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Why modern pcm1704 dac still has to oversample

derekchan

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At the begining of cd play they use 1bit nos zero oversample for d a conversion,the dac called single chip nos r2r dac
I am wondering why pcm1704 exposure 3510 player still has to process 4x or 8x oversample?

what does it differ from 4499ex delta sigma dac if 4499ex has to process oversample for d a conversion which is the nature of delta sigma dac
Does it mean pcm1704 is a r2r dac but not nos dac like msb ?
If i really want to listen nos r2r sound,the 80s to 90s pcm63 1bit nos dac cd player isnt work properly due to old aging hardware component right?

I search google said eversolo a8 feature nos emulation super slow roll off option can bypass digital filter,but it is not the type of msb dac caused the hardware js delta sigma
My question beside digital oversample filter what else oversampling of 4499ex has to process for d a conversion
 
At the begining of cd play they use 1bit nos zero oversample for d a conversion,the dac called single chip nos r2r dac
I am wondering why pcm1704 exposure 3510 player still has to process 4x or 8x oversample?
The 27 year old chip was designed for this and eliminates the need for a variable and sharp analog reconstruction filter allowing for a simple low pass filter that is easy to make.

what does it differ from 4499ex delta sigma dac if 4499ex has to process oversample for d a conversion which is the nature of delta sigma dac
Does it mean pcm1704 is a r2r dac but not nos dac like msb ?
4499ex is also a multibit DAC but with far less bits and a much higher 'upsampling rate'.
If i really want to listen nos r2r sound,the 80s to 90s pcm63 1bit nos dac cd player isnt work properly due to old aging hardware component right?
80's CD players were NOT NOS and had 4x or more oversampling .... because of the earlier explained reconstruction filter.
The fact that some people got the idea to use the DAC chip without a filter in a way it was never intended to be used is because the one starting it had strange ideas about it.
I search google said eversolo a8 feature nos emulation super slow roll off option can bypass digital filter,but it is not the type of msb dac caused the hardware js delta sigma
My question beside digital oversample filter what else oversampling of 4499ex has to process for d a conversion
DS chips can also emulate a 'filterless' (sample and hold) DAC with an absurd amount of (barely audible) distortion and heaps of unwanted HF crap that potentially can kill speakers or 'upset' amplifiers behind it.
Those DACs do NOT comply to the sampling theorem and are NOT HiFi and have a 'variable' roll-off that is music dependent.

Enjoy it... many do.
 
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At the begining of cd play they use 1bit nos zero oversample for d a conversion,the dac called single chip nos r2r dac
No, they were not 1bit delta-sigma. The first were 14-bit R2R DACs. And they also worked better with an oversampling filter.
I am wondering why pcm1704 exposure 3510 player still has to process 4x or 8x oversample?
There is nothing really modern about PCM1704. It’s 28 years old by now ;)

R2R doesn’t mean it will not benefit form oversampling. Every DAC does. As @solderdude points out: it will make the analog filter much simpler to implement, leading to better and more linear performance in the audible band.

If you want NOS sound, just use software based upsamping that emulates this. It will work with every basically every DAC, and costs nothing. Alternatively just EQ the highs down a few dB. Will will have a similar effect.
 
No, they were not 1bit delta-sigma. The first were 14-bit R2R DACs. And they also worked better with an oversampling filter.

There is nothing really modern about PCM1704. It’s 28 years old by now ;)

R2R doesn’t mean it will not benefit form oversampling. Every DAC does. As @solderdude points out: it will make the analog filter much simpler to implement, leading to better and more linear performance in the audible band.

If you want NOS sound, just use software based upsamping that emulates this. It will work with every basically every DAC, and costs nothing. Alternatively just EQ the highs down a few dB. Will will have a similar effect.
Sony's first CDPlayer (CDP 101 in 1982) used a 16 bit R2R chip. Sanyo also used this DAC (DAD 8).
Philips could not make one yet and a smart cookie found that with 14 bit and 4 times oversampling one could get the same resolution AND simplify the reconstruction filter at the same time.
From then on all CDP's used oversampling for obvious reasons.

That first SONY 16 bit NOS chip was not that great in linearity b.t.w. and required a very difficult to properly make steep analog low pass filter which was costly.

Later when the bit and oversampling race was in full bloom (20 bit , 16x oversampling) Sony had a 1-bit chip and at the same time Technics launched MASH (DS with 3.5 bits)
From then on DS chips became popular because of linearity, production costs and only requiring super simple and cheap post filters.
 
