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Encoding old material at 96KHz question

Hi.

I've a question regarding high sampling rate encoding. Let's say that I encode the audio track of an old VHS cassette at 96KHz.

That audio is broadcasted at a frequency of 12KHz, so anything above this is irrelevant (white noise, NTSC horizontal frequency...).

Would the encoded lossless audio of that cassette in 96KHz have relevant data in 96KHz ?
I mean, if I convert the FLAC 96KHz to FLAC 48 KHz, would I lose quality ? Because, it might create some echo or other relevant data in more high frequencies.

I don't personally want to convert old material to 96KHz, but some shared materials are only available in 96KHz.

Thank you

Re: Encoding old material at 96KHz question

Reply #1
"CD quality" (16-bit, 44.1kHz) is better than human hearing.    i.e. If you take a high resolution original and downsample to 16/44, you won't hear a difference in a proper-blind ABX listening test.

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Let's say that I encode the audio track of an old VHS cassette
If you are lucky enough to have VHS Hi-Fi it was the better than vinyl or cassettes!

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but some shared materials are only available in 96KHz.
If you are talking about digitized vinyl there is a myth that analog has "infinite resolution" but in reality analog resolution is limited by noise.  Vinyl can also extend beyond 20kHz (and beyond the 22,050Hz CD limit) but you can't hear that high, what's on records at those frequencies is mostly noise, digital frequency response is flatter in the audible range (where it really counts), and records "struggle" with very-low frequencies whereas digital can go down to DC (zero Hz).

24-bits/96kHz is the "pro studio standard."

Re: Encoding old material at 96KHz question

Reply #2
> "CD quality" (16-bit, 44.1kHz) is better than human hearing.    i.e. If you take a high resolution original and downsample to 16/44, you won't hear a difference in a proper-blind ABX listening test.

I'm prety conviced of that, and I don't like too high quality rips. Human hearing is pretty poor when we think about it.

But that's not really my point. I donwloaded audios from old cassettes, containing recordings of TV Broadcasts with a pretty poor quality (under 15KHz of frequency range). The thing is that those rips are encoded at 96KHz 24-bit.

When I converted those tracks to 48KHz 24-bit and made an audio difference, I still get a relevant track with the all the sound (very low volume and not very audible, but since it's silence, it tends to show that converting from 96KHz to 48KHz; even an old material, will make data dissappear).

I'm not sure though..

Re: Encoding old material at 96KHz question

Reply #3
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But that's not really my point.
OK.  What is your point?

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When I converted those tracks to 48KHz 24-bit and made an audio difference, I still get a relevant track with the all the sound (very low volume and not very audible
If you get dead silence, yes that proves there is no difference.  

If you don't get silence it proves there is a "data" difference but it does NOT prove there is an audible difference.    Just for example, if you add 10 milliseconds of delay to the copy there is no difference in the sound but subtraction will give you a very "loud" comb-filtered result.    A difference in level will also give a non-silent result. 

Or as a more obvious example, you can invert a copy and it will sound identical.   But, if you then subtract from the original you are now "subtracting a negative" (adding) and you'll double the volume and possibly push the levels into clipping (distortion).

 There can be other differences that you can't hear in an ABX test, or if you listen to the high-resolution original today and the downsampled copy tomorrow.

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it tends to show that converting from 96KHz to 48KHz; even an old material, will make data dissappear).
Of course, you are throwing-away half of the samples so you are throwing-away half of the data (assuming the same bit-depth).   In fact, downsampling requires low-pass filtering to prevent aliasing so you are not simply throwing-away half of the samples.


P.S.
The sound of the difference is not the same as the difference in the sound...


Re: Encoding old material at 96KHz question

Reply #4
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If you don't get silence it proves there is a "data" difference but it does NOT prove there is an audible difference.    Just for example, if you add 10 milliseconds of delay to the copy there is no difference in the sound but subtraction will give you a very "loud" comb-filtered result.    A difference in level will also give a non-silent result. 

