Problem with NDISRC, WHIPSINK, and audio

Sat Jul 15 14:23:30 UTC 2023

Thanks Tarun,

we are making progress - this pipeline does startup, but it fails after a few seconds:

gst-launch-1.0  ndisrc ndi-name="NDIPE8 (SIGGEN)" ! ndisrcdemux  name=demux demux.video ! queue ! videoconvert ! whipwebrtcsink name=ws signaller::whip-endpoint="https://director.millicast.com/api/whip/myStreamName",auth-token=“..redacted.." demux.audio ! queue ! audioconvert ! ws.

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…

<it runs fine here for a few seconds with the numbers running>

thread 'tokio-runtime-worker' panicked at 'Signal 'request-aux-sender' of type 'GstWebRTCBin' not found', /home/ubuntu/.cargo/git/checkouts/gtk-rs-core-7be42ca38bd6361c/44c4ab3/glib/src/object.rs:2642:32
note: run with `RUST_BACKTRACE=1` environment variable to display a backtrace
thread '<unnamed>' panicked at 'called `Result::unwrap()` on an `Err` value: PoisonError { .. }', net/webrtc/src/webrtcsink/imp.rs:3013:47
ERROR: from element /GstPipeline:pipeline0/GstWhipWebRTCSink:ws: Panicked: called `Result::unwrap()` on an `Err` value: PoisonError { .. }
thread '<unnamed>' panicked at 'called `Result::unwrap()` on an `Err` value: PoisonError { .. }', net/webrtc/src/webrtcsink/imp.rs:3013Execution ended after 0:00:01.886767934
:47Setting pipeline to NULL ...

ERROR: from element /GstPipeline:pipeline0/GstWhipWebRTCSink:ws: Panicked
ERROR: from element /GstPipeline:pipeline0/GstWhipWebRTCSink:ws: Panicked
ERROR: from element /GstPipeline:pipeline0/GstWhipWebRTCSink:ws: Panicked: called `Result::unwrap()` on an `Err` value: PoisonError { .. }
ERROR: from element /GstPipeline:pipeline0/GstWhipWebRTCSink:ws: Panicked
Freeing pipeline …

> On 15 Jul 2023, at 14:47, Tarun Tej K <tarun4690 at gmail.com> wrote:
> 
>> auth-token=“…redacted…."
> 
>> which doesnt appear to be a supported parameter in whipwebrtcsink so perhaps I cant use that in place of whipsink
> whipwebrtcsink does have 'auth-token' property.
> https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/blob/main/net/webrtc/src/whip_signaller/imp.rs#L577
> 
> Did you try setting it like this `whipwebrtcsink signaller::auth-token="wxyz"` ?
> 
> On Sat, Jul 15, 2023 at 6:44 PM GST Developer <gstreamer at gallery.co.uk> wrote:
>> 
>> Thanks Tarun
>> 
>> The Dolby.io workflow also requires
>> 
>> auth-token=“…redacted…."
>> 
>> which doesnt appear to be a supported parameter in whipwebrtcsink so perhaps I cant use that in place of whipsink
>> 
>> On 15 Jul 2023, at 13:53, Tarun Tej K <tarun4690 at gmail.com> wrote:
>> 
>> 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 <gstreamer at gallery.co.uk> wrote:
>> 
>> 
>> 
>> 
>> On 15 Jul 2023, at 12:36, Tarun Tej K <tarun4690 at gmail.com> 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 <gstreamer at gallery.co.uk> 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, <gstreamer-devel at lists.freedesktop.org> 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 !!!
>> 
>> 
>> 
>> 
>> 
>> 
>> 

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