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body { font-family: Segoe UI; font-size: 12pt; } </style></head><body class="plain"><div>It seems it needs this, which possibly matches ffmpeg's pkt_size flag ? ie udp://192.168.4.43:9999?pkt_size=1316<br /><br /><br />"chopmydata step-size=188 min-size=188 max-size=1316"<br /><br />I still get unstable playback locally<br /><br />on the sender console<br /><br />"10:43:54.198464/mpegtsparse2-0:*E: SRT.d: SND-DROPPED 41 packets - lost delaying for 1038ms"<br /><br />on the receiver console<br /><br />"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"<br /><br />I am getting the same errors eventually for this command.It crashes eventually<br /><br />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"<br /><br />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<br /><br /> 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<br /><br />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.<br /><br />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<br /><br />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<br /><br /><br /><br /><br /> <br /><br /><br /><br /><br /><br /><br /><br /></div>
<div><br /></div>
<div>------ Original Message ------</div>
<div>From: "Nicolas Dufresne" <<a href="mailto:nicolas@ndufresne.ca">nicolas@ndufresne.ca</a>></div>
<div>To: "Daniel Rossi" <<a href="mailto:electroteque@gmail.com">electroteque@gmail.com</a>></div>
<div>Cc: "Discussion of the development of and with GStreamer" <<a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a>></div>
<div>Sent: 7/27/2019 3:48:52 AM</div>
<div>Subject: Re: Re[4]: Send mpeg-ts file source to SRT. Error about payload</div><div><br /></div>
<div id="x8800679b3b89475"><blockquote type="cite" class="cite2">
<div class="plain_line">Le vendredi 26 juillet 2019 à 16:37 +0000, Daniel Rossi a écrit :</div>
<blockquote type="cite" class="cite">
<div class="plain_line"> according to gst-inspect-1.0 filesrc</div>
<div class="plain_line"> </div>
<div class="plain_line"> I have a blocksize option. Do I set this to 1316?</div>
<div class="plain_line"> </div>
<div class="plain_line"> inspecting tsparse doesn't say much, including command usage.</div>
<div class="plain_line"> </div>
<div class="plain_line"> gst-launch-1.0 -v filesrc location=sintel_lang.ts blocksize=1316 ! tsparse ! srtsink uri=srt://:8888/</div>
<div class="plain_line"> </div>
<div class="plain_line"> and the receiver</div>
<div class="plain_line"> </div>
<div class="plain_line"> gst-launch-1.0 srtsrc uri=srt://192.168.4.55:8888 ! decodebin ! autovideosink</div>
<div class="plain_line"> </div>
<div class="plain_line"> I get these errors.</div>
<div class="plain_line"> </div>
<div class="plain_line"> 02:34:56.655957*E: SRT.c: LiveSmoother: payload size: 18800 exceeds maximum allowed 1316</div>
</blockquote>
<div class="plain_line"> </div>
<div class="plain_line">Apparently the parser ignores the input size, just like you ignored my</div>
<div class="plain_line">recommendation for tsparse configuration. Anyway ...</div>
<div class="plain_line"> </div>
<div class="plain_line"># Transmitter</div>
<div class="plain_line">gst-launch-1.0 filesrc location=my.ts ! \</div>
<div class="plain_line"> tsparse set-timestamps=1 smoothing-latency=40000000 ! \</div>
<div class="plain_line"> chopmydata step-size=188 min-size=188 max-size=1316 ! \</div>
<div class="plain_line"> srtsink uri=srt://:8888</div>
<div class="plain_line"> </div>
<div class="plain_line"># Receiver / Player</div>
<div class="plain_line">gst-play-1.0 srt://127.0.0.1:8888</div>
<blockquote type="cite" class="cite">
<div class="plain_line"> </div>
<div class="plain_line"> </div>
<div class="plain_line"> </div>
<div class="plain_line"> ------ Original Message ------</div>
<div class="plain_line"> From: "Nicolas Dufresne" <<a href="mailto:nicolas@ndufresne.ca">nicolas@ndufresne.ca</a>></div>
