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Topic: Encoding old material at 96KHz question (Read 2015 times) previous topic - next topic
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Re: Encoding old material at 96KHz question

Reply #25 –
@greynol : That's pretty much what I was asking in my first post, not for down sampling but recording with an ADC. I didn't get a clear answer yet, but it's hard for me to conceive that an old audio, or a phone conversation, would sound better in 96KHz than in 48Khz. For what I understand, you should capture in 24bit for fidelity, and choose the sample rate according to the audio frequency of the signal, and it's enough to be perfect.

Re: Encoding old material at 96KHz question

Reply #26 –
24 bits for overhead. 16 bits are more than adequate for that material unless you squander them.
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 #27 –
@greynol And what about the digitalized material? It may be safe to change sample rate to 48KHz, but is it safe to change bit depth to 16 bit while it was captured in 24bit ? (It's shitty audio so...).

Re: Encoding old material at 96KHz question

Reply #28 –
With dither the perceived dynamic range of 16 bits can be as high as 110dB, perhaps more. We’re talking permanent hearing loss.  And I haven’t taken into account the ambient noise in the listening environment which only raises the minimum SPL required in order to hear the softest sound. The quietest possible listening environment that includes the necessary air to breathe is around 30dB SPL. Now add 110 dB.

So you tell me, is 16 bits insufficient for consumer audio, “shitty” or otherwise?
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 #29 –
Quote from: andiandi on
it's hard for me to conceive that an old audio, or a phone conversation, would sound better in 96KHz than in 48Khz.
It won't, and a "new" audio won't too. Humans don't hear ultrasound and there won't be any other difference unless there's a huge issue in DAC or in the process of downsampling or recording that was used to create this recording.

Re: Encoding old material at 96KHz question

Reply #30 –
Quote from: magicgoose on
Quote from: andiandi on
it's hard for me to conceive that an old audio, or a phone conversation, would sound better in 96KHz than in 48Khz.
It won't, and a "new" audio won't too. Humans don't hear ultrasound and there won't be any other difference unless there's a huge issue in DAC or in the process of downsampling or recording that was used to create this recording.

Thanks for your reply. That answers my initial post ! I guess I'll do some comparison tests between many downsampling software to have a wider view about those.

Re: Encoding old material at 96KHz question

Reply #31 –
Quote from: andiandi on
you should capture in 24bit for fidelity
In theory , recording in 24 bits sometimes can be useful, because it allows you to leave more headroom to avoid clipping and at the same time not  to drown quiet sounds in ADC noise. But when recording and all processing is finished, there is no need in 24 bits.

Quote from: andiandi on
I guess I'll do some comparison tests between many downsampling software to have a wider view about those.
Here is comparison of different resamplers - http://src.infinitewave.ca/

Re: Encoding old material at 96KHz question

Reply #32 –
Quote
In theory , recording in 24 bits sometimes can be useful, because it allows you to leave more headroom to avoid clipping and at the same time not  to drown quiet sounds in ADC noise. But when recording and all processing is finished, there is no need in 24 bits.

Pretty clear and concise.

Quote
Here is comparison of different resamplers - http://src.infinitewave.ca/

Thanks. Its very helpful.

Re: Encoding old material at 96KHz question

Reply #33 –
@Rollin : I don't know if I should believe http://src.infinitewave.ca/...

It says that DBPoweramp resampler is excellent


However, I tried with DBPoweramp 16.2, and it gives me the same result as DBpoweramp resampler DSP in Foobar2k, so much loss




Is this website really comparing resamplers quality or something else ?

Re: Encoding old material at 96KHz question

Reply #34 –
Quote from: andiandi on
@Rollin : I don't know if I should believe http://src.infinitewave.ca/...

It is a good test, you should believe it.

Quote from: andiandi on
However, I tried with DBPoweramp 16.2, and it gives me the same result as DBpoweramp resampler DSP in Foobar2k, so much loss

Probably some difference in level between them since when you subtract you see an attenuated copy of the original spectrum.  Trying to subtract them is a bad way to test since if you get some difference you then have to rule out all of the different things that could cause it.

