RadioBoss Latency loop back issues

tunnelvision

New member
Here is my setup

Berhinger X1622USB Mixer ( note note using usb running completely analog)

Mics in channel 1 & 2 sending direct signal (subgrounp) to line in on sound card

output of sound card going in to two channels for playback

I have the mic input setup in radioboss set to us my line in on my sound card. When I press the mic button, the volume of the music drops like it should but when I speak into the mic I literally am getting close to 1 sec of latency when monitoring my show. This makes it very hard to announce anything on the radio.

I am using directsound in the mic/line settings

Need some help with this
 
Latency depends mostly on playback buffer - you should decrease it in the playback settings.
Also, is it possible to switch to ASIO/WASAPI playback and recording?
 
we tried the usb connection via the mixer using asio drivers and still get delay.

I have a M-Audio AudioPhile 2496 Soundcard coming tomorrow which is a much better sound card then the one on our motherboard. According to M-Audio this soundcard has zero latency or I would say minimal compared to what we are using.

So here is my setup please correct me if I am doing this wrong

Berhinger 1622USB Mixer
using channels 1 & 2 for microphones - routing direct signal from those two out the sub 1 &2 outputs of the mixer to the line input on the m-audio card

we need to be able to monitor what we are saying over the air and music at the same time so I took the output of the soundcard and put them back into the berhinger mixer's 5 & 6 inputs that are feeding the main output of the mixer to our transmitter.

My only issue that I can see happening is a audio loopback when the mic is on.

is this the right setup
 
even with a high quality sound card still voice latency - if I adjust the output buffers I get skipping music, sometime music skips and plays faster then it original format and still after all of this time still not monitoring in realtime or even close to it.
 
tunnelvision said:
even with a high quality sound card still voice latency - if I adjust the output buffers I get skipping music, sometime music skips and plays faster then it original format and still after all of this time still not monitoring in realtime or even close to it.
It depends not only on sound card. Please see this topic on how to reduce latency to minimum: http://www.djsoft.net/smf/index.php/topic,762.0.html
M-Audio card supports ASIO - you should use it.
 
I think it's impossible to hear 1ms delay :)
Note, that Breakaway processing adding some delay to the sound...
Also, adjust buffer size in the ASIO Control panel - in RadioBOSS, in the Settings->Playback->Auxiliary outputs select your device and click "ASIO CP" button. In the window appeared change the buffer size.
 
I removed the breakaway software completely and made the changes you suggested.

I changed the buffer size to 1024 and the latency is now better however I am without my breakaway broadcasting software.

So instead of having the breakaway software running I moved the output of my m-audio card to 2 channels on my mixer that have great eq and compression thus eliminating the breakaway software.

Too bad all won't play nicely with eachother.

Our transmitter we are going to be using has built in compression and limiting that should tighten up our sound over the radio however out streaming sound is suffering
 
I'm sure there's a way to decrease latency created by Breakaway to minimum. You should contact their support about it... My knowledge of Breakaway is very limited :)
You may also try to further decrease ASIO latency. To 512 samples, for example...
 
actually for your information if you add any plug-in to RadioBoss is causes latency.

I downloaded the full version of Audio Proc from audioproc.ca and the latency is back again if I run it.

So at this point I am completely at a loss
 
The latency added by plugin depends on how much processing it is doing (and how complex processing is). RadioBOSS can't do anything to decrease it...
 
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