Or as a more obvious example, you can invert a copy and it will sound identical.   But, if you then subtract from the original you are now "subtracting a negative" (adding) and you'll double the volume and possibly push the levels into clipping (distortion).

There can be other differences that you can't hear in an ABX test, or if you listen to the high-resolution original today and the downsampled copy tomorrow.

Very interesting.


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Of course, you are throwing-away half of the samples so you are throwing-away half of the data (assuming the same bit-depth).   In fact, downsampling requires low-pass filtering to prevent aliasing so you are not simply throwing-away half of the samples.

It make sense, but I guess low-passing will just decrease quality loss when converting 96KHz -> 48KHz, but it'll still lose something.
e.g. I low pass the 96KHz track frequency to 12KHz (but keep it at 96KHz sampling rate) and then convert it to 48KHz. Would both tracks be the same or not ?

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P.S.
The sound of the difference is not the same as the difference in the sound...

Really ? so is there a better way to tell if two songs are the exact same ? for instance after down sampling.

Re: Encoding old material at 96KHz question

Reply #5
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Of course, you are throwing-away half of the samples so you are throwing-away half of the data (assuming the same bit-depth).   In fact, downsampling requires low-pass filtering to prevent aliasing so you are not simply throwing-away half of the samples.

It make sense, but I guess low-passing will just decrease quality loss when converting 96KHz -> 48KHz, but it'll still lose something.

Assuming you do it correctly you should lose a bunch of ultrasonic noise but nothing else.

e.g. I low pass the 96KHz track frequency to 12KHz (but keep it at 96KHz sampling rate) and then convert it to 48KHz. Would both tracks be the same or not ?

Assuming you mean 20 or 24 KHz, yes they should be the same. 

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Really ? so is there a better way to tell if two songs are the exact same ? for instance after down sampling.

Upsample them to a common sampling rate, align them in time, and then subtract them. 

Re: Encoding old material at 96KHz question

Reply #6
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Assuming you mean 20 or 24 KHz, yes they should be the same.

20 KHz of audio frequency ? but it's already in more than 20KHz, but the track is just 12KHz at its source. From what @DVDdoug said, I understand that you should low pass to the closest frequency to the source and then down sample ?

96KHz is too much in filesize. My goal is to convert a poor quality 96KHz recorded track to 48KHz without any loss at all.

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Upsample them to a common sampling rate, align them in time, and then subtract them.

Alright I'll keep that in mind thanks.

Re: Encoding old material at 96KHz question

Reply #7
My goal is to convert a poor quality 96KHz recorded track to 48KHz without any loss at all.
Convert the 96k track to 48k using foobar2000.  Then do an ABX comparison also in foobar2000 to see if you can tell them apart.

Re: Encoding old material at 96KHz question

Reply #8
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Assuming you mean 20 or 24 KHz, yes they should be the same.

20 KHz of audio frequency ? but it's already in more than 20KHz, but the track is just 12KHz at its source. From what @DVDdoug said, I understand that you should low pass to the closest frequency to the source and then down sample ?

I don't understand the question.  However, if you are downsampling to 48KHz, 24 KHz is the highest low pass you could use, and 20kHz would be a more common one.  You should not use 12 KHz. 

Re: Encoding old material at 96KHz question

Reply #9
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From what @DVDdoug said, I understand that you should low pass to the closest frequency to the source and then down sample ?
That's not exactly what I said...  In this case filtering should make no difference but it will be automatically filtered by the downsampling algorithm.

The Nyquist sampling theory says you can't have audio any higher than half the sample rate.    Or to simplify, you need to sample at least twice per cycle so you can sample the positive half of the waveform at least once and the bottom half at least once per cycle.  If your audio is higher than that you get aliasing (false frequencies somewhere below the Nyquist limit).    So, every ADC (analog-to-digital converter) has a low-pass filter and every downsampling algorithm has a low-pass filter  (i.e. an anti-aliasing filter).

