audio)";
-    else if (media_type == webrtc::MediaType::VIDEO)
+    else if (media_type == MediaType::VIDEO)
       return "(in,video)";
-    else if (media_type == webrtc::MediaType::DATA)
+    else if (media_type == 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::rtcp::SenderReport sr;
   if (!sr.Parse(rtcp_block))
     return;
-  webrtc::MediaType media_type = GetMediaType(sr.sender_ssrc(), direction);
+  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::rtcp::ReceiverReport rr;
   if (!rr.Parse(rtcp_block))
     return;
-  webrtc::MediaType media_type = GetMediaType(rr.sender_ssrc(), direction);
+  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::rtcp::ExtendedReports xr;
   if (!xr.Parse(rtcp_block))
     return;
-  webrtc::MediaType media_type = GetMediaType(xr.sender_ssrc(), direction);
+  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) {
   std::cout << log_timestamp << "\t"
-            << "RTCP_SDES" << StreamInfo(direction, webrtc::MediaType::ANY)
+            << "RTCP_SDES" << StreamInfo(direction, 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::rtcp::Bye bye;
   if (!bye.Parse(rtcp_block))
     return;
-  webrtc::MediaType media_type = GetMediaType(bye.sender_ssrc(), direction);
+  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) {
   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);
+      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);
+      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);
+      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);
+      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 =
+      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) {
   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);
+      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);
+      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);
+      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 30 matching lines @@
     std::cerr << "Error while parsing input file: " << input_file << std::endl;
     return -1;
   }
 
   for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
     if (parsed_stream.GetEventType(i) ==
         webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
       webrtc::rtclog::StreamConfig config;
       parsed_stream.GetVideoReceiveConfig(i, &config);
 
-      global_streams.emplace_back(config.remote_ssrc,
-                                  webrtc::MediaType::VIDEO,
+      global_streams.emplace_back(config.remote_ssrc, MediaType::VIDEO,
                                   webrtc::kIncomingPacket);
-      global_streams.emplace_back(config.local_ssrc,
-                                  webrtc::MediaType::VIDEO,
+      global_streams.emplace_back(config.local_ssrc, MediaType::VIDEO,
                                   webrtc::kOutgoingPacket);
 
       if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_noincoming) {
         std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
                 << "\tssrc=" << config.remote_ssrc
                 << "\tfeedback_ssrc=" << config.local_ssrc << std::endl;
       }
     }
     if (parsed_stream.GetEventType(i) ==
         webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
       webrtc::rtclog::StreamConfig config;
       parsed_stream.GetVideoSendConfig(i, &config);
-      global_streams.emplace_back(config.local_ssrc, webrtc::MediaType::VIDEO,
+      global_streams.emplace_back(config.local_ssrc, MediaType::VIDEO,
                                   webrtc::kOutgoingPacket);
 
-      global_streams.emplace_back(config.rtx_ssrc, webrtc::MediaType::VIDEO,
+      global_streams.emplace_back(config.rtx_ssrc, MediaType::VIDEO,
                                   webrtc::kOutgoingPacket);
 
       if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_nooutgoing) {
         std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
         std::cout << "\tssrcs=" << config.local_ssrc;
         std::cout << "\trtx_ssrcs=" << config.rtx_ssrc;
         std::cout << std::endl;
       }
     }
     if (parsed_stream.GetEventType(i) ==
         webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
       webrtc::rtclog::StreamConfig config;
       parsed_stream.GetAudioReceiveConfig(i, &config);
-      global_streams.emplace_back(config.remote_ssrc,
-                                  webrtc::MediaType::AUDIO,
+      global_streams.emplace_back(config.remote_ssrc, MediaType::AUDIO,
                                   webrtc::kIncomingPacket);
-      global_streams.emplace_back(config.local_ssrc,
-                                  webrtc::MediaType::AUDIO,
+      global_streams.emplace_back(config.local_ssrc, MediaType::AUDIO,
                                   webrtc::kOutgoingPacket);
       if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_noincoming) {
         std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
                   << "\tssrc=" << config.remote_ssrc
                   << "\tfeedback_ssrc=" << config.local_ssrc << std::endl;
       }
     }
     if (parsed_stream.GetEventType(i) ==
         webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
       webrtc::rtclog::StreamConfig config;
       parsed_stream.GetAudioSendConfig(i, &config);
-      global_streams.emplace_back(config.local_ssrc, webrtc::MediaType::AUDIO,
+      global_streams.emplace_back(config.local_ssrc, MediaType::AUDIO,
                                   webrtc::kOutgoingPacket);
       if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_nooutgoing) {
         std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
                   << "\tssrc=" << config.local_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);
+
+      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);
-      media_type = GetMediaType(parsed_header.ssrc, direction);
+      MediaType 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 &&
         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);
+      parsed_stream.GetRtcpPacket(i, &direction, packet, &length);
 
       webrtc::rtcp::CommonHeader rtcp_block;
       const uint8_t* packet_end = packet + length;
       for (const uint8_t* next_block = packet; next_block != packet_end;
            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;
         }
@@ skipping 22 matching lines @@
             PrintPsFeedback(rtcp_block, log_timestamp, direction);
             break;
           default:
             break;
         }
       }
     }
   }
   return 0;
 }