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Sony's first CDPlayer (CDP 101 in 1982) used a 16 bit R2R chip. Sanyo also used this DAC (DAD 8).
Philips could not make one yet and a smart cookie found that with 14 bit and 4 times oversampling one could get the same resolution AND simplify the reconstruction filter at the same time.
From then on all CDP's used oversampling for obvious reasons.

That first SONY 16 bit NOS chip was not that great in linearity b.t.w. and required a very difficult to properly make steep analog low pass filter which was costly.

Later when the bit and oversampling race was in full bloom (20 bit , 16x oversampling) Sony had a 1-bit chip and at the same time Technics launched MASH (DS with 3.5 bits)
From then on DS chips became popular because of linearity, production costs and only requiring super simple and cheap post filters.
Eversolo a8 can bypass oversampling but still not nos dac why? Despite oversampling filter what else does delta sigma must to oversample for d a conversion
 
Eversolo a8 can bypass oversampling but still not nos dac why?
No, it cannot, and nothing in the manual claims anything like it.
Despite oversampling filter what else does delta sigma must to oversample for d a conversion
This question doesn’t make much sense to me. Oversampling is an integral part of delta sigma DACs.
 
Eversolo a8 can bypass oversampling but still not nos dac why? Despite oversampling filter what else does delta sigma must to oversample for d a conversion
No you cannot bypass oversampling but you can select 'super slow' filter.
You basically bypass a (proper or decent) reconstruction filter.
In this case the DAC chip emulates a filterless NOS DAC (you get the same 'staircase' signals and 'kind-of' roll-off)
You can not bypass oversampling as that is how a DS DACs works, it uses a few bit levels but very high sampling rate.
The few bit levels are used so the HF switching levels become smaller in amplitude and thus need less post filtering.

Thus when you select 'super slow' filter it outputs a similar (but actually even 'better quality') waveform as a filterless NOS R2R DAC.
 
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No you cannot bypass oversampling but you can select 'super slow' filter.
You basically bypass a (proper or decent) reconstruction filter.
In this case the DAC chip emulates a filterless NOS DAC (you get the same 'staircase' signals and 'kind-of' roll-off)
You can not bypass oversampling as that is how a DS DACs works, it uses a few bit levels but very high sampling rate.
The few bit levels are used so the HF switching levels become smaller in amplitude and thus need less post filtering.

Thus when you select 'super slow' filter it outputs a similar (but actually even 'better quality') waveform as a filterless NOS R2R DAC.
1.my av receiver built in 9018k2m dac ,does it upscale or upsampling 44.khz cd audio and bluray 24bit 48khz surround to higher hz data?like upsampling to 192khz for internal processing stuff. i tried to ask anthem but they said doesn't process upsampling stuff

2.if i gonna try r2r dac is exposure 3510 cd player a cheapest choice?how could it handle thermal drift like msb dac?

3.pcm1704 is a .pcm end gamer,how about .dsf file? what kind of dac greatest to reproduce dsd file sound? is it 1 bit dac the best?

4.does people use hqplayer for oversampling or upsampling ,then pass data to dac for d/a conversion?
 
1.my av receiver built in 9018k2m dac ,does it upscale or upsampling 44.khz cd audio and bluray 24bit 48khz surround to higher hz data?like upsampling to 192khz for internal processing stuff. i tried to ask anthem but they said doesn't process upsampling stuff
It does not 'upsample' in the sense of what you (think) it means.
It uses DS modulation and offers some filter options.
This means just a few bit levels and very high 'switching rate'
2.if i gonna try r2r dac is exposure 3510 cd player a cheapest choice?
A R2R DAC can be connected to any digital signal that it is designed for, oversampling or not and limited to the max bitrate/depth it can handle.
how could it handle thermal drift like msb dac?
There is nothing to handle.
3.pcm1704 is a .pcm end gamer,how about .dsf file?
It can't play .dsf
what kind of dac greatest to reproduce dsd file sound?
Any DAC that can handle the format.
is it 1 bit dac the best?
It just needs to be handing the DSD files. Regardless if it is real 1 bit or multibit that can switch between levels or not)
4.does people use hqplayer for oversampling or upsampling ,then pass data to dac for d/a conversion?
Yes, that's the only way you can use it.

PCM needs a reconstruction filter to retrieve the original (intended) waveform.

Omitting it means you add distortion and the output signal does NOT resemble what's on the recording.
It is a very silly and unwise thing to do (filterless NOS R2R) but let's face it.... that's human behavior in a nutshell.
 
Why the obsession with NOS? It was never a good idea, and never will be.
But ... but ... but audiophiles rave about this 'more organic', 'more analog', 'better' sound and I need to hear what it is all about.
After-all these owners and reviewers of such devices all agree and can't be wrong.

