Filter brewing for the Soekris R2R

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Joined 2005
Rather than using a file to set the sampling frequency in SoX, you can:

Code:
sox --plot octave -r 352.8k -n output.wav synth 1 noise  sinc  -a140 -M  -21.75k  -t5500 > filter.plt

use the -r to set sampling frequency. -n sets to a null input file, and the synth effect generates 1 second of white noise.

cheers
Paul
 
Disabled Account
Joined 2005
Thanks for these functional parameters, that's the way to get the compromise between phase, delay, aliasing and ringing, eg. 'rate -M sinc -p30 -22050 -t12k' .

If you zoom in on the frequency response with those parameters you'll see the response starts to drop at 16050z (22050-6000hz) , hits -6dB at 22050, and reaches -120dB at 28050z.

So with a bit of tweaking you can tune the filters to meet very specific criteria. Even Team Dogmatix should make brick wall filters with lots of passband ripple.
 
I just found a very nice tool: Resampler-V DSP plugin for Foobar

It's a resampling/upsampling plugin for foobar with a very good interface.
It uses the SoX and SSRC library.

You can change the pass band, stop band, attenuation, phase and see the result in the graphs.
By adjusting the sliders you see live changing the graphs, very nice :)

To get to the GUI: Preferences, Playback, DSP manager, configure selected.

Regards,
Danny
I've played with it, did intermediate filters that looked nice... And found out that there were nearly 15db increase in the 20/30hz region. I wasn't aware that it could affect lowest frequencies that much. Plug-ins users should check the linearity of frequency response after use.
 
I've played with it, did intermediate filters that looked nice... And found out that there were nearly 15db increase in the 20/30hz region. I wasn't aware that it could affect lowest frequencies that much. Plug-ins users should check the linearity of frequency response after use.

:confused: 15db is quite a bass boost, to say the least...
What settings were used?
 
Disabled Account
Joined 2005
It really getting a bit off topic... If you want to discuss PC based apps for playback and filtering without an eye to internal filtering I'd ask that you start another thread.

As a general observation, x4 up sampling 44.1 means that you are still using the internal 176.4 filters. Is that want you really want?

Making filters especially at 44.1 is a matter of compromise. High levels of stopband attenuation and steep filters will cause ripples in the passband.
 
totalCRap2 sounds great Paul. Well balanced sound and tons of detail and soundstage. Reminds me of the type of full sound I used to get from the Proceed PDP3 pcm63 dac. I switch back and forth from NOS and I like each depending on music. I would love it if there is a way to select filters on the fly someday with the DAM!
 
As a general observation, x4 up sampling 44.1 means that you are still using the internal 176.4 filters. Is that want you really want?

Worse, it is an error, as
The intermediate frequency is an explicit parameter in the filter-file. Is it possible to use other intermediate frequencies than 352K/384K?

Is the FIR1 IIR FIR2 sequence mandatory (or could something like FIR1 IIR1 FIR2 IIR2 FIR3, or FIR0 FIR1 IIR FIR2 be used)?

The sequences and intermediate sample rates and fixed in hardware.
 
Great work here Paul. Looking forward to receive my board soon.

As they say "you can prove anything with statistics" Lies, damned lies, and statistics - Wikipedia, the free encyclopedia

Averaged response is great for finding underlying long term behaviours but useless for showing short term peaks in energy.

This is a spectrogram of a short segment of the HD Tracks 192/24 release of Miles Davis' So What.

attachment.php


I've configured so that the lowest energy value displayed is -90dB. FFT size 4096 and uses a Hann window. You can see the trumpet note - which is loud - has harmonics above -90dB out to
35kHz. There are harmonics on the same note which are above -65dB out to 28kHz, and above -55dB out to 23kHz.

While it might be the case that this is above the range of hearing of most people, I'd be loathe to filter detail from a recording because there is at least the possibility it may influence perception of the audio.

Inaudible High-Frequency Sounds Affect Brain Activity: Hypersonic Effect | Journal of Neurophysiology

Of course you are free to use what ever filters you want.

cheers
Paul

Not to mention over what any microphone or tape machine in 1959 was able to capture. But still we find proof of its existence now in 2015. It must be HD magic at work here. Let us just hope its by design, and not artificial the engineer was not able to hear while mixing.
 
That's interesting. If I did understand only analogue recording and playing or HD recording and native HD playing are able to reproduce this music correctly and it's not possible to reproduce from red book standard. Hopefully the large quantity of music does not contain that frequencies. :p
Regards,
Georg
 
Slightly out of scope of what this thread is about. But still, a discussion on what to aim for with our filters might not be a bad idea?

The Hypersonic article has a footnote telling us they have to label it as advertising. And some of the text did actually led me to believe it had an agenda.

But I know for a fact that humans do respond to frequencies the ear can’t make anything out of. I used to live close to a company that made their living sound soundproofing bridges and anything else producing vibrant noise.
People living close to bridges (and windmills) often get sick without any apparent reason. Study show that the low vibration from brides do resonance with some of us. Just as only a few of us can hear perfectly pitch not everyone is prone to these low frequencies.

I’ll be surprised if this shouldn’t also apply to high frequencies as well. But I’m not a believer that these frequencies has any musical interest for me. Rather the opposite I should think.
I couple of year ago I tested myself using headphones with known frequency response. I could hear up to 13.500Hz which is normal for my age at the time. I then moved my test to the physical domain and sacrificed a tweeter with known frequency response to 24kHz. With the tweeter moved around my head I was not able to perceive any sounds over what I tested with my headphones. Admittedly the test was done with normal living room noisefloor.

My thinking using files with hi frequencies content is that if someone perceive them as more musical the reason might be artificial time domain reflection from the listening room. Or sound from vibrant object in the listening room making new harmonic distortion to the end result. Well-designed harmonic distortion do resonate favorably with us humans. But I dont't think we can call it hight fidelity if it’s our listening room that is responsible for it.
 
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