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SACD / DVDA vs PCM

T

ted

Jan 1, 1970
0
Does anybody here understand exactly how these formats work? (i.e.
without marketing hype or hot air)

My understanding is that both SACD and DVDA are both lossy methods of
recording audio, mainly because the one bit serial stream cannot be
recovered into its original analogue source using the simple low pass
filter technology used in playback equipment.

I appreciate that a lot of people claim they sound better, but surely
from a technical point of view, these methods are worse than the
original PCM?

Or am I missing something???

Anybody have a a www pointer where the formats are described in decent
technical detail?? (again, without the hype)


Thanks

Ted
 
T

Tim Wescott

Jan 1, 1970
0
ted said:
Does anybody here understand exactly how these formats work? (i.e.
without marketing hype or hot air)

My understanding is that both SACD and DVDA are both lossy methods of
recording audio, mainly because the one bit serial stream cannot be
recovered into its original analogue source using the simple low pass
filter technology used in playback equipment.

I appreciate that a lot of people claim they sound better, but surely
from a technical point of view, these methods are worse than the
original PCM?

Or am I missing something???

Anybody have a a www pointer where the formats are described in decent
technical detail?? (again, without the hype)


Thanks

Ted

They're not lossy because of the 1-bit conversion back to analog,
they're lossy because you cannot guarantee a compression ratio with
lossless compression. This is because any chunk of data will be more or
less self-correlated, which is what lossless compression algorithms look
for.

As soon as you yield to the real-time requirement of having a fixed data
rate that's less than the original data rate you have locked yourself
into using lossy compression.

Does anybody claim that these are better than the original sampled but
uncompressed data? Most real performance claims are going to be based
on oversampling the data to the ADC, which works because it's much
easier to build a good reliable interpolation filter in digital hardware
than in analog, and sampling the ADC faster eases the requirements on
the necessary analog filters.
 
B

Ben Bradley

Jan 1, 1970
0
Does anybody here understand exactly how these formats work? (i.e.
without marketing hype or hot air)

My understanding is that both SACD and DVDA are both lossy methods of
recording audio, mainly because the one bit serial stream cannot be
recovered into its original analogue source using the simple low pass
filter technology used in playback equipment.

I don't see/buy that argument.
I appreciate that a lot of people claim they sound better, but surely
from a technical point of view, these methods are worse than the
original PCM?

In modern A/D converters, PCM is derived from the 1-bit bitsream
generated by sigma-delta (or is it technically delta-sigma?)
converters, so the 1-bit bitstream is indeed "closer" to the original
analog signal than is PCM.

Here's a page describing it (though it seems to have some hot air):

http://www.cdfreaks.com/article/95/6
Or am I missing something???

Anybody have a a www pointer where the formats are described in decent
technical detail?? (again, without the hype)

IIRC, DVDA has several audio formats available, all PCM at various
bit rates and depths, but there's also lossless compression done to
the PCM. Here's one article on it:

http://www.meridian-audio.com/w_paper/mlp_jap_aes9_1.PDF
 
G

Guy Macon

Jan 1, 1970
0
Tim Wescott said:
...you cannot guarantee a compression ratio with lossless compression.

If the above statement was false you could run the data file through
the compression again and again, eventually compressing it to a single
bit that decompresses back to the original.
 
T

ted

Jan 1, 1970
0
Does anybody here understand exactly how these formats work? (i.e.
I don't see/buy that argument.
Well, look at it this way. SACD samples the audio faster, and
generates 64 bits of samples 44100 times per second, whereas PCM
generates 16 bits. So 64 encoded bits per sample look a lot better
than 16. However, the pulse density decoding process for SACD is a
simple low pass filter on the bit stream. This means that, just like
an equivalent PWM decoder, you can only get 64 discrete analogue
levels per sample (unless there is some hidden technique that can
generate more levels than this), this corresponds to six bit
precision! not much when compared to PCMs 65536 levels for the same
sample interval.

I appreciate that if you SACD encode a slowly moving waveform, the
analogue output after the LP filter can follow it much better as more
density bits are used in the LP pass process. However you are now not
"sampling" at 44100, but at a much lower rate.

To do some simple calculations, a SACD stream 65536 bits long would
generate the same analogue accuracy as a 16 bit PCM signal. However
the input audio voltage level has to be steady over this sample time,
meaning it has to be 46Hz or less. What this means is that SACD can
produce "better" bit accuracy than PCM, but only at audio frequencies
below 46Hz.

I also appreciate all the arguments of what the ear can hear etc, but
this is purely a technical discussion.

Have I got something wrong somewhere??

In modern A/D converters, PCM is derived from the 1-bit bitsream
generated by sigma-delta (or is it technically delta-sigma?)
converters, so the 1-bit bitstream is indeed "closer" to the original
nalog signal than is PCM.

Yes but surely, these A/D converters are also lossy in the sense that
they do not produce the resolution they claim.

The Crystal 24 bit CS5396 96Kbps A/D as a typical example. Although it
claims 24 bit resolution, it is only capable of producing about 17 bit
"real" per-sample resolution. If you put a steady (or slowly moving)
voltage input into a
CS5396, you will get 17 steady bits, the other 7 bits are just noise.
I appreciate the noise is HF random and kind of cancels itself out
when averaged over many samples. But at the claimed 96kbps rate, the
discrete resolution is 17 bits only.

Ted
 
M

martin griffith

Jan 1, 1970
0
Does anybody here understand exactly how these formats work? (i.e.
without marketing hype or hot air)

My understanding is that both SACD and DVDA are both lossy methods of
recording audio, mainly because the one bit serial stream cannot be
recovered into its original analogue source using the simple low pass
filter technology used in playback equipment.

I appreciate that a lot of people claim they sound better, but surely
from a technical point of view, these methods are worse than the
original PCM?

Or am I missing something???

Anybody have a a www pointer where the formats are described in decent
technical detail?? (again, without the hype)


Thanks

Ted
a bit OT but I got this from rec audio pro
There are said to currently be about 730 DVD-A titles.
http://miarroba.com/foros/ver.php?foroid=234433&temaid=1935829

The average DVD-A title therefore sold about 548 copies, per RIAA
statistics (above).
Can anybody be making money with sales like these?

martin



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