Need help on achieving REALLY low latency!

RyanDK · 14th June 2002, 21:25

I'm having a kinda urgent issue with the DNAS-buffer.

I work at a radiostation where I've been looking at the possibility of using a Winamp/Shoutcast solution as a way of transmitting audio from anywhere in our community back to the studio as close to real-time as possible.

My plan A was using a HSCSD-cell phone on a direct line to a Shoutcast server at the station house, where it'd redirect the sound to a WinAmp player (or similar) in the Air studio where the feed would go On-Air.

The problem with the latency - well... it's not easy to have a chat with someone On Air who's approx. 90 seconds behind in audio stream, right?

So...! Is there ANY way at all to reduce the buffer setting in the DNAS or other tweaks that'd allow me to have a window of max. 3 seconds of latency?
I know it'd kill the QoS idea, but since it's a direct line with plenty of bandwidth, I thought I'd deal with that as I went along. :-)

Thanks in advance for any help you can give!
Ryan.

Jay · 14th June 2002, 22:13

there is no way to adjust the buffer in the server.

DJ AmPs · 15th June 2002, 17:29

use the voice chat feature built into MSM messenger or Netmeeting if u want to do real-time on the 'net. No point in re-inventing the wheel. Latency is 0.5 - 1 second. depending on connection speed.

-amps

ZwiebelWurst · 4th July 2002, 17:09

there is a way to reduce the buffer, simply use shoutcast 1.6.0, delay will be approx. 2 sec.

shoutcast-1-6-0

RyanDK · 4th July 2002, 21:30

Without fully testing it, I'd say that both NetMeeting and any other "phone"-appliance over the net is out of the question. They all (as far as I've seen) use ADPCM, a-law or u-law compression which doesn't sound good enough at the bitrate I have available. (43 kbps at the absolute maximum incl. flowbits)

Thanks for the ShoutCast 1.6.0 server. I'll be sure to try it out! :-D

So far I've grown quite fond of the OggVorbis/icecast2 constellation. Now I'll see if the older ShoutCast server can do as well as the vorbis setup. ;-)

R.

14th June 2002, 21:25	#1
RyanDK Junior Member Join Date: Jun 2002 Location: Denmark Posts: 4	Need help on achieving REALLY low latency! I'm having a kinda urgent issue with the DNAS-buffer. I work at a radiostation where I've been looking at the possibility of using a Winamp/Shoutcast solution as a way of transmitting audio from anywhere in our community back to the studio as close to real-time as possible. My plan A was using a HSCSD-cell phone on a direct line to a Shoutcast server at the station house, where it'd redirect the sound to a WinAmp player (or similar) in the Air studio where the feed would go On-Air. The problem with the latency - well... it's not easy to have a chat with someone On Air who's approx. 90 seconds behind in audio stream, right? So...! Is there ANY way at all to reduce the buffer setting in the DNAS or other tweaks that'd allow me to have a window of max. 3 seconds of latency? I know it'd kill the QoS idea, but since it's a direct line with plenty of bandwidth, I thought I'd deal with that as I went along. :-) Thanks in advance for any help you can give! Ryan.

14th June 2002, 22:13	#2
Jay Moderator Alumni Join Date: May 2000 Location: Next Door Posts: 8,942	there is no way to adjust the buffer in the server. -Jay \| Radio Toolbox.com

15th June 2002, 17:29	#3
DJ AmPs Major Dude Join Date: Mar 2002 Location: g Posts: 1,603	use the voice chat feature built into MSM messenger or Netmeeting if u want to do real-time on the 'net. No point in re-inventing the wheel. Latency is 0.5 - 1 second. depending on connection speed. -amps BB 192k Mp3 HIGH DEFINITION \| BB 24k Mp3 \| BB 20k WMA \| BB Community __________________________ My Host - Fast-Serv.com \| Free Shoutcast Hosting - Coming Soon...

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4th July 2002, 17:09	#4
ZwiebelWurst Junior Member Join Date: Jul 2002 Posts: 3	there is a way to reduce the buffer, simply use shoutcast 1.6.0, delay will be approx. 2 sec. shoutcast-1-6-0

4th July 2002, 21:30	#5
RyanDK Junior Member Join Date: Jun 2002 Location: Denmark Posts: 4	Without fully testing it, I'd say that both NetMeeting and any other "phone"-appliance over the net is out of the question. They all (as far as I've seen) use ADPCM, a-law or u-law compression which doesn't sound good enough at the bitrate I have available. (43 kbps at the absolute maximum incl. flowbits) Thanks for the ShoutCast 1.6.0 server. I'll be sure to try it out! :-D So far I've grown quite fond of the OggVorbis/icecast2 constellation. Now I'll see if the older ShoutCast server can do as well as the vorbis setup. ;-) R.