<html> <head> <meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1"> <style type="text/css" style="display:none;"> P {margin-top:0;margin-bottom:0;} </style> </head> <body dir="ltr"> <div style="font-family: Calibri, Helvetica, sans-serif; font-size: 12pt; color: rgb(0, 0, 0);"> Hi </div> <div style="font-family: Calibri, Helvetica, sans-serif; font-size: 12pt; color: rgb(0, 0, 0);"> </div> <div style="font-family: Calibri, Helvetica, sans-serif; font-size: 12pt; color: rgb(0, 0, 0);"> I believe that I have figured out where the mistake lies in... is because I failed to fill in the filtercaps for the video in the reciever portion of the pipeline. Would just need to fill in caps=application/x-rtp to do the magic</div> <div style="font-family: Calibri, Helvetica, sans-serif; font-size: 12pt; color: rgb(0, 0, 0);"> </div> <div style="font-family: Calibri, Helvetica, sans-serif; font-size: 12pt; color: rgb(0, 0, 0);"> Reciever</div> <div style="font-family: Calibri, Helvetica, sans-serif; font-size: 12pt; color: rgb(0, 0, 0);"> </div> <div> <div id="appendonsend"></div> <div style="font-family:Calibri,Helvetica,sans-serif; font-size:12pt; color:rgb(0,0,0)"> gst-launch-1.0 udpsrc port=5555 caps=application/x-rtp ! rtph264depay ! decodebin ! autovideosink</div> <div style="font-family:Calibri,Helvetica,sans-serif; font-size:12pt; color:rgb(0,0,0)"> </div> <div style="font-family:Calibri,Helvetica,sans-serif; font-size:12pt; color:rgb(0,0,0)"> Regards</div> <hr tabindex="-1" style="display:inline-block; width:98%"> <div id="divRplyFwdMsg" dir="ltr">From: gstreamer-devel <gstreamer-devel-bounces@lists.freedesktop.org> on behalf of gstreamer-devel-request@lists.freedesktop.org <gstreamer-devel-request@lists.freedesktop.org> Sent: Friday, December 17, 2021 11:30 PM To: gstreamer-devel@lists.freedesktop.org <gstreamer-devel@lists.freedesktop.org> Subject: gstreamer-devel Digest, Vol 131, Issue 23 <div> </div> </div> <div class="BodyFragment"> <div class="PlainText">Send gstreamer-devel mailing list submissions to gstreamer-devel@lists.freedesktop.org To subscribe or unsubscribe via the World Wide Web, visit <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel"> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> 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. Sunxi NV12 detiling (Giulio Benetti) 2. Re: Gstreamer: No RTP protocol present (Howling wong) ---------------------------------------------------------------------- Message: 1 Date: Fri, 17 Dec 2021 13:31:05 +0100 From: Giulio Benetti <giulio.benetti@benettiengineering.com> To: gstreamer-devel@lists.freedesktop.org, Nicolas Dufresne <nicolas.dufresne@collabora.com> Cc: Devrim GECEGEZER <devrim.gecegezer@genemek.com> Subject: Sunxi NV12 detiling Message-ID: <0362eb7a-7158-c502-eaf6-c61c5c48bcf2@benettiengineering.com> Content-Type: text/plain; charset=utf-8; format=flowed Hello Nicolas, All, I'm dealing with Cedrus on Linux 5.15.7 with Gstreamer 1.19.3.1-git(the latest main branch) and I've got to the point that on A13 and A20 I have to detile the output of Sunxi video-engine, since now it's done in software and this causes: ``` videodecoder gstvideodecoder.c:3670:gst_video_decoder_clip_and_push_buf:<v4l2slh264dec0> Dropping frame due to QoS. ``` using kmssink. And I get an entire frame every 5 seconds. I've found this [1] IRC discussion where you state that kmssink support is missing for GST_VIDEO_FORMAT_NV12_32L32. Basically what I understand is that we need a special treatment for A13 and A20, since from >= A33 we have the support for the untiled output(still not tried, but I have to do it with A64). Where can I begin from to implement the kmssink support for detiling? There is something I can imitate? Can you or someone else point me more or less where to start working? [1]: <a href="https://oftc.irclog.whitequark.org/linux-sunxi/2021-07-13#1626188848-1626188913">https://oftc.irclog.whitequark.org/linux-sunxi/2021-07-13#1626188848-1626188913</a>; Thanks in advance Best regards -- Giulio Benetti Benetti Engineering sas ------------------------------ Message: 2 Date: Fri, 17 Dec 2021 15:30:30 +0000 From: Howling wong <watertreader@hotmail.com> To: "gstreamer-devel@lists.freedesktop.org" <gstreamer-devel@lists.freedesktop.org> Subject: Re: Gstreamer: No RTP protocol present Message-ID: <SI2PR04MB51826EC9B35083EBD1DA334BCB789@SI2PR04MB5182.apcprd04.prod.outlook.com> Content-Type: text/plain; charset="iso-8859-1" Sorry, I hope I got it right this time My message being: My mistake in copying the errorneous pipeline, the sender pipeline should read instead "gst-launch-1.0 videotestsrc ! video/x-raw, width=640, height=480 ! queue ! vpuenc_h264 ! rtph264pay ! udpsink host=192.168.60.5 port=5555" Is there a possibility of corrupt gstreamer installation or hardware? Additional Information: 1. Sender (Embedded system on Arm on Archlinux with Gstreamer 1.14) 2. Reciever (Windows x86-64 running on Windows 10 running Gstreamer 1.18) Would different version of Gstreamer affect the system Regards ________________________________ From: gstreamer-devel <gstreamer-devel-bounces@lists.freedesktop.org> on behalf of gstreamer-devel-request@lists.freedesktop.org <gstreamer-devel-request@lists.freedesktop.org> Sent: Friday, December 17, 2021 7:44 PM To: gstreamer-devel@lists.freedesktop.org <gstreamer-devel@lists.freedesktop.org> Subject: gstreamer-devel Digest, Vol 131, Issue 22 Send gstreamer-devel mailing list submissions to gstreamer-devel@lists.freedesktop.org To subscribe or unsubscribe via the World Wide Web, visit <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel"> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> 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. Re: gstreamer-devel Digest, Vol 131, Issue 21 (Howling wong) 2. explain please (James) ---------------------------------------------------------------------- Message: 1 Date: Fri, 17 Dec 2021 09:48:28 +0000 From: Howling wong <watertreader@hotmail.com> To: "gstreamer-devel@lists.freedesktop.org" <gstreamer-devel@lists.freedesktop.org> Subject: Re: gstreamer-devel Digest, Vol 131, Issue 21 Message-ID: <SI2PR04MB5182F19A20728593C75DA256CB789@SI2PR04MB5182.apcprd04.prod.outlook.com> Content-Type: text/plain; charset="iso-8859-1" Hi it is my mistake ... it should be width=640 instead of width=6400 in writing the post... the pipeline do not work with width=640 ________________________________ From: gstreamer-devel <gstreamer-devel-bounces@lists.freedesktop.org> on behalf of gstreamer-devel-request@lists.freedesktop.org <gstreamer-devel-request@lists.freedesktop.org> Sent: Friday, December 17, 2021 11:22 AM To: gstreamer-devel@lists.freedesktop.org <gstreamer-devel@lists.freedesktop.org> Subject: gstreamer-devel Digest, Vol 131, Issue 21 Send gstreamer-devel mailing list submissions to gstreamer-devel@lists.freedesktop.org To subscribe or unsubscribe via the World Wide Web, visit <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel"> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> 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. Gstreamer: No RTP protocol present (Howling wong) 2. Re: Gstreamer: No RTP protocol present (Robert Hensel) ---------------------------------------------------------------------- Message: 1 Date: Fri, 17 Dec 2021 02:55:33 +0000 From: Howling wong <watertreader@hotmail.com> To: "gstreamer-devel@lists.freedesktop.org" <gstreamer-devel@lists.freedesktop.org> Subject: Gstreamer: No RTP protocol present Message-ID: <SI2PR04MB5182E8B2DF3B9669E94C5BC1CB789@SI2PR04MB5182.apcprd04.prod.outlook.com> Content-Type: text/plain; charset="iso-8859-1" I am having some issues with the following gstreamer command Sender (on embeeded system) gst-launch-1.0 videotestsrc ! video/x-raw, width=6400, height=480 ! queue ! vpuenc_h264 ! rtph264pay ! udpsink host=192.168.60.5 port=5555 Receiver(on windows) gst-launch-1.0 udpsrc port=5555 ! queue ! rtph264depay ! decodebin ! autovideosink But I have got the following response Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Got context from element 'autovideosink0': gst.d3d11.device.handle=context, device=(GstD3D11Device)"\(GstD3D11Device\)\ d3d11device4", adapter=(uint)0, device-id=(uint)6429, vendor-id=(uint)32902, hardware=(boolean)true, description=(string)"Intel\(R\)\ HD\ Graphics\ P530"; Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstSystemClock ERROR: from element /GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0: No RTP format was negotiated. Additional debug info: ../gst-libs/gst/rtp/gstrtpbasedepayload.c(538): gst_rtp_base_depayload_handle_buffer (): /GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0: Input buffers need to have RTP caps set on them. This is usually achieved by setting the 'caps' property of the upstream source element (often udpsrc or appsrc), or by putting a capsfilter element before the depayloader and setting the 'caps' property on that. Also see <a href="http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README"> http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README</a> Execution ended after 0:00:00.019641000 Setting pipeline to NULL ... ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data stream error. Additional debug info: ../libs/gst/base/gstbasesrc.c(3127): gst_base_src_loop (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0 streaming stopped, reason not-negotiated (-4) Freeing pipeline .. The complaint seem to be about the incoming stream is not in rtp format and the rtpdepayh264 should not be placed in the pipeline. This assumption is proven to be correct when i replaced the whole pipeline with a fakesink Receiver gst-launch-1.0 udpsrc port=5555 ! queue ! fakesink The pipeline work. However when i observed the packets exchange in wireshark, it show the communication exchange but the protocol is in udp. Though I know that RTP could be based upon UDP protocol but have thought that Wireshark is entirely capable of showing protocol in RTP format I have thought that the sender has already wrapped the video in rtp format before sending the package out. Like to have some ideas on what is wrong here and how to proceed Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <<a href="https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20211217/4681e121/attachment-0001.htm">https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20211217/4681e121/attachment-0001.htm</a>> ------------------------------ Message: 2 Date: Fri, 17 Dec 2021 14:21:49 +1100 From: Robert Hensel <vk3eht@gmail.com> To: Discussion of the development of and with GStreamer <gstreamer-devel@lists.freedesktop.org> Subject: Re: Gstreamer: No RTP protocol present Message-ID: <CAGRXgDxxb4BrjW+yT9fdudoYdiQjs0JCxpLAJQx6_sMjXHoOMQ@mail.gmail.com> Content-Type: text/plain; charset="utf-8" width=6400? Suggest try with width=640 Rob On Fri, 17 Dec 2021 at 13:55, Howling wong via gstreamer-devel < gstreamer-devel@lists.freedesktop.org> wrote: > I am having some issues with the following gstreamer command > > Sender (on embeeded system) > > gst-launch-1.0 videotestsrc ! video/x-raw, width=6400, height=480 ! queue ! vpuenc_h264 ! rtph264pay ! udpsink host=192.168.60.5 port=5555 > > Receiver(on windows) > > gst-launch-1.0 udpsrc port=5555 ! queue ! rtph264depay ! decodebin ! autovideosink > > But I have got the following response > > > Setting pipeline to PAUSED ... > > Pipeline is live and does not need PREROLL ... > > Got context from element 'autovideosink0': > gst.d3d11.device.handle=context, > device=(GstD3D11Device)"\(GstD3D11Device\)\ d3d11device4", adapter=(uint)0, > device-id=(uint)6429, vendor-id=(uint)32902, hardware=(boolean)true, > description=(string)"Intel\(R\)\ HD\ Graphics\ P530"; > > Pipeline is PREROLLED ... > > Setting pipeline to PLAYING ... > > New clock: GstSystemClock > > ERROR: from element > /GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0: No RTP format was > negotiated. > > Additional debug info: > > ../gst-libs/gst/rtp/gstrtpbasedepayload.c(538): > gst_rtp_base_depayload_handle_buffer (): > /GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0: > > Input buffers need to have RTP caps set on them. This is usually > achieved by setting the 'caps' property of the upstream source element > (often udpsrc or appsrc), or by putting a capsfilter element before the > depayloader and setting the 'caps' property on that. Also see > <a href="http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README"> http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README</a> > Execution ended after 0:00:00.019641000 > Setting pipeline to NULL ... > ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal > data stream error. > Additional debug info: > ../libs/gst/base/gstbasesrc.c(3127): gst_base_src_loop (): > /GstPipeline:pipeline0/GstUDPSrc:udpsrc0 > streaming stopped, reason not-negotiated (-4) > Freeing pipeline .. > > > The complaint seem to be about the incoming stream is not in rtp format > and the rtpdepayh264 should not be placed in the pipeline. This assumption > is proven to be correct when i replaced the whole pipeline with a fakesink > > > Receiver > > gst-launch-1.0 udpsrc port=5555 ! queue ! fakesink > > The pipeline work. However when i observed the packets exchange in > wireshark, it show the communication exchange but the protocol is in udp. > Though I know that RTP could be based upon UDP protocol but have thought > that Wireshark is entirely capable of showing protocol in RTP format > > > I have thought that the sender has already wrapped the video in rtp format > before sending the package out. Like to have some ideas on what is wrong > here and how to proceed > > Regards > > -------------- next part -------------- An HTML attachment was scrubbed... URL: <<a href="https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20211217/fa1c5430/attachment.htm">https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20211217/fa1c5430/attachment.htm</a>> ------------------------------ Subject: Digest Footer _______________________________________________ gstreamer-devel mailing list gstreamer-devel@lists.freedesktop.org <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> ------------------------------ End of gstreamer-devel Digest, Vol 131, Issue 21 ************************************************ -------------- next part -------------- An HTML attachment was scrubbed... URL: <<a href="https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20211217/82502d7e/attachment-0001.htm">https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20211217/82502d7e/attachment-0001.htm</a>> ------------------------------ Message: 2 Date: Fri, 17 Dec 2021 19:20:20 +0800 From: James <jam@tigger.ws> To: Discussion of the development of and with GStreamer <gstreamer-devel@lists.freedesktop.org> Subject: explain please Message-ID: <9EEBD5CC-E6E7-4E7D-A1F0-7F69CAD77D83@tigger.ws> Content-Type: text/plain; charset=us-ascii Can anybody explain: 4-0> Dropping frame due to QoS. start:0:02:11.042561535 deadline:0:02:11.042561535 earliest_time:0:02:11.050640555 James ------------------------------ Subject: Digest Footer _______________________________________________ gstreamer-devel mailing list gstreamer-devel@lists.freedesktop.org <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> ------------------------------ End of gstreamer-devel Digest, Vol 131, Issue 22 ************************************************ -------------- next part -------------- An HTML attachment was scrubbed... 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