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From: "Remco Stoutjesdijk" ¡ Ola ! [ BE AWARE: The following is not about tubes, but about
solid state and even digital stuff. If you can't take it, I strongly suggest you
hit the 'del' button... But I found at Arhus that many people are not fully up
to speed on this while (IMHO) it is very useful to know at least a bit about
digital stuff, in this world of SACD and DVD... ] > It was built in 1992, works very well, and has some
nice features. Yumm, features :) I remember buying 'My First Sony' cd player 'cuz it had so
many gimmicks. Used them for a week, after that play, stop, nxt, prev would do
:-) > Instruction booklet says that the DAC is a "MASH
(1bit)". Ok, now for the science part, pay attention... I'll start
at the very beginning. There are basically two ways of digitally describing audio: a/
By mapping the dynamic range onto a set of numbers. This is called PCM
(Pulse Code Modulation). The total amount of numbers defines the amplitude
resolution, while the amount of samples taken per second defines the maximum
digitizeable frequency (Nyquist criterion: Fmax = Fs/2). We all know the
standard 16bits x 44 kHz by now, but 24 x 96 kHz and 24 x 192 kHz (DVD audio)
are also PCM. Just versions with a larger resolution and frequency range. b/
By describing only the _changes_ in the signal with respect to its last
known value. This is called Delta Modulation. It's actually older than PCM
(invented in 1943) but it took until the end of the '80s until it was actually
implementable. The resolution and frequency range both depend on both the number
of bits used and the number of samples per second, since it's a non-linear
system (see below). As we all know, by the end of the '80s - begin '90s, we
started to see the 'bitstream', '1-bit', 'MASH' and other converters. These are
all forms of Delta Modulation converters (see below). Now why did all the companies just switch over all of a
sudden? Well, apart from the fact that you need something new every
couple of years to keep selling stuff, there was actually a real reason, too.
This has a lot to do with IC technology and computers. The IC-technologies of the '70s and '80s were based on both
bipolar and mosfet devices and had relatively large devices and voltages (10
micrometers is *incredibly huge* for todays standards, while I've also heard
people talk about high voltage supplies when they were talking about 5 Volts).
In these technologies relatively good analog devices were possible, there was
enough dynamic range to allow for some 100 dB of S/N. When the computer industry arose, new CMOS technologies
appeared with ever smaller devices and ever higher switching speeds. Now those
devices are *pure shit* for an analog designer. If you look at the curves of a
.18 micron transistor you'll see that never in this world this thing will make a
nice analog performance. Furthermore, when you make two identical transistors
(or resistors, capacitors), the matching is so poor that linearity is hardly
present. Building a DAC for PCM becomes hard here, you can see why:
The analog output of a PCM DAC is equal to 1*MSB + 1/2*(MSB-1) + 1/4 * (MSB -2)
+ ...+ 1/65536*LSB. This means that the accuracy of the current sources (or
resistors) in the DAC needs to be within 1/65536 tolerance. No need to tell that
is *hard*. Not impossible, Burr-Brown is capable of achieving even 20 bit
tolerance (1/1048576 !!!) by laser-trimming their resistors, but that procedure
makes an IC very expensive. So, what's a chap to do? Well, exploit the merits of the
technology, which is speed. The idea of the 1-bit DAC is to recalculate the 16
bit words into 1-bit words. You can see that you need at least 16 1-bit words to
keep the same amount of information so the sampling frequency goes up by 16 at
least. Now how do you do that, you can't simply cut off the last 15 bits... Here we use what is called a Sigma-Delta modulator. It
consists of a truncator, i.e. a 1-bit comparator (in the 1-bit case, which is
easiest and most abundant), a delay and a subtractor. The first mulit-bit-word
comes in and is truncated to 1 bit in the comparator. The comparator actually
decides if the word is smaller or larger than 1/2 of the dynamic range. The rest
of the word is put through the delay block and subtracted from the second
incoming word. After the subtraction, the second word is also truncated and the
rest of this word goes through the delay block again, and so on. A picture makes
it somewhat clearer, but I suck at ascii-art:
