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Is it possible to stretch 16 bits symmetrically?, instead of padding with zeros?
db1989
post May 6 2014, 19:55
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QUOTE (giro1991 @ May 6 2014, 18:46) *
If you can attentuate a full scale wave, then it must to be possible to boost a full scale wave, in the dsp realm.
This is my point. ^
No one ever denied this except people with woefully incorrect concepts of digital audio, as you have had and continue to have

QUOTE
Furthermore, the reason I started persuing dither earlier on, is because it appears to be the only dsp available that increases bit depth that can be employed during live playback (i.e. foobar). The only one.
Dither does not increase bit depth in any useful way whatsoever

QUOTE (giro1991 @ May 6 2014, 18:47) *
"By what means is my intention." Don't ignore the boosting just because the means is not in sight? Rather restrictive.
Most people can't think of a means, so dismiss this view as rediculous, but there is a very valid reason.
No, trust me: we dismiss this entire thread as ridiculous for a lot of “very valid reason”s

QUOTE
If I have a 32bit wave, entering a resampler, then a new, almost infinite wave is constructed, from this new, near infinite wave I can then sample 48k new points, which I can then send to a multibit NOS dac, which is high performing (don't dismiss boosting just because of your opinion of this either).
You are sampling, at best, intermediate points representing no additional depth/fidelity/quality whatsoever. Big deal! And your silly idea of dither will ensure that not only are you gaining no quality, you’re actually adding noise for no reason. Congratulations. I’ll slip your Nobel Prize for Physics in the mail.

QUOTE
Let me simplify my means even further. The chip on the right, is the digital chip, it is the dsp, that conditions the signal fit for the DAC. I've bypassed this and will be streaming 48k direct to the mulitbit. In essence, I'm replacing the digital filter (lots of distortion) with foobar or "32bit supercomputer from the future", atleast relatively speaking to the dac.
Stop acting like you’re doing anything (A) revolutionary / (B) logical. Just stop. Go away, read some textbooks, and then come back and agree with us about how misguided all your ideas here are.
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xnor
post May 6 2014, 22:41
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giro1991, by disabling that filter you will see square waves on the output of your DAC even if you resample to 48 kHz (which is actually quite an insignificant increase in sampling rate).

You trade in the distortion from the DSP chip with the extra HF roll-off, distortion, ultrasonic images ...

As I said, it would be better to buy a proper DAC...


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Kohlrabi
post May 7 2014, 06:36
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[transl.: It is not only not right, it is not even wrong!]


This post has been edited by Kohlrabi: May 7 2014, 06:38


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giro1991
post May 8 2014, 12:10
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QUOTE (saratoga @ May 6 2014, 19:16) *
For the millionth time, all processing these days is done in floating point, so this happens automatically.

Yes it is always used but is multiplication always used?

QUOTE (saratoga @ May 6 2014, 19:16) *
QUOTE (giro1991 @ May 6 2014, 13:46) *
then it must to be possible to boost a full scale wave, in the dsp realm.

Sure, in floating point, multiply by a value greater than 1.0.

Upon entering 32 float, atleast in foobar, I doubt multiplication is automatic, 16bit depth it is just passed onto 32bit float.

Question is anyone willing to show me a dsp/method that does this multiplication and will do it upto 192dB vst/foobar?
If one does not exist then how easy would it be to make one?

QUOTE (db1989 @ May 6 2014, 19:55) *
QUOTE (giro1991 @ May 6 2014, 18:46) *
If you can attenuate a full scale wave, then it must to be possible to boost a full scale wave, in the dsp realm.
This is my point. ^
No one ever denied this except people with woefully incorrect concepts of digital audio, as you have had and continue to have

No one denied It no, but I'm now asking for an implementation that will do it.
I should have excluded my means and approached hydrogen audio with more understanding on the subject. My bad.
I thought it would have been an obvious answer. It was shown to me through theory and analogies very well (thankyou) and I appreciate this, but I'm really after a useable implementation/dsp/vst that can do it (mulitply).

