| 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..06b250dbbdf172b3949c1c85cbf8244625ba5806 100644
|
| --- a/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
|
| +++ b/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
|
| @@ -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)
|
|
|