"Wally" wrote in message
...
Nath wrote:

It's supposed to increase sound from normal CD's. Dunno if it does
though; I can't disable the HDCD decoder in my HDCD DAC. Although
HDCD discs do sound very nice indeed- but I only have 3 or 4 of em.

I'm not terribly clear on how they work, but I gather the HDCDs use some
sort of compression, which decodes to 20-bit rather than 16. So, I was
wondering if that means that there's a 20-bit D/A and, if so, would that
do
a better job of resolving a 16-bit data stream.


--
Wally
www.makearatherlonglinkthattakesyounowhere.com
Things are always clearer in the cold, post-upload light.



FWIG, it's not compression, it's just a algorithm. The DA converter in my
DAC are 24 bit 8 times oversampling, 192 times oversampling pulse density
modulated single bit DAC (oversampling and noise shaping used to convert 8
times oversampled word data, from digital filter, to 192 times single bit
data)


Jargon from my DAC..

Oversampling & Digital Filtering
Oversampling is a process used extensively in digital to analogue conversion
for two main reasons; a simplified analogue reconstruction filter and the
ability to reduce word lengths without loss of information. The later of
these benefits was made use of in early CD players to achieve the 16-bit
resolution from CDs using only 14-bit DACs available at that time.
Oversampling relies on the fact that the round-off mechanism used spreads
the distortion products due to truncation over the entire oversampling
spectrum. The result is that the distortion in the baseband is only a
fraction of the total. The term oversampling simply means increasing the
sampling rate, and nothing more.

The second advantage of oversampling is that lower order analogue
reconstruction filters can be used on the output of the DAC (see figure 3).
This has the benefit that the analogue filter can be designed to have a
constant group delay (linear phase) over the baseband, which is essential
for accurate reconstruction of the original signal. Since the analogue
filter has little effect in the baseband region, component variations due to
temperature, and over time, do no alter the audio performance.

Oversampling makes use of interpolation to create the extra samples.
However, interpolation alone between two consecutive samples is not
sufficient to attenuate the sampled images at FS through to 7 x FS, hence
the reason for digital filtering. The digital filter makes use of many more
samples to perform the interpolation, the more samples it can use, the
better the interpolation and attenuation of the sampled images. The
attenuation of these images is very important for high performance audio
reproduction. This is why the DAC20 uses a PMD100 eight times oversampling
filter which offers outstanding performance in terms of its passband ripple
( 0.0001 dB) and stopband attenuation ( 120 dB). The filter also has the
ability to decode HDCD encoded CDs, thus allowing the full benefit of
increased dynamic range offered by these CDs to be enjoyed.

HDCD
HDCD is an encoding and decoding process developed by Pacific Microsonics to
provide increased amplitude resolution and to correct for sampling rate
limitations. It is compatible with the CD "Red Book" standard, thus allowing
encoded CDs to be played on equipment not containing HDCD decoding, but
still offering some of the benefits of improved fidelity. However, for the
full benefits of HDCD, the use of a decoder is essential. This is another
reason why the DAC20 uses the PMD100 digital filter which contains HDCD
decoding. The DAC20 will therefore allow the full performance of the growing
number of HDCD encoded CDs to be enjoyed.

HDCD encoded CDs contain a hidden code that is detected by the PMD100 for
automatically switching in the decoder. Since this process is automatic,
there are no controls for HDCD on the front of the DAC20, but there is an
indicator to show when HDCD material is being decoded.

The peak output level from the DAC20 is 6 dB higher for HDCD encoded CDs.
This is due to the increased dynamic range offered by HDCD.

Digital to Analogue Conversion
The Digital to Analogue converter stage is constructed around the SAA7350
and TDA1547 from Philips. The SAA7350 is used to up sample and noise shape
the digital audio data stream, producing a 1-bit Pulse Density Modulated
(PDM) output signal for conversion back to analogue by the 1-bit DAC. It
takes in data from the digital filter at 8 x FS, and then outputs it at 192
x FS. This oversampled data stream is then carefully routed to the TDA1547
1-bit DAC. The clock for the DAC is taken directly from the clean master
clock, which is again carefully routed away from digital audio data tracks.

The layout of the Digital to Analogue Converter stage is as important as the
design itself, due to the high frequency content of both data and clock.
Special RF layout techniques are used to maintain converter accuracy and
achieve optimum performance. The attention to detail also ensures that the
full benefits of single bit DAC are obtained.

The voltage reference to the 1-bit DAC is derived from a forward biased LED.
This produces significantly less noise than a standard voltage reference
diode which relies upon reverse bias breakdown. The reference voltage from
the LED is buffered and heavily filtered to provide a low impedance drive
across a wide frequency range, up into the MHz region.

