| 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..06e51478345c677abb2eb49d6b42460e5eeb5c32 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.");
|
| +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;
|
| @@ -521,9 +521,8 @@ int main(int argc, char* argv[]) {
|
| }
|
| std::cout << std::endl;
|
| }
|
| - if (!FLAG_nortcp &&
|
| - parsed_stream.GetEventType(i) ==
|
| - webrtc::ParsedRtcEventLog::RTCP_EVENT) {
|
| + if (FLAG_rtcp && parsed_stream.GetEventType(i) ==
|
| + webrtc::ParsedRtcEventLog::RTCP_EVENT) {
|
| size_t length;
|
| uint8_t packet[IP_PACKET_SIZE];
|
| webrtc::PacketDirection direction;
|
|
|