Gstreamer: No RTP protocol present (Howling wong)
Howling wong
watertreader at hotmail.com
Sat Dec 18 00:34:47 UTC 2021
Hi Olivier
Thanks for your answer
________________________________
From: gstreamer-devel <gstreamer-devel-bounces at lists.freedesktop.org> on behalf of gstreamer-devel-request at lists.freedesktop.org <gstreamer-devel-request at lists.freedesktop.org>
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Subject: gstreamer-devel Digest, Vol 131, Issue 24
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Today's Topics:
1. Re: Gstreamer: No RTP protocol present (Olivier Cr?te)
2. Re: Sunxi NV12 detiling (Nicolas Dufresne)
3. Re: explain please (vinod kesti)
4. Re: Re: Gstreamer: No RTP protocol present (Howling wong)
(Howling wong)
----------------------------------------------------------------------
Message: 1
Date: Fri, 17 Dec 2021 10:31:01 -0500
From: Olivier Cr?te <olivier.crete at collabora.com>
To: Discussion of the development of and with GStreamer
<gstreamer-devel at lists.freedesktop.org>
Cc: Robert Hensel <vk3eht at gmail.com>
Subject: Re: Gstreamer: No RTP protocol present
Message-ID:
<6fa2bee3a60986a540955de8c83460bb117cf70b.camel at collabora.com>
Content-Type: text/plain; charset="UTF-8"
Hi,
The payloader requires that you tell it which payload type is assigned
to H.264
So you'd want something like this:
gst-launch-1.0 udpsrc port=5555 ! application/x-rtp, encoding-name=H264, payload=96 ! queue ! rtph264depay ! decodebin ! autovideosink
Olivier
On Fri, 2021-12-17 at 14:21 +1100, Robert Hensel via gstreamer-devel
wrote:
> width=6400?
> Suggest try with width=640
>
> Rob
>
> On Fri, 17 Dec 2021 at 13:55, Howling wong via gstreamer-devel
> <gstreamer-devel at 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 seehttp://cgit.freedesktop.org/gstreamer/gst-plugins-
> > good/tree/gst/rtp/README
> > ? ? ?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
> >
--
Olivier Cr?te
olivier.crete at collabora.com
------------------------------
Message: 2
Date: Fri, 17 Dec 2021 10:58:25 -0500
From: Nicolas Dufresne <nicolas.dufresne at collabora.com>
To: Discussion of the development of and with GStreamer
<gstreamer-devel at lists.freedesktop.org>
Cc: Giulio Benetti <giulio.benetti at benettiengineering.com>, Devrim
GECEGEZER <devrim.gecegezer at genemek.com>
Subject: Re: Sunxi NV12 detiling
Message-ID:
<4f0e669e36f7fb73a4322ef020a580c2e198d4aa.camel at collabora.com>
Content-Type: text/plain; charset="UTF-8"
Le vendredi 17 d?cembre 2021 ? 13:31 +0100, Giulio Benetti via gstreamer-devel a
?crit?:
> 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
A bit unrelated I suppose, but I have been blocked on the subject since the
decoder only produces green frames for me (I have a Lime2 A20 to test this). If
you could share your kernel branch, and the board DT that works, I could
possibly try and find the difference and unblock this work on my side.
> 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.
Of course all to be expected, the CPUs are not very fast, it is single threaded
by default (see video convert n-threads property), AND this is non-cached CMA
memory, so that is bare metal IOs, very very slow.
>
> 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?
This is all correct, I have a super old patch here that I started for Exynos 4
DRM, but never finished it. I supposed there is likely a helper now for
DrmModeAddFb2 (with modifiers, unless that helpers only exist for the atomic
API, something on my todo). This should be good hint how to translate these
formats to DRM+Modifiers.
https://gitlab.freedesktop.org/ndufresne/gst-plugins-bad/-/commit/ac14733163d770499d834bce12863fe6c4b1570c
>
> [1]:
> https://oftc.irclog.whitequark.org/linux-sunxi/2021-07-13#1626188848-1626188913;
>
> Thanks in advance
> Best regards
------------------------------
Message: 3
Date: Fri, 17 Dec 2021 16:10:30 +0000 (UTC)
From: vinod kesti <vinodkesti at yahoo.com>
To: Discussion of the development of and with GStreamer
<gstreamer-devel at lists.freedesktop.org>
Cc: James <jam at tigger.ws>
Subject: Re: explain please
Message-ID: <2103047805.2245484.1639757430572 at mail.yahoo.com>
Content-Type: text/plain; charset="utf-8"
Hi?James,
Frame was expected to arrive @?0:02:11.042561535but it arrived on??0:02:11.050640555. Frame arrived late so sink is dropping the frame.