<div class="plain_line"> To: "Daniel Rossi" <<a href="mailto:electroteque@gmail.com">electroteque@gmail.com</a>></div>
<div class="plain_line"> Cc: "Discussion of the development of and with GStreamer" <<a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a>></div>
<div class="plain_line"> Sent: 7/26/2019 10:03:13 PM</div>
<div class="plain_line"> Subject: Re: Re[2]: Send mpeg-ts file source to SRT. Error about payload</div>
<div class="plain_line"> </div>
<div class="plain_line"> ></div>
<div class="plain_line"> > Le jeu. 25 juill. 2019 23 h 30, Daniel Rossi <<a href="mailto:electroteque@gmail.com">electroteque@gmail.com</a>> a écrit :</div>
<div class="plain_line"> > > There is an element called tsparse, but same thing.</div>
<div class="plain_line"> > ></div>
<div class="plain_line"> > > gst-launch-1.0 -v filesrc location =sintel_lang.ts ! tsparse ! srtsink uri=srt://:8888</div>
<div class="plain_line"> ></div>
<div class="plain_line"> > 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.</div>
<div class="plain_line"> ></div>
<div class="plain_line"> > > my pullside for the test is</div>
<div class="plain_line"> > ></div>
<div class="plain_line"> > > gst-launch-1.0 srtsrc uri=srt://192.168.4.55:8888 ! decodebin ! autovideosink</div>
<div class="plain_line"> > ></div>
<div class="plain_line"> > > ------ Original Message ------</div>
<div class="plain_line"> > > From: "Nicolas Dufresne" <<a href="mailto:nicolas@ndufresne.ca">nicolas@ndufresne.ca</a>></div>
<div class="plain_line"> > > To: "Daniel Rossi" <<a href="mailto:electroteque@gmail.com">electroteque@gmail.com</a>>; "Discussion of the development of and with GStreamer" <<a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a>></div>
<div class="plain_line"> > > Sent: 7/26/2019 1:19:54 PM</div>
<div class="plain_line"> > > Subject: Re: Send mpeg-ts file source to SRT. Error about payload</div>
<div class="plain_line"> > ></div>
<div class="plain_line"> > > ></div>
<div class="plain_line"> > > > Le jeu. 25 juill. 2019 22 h 25, Daniel Rossi <<a href="mailto:electroteque@gmail.com">electroteque@gmail.com</a>> a écrit :</div>
<div class="plain_line"> > > > > I'm trying to send an mpeg-ts source over SRT for multi language track testing.</div>
<div class="plain_line"> > > > ></div>
<div class="plain_line"> > > > > When pulling this stream I am getting an internal error.</div>
<div class="plain_line"> > > > ></div>
<div class="plain_line"> > > > > gst-launch-1.0 -v filesrc location =sintel_lang.ts ! rtpstreampay ! srtsink uri=srt://:8888/</div>
<div class="plain_line"> > > > > Setting pipeline to PAUSED ...</div>
<div class="plain_line"> > > > > Pipeline is PREROLLING ...</div>
<div class="plain_line"> > > > > Pipeline is PREROLLED ...</div>
<div class="plain_line"> > > > > Setting pipeline to PLAYING ...</div>
<div class="plain_line"> > > > > New clock: GstSystemClock</div>
<div class="plain_line"> > > > > 12:13:33.532337/filesrc0:src*E: SRT.c: LiveSmoother: payload size: 4098 exceeds maximum allowed 1316</div>
<div class="plain_line"> > > ></div>
<div class="plain_line"> > > > a) why do you use stream pay ?</div>
<div class="plain_line"> > > > b) you might want to use mpegtsparse to timestamp your stream</div>
<div class="plain_line"> > > > c) configure filesrc to read 1316 bytes to fix this error.</div>
<div class="plain_line"> > > ></div>
<div class="plain_line"> > > > ></div>
<div class="plain_line"> > > > > 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 ?</div>
<div class="plain_line"> > > > > _______________________________________________</div>
<div class="plain_line"> > > > > gstreamer-devel mailing list</div>
<div class="plain_line"> > > > > <a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a></div>
<div class="plain_line"> > > > > <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a></div>
</blockquote>
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