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Remove no- prefix from command line flags in rtc_event_log2text and rtc_event_log2rtp_dump and negate their meaning.

Browse Source 
BUG=webrtc:8202

Review-Url: https://codereview.webrtc.org/3008113002
Cr-Commit-Position: refs/heads/master@{#19798}
This commit is contained in:
terelius
2017-09-12 05:57:36 -07:00
committed by Commit Bot
parent 661d94996b
commit c3d2bfd244
2 changed files with 49 additions and 46 deletions
Download Patch File Download Diff File Expand all files Collapse all files

54
webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
View File

@ -27,21 +27,26 @@ namespace {
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 @@ 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 +148,11 @@ 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 +165,8 @@ 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 +185,11 @@ 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)

41
webrtc/logging/rtc_event_log/rtc_event_log2text.cc
View File

@ -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 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;