Have you enabled QOS. If you dont want QOS then set QOS property to false on every sink of the pipeline
Sent from Yahoo Mail. Get the app
On Friday, 17 December, 2021, 09:21:46 am GMT-6, James via gstreamer-devel <gstreamer-devel at lists.freedesktop.org> wrote:
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
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Message: 4
Date: Sat, 18 Dec 2021 00:29:06 +0000
From: Howling wong <watertreader at hotmail.com>
To: "gstreamer-devel at lists.freedesktop.org"
<gstreamer-devel at lists.freedesktop.org>
Subject: Re: Re: Gstreamer: No RTP protocol present (Howling wong)
Message-ID:
<SI2PR04MB51823B945A278BC53B5CBE77CB799 at SI2PR04MB5182.apcprd04.prod.outlook.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi
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
Reciever
gst-launch-1.0 udpsrc port=5555 caps=application/x-rtp ! rtph264depay ! decodebin ! autovideosink
Regards
________________________________
From: gstreamer-devel <gstreamer-devel-bounces at lists.freedesktop.org> on behalf of gstreamer-devel-request at lists.freedesktop.org <gstreamer-devel-request at lists.freedesktop.org>
Sent: Friday, December 17, 2021 11:30 PM
To: gstreamer-devel at lists.freedesktop.org <gstreamer-devel at lists.freedesktop.org>
Subject: gstreamer-devel Digest, Vol 131, Issue 23
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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 at benettiengineering.com>
To: gstreamer-devel at lists.freedesktop.org, Nicolas Dufresne
<nicolas.dufresne at collabora.com>
Cc: Devrim GECEGEZER <devrim.gecegezer at genemek.com>
Subject: Sunxi NV12 detiling
Message-ID:
<0362eb7a-7158-c502-eaf6-c61c5c48bcf2 at 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]:
https://oftc.irclog.whitequark.org/linux-sunxi/2021-07-13#1626188848-1626188913;
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 at hotmail.com>
To: "gstreamer-devel at lists.freedesktop.org"
<gstreamer-devel at lists.freedesktop.org>
Subject: Re: Gstreamer: No RTP protocol present
Message-ID:
<SI2PR04MB51826EC9B35083EBD1DA334BCB789 at 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 at lists.freedesktop.org> on behalf of gstreamer-devel-request at lists.freedesktop.org <gstreamer-devel-request at lists.freedesktop.org>
Sent: Friday, December 17, 2021 7:44 PM
To: gstreamer-devel at lists.freedesktop.org <gstreamer-devel at lists.freedesktop.org>
Subject: gstreamer-devel Digest, Vol 131, Issue 22
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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 at hotmail.com>
To: "gstreamer-devel at lists.freedesktop.org"
<gstreamer-devel at lists.freedesktop.org>
Subject: Re: gstreamer-devel Digest, Vol 131, Issue 21
Message-ID:
<SI2PR04MB5182F19A20728593C75DA256CB789 at 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 at lists.freedesktop.org> on behalf of gstreamer-devel-request at lists.freedesktop.org <gstreamer-devel-request at lists.freedesktop.org>
Sent: Friday, December 17, 2021 11:22 AM
To: gstreamer-devel at lists.freedesktop.org <gstreamer-devel at lists.freedesktop.org>
Subject: gstreamer-devel Digest, Vol 131, Issue 21
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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 at hotmail.com>
To: "gstreamer-devel at lists.freedesktop.org"
<gstreamer-devel at lists.freedesktop.org>
Subject: Gstreamer: No RTP protocol present
Message-ID:
<SI2PR04MB5182E8B2DF3B9669E94C5BC1CB789 at 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 http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README
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
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Message: 2
Date: Fri, 17 Dec 2021 14:21:49 +1100
From: Robert Hensel <vk3eht at gmail.com>
To: Discussion of the development of and with GStreamer
<gstreamer-devel at lists.freedesktop.org>
Subject: Re: Gstreamer: No RTP protocol present
Message-ID:
<CAGRXgDxxb4BrjW+yT9fdudoYdiQjs0JCxpLAJQx6_sMjXHoOMQ at 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 at 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
> http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README
> 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
>
>
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Message: 2
Date: Fri, 17 Dec 2021 19:20:20 +0800
From: James <jam at tigger.ws>
To: Discussion of the development of and with GStreamer
<gstreamer-devel at lists.freedesktop.org>
Subject: explain please
Message-ID: <9EEBD5CC-E6E7-4E7D-A1F0-7F69CAD77D83 at 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
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