Re: Problem with NDISRC, WHIPSINK, and audio
GST Developer via gstreamer-devel <[email protected]>
| Newsgroups | gmane.comp.video.gstreamer.devel |
|---|---|
| Message-ID | <[email protected]> |
Thanks for your reply. Sadly, this doesn't change the behaviour - when attempting to send video *and* audio, it stalls after “Redistribute latency…” So, Just video works: 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. whipsink name=whip auth-token=“…redacted….." whip-endpoint="https://director.millicast.com/api/whip/myStreamName” and If I try *just* audio: gst-launch-1.0 ndisrc ndi-name="NDIPE8 (SIGGEN)" ! ndisrcdemux name=demux demux.audio ! queue ! audioconvert ! opusenc ! rtpopuspay ! 'application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000,encoding-params=(string)2' ! whip. whipsink name=whip auth-token=“...redacted..." whip-endpoint="https://director.millicast.com/api/whip/myStreamName” This also works ! but when I try to combine them : 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. demux.audio ! queue ! audioconvert ! opusenc ! rtpopuspay ! 'application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000,encoding-params=(string)2' ! whipsink name=whip auth-token="…redacted….." whip-endpoint="https://director.millicast.com/api/whip/myStreamName” it stalls after “Redistribute latency…” This does feel like some sort of demux deadlock between the ndisrcdemux and the whipsink. I have pasted the debug output for the stall loop below, which repeats over and over: 0:00:19.425157202 9470 0x557d29708a00 TRACE structure gststructure.c:292:gst_structure_new_id_empty_with_size: created structure 0x557d298c1aa0 0:00:19.425181034 9470 0x557d29708a00 DEBUG structure gststructure.c:2258:gst_structure_parse_field: trying field name 'long-name' 0:00:19.433174494 9470 0x557d29708a00 DEBUG default gstvalue.c:2776:_priv_gst_value_parse_value: trying type name 'string' 0:00:19.433194340 9470 0x557d29708a00 DEBUG structure gststructure.c:2258:gst_structure_parse_field: trying field name 'klass' 0:00:19.433204782 9470 0x557d29708a00 DEBUG default gstvalue.c:2776:_priv_gst_value_parse_value: trying type name 'string' 0:00:19.433216613 9470 0x557d29708a00 DEBUG structure gststructure.c:2258:gst_structure_parse_field: trying field name 'description' 0:00:19.433223154 9470 0x557d29708a00 DEBUG default gstvalue.c:2776:_priv_gst_value_parse_value: trying type name 'string' 0:00:19.433232379 9470 0x557d29708a00 DEBUG structure gststructure.c:2258:gst_structure_parse_field: trying field name 'author' 0:00:19.433246113 9470 0x557d29708a00 DEBUG default gstvalue.c:2776:_priv_gst_value_parse_value: trying type name 'string' 0:00:19.433254115 9470 0x557d29708a00 DEBUG GST_REGISTRY gstregistrychunks.c:627:gst_registry_chunks_load_feature: Element factory : npadtemplates=2 0:00:19.433261379 9470 0x557d29708a00 DEBUG GST_REGISTRY gstregistrychunks.c:526:gst_registry_chunks_load_pad_template: Reading/casting for GstRegistryChunkPadTemplate at address 0x7f3208f72f30 0:00:19.433271543 9470 0x557d29708a00 DEBUG GST_REGISTRY gstregistrychunks.c:540:gst_registry_chunks_load_pad_template: Added pad_template src 0:00:19.433279923 9470 0x557d29708a00 DEBUG GST_REGISTRY gstregistrychunks.c:526:gst_registry_chunks_load_pad_template: Reading/casting for GstRegistryChunkPadTemplate at address 0x7f3208f72fb0 0:00:19.433289253 9470 0x557d29708a00 DEBUG GST_REGISTRY gstregistrychunks.c:540:gst_registry_chunks_load_pad_template: Added pad_template sink 0:00:19.433302373 9470 0x557d29708a00 DEBUG GST_REGISTRY gstregistry.c:592:gst_registry_add_feature:<registry0> adding feature 0x557d298bf6d0 (avdec_wmavoice) 0:00:19.433311993 9470 0x557d29708a00 DEBUG GST_REFCOUNTING gstobject.c:708:gst_object_set_parent:<avdec_wmavoice> set parent (ref and sink) 0:00:19.433322596 9470 0x557d29708a00 TRACE GST_REFCOUNTING gstobject.c:292:gst_object_ref_sink:<avdec_wmavoice> 0x557d298bf6d0 ref_sink 1->2 0:00:19.433330585 9470 0x557d29708a00 LOG GST_REGISTRY gstregistry.c:610:gst_registry_add_feature:<registry0> emitting feature-added for avdec_wmavoice 0:00:19.433341108 9470 0x557d29708a00 DEBUG GST_REGISTRY gstregistrychunks.c:741:gst_registry_chunks_load_feature: Added feature avdec_wmavoice, plugin 0x557d298724d0 libav 0:00:19.433350373 9470 0x557d29708a00 DEBUG GST_REGISTRY gstregistrychunks.c:583:gst_registry_chunks_load_feature: Plugin 'libav' feature 'avdec_ws_snd1' typename : 'GstElementFactory' 0:00:19.433362464 9470 0x557d29708a00 TRACE GST_REFCOUNTING gstobject.c:208:gst_object_init:<GstObject@0x557d298c2050> 0x557d298c2050 new 0:00:19.433374983 9470 0x557d29708a00 LOG GST_REGISTRY gstregistrychunks.c:610:gst_registry_chunks_load_feature: Reading/casting for GstRegistryChunkElementFactory at address 0x7f3208f72ff0 0:00:19.433385348 9470 0x557d29708a00 TRACE structure gststructure.c:292:gst_structure_new_id_empty_with_size: created structure 0x557d298c1e40 0:00:19.433394095 9470 0x557d29708a00 DEBUG structure gststructure.c:2258:gst_structure_parse_field: trying field name 'long-name' 0:00:19.433402420 9470 0x557d29708a00 DEBUG default gstvalue.c:2776:_priv_gst_value_parse_value: trying type name 'string' 0:00:19.433411420 9470 0x557d29708a00 DEBUG structure gststructure.c:2258:gst_structure_parse_field: trying field name 'klass' 0:00:19.433419783 9470 0x557d29708a00 DEBUG default gstvalue.c:2776:_priv_gst_value_parse_value: trying type name 'string' > On 15 Jul 2023, at 01:17, Tarun Tej K <[email protected]> wrote: > > Hi, > > What is the idea behind fixing the sink pad number i.e., sink_0 to video and sink_1 to audio ? The pads would be linked dynamically anyway. > > Can you try the below modified pipeline. I have also added a queue in audio branch after demux. > > 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. demux.audio ! queue ! audioconvert ! opusenc ! rtpopuspay ! 'application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000,encoding-params=(string)2' ! whipsink name=whip auth-token="…redacted….." whip-endpoint="https://director.millicast.com/api/whip/myStreamName” > > > On Sat, 15 Jul, 2023, 03:30 GST Developer via gstreamer-devel, <[email protected] <mailto:[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 !!! >> >> >>