OK, I'm not a great expert but I've read a bit about how these things work, so here are my (partial) definitions of those terms that are confusing you. Some of my answer is taken from the rec.audio.pro FAQ at http://recordist.com/rap-faq/current.

First, to get our knowledge base even, the DA conversion stage takes a clock signal, and on each clock the input from the digital side is converted to an analog voltage. How does it do this? Most converters work by storing chunks of electrical charge in capacitors and then emptying the appropriate capacitors into the output stream on the clock signal. CD audio is two 16-bit words at a rate of 44.1KHz (44100 samples per second).

Now, there are a couple of ways to do this conversion. One is to take your (sixteen-bit) lump of data as one chunk (sixteen inputs) and have sixteen capacitors with 1, 2, 4, 8, and so on up to 32,768 'units' of charge (where a unit is 1/4,096V - i.e the largest bucket holds 2 volts - all of them summed together produce 4 volts).

Now the largest capacitor in this series has to have a tolerance of about 0.01% so it doesn't end up holding 32,126 or 40,000 units of charge, which makes it rather expensive to manufacture. However, you can treat a stream of 16 bit chunks coming at you at 44.1KHz as a stream of single bits coming at you at 11.3MHz. This is what a 1-bit DAC does, essentially. I'm not exactly sure how it works from there - the FAQ doesn't say much - but I thought it worked like a kind of finite state machine, so that after sixteen clocks at 11.3MHz the signal would have 'bounced around' inside the chip enough to get the right voltage on the analog out. But I'll research this further.

Each clock cycle, the output is forced to be some arbitrary voltage, which may not (in the case of very loud percussive hits) be very close to the last voltage output. Don't forget here that it's a two-way street - whatever's on the end of the analog output has its own ideas of electrical conductance, capacitance and inductance. So the DAC does its best to drag the signal on its output stage to where it should be, which may make the signal jerk around a bit.

Think of it like an analog clock with hands on a ratchet mechanism - at 2:59 the hour hand is still pointing at the 2, and one minute later it's pointing to the 3. As the last second ticks over the clock mechanism has to move that hand fairly quickly, so it might jerk a bit when starting to move and shudder a bit before it comes to rest. This sort of thing does happen (in a very minute way) on the audio line.

Noise shaping is basically the process of filtering the jerks and shudders out of the signal that comes out, so it looks (relatively) smooth to the receiver of the analog signal. MGrant's explanation is much better than this, but I'm sticking to it because it may help the people who didn't read the R.A.P. FAQ.

Oversampling can refer to two things, here. Firstly, it could mean reading the bits off the CD a couple of times and working out your best guess of what they actually are. The audio is encoded with an error-correcting system here, which is designed not so much to eliminate the possibility of errors as to work out a good guess of what the value might have been when you do get a bad byte. By re-reading it a couple of times, the microscopic wobble in the CD that caused the phase change that caused the misread might have gone now.

On the other hand, it can mean that you try several different analog outputs of the same digital input and average them to work out the most likely original signal. After all, those capacitors can sometimes not be quite filled, or the clock might have been slightly skewed and have hit the output latch (the bit that tells the converter stage to send on its results to the output line) before the rest of the chip was ready. So oversampling can work here as well.

I'll get to the bottom of 1-bit DACs and get back to you.

Save the whales. Feed the hungry. Free the mallocs.

Edited by PaulWay on 6/9/00 02:52 AM.