New parser for event log. Manually parse the outermost EventStream

webrtc/call/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.h"
 #include "webrtc/call/rtc_event_log.h"
+#include "webrtc/call/rtc_event_log_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/test/rtp_file_writer.h"
 
-// Files generated at build-time by the protobuf compiler.
-#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
-#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
-#else
-#include "webrtc/call/rtc_event_log.pb.h"
-#endif
-
 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,
@@ skipping 48 matching lines @@
     return 0;
   }
   std::string input_file = argv[1];
   std::string output_file = argv[2];
 
   uint32_t ssrc_filter = 0;
   if (!FLAGS_ssrc.empty())
     RTC_CHECK(ParseSsrc(FLAGS_ssrc, &ssrc_filter))
         << "Flag verification has failed.";
 
-  webrtc::rtclog::EventStream event_stream;
-  if (!webrtc::RtcEventLog::ParseRtcEventLog(input_file, &event_stream)) {
+  webrtc::ParsedRtcEventLog parsed_stream;
+  if (!parsed_stream.ParseFile(input_file)) {
     std::cerr << "Error while parsing input file: " << input_file << std::endl;
     return -1;
   }
 
   std::unique_ptr<webrtc::test::RtpFileWriter> rtp_writer(
       webrtc::test::RtpFileWriter::Create(
           webrtc::test::RtpFileWriter::FileFormat::kRtpDump, output_file));
 
   if (!rtp_writer.get()) {
     std::cerr << "Error while opening output file: " << output_file
               << std::endl;
     return -1;
   }
 
-  std::cout << "Found " << event_stream.stream_size()
+  std::cout << "Found " << parsed_stream.GetNumberOfEvents()
             << " events in the input file." << std::endl;
   int rtp_counter = 0, rtcp_counter = 0;
   bool header_only = false;
-  // TODO(ivoc): This can be refactored once the packet interpretation
-  //             functions are finished.
-  for (int i = 0; i < event_stream.stream_size(); i++) {
-    const webrtc::rtclog::Event& event = event_stream.stream(i);
-    if (!FLAGS_nortp && event.has_type() && event.type() == event.RTP_EVENT) {
-      if (event.has_timestamp_us() && event.has_rtp_packet() &&
-          event.rtp_packet().has_header() &&
-          event.rtp_packet().header().size() >= 12 &&
-          event.rtp_packet().has_packet_length() &&
-          event.rtp_packet().has_type()) {
-        const webrtc::rtclog::RtpPacket& rtp_packet = event.rtp_packet();
-        if (FLAGS_noaudio && rtp_packet.type() == webrtc::rtclog::AUDIO)
+  for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
+    // 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);
+      if (packet.original_length > packet.length)
+        header_only = true;
+      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)
+        continue;
+      if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO)
+        continue;
+      if (FLAGS_nodata && media_type == webrtc::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;
-        if (FLAGS_novideo && rtp_packet.type() == webrtc::rtclog::VIDEO)
+      }
+
+      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);
+      // 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)
+        continue;
+      if (FLAGS_novideo && media_type == webrtc::MediaType::VIDEO)
+        continue;
+      if (FLAGS_nodata && media_type == webrtc::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;
-        if (FLAGS_nodata && rtp_packet.type() == webrtc::rtclog::DATA)
-          continue;
-        if (!FLAGS_ssrc.empty()) {
-          const uint32_t packet_ssrc =
-              webrtc::ByteReader<uint32_t>::ReadBigEndian(
-                  reinterpret_cast<const uint8_t*>(rtp_packet.header().data() +
-                                                   8));
-          if (packet_ssrc != ssrc_filter)
-            continue;
-        }
+      }
 
-        webrtc::test::RtpPacket packet;
-        packet.length = rtp_packet.header().size();
-        if (packet.length > packet.kMaxPacketBufferSize) {
-          std::cout << "Skipping packet with size " << packet.length
-                    << ", the maximum supported size is "
-                    << packet.kMaxPacketBufferSize << std::endl;
-          continue;
-        }
-        packet.original_length = rtp_packet.packet_length();
-        if (packet.original_length > packet.length)
-          header_only = true;
-        packet.time_ms = event.timestamp_us() / 1000;
-        memcpy(packet.data, rtp_packet.header().data(), packet.length);
-        rtp_writer->WritePacket(&packet);
-        rtp_counter++;
-      } else {
-        std::cout << "Skipping malformed event." << std::endl;
-      }
-    }
-    if (!FLAGS_nortcp && event.has_type() && event.type() == event.RTCP_EVENT) {
-      if (event.has_timestamp_us() && event.has_rtcp_packet() &&
-          event.rtcp_packet().has_type() &&
-          event.rtcp_packet().has_packet_data() &&
-          event.rtcp_packet().packet_data().size() > 0) {
-        const webrtc::rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
-        if (FLAGS_noaudio && rtcp_packet.type() == webrtc::rtclog::AUDIO)
-          continue;
-        if (FLAGS_novideo && rtcp_packet.type() == webrtc::rtclog::VIDEO)
-          continue;
-        if (FLAGS_nodata && rtcp_packet.type() == webrtc::rtclog::DATA)
-          continue;
-        if (!FLAGS_ssrc.empty()) {
-          const uint32_t packet_ssrc =
-              webrtc::ByteReader<uint32_t>::ReadBigEndian(
-                  reinterpret_cast<const uint8_t*>(
-                      rtcp_packet.packet_data().data() + 4));
-          if (packet_ssrc != ssrc_filter)
-            continue;
-        }
-
-        webrtc::test::RtpPacket packet;
-        packet.length = rtcp_packet.packet_data().size();
-        if (packet.length > packet.kMaxPacketBufferSize) {
-          std::cout << "Skipping packet with size " << packet.length
-                    << ", the maximum supported size is "
-                    << packet.kMaxPacketBufferSize << std::endl;
-          continue;
-        }
-        // For RTCP packets the original_length should be set to 0 in the
-        // RTPdump format.
-        packet.original_length = 0;
-        packet.time_ms = event.timestamp_us() / 1000;
-        memcpy(packet.data, rtcp_packet.packet_data().data(), packet.length);
-        rtp_writer->WritePacket(&packet);
-        rtcp_counter++;
-      } else {
-        std::cout << "Skipping malformed event." << std::endl;
-      }
+      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;
 }