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Old 1st March 2015, 09:48 PM   #261
soekris is offline soekris  Denmark
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Quote:
Originally Posted by zfe View Post
@Soren:
Speaking of "a single bi-quad", a question I always wanted to ask is, can you cascade several IIR bi-quad filders with the DAM?
As you want to use them for the crossover (+ potentially de-emphasis) this should be the case?
If so, how should the headers look like that they are used cascaded.
All Biquads are cascaded, kinda the only option....
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Old 1st March 2015, 09:52 PM   #262
soekris is offline soekris  Denmark
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Quote:
Originally Posted by pos View Post
Regarding IIR de-emphasis, here are some biquad coefficients that can be used in place of the original ones for better accuracy :

Code:
dam1021,352800,8,30,5,1
Deemphasis IIR, 352.8 Khz Samplerate, hiself f=5600 Hz, Q=0.485, gain=-10.1 dB
0.331085040573029
-0.577213330670734
0.251294171647457
1.85212451656979
-0.857290398119544

dam1021,384000,8,30,5,1
Deemphasis IIR, 384 Khz Samplerate, hiself f=5600 Hz, Q=0.485, gain=-10.1 dB
0.329617410213504
-0.581107678819127
0.255876017438852
1.86372441280949
-0.868110161642715
This is as close as I could get with a single biquad.
That said, it looks like variations do exist among CDs on how emphasis was implemented...
Still I think it is a good thing to be as close as possible to the theoretical formula
Good work, thanks. I'll use those in the release filters....
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Old 1st March 2015, 10:43 PM   #263
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Quote:
Originally Posted by zfe View Post
What should be the difference between using the DAC Volume control and using different audio input files with scaled PCM sample values?
None I hope, but I have full control on the input files but no knowledge about the implementation details of the signal processing in the DAC.

Changing the volume, via differently scaled input files, I did (in real world) and it produced the described result as in

But OK I will try the volume controll tomorrw.
The DAC has a 28bit R2R ladder.

The filters are processed FIR1 -> Volume -> FIR2.

If the gain added at FIR1 is clipping audio data then using the volume control to shift the data by 2 or 4 bits will make no difference to clipping.

If the clipping is passing through FIR1 unclipped but then clipping in the R2R shifting the data by 4 bits/-24dB should eliminate the clipping.

That should identify if point where clipping occurs.

I'd test but I'm at work as it's monday morning.
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Old 2nd March 2015, 10:27 AM   #264
TNT is online now TNT  Sweden
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Quote:
Originally Posted by soekris View Post
Good work, thanks. I'll use those in the release filters....
Maybe you should ask for permission

//
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Old 2nd March 2015, 11:01 AM   #265
zfe is offline zfe
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Quote:
Originally Posted by spzzzzkt View Post
The DAC has a 28bit R2R ladder.

The filters are processed FIR1 -> Volume -> FIR2.

If the gain added at FIR1 is clipping audio data then using the volume control to shift the data by 2 or 4 bits will make no difference to clipping.

If the clipping is passing through FIR1 unclipped but then clipping in the R2R shifting the data by 4 bits/-24dB should eliminate the clipping.

That should identify if point where clipping occurs.

I'd test but I'm at work as it's monday morning.
If you use the DAC volume control to compensate the multiplier the output is unclipped.
I tested for all FIR with one tap 1
1.) FIR1 multiplier 2 volume -6dB
2.) FIR1 multiplier 2, FIR2 multiplier 2 volume -12dB

That is to be what you could expect with 32bit fixed point arithmetic in the FPGA and a 28bit ladder. You might get clipping in the "computation" itself if you exaggerate with the multipliers.

P.S. I have the impression your answers get a bit ... hmm tense
Please be aware that English is not my first language, I come from a different cultural background and I am not in engineering nor audio business.
I have no intensions to offend you.
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Old 2nd March 2015, 11:48 AM   #266
pos is offline pos  Europe
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Quote:
Originally Posted by TNT View Post
Maybe you should ask for permission

//
Why? I put them here for anyone to use, explicitly. Soren is of course more than welcome to add them into the official filter
This is nothing fancy really: just a high shelving biquad with specific parameters (shown in the first line of the biquad description).
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Last edited by pos; 2nd March 2015 at 11:52 AM.
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Old 2nd March 2015, 12:19 PM   #267
zfe is offline zfe
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Lets look at some nice pictures
All are from a full amplitude 11025kHz sinus signal 44.1kHz sample rate, measured at the SE-output of the DAM with an oscilloscope and 1MOhm probes. The oscilloscope settings are unchanged.

First all FIR taps set one 1. (so 2 times zero insertion only: at FIR1 and FIR2). Perhaps that should be called the bypass filter instead of "NOS".
Bypass.jpg

Second the NOS filter of pos (FIR1 and FIR2 8 taps 1)
NOS.jpg

And last the stock filter from Soren
fltr.jpg

Quite that what you (or at least I) would expect, except the much higher output level with the last two.
For an explanation of that lets have an amplified look at the first negative spike of the bypass filter (i.e. the raw output of the DAM)
SE1.jpg
SE2.jpg
The time of the, next to linear, negative, rise is 1/(64*44.1kHz) ... fine.
This is the typical diagram of charging (not to full applied voltage) an decharging a capacitor.
So the voltage pulse of the DAM is to short to charge the capacitor (C135) completely. Secondly the decharge time is considerably longer (due to the high resistive load). The other two filters continue "charging" and result in a higher output voltage.

As also modern preamps and amps have very high input resistance, the situation might there be similar.

Last picture, the bypass filter spike at the balanced output (other oscilloscope time base).
Balanced.jpg
You see that even the excellent op amp of the balanced output is not fast enough to reproduce the SE output spike.

I for myself think I will not torture my equipment with the Bypass or NOS filter.

Last edited by zfe; 2nd March 2015 at 12:28 PM.
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Old 2nd March 2015, 12:31 PM   #268
pos is offline pos  Europe
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Quote:
Originally Posted by zfe View Post
Second the NOS filter of pos (FIR1 and FIR2 8 taps 1)
You must be confusing me with someone else
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Old 2nd March 2015, 12:38 PM   #269
zfe is offline zfe
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Quote:
Originally Posted by pos View Post
You must be confusing me with someone else
Excuse me, indeed I do. The NOS filters are the ones proposed by oneclock.
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Old 2nd March 2015, 03:18 PM   #270
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Well , this is the NOS no-filter that other people likes.

The correct filter NOS that people are asking, previously proposed by Soren. I think it's not right to put NOS filter: one sample and seven 0.

Quote:
Originally Posted by soekris View Post
Are you talking about non oversampling mode ? That can easily be done, t.ex. for 44.1:

Enter 8x 1.00000 as filter coefficients, 8 taps and gain set to 1.

The 8 is for 8 times oversampling, use 4 and 2 for other oversampling rates,
just like the 352/384 bypass filters using 1.

Again, I don't recommend non oversampling, and you then need the compensating filter for the 3db loss at 20 khz.
With 11025 Khz. or 22.050 kHz. sinus see bad. With lower frequency signal is more nice. With audio signal the NOS output is more similar original audio. And one sample and seven 0 see bad, no similar to original audio.

I 'm not very interested NOS filter, if I have a good filter oversampling , without saturation or rounding errors. But they are people likes NOS without oversampling filter.

In case I use NOS, I prefer with an IIR filter that amplify the fall of 16 kHz due to rectangles reconstruction. (a sync in frequency). (3 dB. 20 Khz.)
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