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_log2text.cc

 /*
  *  Copyright (c) 2017 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.
  */
 
@@ skipping 22 matching lines @@
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/flags.h"
 
 namespace {
 
-DEFINE_bool(noconfig, false, "Excludes stream configurations.");
-DEFINE_bool(noincoming, false, "Excludes incoming packets.");
-DEFINE_bool(nooutgoing, false, "Excludes outgoing packets.");
+DEFINE_bool(config, true, "Excludes stream configurations.");

[oprypin_webrtc, 2017/09/04 14:48:21]

It seems like the descriptions are misleading now.

Perhaps change all descriptions from
    DEFINE_bool\((\w+), true, "Excludes
to
    DEFINE_bool(\1, true, "Use --no\1 to exclude

[terelius, 2017/09/05 11:26:28]

Changed to "Includes/excludes \1" instead. Is this acceptable to you?

[oprypin_webrtc, 2017/09/05 11:31:52]

The problem is that people will not know how to use this. I don't think it's
clear enough.
--help lists the flag as --config but passing --config has no effect, and it's a
really obscure detail that --noconfig is what you actually need to pass.

[terelius, 2017/09/12 11:16:55]

Oh, I see; the (intuitive) --config=false no longer works. Was there a reason
for removing it? Will we add it back?

Updated the help text according to your preference.

+DEFINE_bool(incoming, true, "Excludes incoming packets.");
+DEFINE_bool(outgoing, true, "Excludes outgoing packets.");
 // TODO(terelius): Note that the media type doesn't work with outgoing packets.
-DEFINE_bool(noaudio, false, "Excludes audio packets.");
+DEFINE_bool(audio, true, "Excludes audio packets.");
 // TODO(terelius): Note that the media type doesn't work with outgoing packets.
-DEFINE_bool(novideo, false, "Excludes video packets.");
+DEFINE_bool(video, true, "Excludes video packets.");
 // TODO(terelius): Note that the media type doesn't work with outgoing packets.
-DEFINE_bool(nodata, false, "Excludes data packets.");
-DEFINE_bool(nortp, false, "Excludes RTP packets.");
-DEFINE_bool(nortcp, false, "Excludes RTCP packets.");
+DEFINE_bool(data, true, "Excludes data packets.");
+DEFINE_bool(rtp, true, "Excludes RTP packets.");
+DEFINE_bool(rtcp, true, "Excludes RTCP packets.");
 // TODO(terelius): Allow a list of SSRCs.
 DEFINE_string(ssrc,
               "",
               "Print only packets with this SSRC (decimal or hex, the latter "
               "starting with 0x).");
 DEFINE_bool(help, false, "Prints this message.");
 
 using MediaType = webrtc::ParsedRtcEventLog::MediaType;
 
 static uint32_t filtered_ssrc = 0;
@@ skipping 13 matching lines @@
     str = str.substr(2);
   }
   std::stringstream ss(str);
   ss >> read_mode >> filtered_ssrc;
   return str.empty() || (!ss.fail() && ss.eof());
 }
 
 bool ExcludePacket(webrtc::PacketDirection direction,
                    MediaType media_type,
                    uint32_t packet_ssrc) {
-  if (FLAG_nooutgoing && direction == webrtc::kOutgoingPacket)
+  if (!FLAG_outgoing && direction == webrtc::kOutgoingPacket)
     return true;
-  if (FLAG_noincoming && direction == webrtc::kIncomingPacket)
+  if (!FLAG_incoming && direction == webrtc::kIncomingPacket)
     return true;
-  if (FLAG_noaudio && media_type == MediaType::AUDIO)
+  if (!FLAG_audio && media_type == MediaType::AUDIO)
     return true;
-  if (FLAG_novideo && media_type == MediaType::VIDEO)
+  if (!FLAG_video && media_type == MediaType::VIDEO)
     return true;
-  if (FLAG_nodata && media_type == MediaType::DATA)
+  if (!FLAG_data && media_type == MediaType::DATA)
     return true;
   if (strlen(FLAG_ssrc) > 0 && packet_ssrc != filtered_ssrc)
     return true;
   return false;
 }
 
 const char* StreamInfo(webrtc::PacketDirection direction,
                        MediaType media_type) {
   if (direction == webrtc::kOutgoingPacket) {
     if (media_type == MediaType::AUDIO)
@@ skipping 273 matching lines @@
 
   webrtc::RtpHeaderExtensionMap default_map = GetDefaultHeaderExtensionMap();
 
   webrtc::ParsedRtcEventLog parsed_stream;
   if (!parsed_stream.ParseFile(input_file)) {
     std::cerr << "Error while parsing input file: " << input_file << std::endl;
     return -1;
   }
 
