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Hi-end DSP based multi-channel integrated Preamp/Crossover/DAC project
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Old 12th October 2015, 10:28 AM   #401
Tranquility Bass is offline Tranquility Bass  Australia
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Hi-end DSP based multi-channel integrated Preamp/Crossover/DAC project
Quote:
Originally Posted by torgeirs View Post
Is it the analog output or the digital output?
For analog it should be two clean sinuses.
Or will the 44.1 -> 192 convertion interpolate the +0dbFS sinus to a clipped signal instead of a clean sinus?
Thank you for the testresults.
Kind regards Torgeir
It's at the output of the DAC. If the signal is clipped before the SRC then it will be clipped after the SRC because the S/PDIF cannot handle anything above FS. I noticed that if I force the output of the sound-blaster to re-sample at 96kHz then neither signal is clipped.

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Old 12th October 2015, 01:13 PM   #402
torgeirs is offline torgeirs  Norway
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The signal is not actually clipped. No values abow 1 or -1!
A perfect dac should output a pure sine at 1.08 volts if a sin at 0dBFS outputs 1 volt.

If you do a FFT on the signal in audacity it is a pure sine.
The magic is in the phase shift so the sine is never sampled at the max values of the sine. But it should of course be output as a sine curve, ref Nyquist.

Such a sine is difficult to sample in a ADC, but as the tc electonics paper shows, it is a common signal after modern mixing and mastering sound prosessing in the digital domain.

Last edited by torgeirs; 12th October 2015 at 01:18 PM.
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Old 12th October 2015, 07:23 PM   #403
masopa is offline masopa  United Kingdom
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Quote:
Originally Posted by labjr View Post
I became interested in the Linkwitz LX521 speaker system. Then they stopped making the ASP board and went to DSP, using a MiniDSP box. I feel the 96 kHz sample rate would be a compromise compared to the analog crossover. Analog can have it's own issues but digital not done properly, to me, is much worse. And having to resample 192khz PCM and DSD to 96khz doesn't seem like a great idea.

I had considered Metric Halo but was told their interface does not quite have the DSP power for a 192khz stereo 4-way crossover. I think they're supposed to have a new and improved box any day now.

This project may renew my interest in that project again.
Well, you don't need to worry about the 96kHz sample rate on the MiniDSP... because it's actually 48kHz. There's always the 4x10 HD though, which is 96 (and £500)

All the more reason why getting something like this would be wonderful. It could be more than the Hypex DSP promised (and failed) to be...
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Old 13th October 2015, 03:47 AM   #404
jbsaunders is offline jbsaunders  United States
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Quote:
Originally Posted by labjr View Post
I became interested in the Linkwitz LX521 speaker system. Then they stopped making the ASP board and went to DSP, using a MiniDSP box. I feel the 96 kHz sample rate would be a compromise compared to the analog crossover. Analog can have it's own issues but digital not done properly, to me, is much worse. And having to resample 192khz PCM and DSD to 96khz doesn't seem like a great idea.

I had considered Metric Halo but was told their interface does not quite have the DSP power for a 192khz stereo 4-way crossover. I think they're supposed to have a new and improved box any day now.

This project may renew my interest in that project again.
So as I anciently await the completion of this project to implement it in my system as an active 4 way crossover, or at least to start.

So at the risk of hi-jacking this thread, I propose a question. I am trying to understand why one would have the need for a word length of 192khz or better. If you can achieve a signal of 48khz, well above human hearing, capping out at 96khz. Why choose too use up computing power (taps) when, as far as I understand, their would be no audible gain? Again this is my understanding, please feel free if you are so inclined to inform me
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Old 13th October 2015, 07:34 AM   #405
Rajapruk is offline Rajapruk  Sweden
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If the source material is 192khz, then resampling is avoided. One reason I guess. If it is a valid reason in the whole picture, I do not know!
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Old 13th October 2015, 08:53 AM   #406
torgeirs is offline torgeirs  Norway
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If the DSP uses different samplingrates the filters has to be altered. So when using 1 samplerate in the DSP, only 1 set of filters are needed.
For digital sources there is not any gain in going to 192kHz exept not bandlimit 192kHz material (i think). At analog in it is the same reason of not bandlimiting the source.
To save computing power at lower frequencies, all except the highest frequency driver, you downsample to save taps. Often the HF over 3000 Hz need little prosessing so the penalty is not that big. What you can hear is another story. (For me 44,1 16 bit is enough:-)

