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..944746a14fd3c035c5a2c17180dc559c9b0173f1 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. |
@@ -349,6 +350,46 @@ 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 |
danilchap
2017/02/08 14:30:50
may be label them remote/local rather than ssrc/rt
terelius
2017/02/08 14:41:05
Do you find local/remote to be more clear? The loc
danilchap
2017/02/08 14:44:44
Is it? as I understand
rtp sender care about local
terelius
2017/02/08 14:49:11
I agree, but since this is a receive stream there
danilchap
2017/02/08 15:41:45
Right,
I'm still not sure about 'rtcp' label - it
terelius
2017/02/08 15:55:42
I'm fine with any label. Do you prefer "ssrc"/"fee
danilchap
2017/02/08 16:00:41
Yes, I think ssrc/feedback_ssrc is better.
|
+ << "\tRTCP_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 |
danilchap
2017/02/08 14:30:50
ditto
|
+ << "\tRTCP_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; |