+ Now the result is a 1-bit stream (hence the name, you guessed it :) which describes the audio signal in terms of differences with respect to the previous sample. This is called a first order sigma delta modulator. Still paying attention? Then you probably wonder WHY the
heck this is all necessary, well, here's the reason: the delay in the discrete
time (z) domain is analogous to an time-derivative in the continuous
time-domain. But it's in the feedback path so the output is the integral of the
original signal. In order to re-construct the original analog output whe only
need to take the time-derivative of this signal. Hence the name sigma delta: the
sigma is the sum part (or subtractor part), the delta is the delay in the
feedback. Why is this tricky feedback part needed? Well, the
comparator is only 1 bit, and thus generates huge amounts of quantization
(=white!) noise (1/12 = -21 dB). By adding the delta (time-derivative) feedback
loop the loop gain is very high for low frequencies and low for high
frequencies. The quantization noise of the comparator is suppressed by the loop
gain. This means at low frequencies there is very little noise, while at higher
frequencies there is a lot. The noise is shaped! Now you get the name 'noise
shaper'! In frequency domain terms: The sigma-delta signal contains
the audio band, then a gap all the way to the sampling frequency (which is at
least 16*44100), then a lot of quantization noise. What's the simplest DAC then?
Yup, a low-pass filter. That's all. What do we have now? We have a DAC that's hardly process
dependant: there's no devices in need of matching. There's no devices which rely
on their analog performance. This has just been a number-crunching experience.
What do we need? Speed, and we have plenty of that. Most Sigma-Delta DACs run at
64*Fs, that is 2.8Mhz. Not a problem *at all* for CMOS (they made the Pentium I
in the same process as current DACs, so it can run at hundreds of Mhz if
needed). So, now we fully exploit the merits of the current IC processes while
we still get a good audio performance (THD+n > 120 dB is possible!). Well, when people got this working they of course wanted
even better versions so they started playing around with the scheme from the
picture. Philips and Sony experimented with putting two or more of these loops
in cascade. Then you get a 2nd, 3rd etc. order sigma delta modulator. Works, but
it's susceptible to (digital) instability, called 'limit cycles'. On the first
1-bitters you can hear small tones when there's no music playing. This is a
limit cycle close to the clock frequency which has created an audible
subharmonic! Technics also played with the scheme but added loops
besides and under the first loop, then summing the results, essentially making a
higher order loop without the danger of instability. They called it Multi stAge
noiSe sHaping (MASH). I'll spare you the details, but pure technically it's not
1-bit anymore but something like 3.5 bits... The 4DAC means there are 2 DAC chips per channel operating in differential mode (1 DAC gets the MSB inverted), so their common-mode errors will cancel. And there you have it! Still awake? Well, then I'll rave on for a bit more: There is virtually
no DAC available anymore today (not even by Burr-Brown, except for old stock)
that is not a Sigma-Delta DAC. That has led to SACD: Sony and Philips have
created a format (Direct Stream Digital, if you paid attention you now also know
what this term means) where the sigma-delta signal is stored immediately,
instead of taking the long road: sigma/delta A/D conversion -> recalculation
to PCM -> storage -> recalculation to sigma/delta -> analog. There is of course something to be said for this approach, instead of extending PCM to the big PCM that's now on DVD-audio. But marketing (and licensing politics and so on) will decide which is the next format to be the successor of CD-audio. Some players have been announced that can play both formats, give me one of those! I've heard 16x44kHz, 24x192kHz and DSD from one dCS
harddisk master demo of the same recording and you may guess which one was my
favorite! I have never heard a more 'analog digital' than the DSD recording. It
was just like vinyl except for the ticks, it send shivers deep, deep down my
spine... until I heard the 6 front channel demo at Philips Research... WHOAAA!
I'm already trying to convince the dCS guys into coming to Arhus next year :) > Are there any - - Find the output of the DAC and make your own postfilter
stage with some tubes, I have very good experiences with doing that to older
players. The DAC and transport are usually OK, the output stage is rotten. - -
Ask Guido Tent about clock modifications, they can also help a lot! Regards, |