hi, i am currently using a handcompiled version 1.17.1 on fedora to play with the new clocksync element. to test cross-host syncing, i have three identical VMs, one acting as the "server", and two acting as "clients". currently, i am trying to get a basic setup to work with udpsink and udpsrc, although eventually, i would love to be able to tcp-stream vorbis. my issues start at the very beginning: the second i start introducing clocksync, things get extremely choppy. so this works fine: gst-launch-1.0 uridecodebin uri="http://[mp3 stream]" ! audioconvert ! queue ! alsasink while this: gst-launch-1.0 uridecodebin uri="http://[mp3 stream]" ! audioconvert ! queue ! clocksync ts-offset=100000 ! alsasink gets extremely choppy (for any value of ts-offset) what i really want is to have this: server [any source] -> [normalize] -> [vorbis enc] -> [tcp server sink] clients: [tcpclientsrc] -> [vorbis dec] -> [clocksync] -> [alsa] tcp or udp multicast doesnt really make a difference now - the point is more that i dont really have a good grasp of the model behind all this to easily build this. also, documentation on how to configure the clock is ... sparse? i would appreciate any simple examples and hints regarding the sync (which is, in my opinion, a really great addition). .rm _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
hi again, i have now come up with this: server: gst-launch-1.0 uridecodebin uri="[mp3 stream]" ! audioconvert ! lamemp3enc ! queue ! tcpserversink host=192.168.71.163 port=4999 client: gst-launch-1.0 tcpclientsrc host=192.168.71.163 port=4999 do-timestamp=true ! decodebin ! audioconvert ! queue ! clocksync ! alsasink sync=false ... and this gives me a playable stream, and sync is ok over a LAN between several VMs ... but that's pure luck and the fact that it's a virtual setup with no wifi or competing traffic involved. i have to say though that even if one client is off (e.g. in the beginning), it usually converges again after a second or two. some issues: - i wanted to use vorbis. not doing thet b/c it seems the problem with the format specification on the client side is not really solvable. - i am not using rtppay / depay here - mostly because i dont know which one to use on mp3 - the second i set "sync" to "true", it essentially stops working - it still plays short pieces every now and then, but no continuous stream i think i am probably making some mistakes. what's missing? .rm On Sat, Jun 27, 2020 at 5:08 PM ruben malchow <[hidden email]> wrote:
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hi again, i have now come up with this: server: gst-launch-1.0 uridecodebin uri="[mp3 stream]" ! audioconvert ! lamemp3enc ! queue ! tcpserversink host=192.168.71.163 port=4999 client: gst-launch-1.0 tcpclientsrc host=192.168.71.163 port=4999 do-timestamp=true ! decodebin ! audioconvert ! queue ! clocksync ! alsasink sync=false ... and this gives me a playable stream, and sync is ok over a LAN between several VMs ... but that's pure luck and the fact that it's a virtual setup with no wifi or competing traffic involved. i have to say though that even if one client is off (e.g. in the beginning), it usually converges again after a second or two. some issues: - i wanted to use vorbis. not doing thet b/c it seems the problem with the format specification on the client side is not really solvable. - i am not using rtppay / depay here - mostly because i dont know which one to use on mp3 - the second i set "sync" to "true", it essentially stops working - it still plays short pieces every now and then, but no continuous stream i think i am probably making some mistakes. what's missing? .rm On Sat, Jun 27, 2020 at 5:08 PM ruben malchow <[hidden email]> wrote:
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