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16x Digital interpolation filter - drive PCM56, PCM58, AD1865 and so on up to 768 kHz
16x Digital interpolation filter - drive PCM56, PCM58, AD1865 and so on up to 768 kHz
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Old 19th April 2019, 07:06 PM   #91
3lite is offline 3lite  Poland
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Join Date: May 2016
Quote:
Originally Posted by Hans Polak View Post
Nyquist can stay at home, we are discussing filtering of an upsampled signal. No aliasing or backfolding takes place.
And in case the digital filter leaves some residue from 22kHz to 24kHz, this will simply appear in the analogue signal after d/a conversion still residing at 22kHz to 24kHz.
I don't think anyone will notice that.


Hans
Yes, you are discussing about an upsampled signal, but upsampling (without low pass filtering) means introducing (deliberately) images of the original signal to increase the sample rate by inserting zeros into the signal. That being the case you have plenty of images / back-folding taking place. In order to fix that you need to low pass the upsampled signal to get rid of it.

The following image should explain it pretty well:

Click the image to open in full size.

chris719 explained very well why normal digital filters aren't very good due to limited amount of resources (taps). They all pretty much allow for some back-folding due to not achieving full attenuation (stop band) by 0.5 fs.

Also, please have a look at this:

Audio DAC shapes

There is hardly any DAC / digital filter there capable of reconstructing 24 kHz sine wave due to the issue pointed out by chris719.

Quote:
Originally Posted by Luke View Post
Hi 3Lite, my intention is to run Ian's I2S to PCM after this as I am building balanced dac. Do you foresee any issues with this configuration?
You do not need Ian's board for anything. It's doing the exact thing my digital filter does already which is splitting I2S into so called "PCM" with both inversed and non-inversed data signals
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Old 19th April 2019, 08:28 PM   #92
Hans Polak is offline Hans Polak  Netherlands
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Location: Blaricum
Quote:
Originally Posted by 3lite View Post
Yes, you are discussing about an upsampled signal, but upsampling (without low pass filtering) means introducing (deliberately) images of the original signal to increase the sample rate by inserting zeros into the signal. That being the case you have plenty of images / back-folding taking place. In order to fix that you need to low pass the upsampled signal to get rid of it.

The following image should explain it pretty well:

Click the image to open in full size.

chris719 explained very well why normal digital filters aren't very good due to limited amount of resources (taps). They all pretty much allow for some back-folding due to not achieving full attenuation (stop band) by 0.5 fs.

Also, please have a look at this:

Audio DAC shapes

There is hardly any DAC / digital filter there capable of reconstructing 24 kHz sine wave due to the issue pointed out by chris719.



You do not need Ian's board for anything. It's doing the exact thing my digital filter does already which is splitting I2S into so called "PCM" with both inversed and non-inversed data signals
I don't want to make some principal case of who's wrong or who is right, but I give it one more final attempt.

Could you explain how any frequency can fold back when the DAC is outputting the signal with the same frequency as the digital filter.
Downfolding or aliasing can only happen when you downsample after filtering like in a studio where Audio is recorded at 384kHz and after digital filtering is down sampled to 44.1Khz.
Or is that what also happens in your filter, in that case you are right.
But if not, if you apply a digital filter or not, as long as the DAC is operating at the same speed, the analoque signal will be exactly like the digital content, but with many mirrors around Fs and convoluted with a Sinc function.
With no digital filter you will have the same performance as a NOS Dac with many ultra sonic mirrors, and in case of a digital filter most primary mirrors will be removed, and a slow analogue filter will remove the mirrors around Fs resulting in a rather clean signal limited to ....kHz.


Hans


P.S. Thank you for the Audio DAC shapes link, a very interesting comparison.
And again, great respect for the filter you made, a colossal job well done.
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Old 19th April 2019, 09:02 PM   #93
Markw4 is offline Markw4  United States
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Location: California
Hans makes an interesting point: Foldback typically occurs during reproduction, not recording. Upsampling should cause HF images to be played back as HF music, not folded back (that is, so long as images do not exceed the upsampled Nyquist).

Last edited by Markw4; 19th April 2019 at 09:07 PM.
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Old 19th April 2019, 09:03 PM   #94
3lite is offline 3lite  Poland
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Join Date: May 2016
Quote:
Originally Posted by Hans Polak View Post
Or is that what also happens in your filter, in that case you are right.
Actually yes, that is how my filter works It is basically a synchronous sample rate converter. Unlike normal digital filters which do have a staged architecture (e.g. one stage of interpolating from 1x -> 2x and the other from 2x -> 8x). Depending on the input stream those stages are omitted to result in selectable interpolation ratio of 2x, 4x or 8x, but they do always interpolate nonetheless (no decimation). In order to achieve similar behavior I have decided to interpolate and decimate at the same time depending on the input stream. I did mention it within the first post

Also, might be worth keeping in mind that you have analogue electronics after the DAC which can cause non-linear or intermodulation distortion and other issues if the input has mirrored images of higher frequency which can in fact cause issues in the audible band.

