Remove no- prefix from command line flags in rtc_event_log2text and rtc_event_log2rtp_dump and nega…

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 <string.h>
 
 #include <iostream>
 #include <memory>
 #include <sstream>
 #include <string>
 
 #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/rtc_base/checks.h"
 #include "webrtc/rtc_base/flags.h"
 #include "webrtc/test/rtp_file_writer.h"
 
 namespace {
 
 using MediaType = webrtc::ParsedRtcEventLog::MediaType;
 
-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_bool(
+    audio,
+    true,
+    "Use --noaudio to exclude audio packets from the converted RTPdump file.");
+DEFINE_bool(
+    video,
+    true,
+    "Use --novideo to exclude video packets from the converted RTPdump file.");
+DEFINE_bool(
+    data,
+    true,
+    "Use --nodata to exclude data packets from the converted RTPdump file.");
+DEFINE_bool(
+    rtp,
+    true,
+    "Use --nortp to exclude RTP packets from the converted RTPdump file.");
+DEFINE_bool(
+    rtcp,
+    true,
+    "Use --nortcp to exclude RTCP packets from the converted RTPdump file.");
 DEFINE_string(ssrc,
               "",
               "Store only packets with this SSRC (decimal or hex, the latter "
               "starting with 0x).");
 DEFINE_bool(help, false, "Prints this message.");
 
 // 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
@@ skipping 60 matching lines @@
 
   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 (!FLAG_nortp &&
+    if (FLAG_rtp &&
         parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
       webrtc::test::RtpPacket packet;
       webrtc::PacketDirection direction;
       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;
 
       webrtc::RTPHeader parsed_header;
       rtp_parser.Parse(&parsed_header);
       MediaType media_type =
           parsed_stream.GetMediaType(parsed_header.ssrc, direction);
-      if (FLAG_noaudio && media_type == MediaType::AUDIO)
+      if (!FLAG_audio && media_type == MediaType::AUDIO)
         continue;
-      if (FLAG_novideo && media_type == MediaType::VIDEO)
+      if (!FLAG_video && media_type == MediaType::VIDEO)
         continue;
-      if (FLAG_nodata && media_type == MediaType::DATA)
+      if (!FLAG_data && media_type == MediaType::DATA)
         continue;
       if (strlen(FLAG_ssrc) > 0) {
         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 (!FLAG_nortcp &&
-        parsed_stream.GetEventType(i) ==
-            webrtc::ParsedRtcEventLog::RTCP_EVENT) {
+    if (FLAG_rtcp && parsed_stream.GetEventType(i) ==
+                         webrtc::ParsedRtcEventLog::RTCP_EVENT) {
       webrtc::test::RtpPacket packet;
       webrtc::PacketDirection direction;
       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;
 
       // Note that |packet_ssrc| is the sender SSRC. An RTCP message may contain
       // report blocks for many streams, thus several SSRCs and they doen't
       // necessarily have to be of the same media type.
       const uint32_t packet_ssrc = webrtc::ByteReader<uint32_t>::ReadBigEndian(
           reinterpret_cast<const uint8_t*>(packet.data + 4));
       MediaType media_type = parsed_stream.GetMediaType(packet_ssrc, direction);
-      if (FLAG_noaudio && media_type == MediaType::AUDIO)
+      if (!FLAG_audio && media_type == MediaType::AUDIO)
         continue;
-      if (FLAG_novideo && media_type == MediaType::VIDEO)
+      if (!FLAG_video && media_type == MediaType::VIDEO)
         continue;
-      if (FLAG_nodata && media_type == MediaType::DATA)
+      if (!FLAG_data && media_type == MediaType::DATA)
         continue;
       if (strlen(FLAG_ssrc) > 0) {
         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;
 }