Send mpeg-ts file source to SRT. Error about payload

Daniel Rossi electroteque at gmail.com
Thu Aug 1 15:38:18 UTC 2019


Are these params currently supported on the uri's ?

https://github.com/Haivision/srt/issues/659#issuecomment-516922869
https://github.com/Haivision/srt/issues/445

I see no reference in the sources about srt-recv:{INT, 0...}



------ Original Message ------
From: "Daniel Rossi" <electroteque at gmail.com>
To: "Nicolas Dufresne" <nicolas at ndufresne.ca>
Cc: "Discussion of the development of and with GStreamer" 
<gstreamer-devel at lists.freedesktop.org>
Sent: 8/1/2019 12:29:54 AM
Subject: Re[11]: Send mpeg-ts file source to SRT. Error about payload

>This was their response in regards to a receiver buffer size. I have 
>not touched libsrt before so I would have no idea what this means
>
>https://github.com/Haivision/srt/issues/659#issuecomment-516865921
>
>------ Original Message ------
>From: "Daniel Rossi" <electroteque at gmail.com>
>To: "Nicolas Dufresne" <nicolas at ndufresne.ca>
>Cc: "Discussion of the development of and with GStreamer" 
><gstreamer-devel at lists.freedesktop.org>
>Sent: 8/1/2019 12:17:57 AM
>Subject: Re[10]: Send mpeg-ts file source to SRT. Error about payload
>
>>Sorry mate I am still fine tuning mpegts sending to wowza, and 
>>consuming both audio and video on a listener receiver eventually from 
>>Wowza.
>>
>>I have made contact to libsrt about the errors.
>>
>>I tried using UDP. I get no error. but calling and listening locally, 
>>it skips frames and lots of blocky picture.
>>
>>gst-launch-1.0 filesrc location=sintel_lang.ts ! tsparse 
>>set-timestamps=1 smoothing-latency=40000000 ! chopmydata step-size=188 
>>min-size=188 max-size=1316 ! udpsink host=127.0.0.1 port=8081
>>
>>gst-play-1.0 udp://127.0.0.1:8081
>>
>>Is there a better pipeline for either loading mpeg-ts or muxing to 
>>mpeg-ts from h264. I've seen nothing much out there. Most examples 
>>just using the videotest source ! I've seen so many cheatsheets but 
>>nothing for what I need to do.
>>
>>I just need a test caller for Wowza ingest to confirm if it sends 
>>multiple tracks correctly. but my production requirements is consuming 
>>a Wowza SRT stream target which is mpeg-ts using a receiver listener. 
>>My tests have been unstable output so far. locally or remotely.
>>
>>
>>
>>
>>------ Original Message ------
>>From: "Nicolas Dufresne" <nicolas at ndufresne.ca>
>>To: "Daniel Rossi" <electroteque at gmail.com>
>>Cc: "Discussion of the development of and with GStreamer" 
>><gstreamer-devel at lists.freedesktop.org>
>>Sent: 7/30/2019 9:10:25 PM
>>Subject: Re: Re[8]: Send mpeg-ts file source to SRT. Error about 
>>payload
>>
>>>
>>>
>>>Le lun. 29 juill. 2019 23 h 48, Daniel Rossi <electroteque at gmail.com> 
>>>a écrit :
>>>>Thankyou. So confirming chopmydata is like "pkt_size=1316" ? It is 
>>>>an obvious libsrt output, so I will have to take it up with them !
>>>
>>>Yes.
>>>
>>>>
>>>>------ Original Message ------
>>>>From: "Nicolas Dufresne" <nicolas at ndufresne.ca>
>>>>To: "Daniel Rossi" <electroteque at gmail.com>
>>>>Cc: "Discussion of the development of and with GStreamer" 
>>>><gstreamer-devel at lists.freedesktop.org>
>>>>Sent: 7/30/2019 2:23:19 AM
>>>>Subject: Re: Re[6]: Send mpeg-ts file source to SRT. Error about 
>>>>payload
>>>>
>>>>>Le lundi 29 juillet 2019 à 15:41 +0000, Daniel Rossi a écrit :
>>>>>>It seems it needs this, which possibly matches ffmpeg's pkt_size 
>>>>>>flag ? ie udp://192.168.4.43:9999?pkt_size=1316
>>>>>>