Besides ... the word 'filter' implies that you remove something and I don't want anything removed. Just look at those glorious needle pulses and square-waves, the lack of the dreaded 'ringing' and 'pre-echo's'
I want.... no need, fast impulses and only NOS does that (and DSD... I have to love DSD as well, more organic, less digital, more natural).

I need to get me some of that 'ideal' impulse response, smoother more analog sound, the way it used to be (vintage sound).
I HAVE to have that and so I am going to try all kinds of R2R, cables, tubes etc. in order to get all of that instead of the 'boring/bland/flat/clinical/dead/cold' digital sound.
Who cares what measurements say ... I listen with my ears ... so do some manufacturers ... they tune their gear to their house sound.

I need... no... must experience that glorious and wonderful house sound of [this or that brand] too. These engineers truly know what they talk about and know how to design 'pleasant organic' sounding gear by using carefully selected, and usually very expensive, components. The combination of those parts is what creates their 'house sound'.
Like a 4 star cook creates flavors.
Magic ... I want the magic... I want to believe and experience what has been perceived by folks with much better ears and gears with superior knowledge about audio because they heard/reviewed almost everything 'high-end' that exists.
Bah ... those scientists and plot gazers with their cheap Chinese gear know NOTHING about good sound. They only look at their (SINAD) meter. They never even heard good sound anyway ...

:facepalm: I'll get my coat ....
 
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'more organic', 'more analog'
Ah yes, how could I be so stupid to forget…

1790503874465.gif


… um, okay…
 
Yeah, good luck with that…
the reference value for your sigma delta is send at 44.1Khz, your sigma delta still runs on n*fs. no oversampling. it will work
 
In theory you could do sigma delta without oversampling...
No... @voodooless is correct but you can create a filter that results in the, shown in the shown by voodooless 'stepped staircase' waveform, by altering the filter in the DS process and that's what the 'discussed super slow' filter setting does.
The oversampling is how DS works (to create the 'in-between' sample values by averaging the very high frequency switched signal).

Ah yes, how could I be so stupid to forget…

View attachment 561242

… um, okay…
Well... a staircase is a totally logic way to climb to 'sonic heights'.
You can clearly see the phenomenal 'speed' in the rising and falling edges which exceeds that of the fundamental sine-wave. :eek:
 
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That first SONY 16 bit NOS chip was not that great in linearity b.t.w. and required a very difficult to properly make steep analog low pass filter which was costly.
Yup, I tested a Yamaha CD-1 with that first DAC from Sony (CX-890), true NOS, but it was indeed filtering, of course, and as you said:

1790525695680.png


Not a too bad attenuation of -90dB, but fully active only at 28kHz though.

But linearity in audio band was difficult, like you said too. This is a zoomed view of the bandwidth, 20-20kHz, with linear frequency scale which I prefer to analyze the effect of a filter:

1790528373029.png


It is within the specs of Yamaha (+-0.5dB), but of course not what we are used to see today.

To compare with the tech of Philips, the below is the TDA1540 (14bits DAC) with its 4x oversampling filter (SAA7030):

1790527751817.png


In 2026, we have:

1790527412019.png


As flat as we want, and even a choice of unnecessary attenuations for those who like. The linear fast filter shows a little attenuation of -0.1dB at 20kHz and that's because I forgot to use the calibration file when I performed the measurement.

Cheers
 
In theory you could do sigma delta without oversampling...
All the architectures and designs I have seen and done for delta-sigma converters "without" oversampling used multiple modulators in parallel to provide oversampling without higher-speed delta-sigma loops. Thus they still fundamentally oversampled the signal, by time-shifting the samples and recombining digitally (usually), relieving the high-speed sampling clock requirements but not the timing (sampling time or aperture time) so their benefit was difficult to realize in practice. I do not know of any commercial products, though it has been a while since I looked, due to various problems implementing the approaches. MASH and other multibit architectures using "conventional" oversampling loops took over the high-resolution market AFAIK.

I did design one using Hadamard coding that was pretty cool, but the front end of all the modulators still needed full signal bandwidth and very tight matching so implementation proved impractical. There was also no power savings using multiple lower-rate modulators, the main desire of the program, just a lot of added complexity. Theory was smacked hard by reality.

Note any DAC can be oversampled, but without the delta-sigma noise modulator, you only gain 0.5 bit for each 2x increase in sampling rate so the benefits are modest (and the digital filters must start out much wider, thus more complicated and power-hungry).
 
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