Re: Encoding old material at 96KHz question

Reply #35 –
Quote
Probably some difference in level between them since when you subtract you see an attenuated copy of the original spectrum.  Trying to subtract them is a bad way to test since if you get some difference you then have to rule out all of the different things that could cause it.

It's a bad way to compare ? Then why with the same tests (inverting and mixing tracks), I get perfect results in the spectrum with Sox (and by the way, Sox is excellent in src.infinitewave.ca) ?

And what is a good way to compare my down sampled 48KHz track and its 96KHz counterpart and be sure that I do not lose anything relevant ?

Re: Encoding old material at 96KHz question

Reply #36 –
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 #37 –
@Wombat I can see with Deltawave that the waveform is slightly changed/deteriorated between the source 96KHZ and the resampled 48KHz. And in Audacity too btw.

But, is the waveform only changed because of the noise cut above 24KHz ? (after the 96KHz -> 48KHz conversion)

So is inaudible noise (>20KHz) also part of the waveform, so that it can be changed through a resample ?

Because right now, I don't know what to think : is my 96 to 48 resample a success, in that case I could delete source files, or is there a downgrade on my audios...

EDIT : I tested with a upsampling, and it's the same. A 44.1KHz tracks upsampled in 96KHz has its waveform changed. Why is that ? does it means deterioration in a bad way ?

Re: Encoding old material at 96KHz question

Reply #38 –
You must decide what you want. Resampling is lossy (except if you multiply the sampling rate by an integer number without interpolation). Is it audible? Down to 32kHz most often not.
In your case I'd downsample everything with SOX or fb2k to 16bit 24kHz, or eventually 16bit 48kHz with dithering, or even use some lossy codec...
Don't look at graphs and don't listen to extracted difference unless you EXACTLY know what you're doing. It is enough to shift one of the files by 1 sample to get huge difference, while in reality there will be none. Graphs may exaggerate things which are inaudible. Don't compare audio with your eyes (you wouldn't compare pictures with your ears would you). If you can't accept a loss that you can't hear, stay lossless.

Re: Encoding old material at 96KHz question

Reply #39 –
Quote from: andiandi on
However, I tried with DBPoweramp 16.2, and it gives me the same result as DBpoweramp resampler DSP in Foobar2k, so much loss
IIRC, the dbpoweramp/SSRC resampler causes some kind of sub-sample shift, which makes it impossible to use the difference method. Someone mentioned it on another resampler thread. It does not mean it is worse quality.
Proverb for Paranoids: "If they can get you asking the wrong questions, they don't have to worry about answers."
-T. Pynchon (Gravity's Rainbow)

Re: Encoding old material at 96KHz question

Reply #40 –
Quote from: sTisTi on
IIRC, the dbpoweramp/SSRC resampler causes some kind of sub-sample shift, which makes it impossible to use the difference method. Someone mentioned it on another resampler thread. It does not mean it is worse quality.
Exactly this should be compensated by the deltawave software. Normaly if you read the results from resampling it shows the difference very well. This is pretty much the noisefloor from dither or/and bit reduction.
Edit: attached 24-96 to 16-44.1, foobar SSRC
Is troll-adiposity coming from feederism?
With 24bit music you can listen to silence much louder!

Re: Encoding old material at 96KHz question

Reply #41 –
Quote from: sTisTi on
Quote from: andiandi on
However, I tried with DBPoweramp 16.2, and it gives me the same result as DBpoweramp resampler DSP in Foobar2k, so much loss
IIRC, the dbpoweramp/SSRC resampler causes some kind of sub-sample shift, which makes it impossible to use the difference method. Someone mentioned it on another resampler thread. It does not mean it is worse quality.

Alright I understand better. I'll stick to SOX though, he's good everywhere, with every test.

Quote
Exactly this should be compensated by the deltawave software. Normaly if you read the results from resampling it shows the difference very well. This is pretty much the noisefloor from dither or/and bit reduction.
Edit: attached 24-96 to 16-44.1, foobar SSRC

I see... I think that for my source the difference between 96KHz and 48/32 even 24KHz would be inaudible. But what about 24-Bit to 16-Bit ?. In your case, deltawave shows a clean result. But look what I have with my source (96KHz 24Bit to 96KHz 16-bit, with Sox, Dither and no dither)



First and foremost, do you confirm that a comparison with deltawave is more reliable than inverting in Audacity + Spek ? (in which I get NO difference between 24/16 bit, as if it was not deteriorated)



Also, are those differences relevant then ?