This again is essential for accurate conversion.

Noise Shaping and 1-Bit DAC
The process of oversampling and how it can be used to reduce the word length
has been described briefly. For each doubling of the sampling frequency, one
bit can be dropped from the word length. If this process is taken far enough
the word length can be reduced to a single bit and a 1-bit DAC can be used
to convert the samples back into analogue. Unfortunately, to reduce a 16-bit
word down using this process would require an oversampling rate up in the
GHz region. Since this is totally impractical, a compromise is made and a
process called noise-shaping is used to get down to a single bit at sensible
oversampling rates. Present noise shapers run at rates from 64 x up to 192 x
FS. The higher the sampling rate the better, for obvious reasons. This is
why the DAC20 uses an SAA7350 which runs at 192 x FS.

Noise-shaping is used to reduce the noise in the baseband, which has
resulted from the inability to oversample at a high enough rate, by moving
it up to frequencies outside of the baseband. The SAA7350 uses 3rd order
noise-shaping to achieve this objective. The noise shaped 1-bit waveform is
processed in the DAC20 by the TDA1547 1-bit DAC. This part has been chosen
for its excellent performance and linearity.

Prerequisites for Accurate Digital to Analogue Conversion
In order for the original signal to be reproduced accurately from the
digital data, it is essential that the data is error free and the clock used
for the conversion process is precisely timed. Clock purity is a particular
area where great attention to detail is paid in all TAG McLaren Audio
digital audio products. It ensures that the reproduced audio is precise in
every detail, making the experience life-like.

It is well known that the digital audio interface has its limitations, and
that it is the main jitter contributor in outboard DACs. The reason for this
is that the recovered clock, extracted from the data stream using a phase
locked loop (PLL), tends to be modulated by the digital audio data samples.
One would argue that if this is the case, then single box CD players must be
better. This would certainly be true if nothing was done to remove the
jitter from the clock.

Jitter and Jitter Reduction
The term timing jitter is loosely used to describe timing errors associated
with digital signal transitions relative to their ideal positions. For
clocks, timing jitter can be considered as a type of phase or frequency
modulation. The clock recovered from the incoming data stream tends to be
modulated by the data, an inherent weakness of the AES/EBU and SPDIF
interface. This correlated modulation also tends to be worse for low level
audio signals, where all the data bits tend to be changing from all ones to
all zeros and vice versa.

The DAC20 contains three PLLs for clock recovery, although generally, we
only talk about two. The first, which is part of the digital audio receiver
(CS8412), has a high gain VCO and a fairly wide closed loop bandwidth. The
wide bandwidth is essential for correct operation of the receiver, but does
not provide for good jitter rejection. The second PLL has a lower closed
loop bandwidth and performs a given amount of jitter rejection, whilst at
the same time up-converting the 256 x FS clock to 384 x FS. The clock from
this PLL is used for data conversion when the Master Clock PLL cannot lock.
The third PLL, which generates the Master Clock, is where the ultimate clock
performance is achieved using Voltage Controlled Crystal Oscillators.

Master Clock PLL
The DAC20 is designed for optimum performance at each of the three standard
sampling rates; 32, 44.1, and 48 kHz. At these sampling rates the DAC20 will
operate in crystal lock mode using the Master Clock PLL.

Analogue Reconstruction Filters
The DAC20 uses reconstruction filters to remove the high frequency digital
images. Whilst these images are outside of the human hearing frequency
range, and thus cannot be heard, it is still essential that they are
removed. This is because other equipment in the audio path (preamplifier and
power amplifier) could, by means of inter-modulation, introduce spurious
signals into the baseband, thus ruining the reproduction.

The filters are a five pole design offering good attenuation in the stopband
whilst maintaining a constant group delay (linear phase) in the passband.
The first stage makes use of OPA627s from Burr Brown, whilst the remaining
stages make use of OPA134s, also from Burr Brown. High accuracy, low
temperature coefficient, Vishay metal film resistors are also used, along wi
th Wima FKP2 (polypropylene) capacitors. These components ensure both
accurate filtering characteristics and audiophile performance.

The output stage is direct coupled, using a servo to remove DC offsets. The
servo provides a low frequency break point of around 0.1Hz, which is
essential for maintaining the phase information of transient signals. The
servo amplifier is an OPA627, the same as is used in the first stage of the
filter.

The final output stage provides a low impedance (100R) drive to the phono
output sockets, thus allowing long cable lengths to be driven. The outputs
are also stable for all capacitive loads, making them suitable for use with
all types of audiophile interconnects.