>>>>>>
>>>>>>"chopmydata step-size=188 min-size=188 max-size=1316"
>>>>>>
>>>>>>I still get unstable playback locally
>>>>>>
>>>>>>on the sender console
>>>>>>
>>>>>>"10:43:54.198464/mpegtsparse2-0:*E: SRT.d: SND-DROPPED 41 packets 
>>>>>>- lost delaying for 1038ms"
>>>>>>
>>>>>>on the receiver console
>>>>>>
>>>>>>"10:30:08.957619*E: SRT.c: %229645152:No room to store incoming 
>>>>>>packet: offset=8907 avail=6437 ack.seq=59978716 pkt.seq=59987623 
>>>>>>rcv-remain=1754"
>>>>>>
>>>>>>I am getting the same errors eventually for this command.It 
>>>>>>crashes eventually
>>>>>>
>>>>>>gst-launch-1.0 videotestsrc ! video/x-raw, height=360, width=640 ! 
>>>>>>videoconvert ! x264enc tune=zerolatency ! video/x-h264, 
>>>>>>profile=high ! mpegtsmux ! srtsink uri="srt://192.168.4.43:8081"
>>>>>
>>>>>You should add is-live=1 to videotestsrc, though I agree something
>>>>>seems not too robust here.
>>>>>
>>>>>>
>>>>>>Playing it back in VLC, the picture has artifacts and it's 
>>>>>>skipping. No logs on the sender however. It's for udpsink also. So 
>>>>>>might be my source file ? It was converted with ffmpeg first
>>>>>>
>>>>>>ffmpeg -i sintel_lang_2000k.mp4 -codec:v copy -codec:a copy -map 0 
>>>>>>-streamid 0:50 -streamid 1:52 -streamid 2:53 -streamid 3:54 
>>>>>>-streamid 4:55 -streamid 5:56 -f mpegts sintel_lang.ts
>>>>>>
>>>>>>My proof of concept seems to work however. PID's of the audio are 
>>>>>>sent with the stream for individual ingesting in Wowza over SRT.
>>>>>>
>>>>>>Is there specific documenation for sending mpeg-ts or converting 
>>>>>>from h264 first ? With ffmpeg I have been doing this for my udp 
>>>>>>specific tests
>>>>>>
>>>>>>ffmpeg -re -i sintel_lang_2000k.mp4 -codec copy -bsf:v 
>>>>>>h264_mp4toannexb -map 0 -streamid 0:50 -streamid 1:52 -streamid 
>>>>>>2:53 -streamid 3:54 -streamid 4:55 -streamid 5:56 -f mpegts 
>>>>>>udp://192.168.4.43:10000?pkt_size=1316
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>------ Original Message ------
>>>>>>From: "Nicolas Dufresne" <nicolas at ndufresne.ca>
>>>>>>To: "Daniel Rossi" <electroteque at gmail.com>
>>>>>>Cc: "Discussion of the development of and with GStreamer" 
>>>>>><gstreamer-devel at lists.freedesktop.org>
>>>>>>Sent: 7/27/2019 3:48:52 AM
>>>>>>Subject: Re: Re[4]: Send mpeg-ts file source to SRT. Error about 
>>>>>>payload
>>>>>>
>>>>>> > Le vendredi 26 juillet 2019 à 16:37 +0000, Daniel Rossi a écrit 
>>>>>>:
>>>>>> > > according to gst-inspect-1.0 filesrc
>>>>>> > >
>>>>>> > > I have a blocksize option. Do I set this to 1316?
>>>>>> > >
>>>>>> > > inspecting tsparse doesn't say much, including command usage.
>>>>>> > >
>>>>>> > > gst-launch-1.0 -v filesrc location=sintel_lang.ts 
>>>>>>blocksize=1316 ! tsparse ! srtsink uri=srt://:8888/
>>>>>> > >
>>>>>> > > and the receiver
>>>>>> > >
>>>>>> > > gst-launch-1.0 srtsrc uri=srt://192.168.4.55:8888 ! decodebin 
>>>>>>! autovideosink
>>>>>> > >
>>>>>> > > I get these errors.
>>>>>> > >
>>>>>> > > 02:34:56.655957*E: SRT.c: LiveSmoother: payload size: 18800 
>>>>>>exceeds maximum allowed 1316
>>>>>> >
>>>>>> >
>>>>>> > Apparently the parser ignores the input size, just like you 
>>>>>>ignored my
>>>>>> > recommendation for tsparse configuration. Anyway ...
>>>>>> >
>>>>>> > # Transmitter
>>>>>> > gst-launch-1.0 filesrc location=my.ts ! \
>>>>>> > tsparse set-timestamps=1 smoothing-latency=40000000 ! \
>>>>>> > chopmydata step-size=188 min-size=188 max-size=1316 ! \
>>>>>> > srtsink uri=srt://:8888
>>>>>> >
>>>>>> > # Receiver / Player
