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pschelbert:
Hi
in foobar2000 there are two nice components for re sampling.

Resampler-V:
there is in foobar2000 a component which is called
Resampler-V
that one can do SSRC or SOX, passband, stopban attenuation, 3 Phase setting: linear. minimum, intermediate
Great is that you can see the plot of the frequency response and pulse response.
Wish there: Continuous phase adjust (not just linear, intermediate, minimum)

Resampler (SOX) mod2:
There is another one, which has only
-Quality
-Passband
-Phase (but continuous from linear to minimum)


Application:

To circumvent the internal DAC-filter.
If you play a CD at 44.1kHz, the DAC does set his internal filter to 44.1kHz. It may be that you do not like the filter (because its linear, not step enough , to steep etc.), or want to try a filter with another phase response.

How to do:
Upsample the signal for example to 192kHz with an upsampler which allows to select frequency and phase response (from linear to minimum).
Play the file. The DAC is setting his internal filter now for 192kHz, which has now as good as no effect in the range of up to 20kHz. The upsample-filter will by far dominate.
Result: The DAC internal filter is circumvented!!

Nicest solution would be: Integrate Resampler-V into JRiver MC22 and add continuous phase adjust to it.

Peter

marko:

--- Quote from: pschelbert on July 19, 2016, 02:32:34 am ---Hi
in foobar2000 there are two nice components for re sampling.

Resampler-V:
there is in foobar2000 a component which is called
Resampler-V
that one can do SSRC or SOX, passband, stopban attenuation, 3 Phase setting: linear. minimum, intermediate
Great is that you can see the plot of the frequency response and pulse response.
Wish there: Continuous phase adjust (not just linear, intermediate, minimum)

Resampler (SOX) mod2:
There is another one, which has only
-Quality
-Passband
-Phase (but continuous from linear to minimum)


Application:

To circumvent the internal DAC-filter.
If you play a CD at 44.1kHz, the DAC does set his internal filter to 44.1kHz. It may be that you do not like the filter (because its linear, not step enough , to steep etc.), or want to try a filter with another phase response.

How to do:
Upsample the signal for example to 192kHz with an upsampler which allows to select frequency and phase response (from linear to minimum).
Play the file. The DAC is setting his internal filter now for 192kHz, which has now as good as no effect in the range of up to 20kHz. The upsample-filter will by far dominate.
Result: The DAC internal filter is circumvented!!

Nicest solution would be: Integrate Resampler-V into JRiver MC22 and add continuous phase adjust to it.

Peter

--- End quote ---
I don't mean to be rude, so apologise in advance if this is taken that way, but... seriously?

I just read that whole post and understood not one single part of it.
Do the people who do understand it actually enjoy listening to music?
What I mean is, do you hear it, and it makes you happy because it reminds you of your first girlfriend, or of the time you saw them live at whatever stadium with all your mates... or...

Do you hear it and it makes you happy because you can tell it has continuous phase adjust, and if you could tell it was just linear, it make you mad?
Is this what SOX is all about?

I'm just a bit bemused and clearly way out of my depth here. Still, you should try reading some of this stuff from the outside looking in. It's mad.

-marko.

kstuart:

--- Quote ---This will do the job of getting rid of the DAC-internal filter (mostly)
--- End quote ---

The upsampling in the recent Schiit multibit DACs is better than anything you can do on PC, so you only want to bypass the DAC if it has poor resampling.

JimH:

--- Quote from: kstuart on July 19, 2016, 11:23:48 am ---The upsampling in the recent Schiit multibit DACs is better than anything you can do on PC, so you only want to bypass the DAC if it has poor resampling.

--- End quote ---
That's a bold statement.

kstuart:

--- Quote from: JimH on July 19, 2016, 12:16:52 pm ---That's a bold statement.

--- End quote ---
From their web site:
"... time- and frequency-domain optimized digital filter with a true closed-form solution. This means it retains all the original samples, performing a true interpolation. This digital filter gives you the best of both NOS (all original samples retained) and upsampling (easier filtering of out-of-band noise) designs."

Longer explanation from the designer:
"The below are the claims of the digital filter:

1. The filter is absolutely proprietary.

2. The development tools and coefficient calculator to derive the above filters are also proprietary.

3. The math involved in developing the filter and calculating has a closed form solution. It is not an approximation, as all other filters I have studied (most, if not all of them). Therefore, all of the original samples are output. This could be referred to fairly as bit perfect; what comes in goes out.

4. Oversimplified, however essentially correct: The filter is also time domain optimized which means the phase info in the original samples are averaged in the time domain with the filter generated interpolated samples to for corrected minimum phase shift as a function of frequency from DC to the percentage of nyquist - in our case .968. Time domain is well defined at DC - the playback device behaves as a window fan at DC - it either blows (in phase) or sucks (out). It is our time domain optimization that gives the uncanny sonic hologram that only Thetas and Schiit Multibits do. (It also allows the filter to disappear. Has to be heard to understand.) Since lower frequency wavelengths are measured in tens of feet, placement in image gets increasingly wrong as a function of decreasing frequency in non time domain optimized recordings - these keep the listener's ability to hear the venue - not to mention the sum of all of the phase errors in the microphones, mixing boards, eq, etc on the record side. An absolute phase switch is of little to no value in a non time domain optimized, stochastic time domain replay system. It makes a huge difference with an Schiit Yggdrasil DAC.

5. This is combined with a frequency domain optimization which does not otherwise affect the phase optimization. The 0.968 of nyquist also gives us a small advantage that none of the off-the shelf FIR filters (0.907) provide: frequency response out to 21.344KHz, 42.688KHz, 85.3776KHz, and 170.5772KHz bandwidth for native 1,2,4, and 8x 44.1KHz SR multiple recordings - the 48KHz table is 23.232, 46.464, 92.868, and 185.856KHz respectively for 1,2,4, and 8x. This was the portion of the filter that had the divide by zero problem which John Lediaev worked out in 1983, to combine with #4 above AND retain the original samples.

This is what the competition offers:

Frequency domain optimization FIR filters with Parks-McClellan optimization. The development tools for these types of filters can be downloaded for a price range of free to $300 on the internet. Parks-McClellan is the goto filter optimization for audio design. These filters are derived with no closed form math; only successive approximation. The original samples are lost. The output is approximated. An educated guess. This optimization is ubiquitous in the front end of delta sigma dacs as well as standalone digital filters. While there is no inherent phase shift within Parks-McClellan filters, there is no optimization of phase either. The listener is left with what remains from the mixing boards, transducers, brick-wall filters, etc which can and usually do destroy proper phase/position information. Finally, it is processor efficient and economical to implement. Read cheap.

Any avoidance of the Parks-McClellan pablum requires a lot of original DSP work. Am I a prophet who received the tablets from God or some other high-end audio drivel. Hell, no. I was the producer and director of this project and worked with Dave Kerstetter (hardware-software), John Lediaev (Math), Tom Lippiat (DSP Code), Warren Goldman (Coefficient Generator and development tools) for a total of 15 or so man years. These folks either taught math at The University of Iowa, Computer Science at Carnegie-Mellon University, worked at think tanks like the Rand Corporation – you get the idea. We did this for no money - What we all had in common was that we loved audio. All other audio pros were interested in Parks-McClellan and pointed and laughed at us. That's the way it happened.

It was worth it, every hour, day, and year. So go for it if you want. For what it is, it is not a lot of money."

FYI, this guy has been doing this longer than anyone - he designed the first separate DAC product, roughly 30 years ago.

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