Sampling with the AWE - asking for trouble

For those who think that 16 bits always are 16 bits

Written by Mathias C. Hjelt
V1.1 13 Jun 95

Introduction

To make a long story short enough for being written in less than an hour, let me put it like this: 16 bits at 44.1kHz does not automatically mean you've got CD-quality, no matter what the PC sound card manufacturers and advertisers say.

This stuff deals with things like why 16 bits are not always 16, and why you don't have to worry about Nyquist's theorem when sampling with the AWE. Note: this is only about the ADC, and has got nothing to do with the EMU and the rest of the AWE.

Frequency response

Nyquist's theorem says that the sampling frequency must be at least twice the highest frequency of the signal that is to be digitized / reproduced. Since the AWE's maximum sampling frequency is 45.45kHz, the highest frequency it can sample/play should be close to 22.72kHz. Unfortunately, there are things like cheap AD converters and way too efficient low-pass filters which make sure these frequencies never get far.

Unfortunately I don't have any fancy frequency response graphs or exact figures to show here. Many computer mags that have tested the AWE have reported that high end is being attenuated pretty much above 15kHz, but I doubt that. My own brief tests showed that an 18kHz signal is somewhat attenuated, but definitely not down by 12dB. However, if you're looking for the ultimate high-end, this card won't be of much use.

The low end (bass, that is) is also mistreated. Somewhere between 20 and 50 Hz there's a slope that chomps up several dBs. Below that the level is pretty constant at a few -dB. However, these frequencies are seldom taken into account in "real" tests (in computer mags, that is) since 50Hz is usually where the graphs start.

In any case, the playback side of the AWE has got much better frequency response than this darned recording part. But how bad is this? Does it make the card useless? There are - as always - two answers: yes and no. For most users, no, for the audiophiles and other serious dudes, sort of. If you just want to sample your own sounds and use them with the AWE's EMU, the recording quality will be sufficient, and most regular music applications will work fine. But if you want to do real accurate signal processing, research etc, you'd probably want a much cleaner signal and a much better frequency response. But that's not what the AWE is all about, so why feel sad?

Noise fighting - an impossible but not useless task

When sampling with the AWE, make sure you use the line input, never the mic input, and see to it that you always use recording gain x1. This does of course mean that you can't sample right off a mic, and that you'll run into trouble when sampling stuff that has got rather weak signal level, but instead you'll be rewarded with as good SNR (signal-to-noise ratio) as possible. Note that "as good as" is not equal to "good".

If you want to sample something using a mike or some other low-level audio device, you need to use an external pre-amplifier to be able to get a sufficiently high signal level for the AWE's line input. It is general knowledge that you always should use as much as possible of a sample's amplitude range (but yet not have the peaks clipped), and this is why you need to make sure you've got enough volume on the input. A suitable pre-amp may be a mixer or even a guitar pre-amp. Anything that can amplify weak signals without adding noise will do fine.

Why not use the mic input? Forget it - it's flooded with noise. Why not increase the recording gain factor to x2 or x4? Now that's a better question. The problem is that when you increase the gain, you also increase the background noise level. With background noise, I mean the crap that you get when you sample stuff without any inputs enabled, or alternatively with the inputs enabled but tied to ground. I did these tests using both methods, running at 44.1kHz, 16 bits, mono:

Gain x1: Gain x2: Gain x4: Quite embarrasing, isn't it? The only (poor) consolation we've got is that the noise level of gain x1 won't bother that much when it is played through the EMU. It is, in fact, left out almost completely. For an explanation of that, check out the page about why the EMU isn't pro.

In addition to all this, the ADC has got a rather stupid DC offset error, which means that a zero signal does not result in a zero sample. The offset isn't even the same on the left and the right channel when doing stereo sampling. An offset error like this results in asymmetric clipping when the inputs are overloaded, and in silent pops at the beginning & end of the sample when it is played.