<html><head><meta http-equiv="Content-Type" content="text/html; charset=UTF-8"/></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;" class="">Solved this with help from Tim on IRC.<div class=""> </div><div class="">Needed to add `-lgsttag-1.0` to my make command.</div><div class=""> <div class=""> <div style="caret-color: rgb(0, 0, 0); color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; font-style: normal; font-variant-caps: normal; font-weight: normal; letter-spacing: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; text-decoration: none;">David Walter –––––––––––––––––––––––––––––––– </div> </div> <div> <blockquote type="cite" class=""><div class="">On Oct 5, 2021, at 12:43 AM, <a href="mailto:gstreamer-devel-request@lists.freedesktop.org" class="">gstreamer-devel-request@lists.freedesktop.org</a> wrote:</div> <div class=""><div class="">Send gstreamer-devel mailing list submissions to <a href="mailto:gstreamer-devel@lists.freedesktop.org" class="">gstreamer-devel@lists.freedesktop.org</a> To subscribe or unsubscribe via the World Wide Web, visit https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel or, via email, send a message with subject or body 'help' to gstreamer-devel-request@lists.freedesktop.org You can reach the person managing the list at gstreamer-devel-owner@lists.freedesktop.org When replying, please edit your Subject line so it is more specific than "Re: Contents of gstreamer-devel digest..." Today's Topics: 1. webrtcbin do-nack: rtprtxsend says "requested seqnum has not been transmitted yet in the original stream" (Michiel Konstapel) 2. Re: rtsp-server (???) 3. gst_tag_xmp_list_schemas() linker error on Ubuntu (David Walter) 4. Screen capture, select the right screen. (Davide Perini) 5. Changing buffer PTS/DTS in Python (Marianna Smidth Buschle) ---------------------------------------------------------------------- Message: 1 Date: Mon, 4 Oct 2021 14:31:39 +0200 From: Michiel Konstapel <michiel@aanmelder.nl> To: Discussion of the development of and with GStreamer <gstreamer-devel@lists.freedesktop.org> Subject: webrtcbin do-nack: rtprtxsend says "requested seqnum has not been transmitted yet in the original stream" Message-ID: <d841acc5-4d39-aaa1-ffdf-50fdb2a4c7b7@aanmelder.nl> Content-Type: text/plain; charset=utf-8; format=flowed I am trying to improve WebRTC video quality in less than optimal network conditions, sending audio and video from my pipeline to a browser. My first thought was to try and enable do-nack on the transceivers. Or for now, only the video transceiver, because Chrome doesn't support NACK for audio? I am using tc to simulate 1% packet loss on the link, which has a noticeable impact on the video (both visually (dropped frames and smearing) and in terms of the packetsLost statistic). However, when I enable "do-nack", I see no improvement, and in the logs I see: rtprtxsend gstrtprtxsend.c:651:gst_rtp_rtx_send_sink_event:<rtprtxsend0> Payload 96 not in rtx-pt-map rtprtxsend gstrtprtxsend.c:655:gst_rtp_rtx_send_sink_event:<rtprtxsend0> got caps for payload: 96->-1, ssrc: 715396252->3436539345 : application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, packetization-mode=(string)1, profile-level-id=(string)42c028, sprop-parameter-sets=(string)"Z0LAKJWgHgCJ+XAWoCAgKAAAH0AABhqEIA\=\=\,aM48gA\=\=", payload=(int)96, ssrc=(uint)715396252, timestamp-offset=(uint)4223414689, seqnum-offset=(uint)25976, a-framerate=(string)25 on startup, and then rtprtxsend gstrtprtxsend.c:524:gst_rtp_rtx_send_src_event:<rtprtxsend0> requested seqnum 25100 has not been transmitted yet in the original stream; either the remote end is not configured correctly, or the source is too slow This repeats ~10 times for each sequence number. This is the generated offer: v=0 o=- 3673290479767803662 0 IN IP4 0.0.0.0 s=- t=0 0 a=ice-options:trickle m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 c=IN IP4 0.0.0.0 a=setup:actpass a=ice-ufrag:GBQSDSR0t2BjpCKzhBy6D5TWI5PIwHcL a=ice-pwd:gVq+L+I5gtsYST5+0Ko1qiou+cmTMBek a=rtcp-mux a=rtcp-rsize a=sendonly a=rtpmap:96 H264/90000 a=rtcp-fb:96 nack pli a=framerate:25 a=fmtp:96 packetization-mode=1;profile-level-id=42c028;sprop-parameter-sets=Z0LAKJWgHgCJ+XAWoCAgKAAAH0AABhqEIA==,aM48gA== a=rtpmap:97 red/90000 a=rtpmap:98 ulpfec/90000 a=ssrc:694513671 msid:user119389484@host-196b6568 webrtctransceiver2 a=ssrc:694513671 cname:user119389484@host-196b6568 a=mid:video0 a=fingerprint:sha-256 66:39:E9:A0:C1:C8:83:09:D0:3D:CD:A6:A7:1E:7B:A3:E4:DE:62:B0:20:08:98:16:74:F6:48:AD:49:BB:B9:0F m=audio 9 UDP/TLS/RTP/SAVPF 96 c=IN IP4 0.0.0.0 a=setup:actpass a=ice-ufrag:DWJDEBPdaEDqrvwWH4taF+4IxwzJUbtt a=ice-pwd:2TLZ/cUYHrYWec9NL2Q08z2X2N5pQEy6 a=rtcp-mux a=rtcp-rsize a=sendonly a=rtpmap:96 OPUS/48000/2 a=rtcp-fb:96 nack pli a=fmtp:96 sprop-maxcapturerate=48000;sprop-stereo=1 a=ssrc:1591833598 msid:user119389484@host-196b6568 webrtctransceiver3 a=ssrc:1591833598 cname:user119389484@host-196b6568 a=mid:audio1 a=fingerprint:sha-256 66:39:E9:A0:C1:C8:83:09:D0:3D:CD:A6:A7:1E:7B:A3:E4:DE:62:B0:20:08:98:16:74:F6:48:AD:49:BB:B9:0F This appears to also be offering forward error correction (red, ulpfec)? However, I have not enabled that (yet) but I was going to try that next. One thing that worries/confuses me, is that it is using payload types 96/97/98 for video, and 96 for