Index: webrtc/logging/rtc_event_log/rtc_event_log2text.cc |
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log2text.cc b/webrtc/logging/rtc_event_log/rtc_event_log2text.cc |
index bba6aceee8a6ccb1b7e006405a987ee1b9f8bdd3..e49e0833ddbd92a696e87a0c83f88c538e219db1 100644 |
--- a/webrtc/logging/rtc_event_log/rtc_event_log2text.cc |
+++ b/webrtc/logging/rtc_event_log/rtc_event_log2text.cc |
@@ -40,17 +40,17 @@ |
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.
terelius
2017/09/05 11:26:28
Changed to "Includes/excludes \1" instead. Is this
oprypin_webrtc
2017/09/05 11:31:52
The problem is that people will not know how to us
terelius
2017/09/12 11:16:55
Oh, I see; the (intuitive) --config=false no longe
|
+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, |
"", |
@@ -84,15 +84,15 @@ bool ParseSsrc(std::string str) { |
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; |
@@ -386,7 +386,7 @@ int main(int argc, char* argv[]) { |
} |
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 = |
@@ -407,7 +407,7 @@ int main(int argc, char* argv[]) { |
} |
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 = |
@@ -430,7 +430,7 @@ int main(int argc, char* argv[]) { |
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 = |
@@ -451,7 +451,7 @@ int main(int argc, char* argv[]) { |
} |
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); |
@@ -470,7 +470,7 @@ int main(int argc, char* argv[]) { |
} |
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; |
@@ -521,7 +521,7 @@ int main(int argc, char* argv[]) { |
} |
std::cout << std::endl; |
} |
- if (!FLAG_nortcp && |
+ if (!!FLAG_rtcp && |
parsed_stream.GetEventType(i) == |
webrtc::ParsedRtcEventLog::RTCP_EVENT) { |
size_t length; |