Tool to convert RtcEventLog files to RtpDump format.

webrtc/video/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 <string>
+
+#include "gflags/gflags.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/test/rtp_file_writer.h"
+#include "webrtc/video/rtc_event_log.h"
+
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/video/rtc_event_log.pb.h"
+#else
+#include "webrtc/video/rtc_event_log.pb.h"
+#endif
+
+DEFINE_bool(audio_only,

[hlundin-webrtc, 2015/08/17 11:14:34]

Feature request: Can you add a flag to store only packets with a specific SSRC?
There is a similar feature in NetEqRtpPlay.

[ivoc, 2015/08/21 14:30:57]

Done.

+            false,
+            "Store only audio packets in the converted "
+            "RTPdump file.");
+DEFINE_bool(video_only,
+            false,
+            "Store only video packets in the converted "
+            "RTPdump file.");
+DEFINE_bool(data_only,
+            false,
+            "Store only data packets in the converted "
+            "RTPdump file.");
+
+// This utility will convert a stored event log to the rtpdump format.
+int main(int argc, char* argv[]) {
+  std::string program_name = argv[0];
+  std::string usage =
+      "Tool for converting an RtcEventLog file to an RTP dump file.\n"
+      "Run " +
+      program_name +
+      " --helpshort for usage.\n"
+      "Example usage:\n" +
+      program_name + " input.rel output.rtp\n";
+  google::SetUsageMessage(usage);
+  google::ParseCommandLineFlags(&argc, &argv, true);
+
+  if (argc != 3) {
+    std::cout << google::ProgramUsage();
+    return 0;
+  }
+  std::string input_file = argv[1];
+  std::string output_file = argv[2];
+
+  webrtc::rtclog::EventStream event_stream;
+  if (!webrtc::RtcEventLog::ParseRtcEventLog(input_file, &event_stream)) {
+    std::cerr << "Error while parsing input file: " << input_file << std::endl;
+    return -1;
+  }
+
+  rtc::scoped_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()
+            << " events in the input file." << std::endl;
+  int rtp_counter = 0;
+  // 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);

[hlundin-webrtc, 2015/08/17 11:14:34]

const auto& ? Or is the compiler unable to deduce the type in this case?

[terelius, 2015/08/17 11:39:06]

Personally, I like being explicit with the types. I guess it is a matter of
project conventions though...

[ivoc, 2015/08/21 14:30:57]

I tend to agree with terelius@ on this one, let me know if you think it should
change to auto.

[hlundin-webrtc, 2015/08/26 09:24:37]

Acknowledged.

+    if (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().has_packet_length() &&
+          event.rtp_packet().has_type()) {
+        if (FLAGS_audio_only &&
+            event.rtp_packet().type() != webrtc::rtclog::AUDIO)
+          continue;
+        if (FLAGS_video_only &&
+            event.rtp_packet().type() != webrtc::rtclog::VIDEO)
+          continue;
+        if (FLAGS_data_only &&
+            event.rtp_packet().type() != webrtc::rtclog::DATA)
+          continue;
+        webrtc::test::RtpPacket packet;
+        packet.length = event.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 = event.rtp_packet().packet_length();

[hlundin-webrtc, 2015/08/17 11:14:33]

I think it would be handy to know if the output is a header-only RTP dump or if
the packets are complete with payloads. Suggestion: declare a bool header_only
before the for loop and initialize to false. If any packet has an
original_length > length, then set it to true. Print out to screen after loop.

[ivoc, 2015/08/21 14:30:57]

Done.

+        packet.time_ms = event.timestamp_us() / 1000;
+        memcpy(packet.data, event.rtp_packet().header().data(), packet.length);
+        rtp_writer->WritePacket(&packet);
+        rtp_counter++;
+      } else {
+        std::cout << "Skipping malformed event." << std::endl;
+      }
+    }
+  }
+  std::cout << "Wrote " << rtp_counter << " RTP packets to the output file."
+            << std::endl;
+  return 0;
+}