Re: Problem with NDISRC, WHIPSINK, and audio
Tarun Tej K via gstreamer-devel <[email protected]>
| Newsgroups | gmane.comp.video.gstreamer.devel |
|---|---|
| Message-ID | <CAMvyGd8D4debBB+MbMvCk7Q4XeU39zP+f4z0V8vxmgGkUZtQSw@mail.gmail.com> |
> The same ndisrc / ndisrcdemux works fine into a webrtcsink pipeline with A+V We now have a whip signaller implementation in the webrtcsink - https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1168. But I don't think this is a part of a release yet, so you will have to set up an uninstalled version. On Sat, Jul 15, 2023 at 5:18 PM GST Developer <[email protected]> wrote: > > > > > On 15 Jul 2023, at 12:36, Tarun Tej K <[email protected]> wrote: > > > > Ok. I am not familiar with ndisrcdemux, does this mean the buffers are > > being dropped at the demuxer? > > 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. > > 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. > > > Also what is the GST_DEBUG value set in the environment > > in the log, it was *.7 I believe, the TL:DR was *.3 I think. > > > and gstreamer > > version you are using? > > 1.20.3 > > Thanks ! > > > > > > On Sat, Jul 15, 2023 at 4:45 PM GST Developer <[email protected]> wrote: > >> > >> besides the full log (second one) that i shared, http://www.gallery.co.uk/gstlog2.txt.zip > >> > >> TL:DR this appears to be the crux of the issue: > >> > >> Pipeline is live and does not need PREROLL ... > >> 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 > >> Pipeline is PREROLLED ... > >> Setting pipeline to PLAYING ... > >> New clock: GstSystemClock > >> Redistribute latency... > >> Redistribute latency... > >> Redistribute latency... > >> 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 > >> 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 > >> 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 > >> > >> > >> > >>> > >>> > >>> On Sat, 15 Jul, 2023, 03:30 GST Developer via gstreamer-devel, <[email protected]> wrote: > >>>> > >>>> Hi Folks. > >>>> > >>>> I am attempting use gstreamer to send content from an NDI Source to a WHIP end point at Dolby.io. > >>>> > >>>> If I just send VIDEO, its working fine: > >>>> > >>>> 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” > >>>> > >>>> This works FINE !! we get the signal to the WHIP server and the end to end latency is about 2 seconds. > >>>> > >>>> Now I want to add AUDIO to that pipeline, and I am trying: > >>>> > >>>> 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” > >>>> > >>>> Now, it starts up with: > >>>> > >>>> Setting pipeline to PAUSED ... > >>>> Pipeline is live and does not need PREROLL ... > >>>> Pipeline is PREROLLED ... > >>>> Setting pipeline to PLAYING ... > >>>> New clock: GstSystemClock > >>>> Redistribute latency... > >>>> Redistribute latency... > >>>> Redistribute latency… > >>>> > >>>> But it stalls right there, and never gets to counting time, and we dont see anything arrive at the WHIP server. > >>>> > >>>> Might anyone know what I am doing wrong ? > >>>> > >>>> Many thanks !!! > >>>> > >>>> > >>>> > >>> > >> >