<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body style="overflow-wrap: break-word; -webkit-nbsp-mode: space; line-break: after-white-space;">Thanks Tarun<div><br></div><div>The <a href="http://Dolby.io">Dolby.io</a> workflow also requires </div><div><br></div><div>auth-token=“…redacted…."</div><div><br></div><div>which doesnt appear to be a supported parameter in whipwebrtcsink so perhaps I cant use that in place of whipsink<br>
<div><br><blockquote type="cite"><div>On 15 Jul 2023, at 13:53, Tarun Tej K <tarun4690@gmail.com> wrote:</div><br class="Apple-interchange-newline"><div><div><blockquote type="cite">The same ndisrc / ndisrcdemux works fine into a webrtcsink pipeline with A+V<br></blockquote>We now have a whip signaller implementation in the webrtcsink -<br>https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1168.<br>But I don't think this is a part of a release yet, so you will have to<br>set up an uninstalled version.<br><br>On Sat, Jul 15, 2023 at 5:18 PM GST Developer <gstreamer@gallery.co.uk> wrote:<br><blockquote type="cite"><br><br><br><blockquote type="cite">On 15 Jul 2023, at 12:36, Tarun Tej K <tarun4690@gmail.com> wrote:<br><br>Ok. I am not familiar with ndisrcdemux, does this mean the buffers are<br>being dropped at the demuxer?<br></blockquote><br>This also new to me, but its certainly warning that the input buffer is filling, presumably because its not draining. I did try with a much larger receive buffer but it does the same thing, so this does look like some sort of deadlock.<br><br>The same ndisrc / ndisrcdemux works fine into a webrtcsink pipeline with A+V, so it looks like some quirk where it doesnt like whipsink in A+V workflow. I can use gst videotestsrc and audiotestsrc into whipsink fine, so I know there is nothing fundamentally wrong with either of the components, ndisrc, ndisrcdemux and whipsink.<br><br><blockquote type="cite">Also what is the GST_DEBUG value set in the environment<br></blockquote><br>in the log, it was *.7 I believe, the TL:DR was *.3 I think.<br><br><blockquote type="cite">and gstreamer<br>version you are using?<br></blockquote><br>1.20.3<br><br>Thanks !<br><br><br><blockquote type="cite"><br>On Sat, Jul 15, 2023 at 4:45 PM GST Developer <gstreamer@gallery.co.uk> wrote:<br><blockquote type="cite"><br>besides the full log (second one) that i shared, http://www.gallery.co.uk/gstlog2.txt.zip<br><br>TL:DR this appears to be the crux of the issue:<br><br>Pipeline is live and does not need PREROLL ...<br>0:00:00.036747546 11562 0x558781cb5400 FIXME default gstutils.c:4025:gst_pad_create_stream_id_internal:<ndisrc0:src> Creating random stream-id, consider implementing a deterministic way of creating a stream-id<br>Pipeline is PREROLLED ...<br>Setting pipeline to PLAYING ...<br>New clock: GstSystemClock<br>Redistribute latency...<br>Redistribute latency...<br>Redistribute latency...<br>0:00:01.220267971 11562 0x7fe028007330 WARN ndireceiver net/ndi/src/ndisrc/receiver.rs:850:gstndi::ndisrc::receiver::Receiver::receive_thread:<ndisrc0> Dropping old buffer -- queue has 11 items<br>0:00:01.232472921 11562 0x7fe028007330 WARN ndireceiver net/ndi/src/ndisrc/receiver.rs:850:gstndi::ndisrc::receiver::Receiver::receive_thread:<ndisrc0> Dropping old buffer -- queue has 11 items<br>0:00:01.239202906 11562 0x7fe028007330 WARN ndireceiver net/ndi/src/ndisrc/receiver.rs:850:gstndi::ndisrc::receiver::Receiver::receive_thread:<ndisrc0> Dropping old buffer -- queue has 11 items<br><br><br><br><blockquote type="cite"><br><br>On Sat, 15 Jul, 2023, 03:30 GST Developer via gstreamer-devel, <gstreamer-devel@lists.freedesktop.org> wrote:<br><blockquote type="cite"><br>Hi Folks.<br><br>I am attempting use gstreamer to send content from an NDI Source to a WHIP end point at Dolby.io.<br><br>If I just send VIDEO, its working fine:<br><br>gst-launch-1.0 ndisrc ndi-name="NDIPE8 (SIGGEN)" ! ndisrcdemux name=demux demux.video ! queue ! videoconvert ! x264enc ! video/x-h264,format=byte-stream,profile=baseline ! rtph264pay ! 'application/x-rtp,media=video,encoding-name=H264,payload=97,clock-rate=90000' ! whip.sink_0 whipsink name=whip auth-token=“…redacted….." whip-endpoint="https://director.millicast.com/api/whip/myStreamName”<br><br>This works FINE !! we get the signal to the WHIP server and the end to end latency is about 2 seconds.<br><br>Now I want to add AUDIO to that pipeline, and I am trying:<br><br>gst-launch-1.0 ndisrc ndi-name="NDIPE8 (SIGGEN)" ! ndisrcdemux name=demux demux.video ! queue ! videoconvert ! x264enc ! video/x-h264,format=byte-stream,profile=baseline ! rtph264pay ! 'application/x-rtp,media=video,encoding-name=H264,payload=97,clock-rate=90000' ! whip.sink_0 demux.audio ! audioconvert ! opusenc ! rtpopuspay ! 'application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000,encoding-params=(string)2' ! whip.sink_1 whipsink name=whip auth-token="…redacted….." whip-endpoint="https://director.millicast.com/api/whip/myStreamName”<br><br>Now, it starts up with:<br><br>Setting pipeline to PAUSED ...<br>Pipeline is live and does not need PREROLL ...<br>Pipeline is PREROLLED ...<br>Setting pipeline to PLAYING ...<br>New clock: GstSystemClock<br>Redistribute latency...<br>Redistribute latency...<br>Redistribute latency…<br><br>But it stalls right there, and never gets to counting time, and we dont see anything arrive at the WHIP server.<br><br>Might anyone know what I am doing wrong ?<br><br>Many thanks !!!<br><br><br><br></blockquote><br></blockquote><br></blockquote></blockquote><br></blockquote></div></div></blockquote></div><br></div></body></html>