audio, too. Is the duplicate 96 a problem? And, where is it getting these numbers? The RTP packets I am using as "input" for webrtcbin have 96 for audio and 97 for video. In short: how do I use NACKs or FEC to handle packet loss from webrtcbin to a browser? Kind regards, Michiel ------------------------------ Message: 2 Date: Mon, 4 Oct 2021 21:51:24 +0900 From: ??? <byunghun.lee@fainders.ai> To: Discussion of the development of and with GStreamer <gstreamer-devel@lists.freedesktop.org> Cc: jam@tigger.ws Subject: Re: rtsp-server Message-ID: <CADirASCPgDKNdR6Cq-zh4miOGmi-Ce8j4r+M3g+RnFj=ECWGqA@mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hi Then the server pipeline reads from the pipesrc?s and every new connection uses a new src? James I'm new on gstreamer. So maybe my saying could be useless but anyway I had got same error when I tried connect multiple client into a single usb source with rtsp-server. I thought this error means, multiple pipeline tried to make source from a single usb source. So after with some research, in github of rtsp-server, I found that If you don't set gst media factory shared option to true, then every connection will make new pipeline. Even you set the option true, in my case there have identical pts error, but anyway with some other muxer or elements I can work around that error. I hope this work for you -------------- next part -------------- An HTML attachment was scrubbed... URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20211004/6b84aa46/attachment-0001.htm> ------------------------------ Message: 3 Date: Mon, 04 Oct 2021 21:19:44 +0000 From: David Walter <dlwalter@protonmail.com> To: Discussion of the development of and with GStreamer <gstreamer-devel@lists.freedesktop.org> Subject: gst_tag_xmp_list_schemas() linker error on Ubuntu Message-ID: <70B8BEBB-77C5-4229-AEF5-8702DC4E1E74@protonmail.com> Content-Type: text/plain; charset="utf-8" Gstreamer 1.14.4 on a Jetson Nano running Ubuntu 18.04. I have no problem building and running a basic videotestsrc pipeline. I am trying to add a custom XMP schema, starting with just listing the schemas using [gst_tag_xmp_list_schemas](https://gstreamer.freedesktop.org/documentation/tag/gsttagxmp.html?gi-language=c#gst_tag_xmp_list_schemas) but get a ?undefined reference to ?gst_tag_xmp_list_schemas?. I am not able to access any of the functions in [gstxmptag.c](https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/blob/1.14.4/gst-libs/gst/tag/gstxmptag.c) either. Is this included in libgstreamer-plugins-base1.0-0? -------------- next part -------------- An HTML attachment was scrubbed... URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20211004/e39bcf2a/attachment-0001.htm> ------------------------------ Message: 4 Date: Tue, 5 Oct 2021 09:24:46 +0200 From: Davide Perini <perini.davide@dpsoftware.org> To: "gstreamer-devel@lists.freedesktop.org" <gstreamer-devel@lists.freedesktop.org> Subject: Screen capture, select the right screen. Message-ID: <50c2da8a-4078-b3e1-2003-a6b93308ef91@dpsoftware.org> Content-Type: text/plain; charset="iso-8859-15"; Format="flowed" Hi all, I'm using this command to screen capture ./gst-launch-1.0 d3d11desktopdupsrc monitor-index=1 ! d3d11convert ! d3d11download ! autovideosink I have noticed that the "monitor-index" does not represent the real display number on Windows. If I rearrange the monitor order in Windows, monitor index on gstreamer does not change. How can I know how to record the left and right screen if that number does not represent the "numbers I see in Windows"? Thank you Davide -------------- next part -------------- An HTML attachment was scrubbed... URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20211005/96dfeb68/attachment-0001.htm> ------------------------------ Message: 5 Date: Tue, 5 Oct 2021 09:43:46 +0200 From: Marianna Smidth Buschle <msb@qtec.com> To: gstreamer-devel@lists.freedesktop.org Subject: Changing buffer PTS/DTS in Python Message-ID: <24676185-66b4-745f-9e09-069cac0452ff@qtec.com> Content-Type: text/plain; charset=utf-8; format=flowed Hi, I need to manipulate some buffer timestamps in python. But the Gst.Buffer.make_writable() function does not exist: https://gitlab.freedesktop.org/gstreamer/gst-python/-/issues/18 So I don't suppose it is safe to do: "buf.pts = Gst.CLOCK_TIME_NONE"? Are there any workarounds? I have been trying to cook a solution using ctypes. But I can't seem to make the C pointer to GstBuffer* I receive back into a python Gst.Buffer. It seems like Gst.Buffer is not compatible with ctypes: https://gitlab.freedesktop.org/gstreamer/gst-python/-/issues/46 So I started trying to duplicate the GstBuffer* in python as a ctypes compatible class/struct. But then I also need to implement the GstMiniObject the same way, and I'm afraid I will find even more dependencies in the mini object. -- Best regards / Med venlig hilsen ?Marianna Smidth Buschle? ------------------------------ Subject: Digest Footer _______________________________________________ gstreamer-devel mailing list gstreamer-devel@lists.freedesktop.org https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel ------------------------------ End of gstreamer-devel Digest, Vol 129, Issue 6 *********************************************** </div></div></blockquote></div> </div></body></html>