Moved RtcEventLog files from call/ to logging/ (new top level dir)

webrtc/call/rtc_event_log2rtp_dump.cc

Index: webrtc/call/rtc_event_log2rtp_dump.cc
diff --git a/webrtc/call/rtc_event_log2rtp_dump.cc b/webrtc/call/rtc_event_log2rtp_dump.cc
deleted file mode 100644
index 5733cfa31d23d3f6a550a2f79f154fb7eeeb1f60..0000000000000000000000000000000000000000
--- a/webrtc/call/rtc_event_log2rtp_dump.cc
+++ /dev/null
@@ -1,186 +0,0 @@
-/*
- *  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"
-
-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).");
-
-// 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 &&
-      (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 >> *ssrc;
-  return str.empty() || (!ss.fail() && ss.eof());
-}
-
-}  // namespace
-
-// 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];
-
-  uint32_t ssrc_filter = 0;
-  if (!FLAGS_ssrc.empty())
-    RTC_CHECK(ParseSsrc(FLAGS_ssrc, &ssrc_filter))
-        << "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;
-  }
-
-  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 " << 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++) {
-    // 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;
-      }
-
-      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;
-      }
-
-      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;
-}