   for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
-    if (!FLAG_noconfig && !FLAG_novideo && !FLAG_noincoming &&
+    if (FLAG_config && !!FLAG_video && !!FLAG_incoming &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
       webrtc::rtclog::StreamConfig config =
           parsed_stream.GetVideoReceiveConfig(i);
       std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
                 << "\tssrc=" << config.remote_ssrc
                 << "\tfeedback_ssrc=" << config.local_ssrc;
       std::cout << "\textensions={";
       for (const auto& extension : config.rtp_extensions) {
         std::cout << extension.ToString() << ",";
       }
       std::cout << "}";
       std::cout << "\tcodecs={";
       for (const auto& codec : config.codecs) {
         std::cout << "{name: " << codec.payload_name
                   << ", payload_type: " << codec.payload_type
                   << ", rtx_payload_type: " << codec.rtx_payload_type << "}";
       }
       std::cout << "}" << std::endl;
     }
-    if (!FLAG_noconfig && !FLAG_novideo && !FLAG_nooutgoing &&
+    if (!!FLAG_config && !!FLAG_video && !!FLAG_outgoing &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
       std::vector<webrtc::rtclog::StreamConfig> configs =
           parsed_stream.GetVideoSendConfig(i);
       for (const auto& config : configs) {
         std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
         std::cout << "\tssrcs=" << config.local_ssrc;
         std::cout << "\trtx_ssrcs=" << config.rtx_ssrc;
         std::cout << "\textensions={";
         for (const auto& extension : config.rtp_extensions) {
           std::cout << extension.ToString() << ",";
         }
         std::cout << "}";
         std::cout << "\tcodecs={";
         for (const auto& codec : config.codecs) {
           std::cout << "{name: " << codec.payload_name
                     << ", payload_type: " << codec.payload_type
                     << ", rtx_payload_type: " << codec.rtx_payload_type << "}";
         }
         std::cout << "}" << std::endl;
       }
     }
-    if (!FLAG_noconfig && !FLAG_noaudio && !FLAG_noincoming &&
+    if (!!FLAG_config && !!FLAG_audio && !!FLAG_incoming &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
       webrtc::rtclog::StreamConfig config =
           parsed_stream.GetAudioReceiveConfig(i);
       std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
                 << "\tssrc=" << config.remote_ssrc
                 << "\tfeedback_ssrc=" << config.local_ssrc;
       std::cout << "\textensions={";
       for (const auto& extension : config.rtp_extensions) {
         std::cout << extension.ToString() << ",";
       }
       std::cout << "}";
       std::cout << "\tcodecs={";
       for (const auto& codec : config.codecs) {
         std::cout << "{name: " << codec.payload_name
                   << ", payload_type: " << codec.payload_type
                   << ", rtx_payload_type: " << codec.rtx_payload_type << "}";
       }
       std::cout << "}" << std::endl;
     }
-    if (!FLAG_noconfig && !FLAG_noaudio && !FLAG_nooutgoing &&
+    if (!!FLAG_config && !!FLAG_audio && !!FLAG_outgoing &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
       webrtc::rtclog::StreamConfig config = parsed_stream.GetAudioSendConfig(i);
       std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
                 << "\tssrc=" << config.local_ssrc;
       std::cout << "\textensions={";
       for (const auto& extension : config.rtp_extensions) {
         std::cout << extension.ToString() << ",";
       }
       std::cout << "}";
       std::cout << "\tcodecs={";
       for (const auto& codec : config.codecs) {
         std::cout << "{name: " << codec.payload_name
                   << ", payload_type: " << codec.payload_type
                   << ", rtx_payload_type: " << codec.rtx_payload_type << "}";
       }
       std::cout << "}" << std::endl;
     }
-    if (!FLAG_nortp &&
+    if (!!FLAG_rtp &&
         parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
       size_t header_length;
       size_t total_length;
       uint8_t header[IP_PACKET_SIZE];
       webrtc::PacketDirection direction;
       webrtc::RtpHeaderExtensionMap* extension_map = parsed_stream.GetRtpHeader(
           i, &direction, header, &header_length, &total_length);
 
       if (extension_map == nullptr)
         extension_map = &default_map;
@@ skipping 30 matching lines @@
       }
       if (parsed_header.extension.hasTransmissionTimeOffset) {
         std::cout << "\tTransmTimeOffset="
                   << parsed_header.extension.transmissionTimeOffset;
       }
       if (parsed_header.extension.hasAudioLevel) {
         std::cout << "\tAudioLevel=" << parsed_header.extension.audioLevel;
       }
       std::cout << std::endl;
     }
-    if (!FLAG_nortcp &&
+    if (!!FLAG_rtcp &&
         parsed_stream.GetEventType(i) ==
             webrtc::ParsedRtcEventLog::RTCP_EVENT) {
       size_t length;
       uint8_t packet[IP_PACKET_SIZE];
       webrtc::PacketDirection direction;
       parsed_stream.GetRtcpPacket(i, &direction, packet, &length);
 
       webrtc::rtcp::CommonHeader rtcp_block;
       const uint8_t* packet_end = packet + length;
       for (const uint8_t* next_block = packet; next_block != packet_end;
@@ skipping 32 matching lines @@
                             direction);
             break;
           default:
             break;
         }
       }
     }
   }
   return 0;
 }