QUOTE
Dither does not increase bit depth in any useful way whatsoever

But it still increases it, which is as close to "multiplication" I could find in dsp form. (before I knew about simply 'multiplication).
The link I posted to 32bit dither code is as close as I got, but it isn't in useable form.

QUOTE
You are sampling, at best, intermediate points representing no additional depth/fidelity/quality whatsoever. Big deal! And your silly idea of dither will ensure that not only are you gaining no quality, you’re actually adding noise for no reason.

I know I'm not adding fidelity.
I can however "multiply" 16 to 32bit, upsample/sample, then dither down to any depth with more precision (regardless of output rate)
I (now) do not intend to use dither to upconvert (my mistake), but to convert from 32bit to a lower depth.
It is clear how its used in quantization (higher depth to lower) which believe it or not, will favour more precision. (4bn slices)
What confused matters here was the fact that dither can also multiply (A simple line of code in the dither dsp i'm sure)
So, aplogies for that.
Also, at such depths dither will be inaudible.

QUOTE (xnor @ May 6 2014, 22:41) *
giro1991, by disabling that filter you will see square waves on the output of your DAC even if you resample to 48 kHz (which is actually quite an insignificant increase in sampling rate).

You trade in the distortion from the DSP chip with the extra HF roll-off, distortion, ultrasonic images ...

As I said, it would be better to buy a proper DAC...


Don't tell me 48 thousand adjustments per second is not enough to reproduce sound.
The fact that there will now be content beyond 44.1 upto 48k (added by the last dithering down stage at output), dismisses ultrasonic/aliasing issues altogether - which by the way, in playback systems were only ever associated with the problematic 44.1 from the get go - which is very true only on the playback side - similar terms, different methods get confused with recording side).

Oversampling in playback systems would not be needed at all if CDs were 48000 from the start I believe.
That and the fact 44.1 sample rate was a compromise over space (play/runtime) regarding Redbook.

This post has been edited by giro1991: May 8 2014, 12:32


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Porcus
post May 8 2014, 12:32
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QUOTE (Thad E Ginathom @ May 2 2014, 21:14) *
Can this 8-page thread be symmetrical stretched to 16 pages? Would that improve the resolution? What will happen to the SNR?


It could improve readability, if you have applied too much vertical compression in order to get each page into a widescreen monitor.

I wonder whether the OP is looking for a way to stretch dynamic range.


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pdq
post May 8 2014, 13:23
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QUOTE (giro1991 @ May 8 2014, 07:10) *
QUOTE (saratoga @ May 6 2014, 19:16) *
For the millionth time, all processing these days is done in floating point, so this happens automatically.

Yes it is always used but is multiplication always used?

QUOTE (saratoga @ May 6 2014, 19:16) *
QUOTE (giro1991 @ May 6 2014, 13:46) *
then it must to be possible to boost a full scale wave, in the dsp realm.

Sure, in floating point, multiply by a value greater than 1.0.

Upon entering 32 float, atleast in foobar, I doubt multiplication is automatic, 16bit depth it is just passed onto 32bit float.

You are quite wrong. Each integer format and each floating point format has its own concept of what is full scale. In 8-bit integer the range is 0 to 255. In 16-bit integer the range is -32768 to +32767. In all floating point formats the range is -1.0 to 1.0.

Any conversion between any of these requires some kind of multiplication, and possibly an offset as well. All audio software handles this correctly.
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xnor
post May 8 2014, 14:38
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QUOTE (giro1991 @ May 8 2014, 13:10) *
Question is anyone willing to show me a dsp/method that does this multiplication and will do it upto 192dB vst/foobar?
If one does not exist then how easy would it be to make one?

By the time the samples hit any plugin they have already been converted to 32-bit floating point samples by foobar2000 automatically.
The plugin gets 32-bit floating point, usually processes them with at least 32-bit floating point precision and outputs 32-bit floating point samples.
As such, any plugin could achieve a much higher dynamic range than 192 dB.


QUOTE (giro1991 @ May 8 2014, 13:10) *
Don't tell me 48 thousand adjustments per second is not enough to reproduce sound.

With your DAC it isn't, for high fidelity anyway.

