| 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 32d5ae5f7a4dfa1827ae70f96b6d9dbbf9943710..cddd91c53c79f8856f6bbbb985efd41f21b03f44 100644
|
| --- a/webrtc/logging/rtc_event_log/rtc_event_log2text.cc
|
| +++ b/webrtc/logging/rtc_event_log/rtc_event_log2text.cc
|
| @@ -37,6 +37,7 @@
|
|
|
| namespace {
|
|
|
| +DEFINE_bool(noconfig, true, "Excludes stream configurations.");
|
| DEFINE_bool(noincoming, false, "Excludes incoming packets.");
|
| DEFINE_bool(nooutgoing, false, "Excludes outgoing packets.");
|
| // TODO(terelius): Note that the media type doesn't work with outgoing packets.
|
| @@ -128,7 +129,7 @@ void PrintSenderReport(const webrtc::rtcp::CommonHeader& rtcp_block,
|
| return;
|
| std::cout << log_timestamp << "\t"
|
| << "RTCP_SR" << StreamInfo(direction, media_type)
|
| - << "\tSSRC=" << sr.sender_ssrc()
|
| + << "\tssrc=" << sr.sender_ssrc()
|
| << "\ttimestamp=" << sr.rtp_timestamp() << std::endl;
|
| }
|
|
|
| @@ -143,7 +144,7 @@ void PrintReceiverReport(const webrtc::rtcp::CommonHeader& rtcp_block,
|
| return;
|
| std::cout << log_timestamp << "\t"
|
| << "RTCP_RR" << StreamInfo(direction, media_type)
|
| - << "\tSSRC=" << rr.sender_ssrc() << std::endl;
|
| + << "\tssrc=" << rr.sender_ssrc() << std::endl;
|
| }
|
|
|
| void PrintXr(const webrtc::rtcp::CommonHeader& rtcp_block,
|
| @@ -157,7 +158,7 @@ void PrintXr(const webrtc::rtcp::CommonHeader& rtcp_block,
|
| return;
|
| std::cout << log_timestamp << "\t"
|
| << "RTCP_XR" << StreamInfo(direction, media_type)
|
| - << "\tSSRC=" << xr.sender_ssrc() << std::endl;
|
| + << "\tssrc=" << xr.sender_ssrc() << std::endl;
|
| }
|
|
|
| void PrintSdes(const webrtc::rtcp::CommonHeader& rtcp_block,
|
| @@ -180,14 +181,13 @@ void PrintBye(const webrtc::rtcp::CommonHeader& rtcp_block,
|
| return;
|
| std::cout << log_timestamp << "\t"
|
| << "RTCP_BYE" << StreamInfo(direction, media_type)
|
| - << "\tSSRC=" << bye.sender_ssrc() << std::endl;
|
| + << "\tssrc=" << bye.sender_ssrc() << std::endl;
|
| }
|
|
|
| void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
|
| uint64_t log_timestamp,
|
| webrtc::PacketDirection direction,
|
| webrtc::MediaType media_type) {
|
| - std::cout << "Rtp feedback found";
|
| switch (rtcp_block.fmt()) {
|
| case webrtc::rtcp::Nack::kFeedbackMessageType: {
|
| webrtc::rtcp::Nack nack;
|
| @@ -197,7 +197,7 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
|
| return;
|
| std::cout << log_timestamp << "\t"
|
| << "RTCP_NACK" << StreamInfo(direction, media_type)
|
| - << "\tSSRC=" << nack.sender_ssrc() << std::endl;
|
| + << "\tssrc=" << nack.sender_ssrc() << std::endl;
|
| break;
|
| }
|
| case webrtc::rtcp::Tmmbr::kFeedbackMessageType: {
|
| @@ -208,7 +208,7 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
|
| return;
|
| std::cout << log_timestamp << "\t"
|
| << "RTCP_TMMBR" << StreamInfo(direction, media_type)
|
| - << "\tSSRC=" << tmmbr.sender_ssrc() << std::endl;
|
| + << "\tssrc=" << tmmbr.sender_ssrc() << std::endl;
|
| break;
|
| }
|
| case webrtc::rtcp::Tmmbn::kFeedbackMessageType: {
|
| @@ -219,7 +219,7 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
|
| return;
|
| std::cout << log_timestamp << "\t"
|
| << "RTCP_TMMBN" << StreamInfo(direction, media_type)
|
| - << "\tSSRC=" << tmmbn.sender_ssrc() << std::endl;
|
| + << "\tssrc=" << tmmbn.sender_ssrc() << std::endl;
|
| break;
|
| }
|
| case webrtc::rtcp::RapidResyncRequest::kFeedbackMessageType: {
|
| @@ -230,7 +230,7 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
|
| return;
|
| std::cout << log_timestamp << "\t"
|
| << "RTCP_SRREQ" << StreamInfo(direction, media_type)
|
| - << "\tSSRC=" << sr_req.sender_ssrc() << std::endl;
|
| + << "\tssrc=" << sr_req.sender_ssrc() << std::endl;
|
| break;
|
| }
|
| case webrtc::rtcp::TransportFeedback::kFeedbackMessageType: {
|
| @@ -242,11 +242,10 @@ void PrintRtpFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
|
| return;
|
| std::cout << log_timestamp << "\t"
|
| << "RTCP_NEWFB" << StreamInfo(direction, media_type)
|
| - << "\tSSRC=" << transport_feedback.sender_ssrc() << std::endl;
|
| + << "\tssrc=" << transport_feedback.sender_ssrc() << std::endl;
|
| break;
|
| }
|
| default:
|
| - RTC_DCHECK(false);
|
| break;
|
| }
|
| }
|
| @@ -264,7 +263,7 @@ void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
|
