Path: utzoo!attcan!uunet!cs.utexas.edu!samsung!munnari.oz.au!sirius.ucs.adelaide.edu.au!hydra!francis
From: francis@cs.ua.oz.au (Francis Vaughan)
Newsgroups: sci.electronics
Subject: Re: Digital/Analog converter help needed
Message-ID: <1168@sirius.ucs.adelaide.edu.au>
Date: 20 Jul 90 01:55:18 GMT
References: <3550.26a1d2d7@ccvax.ucd.ie> <1839@island.uu.net>
Sender: news@ucs.adelaide.edu.au
Reply-To: francis@cs.ua.oz.au
Organization: Adelaide Univerity, Computer Science
Lines: 63

In article <1839@island.uu.net>, rich@island.uu.net (Rich Fanning) writes:
|>In article <3550.26a1d2d7@ccvax.ucd.ie> b_haughey@ccvax.ucd.ie (Brian
J Haughey) writes:
|>>What I am looking for is info on D-A converters that can directly drive
|>>speakers. My query is : would the scaling factors of such a D-A have
|>>to change as the volume went from very low to very high ?
|>
|>I'm not exactly sure what you want here.  You say "D-A converters that
|>can directly drive speakers".  By this do you mean a "digital amplifier"?
|>
|>There are at least two ways to do this: convert D/A at a one-volt, high
|>impedance level, and amplify in analog.  Or amplify the digital signal
pulses,
|>send through a filter to round off all those nasty sharp edges, and out
|>to the speaker.
|>
|>Either way, it would probably be desirable to multiply the 16-bit audio
|>coming in by a scale factor which represents the "volume".
|>Given fast enough hardware, it would probably be reasonable to do a
|>"brute force" approach:  represent the volume by a 16-bit value, and multiply
|>the 16-bit audio signal to get a 32 bit result.  Take the top 16 bits,
|>and there's your scaled digital signal.
|>
|>No doubt, there are more efficient ways to do it.  And a simple multiply
|>does not take care of rounding problems.  How do professional digital mixers
|>do it?

Uh No. If you take the top 16 bits off a 32 bit result you will only get
the full 16 bit resolution of the source material at full volume. If you
play the system at ordinary listening levels (say 30db down) you will end
up discarding nearly half the bits. Leaving you with say 9 bits of music
and 5 bits of 0s. This would result in a drastic reduction in quality.
The Denon Audio Test disk demonstrates this very well. They have a sequence
of tracks of orchestral music recorded on the disk at differing levels of
attenuation. If you normalise the volume on playback for the different tracks
the difference in sound quality is remarkable. Tracks that are recorded 60db
down sound worse than telephone lines. The difference between 0db and 10db
is greater than the difference between a high end CD and a mid to low range
cassette player. 

To do what you suggest you would need a 32 bit DA. Some DA converters allow
you to supply an external reference voltage which coresponds to FFFF. If 
you vary this the output is essentially scaled, but the range of useful
voltages
is typicly only 3:1 and therefore of little use as a volume control. 

Other factors work against you in building a power DA. You need a
reconstruction
filter on the output. On good CD players they will often use Bessel or similar
filters. These loose power. You would have to make a filter capable of taking
the full output power of the DA and you would not want to compromise other
factors of the output stage (like output impeadance). Bessel would be rightout.
It would also be very expensive, a similar exercise to building passive speaker
crossovers.

Building a Class-D output stage may be possible but these have always sounded
really bad to average. They are pulse width modulated switches. Getting rid
of the hash and intermodulation products is still a big problem.
Hence the reconstruction filter. If you want to build a digital speaker you are
really still left with an analogue ouput stage and analogue attenuation after
the DA. 

							Francis Vaughan