Practically speaking, if our audio only goes to 12kHz you don't need to low-pass filter (when going from 96kHz to 48kHz) but every downsampling algorithm filters and all of the data is "mathematically" altered even if it's not altered significantly enough to alter the sound.

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Really ?
Try the test I suggested.   Delay an exact copy by 10 milliseconds and you of course it will sound exactly the same (when played by itself).  There is no difference in sound.    

Then subtract it from the original.    You'll get a VERY LOUD weird sounding difference file.  That's the sound of the difference.

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so is there a better way to tell if two songs are the exact same ? for instance after down sampling.
An ABX test can help tell you if they sound  the same.   Of course, the data can't be the same if one file has twice the number of samples..

Re: Encoding old material at 96KHz question

Reply #10
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Convert the 96k track to 48k using foobar2000.  Then do an ABX comparison also in foobar2000 to see if you can tell them apart.

I tried and can't tell any difference between both tracks. To me they're all the same when I listen to them. But not really (see below)

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I don't understand the question.  However, if you are downsampling to 48KHz, 24 KHz is the highest low pass you could use, and 20kHz would be a more common one.  You should not use 12 KHz.

But is it impossible ? I saw VHS audio rips with a 15KHz cutoff, but in 48KHz and they sound fine.

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Practically speaking, if our audio only goes to 12kHz you don't need to low-pass filter (when going from 96kHz to 48kHz) but every downsampling algorithm filters and all of the data is "mathematically" altered even if it's not altered significantly enough to alter the sound.

In my case, I down sampled using a dsp in foobar, and the difference is pretty clear : I lose "relevant" data (it's basically the full track in low volume and different equalization). It's the same if I upsample again to 96KHz.



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Try the test I suggested.   Delay an exact copy by 10 milliseconds and you of course it will sound exactly the same (when played by itself).  There is no difference in sound.   

Then subtract it from the original.    You'll get a VERY LOUD weird sounding difference file.  That's the sound of the difference.

You're right. With sox, it gives me an high frequency sound difference.


I think I begin to go off topic so I'll clarify my point.

1. Let's say that I want to archive some rare audio from VHS cassettes, and that the only encoded files that I find online are in 96KHz FLAC, is it "over" ? Is there no way to archive those audio in a lower sampling rate with the exact same quality, meaning that the difference of those files is pure silence ?

2. And also : if I rip the audio of a VHS in 96KHz, and then I rip the exact same audio at 48KHz : would the 48KHz sounds the same since the full audio is captured in 48KHz and it's not a down sample ?

I mean would it be perfectly faithful to the source, contrary to a converted 96KHz -> 48KHz even in both are in 48KHz ?

Re: Encoding old material at 96KHz question

Reply #11
1. Let's say that I want to archive some rare audio from VHS cassettes, and that the only encoded files that I find online are in 96KHz FLAC, is it "over" ? Is there no way to archive those audio in a lower sampling rate with the exact same quality, meaning that the difference of those files is pure silence ?

You're not going to get pure silence since you're removing all of the ultrasonic noise in the original file.

2. And also : if I rip the audio of a VHS in 96KHz, and then I rip the exact same audio at 48KHz : would the 48KHz sounds the same since the full audio is captured in 48KHz and it's not a down sample ?

These both do exactly the same thing, so you will not hear any difference between them.

Re: Encoding old material at 96KHz question

Reply #12
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You're not going to get pure silence since you're removing all of the ultrasonic noise in the original file.

You speak about "ultrasonic noise" but if you look at the audacity pic above, it's clearly not ultrasonic. I can hear the full track and voices but in low volume. I'd be OK if it was just white noise or ultrasonic sounds.

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These both do exactly the same thing, so you will not hear any difference between them.

The same thing ? so you're telling that ripping a low quality old muffled track in 96KHz is better than ripping it in 48KHz ? so even in 192KHz it'd be better ? How is that ? The track is very very low quality and I still lose relevant data when down sampling...

