Remove no- prefix from command line flags in rtc_event_log2text and rtc_event_log2rtp_dump and negate their meaning.
BUG=webrtc:8202
Review-Url: https://codereview.webrtc.org/3008113002
Cr-Original-Commit-Position: refs/heads/master@{#19798}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: c3d2bfd2443a55f22d84f15f5f64e2ea80daba19
diff --git a/logging/rtc_event_log/rtc_event_log2rtp_dump.cc b/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
index 4275e59..06b250d 100644
--- a/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
+++ b/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
@@ -27,21 +27,26 @@
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 "
@@ -122,7 +127,7 @@
// 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 +148,11 @@
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 +165,8 @@
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 +185,11 @@
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)
diff --git a/logging/rtc_event_log/rtc_event_log2text.cc b/logging/rtc_event_log/rtc_event_log2text.cc
index bba6ace..0f9d674 100644
--- a/logging/rtc_event_log/rtc_event_log2text.cc
+++ b/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, "Use --noconfig to exclude stream configurations.");
+DEFINE_bool(incoming, true, "Use --noincoming to exclude incoming packets.");
+DEFINE_bool(outgoing, true, "Use --nooutgoing to exclude 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, "Use --noaudio to exclude 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, "Use --novideo to exclude 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, "Use --nodata to exclude data packets.");
+DEFINE_bool(rtp, true, "Use --nortp to exclude RTP packets.");
+DEFINE_bool(rtcp, true, "Use --nortcp to exclude RTCP packets.");
// TODO(terelius): Allow a list of SSRCs.
DEFINE_string(ssrc,
"",
@@ -84,15 +84,15 @@
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 @@
}
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 @@
}
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 @@
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 @@
}
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 @@
}
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 @@
}
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;
