gst_tag_xmp_list_schemas() linker error on Ubuntu

Tue Oct 5 15:50:43 UTC 2021

Solved this with help from Tim on IRC.

Needed to add `-lgsttag-1.0` to my make command.

David Walter
––––––––––––––––––––––––––––––––

> On Oct 5, 2021, at 12:43 AM, gstreamer-devel-request at lists.freedesktop.org wrote:
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> 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 at aanmelder.nl>
> To: Discussion of the development of and with GStreamer
> <gstreamer-devel at 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 at 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 at host-196b6568 webrtctransceiver2
> a=ssrc:694513671 cname:user119389484 at 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 at host-196b6568 webrtctransceiver3
> a=ssrc:1591833598 cname:user119389484 at 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 at fainders.ai>
> To: Discussion of the development of and with GStreamer
> <gstreamer-devel at lists.freedesktop.org>
> Cc: jam at tigger.ws
> Subject: Re: rtsp-server
> Message-ID:
> <CADirASCPgDKNdR6Cq-zh4miOGmi-Ce8j4r+M3g+RnFj=ECWGqA at 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
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>
> ------------------------------
>
> Message: 3
> Date: Mon, 04 Oct 2021 21:19:44 +0000
> From: David Walter <dlwalter at protonmail.com>
> To: Discussion of the development of and with GStreamer
> <gstreamer-devel at lists.freedesktop.org>
> Subject: gst_tag_xmp_list_schemas() linker error on Ubuntu
> Message-ID: <70B8BEBB-77C5-4229-AEF5-8702DC4E1E74 at 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?
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> ------------------------------
>
> Message: 4
> Date: Tue, 5 Oct 2021 09:24:46 +0200
> From: Davide Perini <perini.davide at dpsoftware.org>
> To: "gstreamer-devel at lists.freedesktop.org"
> <gstreamer-devel at lists.freedesktop.org>
> Subject: Screen capture, select the right screen.
> Message-ID: <50c2da8a-4078-b3e1-2003-a6b93308ef91 at 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
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> ------------------------------
>
> Message: 5
> Date: Tue, 5 Oct 2021 09:43:46 +0200
> From: Marianna Smidth Buschle <msb at qtec.com>
> To: gstreamer-devel at lists.freedesktop.org
> Subject: Changing buffer PTS/DTS in Python
> Message-ID: <24676185-66b4-745f-9e09-069cac0452ff at 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?
>
> ------------------------------
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