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 && |