| return;
|
| std::cout << log_timestamp << "\t"
|
| << "RTCP_PLI" << StreamInfo(direction, media_type)
|
| - << "\tSSRC=" << pli.sender_ssrc() << std::endl;
|
| + << "\tssrc=" << pli.sender_ssrc() << std::endl;
|
| break;
|
| }
|
| case webrtc::rtcp::Sli::kFeedbackMessageType: {
|
| @@ -275,7 +274,7 @@ void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
|
| return;
|
| std::cout << log_timestamp << "\t"
|
| << "RTCP_SLI" << StreamInfo(direction, media_type)
|
| - << "\tSSRC=" << sli.sender_ssrc() << std::endl;
|
| + << "\tssrc=" << sli.sender_ssrc() << std::endl;
|
| break;
|
| }
|
| case webrtc::rtcp::Rpsi::kFeedbackMessageType: {
|
| @@ -286,7 +285,7 @@ void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
|
| return;
|
| std::cout << log_timestamp << "\t"
|
| << "RTCP_RPSI" << StreamInfo(direction, media_type)
|
| - << "\tSSRC=" << rpsi.sender_ssrc() << std::endl;
|
| + << "\tssrc=" << rpsi.sender_ssrc() << std::endl;
|
| break;
|
| }
|
| case webrtc::rtcp::Fir::kFeedbackMessageType: {
|
| @@ -297,7 +296,7 @@ void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
|
| return;
|
| std::cout << log_timestamp << "\t"
|
| << "RTCP_FIR" << StreamInfo(direction, media_type)
|
| - << "\tSSRC=" << fir.sender_ssrc() << std::endl;
|
| + << "\tssrc=" << fir.sender_ssrc() << std::endl;
|
| break;
|
| }
|
| case webrtc::rtcp::Remb::kFeedbackMessageType: {
|
| @@ -308,7 +307,7 @@ void PrintPsFeedback(const webrtc::rtcp::CommonHeader& rtcp_block,
|
| return;
|
| std::cout << log_timestamp << "\t"
|
| << "RTCP_REMB" << StreamInfo(direction, media_type)
|
| - << "\tSSRC=" << remb.sender_ssrc() << std::endl;
|
| + << "\tssrc=" << remb.sender_ssrc() << std::endl;
|
| break;
|
| }
|
| default:
|
| @@ -349,6 +348,46 @@ int main(int argc, char* argv[]) {
|
| }
|
|
|
| for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
|
| + if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_noincoming &&
|
| + parsed_stream.GetEventType(i) ==
|
| + webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT) {
|
| + webrtc::VideoReceiveStream::Config config(nullptr);
|
| + parsed_stream.GetVideoReceiveConfig(i, &config);
|
| + std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_RECV_CONFIG"
|
| + << "\tssrc=" << config.rtp.remote_ssrc
|
| + << "\tfeedback_ssrc=" << config.rtp.local_ssrc << std::endl;
|
| + }
|
| + if (!FLAGS_noconfig && !FLAGS_novideo && !FLAGS_nooutgoing &&
|
| + parsed_stream.GetEventType(i) ==
|
| + webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT) {
|
| + webrtc::VideoSendStream::Config config(nullptr);
|
| + parsed_stream.GetVideoSendConfig(i, &config);
|
| + std::cout << parsed_stream.GetTimestamp(i) << "\tVIDEO_SEND_CONFIG";
|
| + std::cout << "\tssrcs=";
|
| + for (const auto& ssrc : config.rtp.ssrcs)
|
| + std::cout << ssrc << ',';
|
| + std::cout << "\trtx_ssrcs=";
|
| + for (const auto& ssrc : config.rtp.rtx.ssrcs)
|
| + std::cout << ssrc << ',';
|
| + std::cout << std::endl;
|
| + }
|
| + if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_noincoming &&
|
| + parsed_stream.GetEventType(i) ==
|
| + webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT) {
|
| + webrtc::AudioReceiveStream::Config config;
|
| + parsed_stream.GetAudioReceiveConfig(i, &config);
|
| + std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_RECV_CONFIG"
|
| + << "\tssrc=" << config.rtp.remote_ssrc
|
| + << "\tfeedback_ssrc=" << config.rtp.local_ssrc << std::endl;
|
| + }
|
| + if (!FLAGS_noconfig && !FLAGS_noaudio && !FLAGS_nooutgoing &&
|
| + parsed_stream.GetEventType(i) ==
|
| + webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT) {
|
| + webrtc::AudioSendStream::Config config(nullptr);
|
| + parsed_stream.GetAudioSendConfig(i, &config);
|
| + std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_SEND_CONFIG"
|
| + << "\tssrc=" << config.rtp.ssrc << std::endl;
|
| + }
|
| if (!FLAGS_nortp &&
|
| parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
|
| size_t header_length;
|
| @@ -369,7 +408,7 @@ int main(int argc, char* argv[]) {
|
|
|
| std::cout << parsed_stream.GetTimestamp(i) << "\tRTP"
|
| << StreamInfo(direction, media_type)
|
| - << "\tSSRC=" << parsed_header.ssrc
|
| + << "\tssrc=" << parsed_header.ssrc
|
| << "\ttimestamp=" << parsed_header.timestamp << std::endl;
|
| }
|
| if (!FLAGS_nortcp &&
|
|
|