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What is the point of upsampling?

Holmz

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If I may ask, is there any website or source I can visit to hear what imaging artifacts even are? Perhaps even an artifically induced test just to demonstrate what such a thing even sounds like?

 

valerianf

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"There is no value in upsampling".
Yes and no.
No: if it is just a DAC connected to a streamer as input and that provides an analog signal that will goes through an analog amplifier.

Yes: if it is a DAC or ADC that are connected to a DSP chain (i.e.: an AVR).
The DSP is a digital machine that will change the sound by alteration of each digital sample.
Smaller is the size of the digital sample (quantification), better is the end result.
The problem is that the DPS is working at the speed of the input ADC (for an analog input) or at the computer audio standard sampling rate (for a digital input).
So yes, we need an input digital oversampling at 96khz (ADC if analog input), a DPS working at 96khz and an output DAC running at 96 khz.
I am dreaming of such an AVR running at 96khz or higher, as room correction becomes a standard feature.

Alas, even if we feed the nowadays AVR with an HD digital source (some streaming are at a 196khz sampling rate), the AVR will run the room correction at 48khz!
In that case, for sure we will not hear any improvement from a CD quality (44.1khz).

If somebody has found an AVR that has a DSP running at a 96 khz sampling rate with any of the input sources, it would be interesting to get some user feedback.
 

Mnyb

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Is it really true that there is no value in upsampling? For example, a CD signal sampled at 44.10kHz has a nominal maximum usable frequency of 22.05kHz. If we want a bandwidth out to 20kHz, then the analog filter will need to roll off at a rate of 680dB/octave or so (96dB of attenuation over 2.05kHz). That's very steep. Hence, wouldn't the analog filter produce a lot of phase shift within the audio band?

On the other hand, upsampling by 4x would require an analog filter that covers the range from 20kHz to 88.2kHz while producing the same final 96dB of attenuation. That's only about 45dB/octave, more than an order of magnitude less than the original filter with no oversampling. This can permit the use of reconstruction filters that produce less phase shift in the audio band.

Of course, on a sine wave test, the oversampled waveform will be more or less identical to the original waveform, subject to some variation due to the arithmetic that has been performed on it in the digital domain by the digital filtering that's required.
I think the discussion is about external uppsamplers in software, all DAC chips does this internally anyway
 

Julf

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So yes, we need an input digital oversampling at 96khz (ADC if analog input), a DPS working at 96khz and an output DAC running at 96 khz.

Why? You need some extra bits (resolution) to deal with rounding errors, but pretty much all DSPs do that anyway.

Alas, even if we feed the nowadays AVR with an HD digital source (some streaming are at a 196khz sampling rate), the AVR will run the room correction at 48khz!
In that case, for sure we will not hear any improvement from a CD quality (44.1khz).
You are making the fallacious assumption that the improvement from a higher sample rate is audible.
 

Blumlein 88

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Why? You need some extra bits (resolution) to deal with rounding errors, but pretty much all DSPs do that anyway.


You are making the fallacious assumption that the improvement from a higher sample rate is audible.
You run across this issue in DAW software. Some effects really need upsampling not to sound noticeable in unwanted effects. I don't know enough to know why. I suspect it is leftover code from back when they were really trying to use as few resources as possible. In other words bad algorithms. That was one reason some early digital mixing/mastering really sounded better at 96 khz. Now it isn't needed, most DAWs for the effects where that mattters will upsample do the DSP, and downsample the result automatically. So it had some truth at one time, and probably doesn't anymore.

Many did make the false assumption the higher sample rate caused the audible improvement. Indirectly it did, but with proper processing it would no longer matter.
 

Julf

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You run across this issue in DAW software. Some effects really need upsampling not to sound noticeable in unwanted effects. I don't know enough to know why. I suspect it is leftover code from back when they were really trying to use as few resources as possible. In other words bad algorithms.
Indeed. And many DAWs do time and pitch shifting (basically changing sample rate), so need "headroom" on the sample rate. Room correction DSP doesn't.
 

voodooless

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You run across this issue in DAW software. Some effects really need upsampling not to sound noticeable in unwanted effects. I don't know enough to know why.
This gives a fairly good overview:

 

Julf

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mansr

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You run across this issue in DAW software. Some effects really need upsampling not to sound noticeable in unwanted effects. I don't know enough to know why.
Non-linear effects create harmonics of the input frequencies. Any harmonics above the Nyquist frequency will fold back as aliases. Operating at a higher sample rate reduces the amount of such aliasing since less of generated harmonic content then falls above Nyquist.
 

Sokel

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Non-linear effects create harmonics of the input frequencies. Any harmonics above the Nyquist frequency will fold back as aliases. Operating at a higher sample rate reduces the amount of such aliasing since less of generated harmonic content then falls above Nyquist.
So,practically in every day home use:
Do or don't use upsampling?
 

Julf

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So,practically in every day home use:
Do or don't use upsampling?
Unless you are doing weird studio effects, it doesn't matter.
 

Julf

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Julf

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Unless you are looking for a more accurate analogue reconstruction.
Have you not understood the Shannon-Nyquist sampling theorem?
 
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