I have seen something like this already when I was testing filter coefficients with higher passband and the only difference were the coefficients and the allowed passband. Certain artifacts did appear within audible band. The funny fact is that throughout this project I have seen artifacts within audible band due to different things (dithering, clocking, allowed passband and so on) and I couldn't really explain them myself except for trying a different approach to fix them.

Last edited by 3lite; 19th April 2019 at 09:30 PM.
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Old 19th April 2019, 09:38 PM   #95
chris719 is offline chris719  United States
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Join Date: Jun 2004
Location: Connecticut
16x Digital interpolation filter - drive PCM56, PCM58, AD1865 and so on up to 768 kHz
Yes, you’ll have a hard time finding an audio ADC that is in the stopband by Fs/2 as well, so it’s probably in your best interest to remove the transition band of the ADC filter.
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Old 19th April 2019, 10:51 PM   #96
dreamth is online now dreamth  Romania
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You said about PCM1792 that is not quite right...
Well, I have an Audient ID22 which is equiped with this dacs and i couldn't find a single musician or sound engineer around me to tell that this box doesn't sound right, on the contrary they were quite pleased with it and in fact Audient ID22 had the best sales on Amazon in 2017, even better than RME and others.Nobody complains about how that thing is recording or playing.
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Old 20th April 2019, 09:04 AM   #97
chris719 is offline chris719  United States
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Join Date: Jun 2004
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16x Digital interpolation filter - drive PCM56, PCM58, AD1865 and so on up to 768 kHz
Quote:
Originally Posted by dreamth View Post
You said about PCM1792 that is not quite right...
Well, I have an Audient ID22 which is equiped with this dacs and i couldn't find a single musician or sound engineer around me to tell that this box doesn't sound right, on the contrary they were quite pleased with it and in fact Audient ID22 had the best sales on Amazon in 2017, even better than RME and others.Nobody complains about how that thing is recording or playing.
The problem isn’t unique to PCM1792; I picked it because it’s a textbook example. It’s also of arguable importance. Almost every DAC that has an integrated linear phase OS filter, or external filter ASIC like DF1704, is not into the stopband by Fs/2 but Fs*0.536 (about 24 kHz for 44.1 sampling rate). Why is that an issue? Because the filters in the ADCs used to record almost all music do the same thing. This allows aliasing in the region of 20 kHz to 22.050 kHz if there is any signal content between 22.050 kHz and 24 kHz after the analog filter preceding the ADC.

This is the original point I was trying to make (incorrectly) in my reply to Hans, as it’s more applicable to ADCs and filters that perform decimation like 3lites. It would be best to have a filter in the DAC that is in the stopband earlier, much closer to 20 kHz if you want to get rid of the aliased region that came out of the ADC.


You can see from the graphs on the first page that 3lite did it all correctly .

Last edited by chris719; 20th April 2019 at 09:15 AM.
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Old 20th April 2019, 09:11 AM   #98
chris719 is offline chris719  United States
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Join Date: Jun 2004
Location: Connecticut
16x Digital interpolation filter - drive PCM56, PCM58, AD1865 and so on up to 768 kHz
Quote:
Originally Posted by Hans Polak View Post
I don't want to make some principal case of who's wrong or who is right, but I give it one more final attempt.

Could you explain how any frequency can fold back when the DAC is outputting the signal with the same frequency as the digital filter.
Downfolding or aliasing can only happen when you downsample after filtering like in a studio where Audio is recorded at 384kHz and after digital filtering is down sampled to 44.1Khz.
Or is that what also happens in your filter, in that case you are right.
But if not, if you apply a digital filter or not, as long as the DAC is operating at the same speed, the analoque signal will be exactly like the digital content, but with many mirrors around Fs and convoluted with a Sinc function.
With no digital filter you will have the same performance as a NOS Dac with many ultra sonic mirrors, and in case of a digital filter most primary mirrors will be removed, and a slow analogue filter will remove the mirrors around Fs resulting in a rather clean signal limited to ....kHz.


Hans


P.S. Thank you for the Audio DAC shapes link, a very interesting comparison.
And again, great respect for the filter you made, a colossal job well done.
You’re right, I knew what I wanted to say conceptually but got it wrong. Actually the funny thing is we both missed that this filter is basically a SSRC and does decimate, which made me right for the wrong reason.
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Old 20th April 2019, 11:25 AM   #99
dreamth is online now dreamth  Romania
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Is it just me in the entire world who happened to find a CD from the 80's , played it on a 90's player and was completely blown away by the quality of that record?
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Old 20th April 2019, 11:27 AM   #100
merlin el mago is offline merlin el mago  Europe
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Quality of the record is the 1st for the best sound quality, no matter the rest if you don't have a best quality record.
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