Removed RtcEventLog deps to call:call_interfaces.

webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc

 /*
  *  Copyright (c) 2015 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 <memory>
 #include <sstream>
 #include <string>
 
 #include "gflags/gflags.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/call/call.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/test/rtp_file_writer.h"
 
 namespace {
 
 DEFINE_bool(noaudio,
             false,
             "Excludes audio packets from the converted RTPdump file.");
 DEFINE_bool(novideo,
             false,
             "Excludes video packets from the converted RTPdump file.");
 DEFINE_bool(nodata,
             false,
             "Excludes data packets from the converted RTPdump file.");
 DEFINE_bool(nortp,
             false,
             "Excludes RTP packets from the converted RTPdump file.");
 DEFINE_bool(nortcp,
             false,
             "Excludes RTCP packets from the converted RTPdump file.");
 DEFINE_string(ssrc,
               "",
               "Store only packets with this SSRC (decimal or hex, the latter "
               "starting with 0x).");
 
+enum class MediaType { ANY, AUDIO, VIDEO, DATA };
+
+// Struct used for storing SSRCs used in a Stream.
+struct Stream {

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

If this struct is used by all/most tools, maybe the definition should be moved
to the parser?

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

Done.

+  Stream(uint32_t ssrc, MediaType media_type, webrtc::PacketDirection direction)
+      : ssrc(ssrc), media_type(media_type), direction(direction) {}
+  uint32_t ssrc;
+  MediaType media_type;
+  webrtc::PacketDirection direction;
+};
+
+// All configured streams found in the event log.
+std::vector<Stream> global_streams;

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

Another alternative is to make this a part of the parser, so you can ask the
parser whether a packet is adio or video. I'm fine with either way though.

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

Done.

+
+// Returns the MediaType for registered SSRCs. Search from the end to use last
+// registered types first.
+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 MediaType::ANY;
+}
+
 // Parses the input string for a valid SSRC. If a valid SSRC is found, it is
 // written to the output variable |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, uint32_t* ssrc) {
   // If the input string starts with 0x or 0X it indicates a hexadecimal number.
   auto read_mode = std::dec;
   if (str.size() > 2 &&
@@ skipping 47 matching lines @@
     std::cerr << "Error while opening output file: " << output_file
               << std::endl;
     return -1;
   }
 
   std::cout << "Found " << parsed_stream.GetNumberOfEvents()
             << " events in the input file." << std::endl;
   int rtp_counter = 0, rtcp_counter = 0;
   bool header_only = false;
   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, MediaType::VIDEO,
+                                  webrtc::kIncomingPacket);
+      global_streams.emplace_back(config.local_ssrc, MediaType::VIDEO,
+                                  webrtc::kOutgoingPacket);
+    }
+    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, MediaType::VIDEO,
+                                  webrtc::kOutgoingPacket);
+
+      global_streams.emplace_back(config.rtx_ssrc, MediaType::VIDEO,
+                                  webrtc::kOutgoingPacket);
+    }
+    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, MediaType::AUDIO,
+                                  webrtc::kIncomingPacket);
+      global_streams.emplace_back(config.local_ssrc, MediaType::AUDIO,
+                                  webrtc::kOutgoingPacket);
+    }
+    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, MediaType::AUDIO,
+                                  webrtc::kOutgoingPacket);
+    }
+
     // The parsed_stream will assert if the protobuf event is missing
     // some required fields and we attempt to access them. We could consider
     // a softer failure option, but it does not seem useful to generate
     // RTP dumps based on broken event logs.
     if (!FLAGS_nortp &&
         parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
       webrtc::test::RtpPacket packet;
       webrtc::PacketDirection direction;
-      webrtc::MediaType media_type;
-      parsed_stream.GetRtpHeader(i, &direction, &media_type, packet.data,
-                                 &packet.length, &packet.original_length);
+      parsed_stream.GetRtpHeader(i, &direction, packet.data, &packet.length,
+                                 &packet.original_length);
       if (packet.original_length > packet.length)
         header_only = true;
       packet.time_ms = parsed_stream.GetTimestamp(i) / 1000;
 
+      webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet.data,
+                                                     packet.length);
+
       // TODO(terelius): Maybe add a flag to dump outgoing traffic instead?
       if (direction == webrtc::kOutgoingPacket)
         continue;
-      if (FLAGS_noaudio && media_type == webrtc::MediaType::AUDIO)
+
+      webrtc::RTPHeader parsed_header;
+      rtp_parser.Parse(&parsed_header);
+      MediaType media_type = GetMediaType(parsed_header.ssrc, direction);
+      if (FLAGS_noaudio && media_type == MediaType::AUDIO)
         continue;
-      if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO)
+      if (FLAGS_novideo && media_type == MediaType::VIDEO)
         continue;
-      if (FLAGS_nodata && media_type == webrtc::MediaType::DATA)
+      if (FLAGS_nodata && media_type == MediaType::DATA)
         continue;
       if (!FLAGS_ssrc.empty()) {
         const uint32_t packet_ssrc =
             webrtc::ByteReader<uint32_t>::ReadBigEndian(
                 reinterpret_cast<const uint8_t*>(packet.data + 8));
         if (packet_ssrc != ssrc_filter)
           continue;
       }
 
       rtp_writer->WritePacket(&packet);
       rtp_counter++;
     }
     if (!FLAGS_nortcp &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::RTCP_EVENT) {
       webrtc::test::RtpPacket packet;
       webrtc::PacketDirection direction;
-      webrtc::MediaType media_type;
-      parsed_stream.GetRtcpPacket(i, &direction, &media_type, packet.data,
-                                  &packet.length);
+      parsed_stream.GetRtcpPacket(i, &direction, packet.data, &packet.length);
       // For RTCP packets the original_length should be set to 0 in the
       // RTPdump format.
       packet.original_length = 0;
       packet.time_ms = parsed_stream.GetTimestamp(i) / 1000;
 
       // TODO(terelius): Maybe add a flag to dump outgoing traffic instead?
       if (direction == webrtc::kOutgoingPacket)
         continue;
-      if (FLAGS_noaudio && media_type == webrtc::MediaType::AUDIO)
+
+      const uint32_t packet_ssrc = webrtc::ByteReader<uint32_t>::ReadBigEndian(
+          reinterpret_cast<const uint8_t*>(packet.data + 4));
+      MediaType media_type = GetMediaType(packet_ssrc, direction);

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

Please add a TODO or comment documenting that media types for RTCP doesn't work.
Or maybe we should remove the RTCP filtering completely? 

+      if (FLAGS_noaudio && media_type == MediaType::AUDIO)
         continue;
-      if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO)
+      if (FLAGS_novideo && media_type == MediaType::VIDEO)
         continue;
-      if (FLAGS_nodata && media_type == webrtc::MediaType::DATA)
+      if (FLAGS_nodata && media_type == MediaType::DATA)
         continue;
       if (!FLAGS_ssrc.empty()) {
-        const uint32_t packet_ssrc =
-            webrtc::ByteReader<uint32_t>::ReadBigEndian(
-                reinterpret_cast<const uint8_t*>(packet.data + 4));
         if (packet_ssrc != ssrc_filter)
           continue;
       }
 
       rtp_writer->WritePacket(&packet);
       rtcp_counter++;
     }
   }
   std::cout << "Wrote " << rtp_counter << (header_only ? " header-only" : "")
             << " RTP packets and " << rtcp_counter << " RTCP packets to the "
             << "output file." << std::endl;
   return 0;
 }