Don't get me wrong. You can wonderfully store audible frequencies even with 44.1 kHz, but when it comes to reproduction your non-oversampling DAC will not perform very well.

If you want to reconstruct a sampled waveform then you need to interpolate (that's basic knowledge, see sampling theorem). You just increase the sampling rate by less than 9%, so the DAC will still very much follow the samples and output square waves.
Without knowing the details of your DAC, I cannot tell if it has steep and complex analog low pass filters or not. If it doesn't then you get square waves. If it does then you have lots of phase shift even in the audible range.


QUOTE (giro1991 @ May 8 2014, 13:10) *
The fact that there will now be content beyond 44.1 upto 48k (added by the last dithering down stage at output), dismisses ultrasonic/aliasing issues altogether - which by the way, in playback systems were only ever associated with the problematic 44.1 from the get go - which is very true only on the playback side - similar terms, different methods get confused with recording side).

No, it does not dismiss those issues. Even if you have a perfect resampler that removes images from 22.05 to 24 kHz, you said yourself that you are adding shaped noise again. Also, images will still be created at ~26 kHz upward. Their suppression will depends on the analog low pass filters of your DAC.

This is also not only a D/A-conversion problem, see sampling theorem.


QUOTE (giro1991 @ May 8 2014, 13:10) *
Oversampling in playback systems would not be needed at all if CDs were 48000 from the start I believe.

Well, you're wrong.
Oversampling is still used with even 96 kHz and higher for various reasons, like making the analog low pass filters in the DAC very simple and cheap, reducing phase shift in the audible range, ...

This post has been edited by xnor: May 8 2014, 14:40


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Kohlrabi
post May 8 2014, 14:50
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QUOTE (giro1991 @ May 8 2014, 13:10) *
Yes it is always used but is multiplication always used?
QUOTE (giro1991 @ May 8 2014, 13:10) *
Upon entering 32 float, atleast in foobar, I doubt multiplication is automatic, 16bit depth it is just passed onto 32bit float.
QUOTE (giro1991 @ May 8 2014, 13:10) *
Question is anyone willing to show me a dsp/method that does this multiplication and will do it upto 192dB vst/foobar?
QUOTE (giro1991 @ May 8 2014, 13:10) *
I thought it would have been an obvious answer. It was shown to me through theory and analogies very well (thankyou) and I appreciate this, but I'm really after a useable implementation/dsp/vst that can do it (mulitply).
Contrary to what you might believe, multiplication is a pretty common operation in IT and programming. How numbers are represented and converted has been mentioned several times here.

QUOTE (giro1991 @ May 8 2014, 13:10) *
I can however "multiply" 16 to 32bit, upsample/sample, then dither down to any depth with more precision (regardless of output rate)
Sampling and upsampling are absolutely not the same thing. I won't accept any more humpty-dumptyisms here. If a signal contains sufficient information in the first place, a signal sampled with 32 bits will have a lot more information, compared to the same signal sampled with 16 bits and then upsampled to 32 bits, which will not have this information. This is all regardless of dither.

QUOTE (giro1991 @ May 8 2014, 13:10) *
Don't tell me 48 thousand adjustments per second is not enough to reproduce sound.
The fact that there will now be content beyond 44.1 upto 48k (added by the last dithering down stage at output), dismisses ultrasonic/aliasing issues altogether - which by the way, in playback systems were only ever associated with the problematic 44.1 from the get go - which is very true only on the playback side - similar terms, different methods get confused with recording side).
Playing with the digital audio buzzword generator again?

QUOTE (giro1991 @ May 8 2014, 13:10) *
My bad.
QUOTE (giro1991 @ May 8 2014, 13:10) *
So, aplogies for that.
It is no longer acceptable that you continue to ignore all answers given here, and just continue to post nonsense. It doesn't do do just write "my bad" and keep going. Nobody cares for your apologies, and I cannot in good conscience have regulars waste any more time on this thread. Hence, this thread is now closed.


Congratulations, db1989 will hand out your reward. emot-toot.gif

This post has been edited by Kohlrabi: May 8 2014, 15:06


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