Regarding the resampling of 192khz. I dont know what is intended here, but some ASRC chips reclock all inputs and thereby introduce some signal prosessing on all input samplerates. This is done so that the DSP controls the samplerate and the filters don't alter characteristics with small changes in input samplerate.

Last edited by torgeirs; 13th October 2015 at 09:01 AM.
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Old 14th October 2015, 04:07 AM   #407
labjr is offline labjr  United States
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Quote:
Originally Posted by jbsaunders View Post
So as I anciently await the completion of this project to implement it in my system as an active 4 way crossover, or at least to start.

So at the risk of hi-jacking this thread, I propose a question. I am trying to understand why one would have the need for a word length of 192khz or better. If you can achieve a signal of 48khz, well above human hearing, capping out at 96khz. Why choose too use up computing power (taps) when, as far as I understand, their would be no audible gain? Again this is my understanding, please feel free if you are so inclined to inform me
I hear the difference. To me, 192kHz sounds better than lower sample rates. I don't want to debate the subject on this thread. It's been beat to death in countless other threads and forums
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Old 14th October 2015, 05:52 AM   #408
jbsaunders is offline jbsaunders  United States
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Of course I meant "anxiously" Thanks for your thoughts, now back to the subject at hand......
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Old 15th October 2015, 09:24 PM   #409
jbsaunders is offline jbsaunders  United States
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Quote:
Originally Posted by torgeirs View Post
If the DSP uses different samplingrates the filters has to be altered. So when using 1 samplerate in the DSP, only 1 set of filters are needed.
For digital sources there is not any gain in going to 192kHz exept not bandlimit 192kHz material (i think). At analog in it is the same reason of not bandlimiting the source.
To save computing power at lower frequencies, all except the highest frequency driver, you downsample to save taps. Often the HF over 3000 Hz need little prosessing so the penalty is not that big. What you can hear is another story. (For me 44,1 16 bit is enough:-)

Regarding the resampling of 192khz. I dont know what is intended here, but some ASRC chips reclock all inputs and thereby introduce some signal prosessing on all input samplerates. This is done so that the DSP controls the samplerate and the filters don't alter characteristics with small changes in input samplerate.
Thinking about your comment, 44.1/16 is good enough, I believe that is because most music that is listened to is a 44.1/ 16 recording. If you can find a recording that is captured at a higher rate say 24/96, and never down sampled, I believe you can her the difference. Higher dynamic range/ higher frincquncy response.
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Old 16th October 2015, 08:51 AM   #410
torgeirs is offline torgeirs  Norway
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By "For me" I ment "For me personally". I don't hear much over 15k. And I also believe in masking. So maybe I hear single 18k signals but not in the presence of 10k signals with higher intensity.
I don't doubt that there are some lucky ones out there that has ears, physically out of the ordinary, that can hear 20k+
As to the 16 bit, you need more bits for prosessing, but 96 db is pretty high dynamic range before the DAC. I don't know any microphones with much better S/N. And when playing 120 dB, 25dB noise is hardly anoying.
On top of that you can noise shape so actual dunamic range is 100+ dB.
The difference I have heard between hires and normal records I think comes from different mastering of the material.
To be fear i haven't tested much on live "unmastered" material like this:
DPA Microphones :: stereo-recordings
But I am satisfied with the music I like and all the energy in that music is under 20k.
But as a final remark: I am not against having some margins so doubling or quadruple bandwith is not a bad idea when DSP resources are high and DSP is cheap.

Last edited by torgeirs; 16th October 2015 at 08:54 AM.
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