PCM1704 or newer chips?

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Why not PMD200 ?

Because I don't have them. But I have the PMD100s.

Why not Wadia ?

a) I bet they cost a fortune.
b) It's no DIY fun.
c) I bet ipod is not lossless (i.e. I prefer WAV; my prejudice).

But what's wrong with SD cards ?

I would have wanted USB sticks, but someone else has to invite a player with USB first.
(Well, actually there is one -- TEAC WAP2200, but it outputs SPDIF..... What a pity !!).

;)


Patrick
 
Serious mode of ALPHA (Adaptive Line Pattern Harmonized algorithm)

Hello inertial

>You speak of Joke, but I remember you have won various "awards" for your Alpha Processor.

Did you forget my comment, that ALPHA has 2 function.

The first one is revises 1LSB stepped wave pattern, this is not joke.<---Adaptive Line Pattern Harmonized algorithm

And the second is, we can get no ringing impulse, much more beautiful than Slow Rolloff when we played a test CD.<---Automatic Low Pass filter Harmonic Adjustment
 

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Oh! I'm sorry, 24 - 16 =�@8

I'm sorry, 24bit - 16bit = 8bit cut

This signal is showing if we edit 24bit digital data to CD's 16bit various method, how become neighborhood waveform of 1LSB /65535 steps 16bit(fs=44.1kHz).

In the figure, ALPHA off position is as same as common D/A converter unit with conventional digital filter.
 

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I can show you this all

This is ALPHA Processing Signal Flow, Adaptive Line Pattern Harmonized Argolithm and Automatic Low Pass filter Harmonic Adjustment.

If you create Automatic Low Pass filter Harmonic Adjustment function, please count ZERO data before rising over 2LSB.

** Adaptive Line Pattern Harmonized Argolithm is only interpolate in case of several sample points were same data, after step up 1LSB.
If over 2LSB step up, ALPHA is not moving.
 

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Dear Nagaesan,

My Codename is EUVL and not Evil.
EUVL stands for something else, much less evil.
But you can call me Patrick. ;)

That report I found through Google.
Thought someone else might be interested.
I doubt the chip has changed since.


Regards,
Patrick


PS BTW your website has so many interesting things but no one can find them from the index page.
Would you perhaps care to make a sitemap to make life easier ?
 
very interesting thread.
Comparing the PCM1704 with the other here mentioned types like AD1955, PCM1792 (= PCM1794 but SPI contr. modes) and AKM4396 isn't possible - so I think.

The PCM1704 is probably the only still available multibit DAC IC with not "indoor" digital filter and I/V conversion units (in opposite to the other mentioned types).

This means, the individual sonic performance is very dark dependent of the individual I/V circuit topology and the filter characteristic of the dig.-Filter.

For example there are various digital filters like DF1704, PMD100, PMD200 or some DSP based versions like e. g. used by Krell SBP64X or SBP32X.
The same situation is to observe by the I/U converter: From only a resistor to very complex discrete tube-, MOSFET and OP-AMP based I/V solutions are to find. By commercial products unfortunately in most cases integrated OP-Amps in use.

In previous years I ask me often, why in commercial products like cd players and DA converter devices equipped with low cost sigma delta or continue calibration DAC's the sonic performance was mostly better than such devices equipped with Burr-Brown multibit DACs like PCM63P, PCM1702 and PCM1704.
The reason may actually be only the previously described, because the highest standards by multibit converter are necessary for the I/U conversion.
About
http://www.diyaudio.com/forums/digital-line-level/34324-best-opamp-i-v-conversion-dac.html
is some information to find about this.

On the other hand there must be to reach the highest sonic performance only by the still available PCM1704, if I use both for the DF unit and the IU (IV) unit ultimate solutions, because this is unfortunately not possible by all versions of one-bit or continue calibration DAC's.

I want to create a own PCM1704-DAC with the use of various I/U topologies and various digital filter units so as also an additional upsampler unit and a non oversampling (zero-oversampling) mode. A simplified schematic (block diagram) I will post here in the next time.
BTW - the old "D1" from Mr. Nelson Pass is still very hard to beat in all respects.
http://www.passlabs.com/pdfs/old product manuals/d1_om.pdf
 
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you completely forgot about offline upsampling on computer,
we can manually upsample 44.1 to 88.2 , or 176.4 for use with an r2r chip & computer, its so easy to do better than nos, no need for 'DSP'.

Too much myth around upsampling really, not that interesting, also IV not interesting imho. Parallel and time interleaved DA stuff is interesting. : D
 
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My Name is Achim, from Cologne / Germany


Sorry,



i found this older thread about PCM1704.
and i saw that you have speek about Denon DA-S1 / DP-S1


i am an owner of this two Denons.


i want to modify the DA-S1 I/V sektion with analog output


which is the best way, for audiophile music ?


Sorry, my english is not so good....



best regards
Achim
 
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