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Upsampling 16/44.1 collection a good idea?

Sounds ideal as a vinyl replacement activity.
I am reminded of the time that a friend visited with his ‘XXX Hi-End ‘ software, don’t get excited it was just playback software.
We spent nearly forty minutes fiddling with it but couldn’t get it to emit one note of music.
Keith
 
Sounds ideal as a vinyl replacement activity.
I am reminded of the time that a friend visited with his ‘XXX Hi-End ‘ software, don’t get excited it was just playback software.
We spent nearly forty minutes fiddling with it but couldn’t get it to emit one note of music.
Keith
I've had many debates with Peter, the author of XXX Hi-End in the past. He even sent me one of his (complex!) USB cables to test at one point. He wasn't too happy when I found that it produced no measurable effect, except for some additional noise in some configurations. I did try his player, too. It was a bit clunky to configure, but it did work for me. Can't say that it was doing something special, but it did run and play music, though the interface was particularly ugly, in my opinion :)
 
Did you ever try his ‘H’ shaped dac which always reminded me of. ’Rimmer’ from Red Dwarf .
Keith
 
Did you ever try his ‘H’ shaped dac which always reminded me of. ’Rimmer’ from Red Dwarf .
Keith
I've seen it and heard from others (and Peter) about it, but have never tested one. The claim was that it was more immune to noise generated by the PC by virtue of better isolation and low CPU usage when used with the XXX Hi-end player. Peter was trying to prove that even the act of moving a mouse on the PC would cause noise to be added for more mainstream DACs by the extra CPU utilization, while his DAC was immune.
 
PGGB is the direct result of lack of critical thinking. The author claims the error of -600dB or better for his reconstruction algorithm.
Just had a closer look at PGGB and yes, this is completely insane, throwing quad-double (256bit) math at simple audio problems. A solution in search of a problem.

For the fun of it and as I already have my own time-domain convolver in double precision, I've used a brickwall filter kernel made with rePhase for a test run of a 2x upsampler. With that filter kernel (Albrecht-11 term, -6dB @ 21500Hz for 88.2kHz target sample rate, 2048 taps, -300dB down at fs/2 and above) -- and not using any dithering for that test -- I could achieve sample identity for a full-scale signal in the 24bit integer domain for any frequency up to 21100Hz, that is, the upsampling contains all the original sample values except for an ultra narrow range at the top end. For music tracks, with much lower HF content, the match would be even better. As such, it can't get any better than this, me thinks.... and it's already overkill ;-)
 
Absolutely a bad idea!!!! DON'T Upsample Your Music Files!

I cant emphasize this enough! Don't do it! Don't do it! Don't do it!


Fixed-point upsampling will create distortion whenever an intersample peak exceeding 0 dBFS occurs. This distortion is often audible because it is quite severe when it occurs. On many CD recordings, this can occur multiple times per second.

See my whitepaper on this topic here:
Intersample Overs in CD Recordings

Intersample peaks can reach +3.01 dBFS. Recordings that are not overly compressed, or overly loud, may contain more intersample overs than recordings that are mastered for maximum loudness. In other words, your best recordings may sustain the highest damage from upsampling the files.

If a DAC is properly designed (with enough DSP headroom and enough analog headroom), the intersample peaks exceeding 0 dBFS can be rendered without distortion. Please note that most DACs clip intersample peaks because they do not have any headroom above 0 dBFS. When this clipping occurs, it is not harmonic distortion. Instead, each clip produces a burst of broadband noise. In other words, you will hear short percussive bursts of noise at every occurrence of an overload. This is not like the sound of an amplifier clipping. It may correspond to the beat of the music and may be mistaken for a percussion instrument, but it is a sound that was not in the original recording. This percussive noise artifact is easy to hear once you learn what to listen for.

Upsampling can be done without causing damage if the audio level is reduced by at least 3 dB before upsampling. We do this within our Benchmark DAC2 and DAC3 converters. It can also be done correctly within a floating point environment, but again the level will need to be reduced by at least 3 dB before saving it back to a fixed point format. For a variety of very good reasons, music recordings are only distributed in fixed-point formats. If you upsample these without a level reduction, you will do permanent damage.

Fixed-point to fixed-point upsampling "bakes" intersample DSP overloads into the upsampled product. This distortion is audible and it cannot be removed. Upsampling will damage the sound of your recordings unless you reduce the signal level.

Beware of upsampled versions of 44.1 kHz recordings. These usually have baked-in distortion. Find the 44.1/16 originals.

Unfortunately some online music services provide "high-resolution" versions of 44.1 kHz recordings that have been upsampled directly from the 44.1 kHz originals without reducing the level. Avoid these "high-resolution" versions.
Very interesting post.

I still use my Benchmark DAC1 at home. I guess I can do the attenuation on my PC and still upsample with Roon. I have found that the IMD increases using Wasapi, which has a fixed volume.
 
Upsampling can be done without causing damage if the audio level is reduced by at least 3 dB before upsampling. We do this within our
Or by upsampling into a bit depth at least 3db deeper. This is not an unknown phenomena. It simply needs to be considered in the process.

However id suggest not upsampling because it is a pointless exercise from an audibility point of view.
 
Or by upsampling into a bit depth at least 3db deeper. This is not an unknown phenomena. It simply needs to be considered in the process.

However id suggest not upsampling because it is a pointless exercise from an audibility point of view.
I'm trying some DSP, like my own FIR filters. Do you think it is still pointless?

I'm just curious, not an expert in signal processing.
 
I'm trying some DSP, like my own FIR filters. Do you think it is still pointless?

I'm just curious, not an expert in signal processing.
Upsampling even in that situation is pointless. Having extra precision (24 or even 32 bits) during processing, on the other hand, is a good idea.
 
I think if I'm going to process a signal it is a good idea to have better resolution in time and amplitude by interpolation before processing.

I insist I'm not a signal processing expert but looks right to my intuition and after any change I perform some measurements. If my preconceptions measure worse I get back too the original solution wondering why.
 
I think if I'm going to process a signal it is a good idea to have better resolution in time and amplitude by interpolation before processing.

I insist I'm not a signal processing expert but looks right to my intuition and after any change I perform some measurements. If my preconceptions measure worse I get back too the original solution wondering why.
Intuition is really not very useful when talking about digital signal theory, but yes, checking results by proper measurements is the way to go.
 
Having extra precision (24 or even 32 bits) during processing, on the other hand, is a good idea.
But even then, getting rid of the 16 bit quantisation noise is not possible. That is already in the signal.
 
But even then, getting rid of the 16 bit quantisation noise is not possible. That is already in the signal.
Indeed. You want the extra precision just to avoid adding more noise.
 
I'm trying some DSP, like my own FIR filters. Do you think it is still pointless?

I'm just curious, not an expert in signal processing.
Yes, except for your own interest. As long as the filters you are comparing are sufficiently fast with sufficient attenuation (as the default filters on most DACs are) then they won’t sound different.
 
Indeed. You want the extra precision just to avoid adding more noise.
Thus further illustrating the pointlessness. Further processing will just add more noise and distortion - albeit inaudible as long as sufficient precision in the bit depth is used.
 
Yes, except for your own interest. As long as the filters you are comparing are sufficiently fast with sufficient attenuation (as the default filters on most DACs are) then they won’t sound different.
They are not lo-pass filters. They equalise magnitude and phase in the audible region.
 
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