So if the VHS track is mono, muffled, buzzy, and from the 80's, it'd still be closer to the source to rip it in 96KHz than 48KHz ?


Re: Encoding old material at 96KHz question

Reply #13
Not sure what to tell you.  I just took a 24/96 track and used foobar2000 to convert it to 24/48.  You can see the "mix" below; no difference at all.  If you couldn't ABX it, why are you worried?  Certainly 24/48 can faithfully capture everything on a VHS audio track.


Re: Encoding old material at 96KHz question

Reply #14
@Apesbrain : Maybe your track is a upsample of a 48KHz track...

Because I don't have the same result with a track captured at 96KHz as you can see on the screenshot above.

Re: Encoding old material at 96KHz question

Reply #15
Perhaps you can provide a sample of each version of the same portion of the track that is no longer than 30 seconds in order to comply with our rules.

I am certain there is an error somewhere in your process.
Is 24-bit/192kHz good enough for your lo-fi vinyl, or do you need 32/384?

Re: Encoding old material at 96KHz question

Reply #16
You may play around with deltawave https://deltaw.org/
Is troll-adiposity coming from feederism?
With 24bit music you can listen to silence much louder!

Re: Encoding old material at 96KHz question

Reply #17
@greynol Sure. Here is a 96KHz sample attached. You can try converting it 48KHz, there will be quality loss :/

@Wombat I'm not familiar with that software but I'll try thanks.

Re: Encoding old material at 96KHz question

Reply #18
It's your process.  All I got was noise averaging at less than -130 dB.

Re: Encoding old material at 96KHz question

Reply #19
@Chibisteven How did you do then ?!

Because I just down sampled with rather fooobar2000 or eac3to, and then make the difference with track invert in Audacity + Mix.
And also with sox difference command.

All the results are the same.

EDIT : I tried with your 48KHz track and the difference is pure silence. How did you down sampled ? is there different way of doing it ?

Re: Encoding old material at 96KHz question

Reply #20
@Chibisteven How did you do then ?!

Because I just down sampled with rather fooobar2000 or eac3to, and then make the difference with track invert in Audacity + Mix.
And also with sox difference command.

All the results are the same.

EDIT : I tried with your 48KHz track and the difference is pure silence. How did you down sampled ? is there different way of doing it ?

I downsampled using the most recent version of SoX (0.8.7) for foobar2000 which defaults to Best Quality.  Dragged both into the most recent version of Audacity for which I have the resampling set to Best Quality (slowest) and the dither turned off and inverted the 48 KHz one.

Re: Encoding old material at 96KHz question

Reply #21
@Chibisteven

Okay I made some test and I came to a strange conclusion to me : the down sample can be "perfect" with a software, and terrible with another.

Look at the difference with z down sample made by DBpoweramp DSP in Foobar :



And now with TAudioConverter (I get the same results with Sox plugins for foobar or Audacity internal down sample):



What the heck ?

Can you confirm that DBPoweramp plugin do this with the sample I attached before ?

It answer one of my question. I always wondered why sometimes, the differences with two files were pure silence and sometime not. It's just that I don't stick to one software to down sample...


Re: Encoding old material at 96KHz question

Reply #22
With a competent resampler, yes, it is perfectly fine.
Is 24-bit/192kHz good enough for your lo-fi vinyl, or do you need 32/384?

Re: Encoding old material at 96KHz question

Reply #23
@greynol : I really thought until now that resampling was just a mathematical operation, similar with every software. So is there well known competent resamplers ? On my tests, Audacity, Taudio and SOX gave the same results but who knows, maybe I should stick to best existing resampler to archive my files.

Re: Encoding old material at 96KHz question

Reply #24
Not all filter designs are the same.  I rarely resample and I can’t say I’ve done any extensive testing.  Resampling a signal that is already band-limited to 12k down to 32k shouldn’t be very challenging.  There is plenty of room in the transition band.
Is 24-bit/192kHz good enough for your lo-fi vinyl, or do you need 32/384?

 
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