I'm in the market for a new soundcard, usb. I was reading reviews of a few cards I thought was interesting,
many people complained about latency, now, I'm not really sure they think of the same type of latency as me.

When I start up a DAW, hook up my keyboard, choose sample frequency, resulution, latency in samples, my DAW
tells me "you got xx ms latency there" ok.

Is there any way to know how low the latency can be in ms at sample freq xx when I have these figures?

RMAA Measurements

Device: ASIO E-MU 0202 | USB

Features:
Input channels: 2
Output channels: 2
Input latency: 2138
Output latency: 2714
Min buffer size: 384
Max buffer size: 96000
Preferred buffer size: 1920
Granularity: 192
ASIOOutputReady - supported



Device: ASIO E-MU 0404 | USB

Features:
Input channels: 4
Output channels: 4
Input latency: 22083
Output latency: 22216
Min buffer size: 88
Max buffer size: 22000
Preferred buffer size: 22000
Granularity: 44
ASIOOutputReady - supported



numbers found here http://ixbtlabs.com/sound.html#subsec110

Can latency be calculated with this?
If not, what determines latency (and can anyone guess what I would be with these cards @ 16/44,1 and 24/48)?

What I mean is latency from when I push a key to when I hear sound. I hope you understand.

If you're using a MIDI connection from the keyboard to your computer and sound is being generated by software, there may be a computer/software delay in addition to the soundcard/buffer delay.

If you are running audio from your keyboard into your computer, you should be able to monitor the audio directly (one way of another) and delay in the digital path shouldn't matter.


I'm not sure, but it appears to be listed as the number of samples... And, it appears to be related to "preferred buffer size". Of course, a buffer IS a delay and to minimize delay you have to minimize the buffer size.

If I'm right and latency is given as a number of samples, a 22,000 sample latency at 44.1kHz is about half a second!!!!

If you reduce the buffer size too much, the output buffer (playback) can "run dry" (buffer underrun), or your input buffer (recording) can overflow and you'll get a glitch. It depends on how "busy" your CPU & databus are... If you're simply recording or playing back and the computer isn't doing anything else, you can get-away with a small buffer. (But, if you're simply recording or playing back, you are not even aware of the delay.) If you are recording and playing back at the same, while doing effects-processing, surfing the Net, etc., you need bigger buffers.

Some E-Mu devices have "zero latency" monitoring which means the sound is routed in-and-out of the device without passing through the computer (for monitoring purposes).

DAWs usually have delay compensation to help keep everything in sync, but this does NOT help when monitoring yourself in real time.

No problems to playback (from NI Beatport SYNC) using 1.5 ms on ASIO mode ... no issues even while surfing the web.

Recording instrument (say guitar) wet with VST effect (Amplitube) can be done using 5-10 ms ("software monitoring").

These results from a P4/2,67GHz/XP SP3 system ... (Windows 7 was "too heavy" system to get good enough results from E-MU 0404 USB in this old PC ... ).


jiitee