>>>>>> > gst-play-1.0 srt://127.0.0.1:8888
>>>>>> > >
>>>>>> > >
>>>>>> > >
>>>>>> > > ------ Original Message ------
>>>>>> > > From: "Nicolas Dufresne" <nicolas at ndufresne.ca>
>>>>>> > > To: "Daniel Rossi" <electroteque at gmail.com>
>>>>>> > > Cc: "Discussion of the development of and with GStreamer" 
>>>>>><gstreamer-devel at lists.freedesktop.org>
>>>>>> > > Sent: 7/26/2019 10:03:13 PM
>>>>>> > > Subject: Re: Re[2]: Send mpeg-ts file source to SRT. Error 
>>>>>>about payload
>>>>>> > >
>>>>>> > > >
>>>>>> > > > Le jeu. 25 juill. 2019 23 h 30, Daniel Rossi 
>>>>>><electroteque at gmail.com> a écrit :
>>>>>> > > > > There is an element called tsparse, but same thing.
>>>>>> > > > >
>>>>>> > > > > gst-launch-1.0 -v filesrc location =sintel_lang.ts ! 
>>>>>>tsparse ! srtsink uri=srt://:8888
>>>>>> > > >
>>>>>> > > > You should use gst-inspect-1.0 to learn about the 
>>>>>>configuration for filesrc and tsparse (I'm typing this from 
>>>>>>memory, and there exist in usage of mpegts and ts as element name 
>>>>>>prefix). File source has an option to configure the read size, 
>>>>>>these needs to be multiple of 188 and max to 1316. The ts parse as 
>>>>>>an option to add and smooth timestamp, these need to be 
>>>>>>configured.
>>>>>> > > >
>>>>>> > > > > my pullside for the test is
>>>>>> > > > >
>>>>>> > > > > gst-launch-1.0 srtsrc uri=srt://192.168.4.55:8888 ! 
>>>>>>decodebin ! autovideosink
>>>>>> > > > >
>>>>>> > > > > ------ Original Message ------
>>>>>> > > > > From: "Nicolas Dufresne" <nicolas at ndufresne.ca>
>>>>>> > > > > To: "Daniel Rossi" <electroteque at gmail.com>; "Discussion 
>>>>>>of the development of and with GStreamer" 
>>>>>><gstreamer-devel at lists.freedesktop.org>
>>>>>> > > > > Sent: 7/26/2019 1:19:54 PM
>>>>>> > > > > Subject: Re: Send mpeg-ts file source to SRT. Error about 
>>>>>>payload
>>>>>> > > > >
>>>>>> > > > > >
>>>>>> > > > > > Le jeu. 25 juill. 2019 22 h 25, Daniel Rossi 
>>>>>><electroteque at gmail.com> a écrit :
>>>>>> > > > > > > I'm trying to send an mpeg-ts source over SRT for 
>>>>>>multi language track testing.
>>>>>> > > > > > >
>>>>>> > > > > > > When pulling this stream I am getting an internal 
>>>>>>error.
>>>>>> > > > > > >
>>>>>> > > > > > > gst-launch-1.0 -v filesrc location =sintel_lang.ts ! 
>>>>>>rtpstreampay ! srtsink uri=srt://:8888/
>>>>>> > > > > > > Setting pipeline to PAUSED ...
>>>>>> > > > > > > Pipeline is PREROLLING ...
>>>>>> > > > > > > Pipeline is PREROLLED ...
>>>>>> > > > > > > Setting pipeline to PLAYING ...
>>>>>> > > > > > > New clock: GstSystemClock
>>>>>> > > > > > > 12:13:33.532337/filesrc0:src*E: SRT.c: LiveSmoother: 
>>>>>>payload size: 4098 exceeds maximum allowed 1316
>>>>>> > > > > >
>>>>>> > > > > > a) why do you use stream pay ?
>>>>>> > > > > > b) you might want to use mpegtsparse to timestamp your 
>>>>>>stream
>>>>>> > > > > > c) configure filesrc to read 1316 bytes to fix this 
>>>>>>error.
>>>>>> > > > > >
>>>>>> > > > > > >
>>>>>> > > > > > > Is there also a way to pipeline a h264 file with multi 
>>>>>>audio tracks through mpeg-ts and set PID numbers for each track 
>>>>>>instead of outputting to TS via ffmpeg first ?
>>>>>> > > > > > > _______________________________________________
>>>>>> > > > > > > gstreamer-devel mailing list
>>>>>> > > > > > > gstreamer-devel at lists.freedesktop.org
>>>>>> > > > > > > 
>>>>>>https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20190801/ee40ac9f/attachment-0001.html>


More information about the gstreamer-devel mailing list