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

Index: webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc b/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
index 4275e5933ffe5e17af6b7705e0e84da6c13791d3..08a68b76114183ff98289a4d4ce9aa87213bb5f8 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
@@ -27,20 +27,18 @@ namespace {
 
 using MediaType = webrtc::ParsedRtcEventLog::MediaType;
 
-DEFINE_bool(noaudio,
-            false,
+DEFINE_bool(audio,
+            true,
             "Excludes audio packets from the converted RTPdump file.");

[ivoc, 2017/09/05 10:59:20]

Please update the descriptions.

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

Done.

-DEFINE_bool(novideo,
-            false,
+DEFINE_bool(video,
+            true,
             "Excludes video packets from the converted RTPdump file.");
-DEFINE_bool(nodata,
-            false,
+DEFINE_bool(data,
+            true,
             "Excludes data packets from the converted RTPdump file.");
-DEFINE_bool(nortp,
-            false,
-            "Excludes RTP packets from the converted RTPdump file.");
-DEFINE_bool(nortcp,
-            false,
+DEFINE_bool(rtp, true, "Excludes RTP packets from the converted RTPdump file.");
+DEFINE_bool(rtcp,
+            true,
             "Excludes RTCP packets from the converted RTPdump file.");
 DEFINE_string(ssrc,
               "",
@@ -122,7 +120,7 @@ int main(int argc, char* argv[]) {
     // 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;
@@ -143,11 +141,11 @@ int main(int argc, char* argv[]) {
       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 =
@@ -160,9 +158,8 @@ int main(int argc, char* argv[]) {
       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);
@@ -181,11 +178,11 @@ int main(int argc, char* argv[]) {
       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)