Re: Encoding old material at 96KHz question

Reply #42 –
The way you're doing subtraction is not reliable compared to almost any other test.

Re: Encoding old material at 96KHz question

Reply #43 –
@andiandi
SoX resampler has no intersample drift so is much easier to compare.
Most of all don't expect to find out anything new. Transparency from resampler software is a long solved problem.
Is troll-adiposity coming from feederism?
With 24bit music you can listen to silence much louder!

Re: Encoding old material at 96KHz question

Reply #44 –
24 vs 16:  I already told you.

With that recording you don’t even need to use dither, not that you shouldn’t use dither.

What’s with all the fear uncertainty and doubt?  You will never tell the difference with either of the resamplers you tried. This obsessing really isn’t worth your time.

44.1/16 is what I would recommend, or go lossy. If opus use 48k, 44.1 with the others since they are tuned at that sample rate. 48k is native in opus.
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 #45 –
Quote from: saratoga on
The way you're doing subtraction is not reliable compared to almost any other test.

Ok got it.

Quote from: Wombat on
@andiandi
SoX resampler has no intersample drift so is much easier to compare.
Most of all don't expect to find out anything new. Transparency from resampler software is a long solved problem.

I'll stick with SoX, I don't want to get on all that tbh.

Quote from: greynol on
24 vs 16:  I already told you.

With that recording you don’t even need to use dither, not that you shouldn’t use dither.

What’s with all the fear uncertainty and doubt?  You will never tell the difference with either of the resamplers you tried. This obsessing really isn’t worth your time.

44.1/16 is what I would recommend, or go lossy. If opus use 48k, 44.1 with the others since these are how those lossy codecs are tuned. 48k is native in opus.


It's just that it's rare material, but i'm sick of keeping 800GB files on my hard drive. That's why I wanted to be sure before doing anything.

Btw, I just did some ABX tests, and it proves that what have been said before is true. Hearing is so much better than looking.

I compared the source in 96KHz with resamples; Here are the results :

96KHz 24-bit = 48KHz 24-bit = 48KHz 16-bit = 32KHz 24-bit = 32KHz 16-bit.

However 24KHz 24-bit is downgraded. I got 16/16 right on ABX, while I couldn't tell other tracks apart (I tried hard, but it's impossible, even with a dB gain)

So I'll stick to 32KHz 16-bit.

I have a last question though. I read somewhere that audio frequencies needed some "free space". Since my sources have been broadcasted just below 12KHz (maybe at 11.3KHz, 11.8KHz, idk), does it explain why there is a deterioration when resampling at 24KHz (24KHz rate= 12KHz frequency, so "too" close to the maximum).

Re: Encoding old material at 96KHz question

Reply #46 –
So, assuming stereo, you went down from 4608 kbps to 1024 kbps - from 800GB to ~178GB.
Now, ABX your files with Opus@128 kbps and consider shaving off another ~155GB down to ~22GB...
Or maybe 96 kbps? Or...even...95kbps  :D

Re: Encoding old material at 96KHz question

Reply #47 –
Quote from: rutra80 on
So, assuming stereo, you went down from 4608 kbps to 1024 kbps - from 800GB to ~178GB.
Now, ABX your files with Opus@128 kbps and consider shaving off another ~155GB down to ~22GB...
Or maybe 96 kbps? Or...even...95kbps  :D

Yeah, but actually I need those audio for editing so I prefer to keep them lossless :)
But going from 96KHz 24-Bit to 32KHz 16-bit will free so much space.

Re: Encoding old material at 96KHz question

Reply #48 –
I ask my question again for those who missed it :

I compared the source in 96KHz with resamples; Here are the results :

96KHz 24-bit = 48KHz 24-bit = 48KHz 16-bit = 32KHz 24-bit = 32KHz 16-bit.

However 24KHz 24-bit is downgraded. I got 16/16 right on ABX, while I couldn't tell other tracks apart (I tried hard, but it's impossible, even with a dB gain)

So I'll stick to 32KHz 16-bit.

I have a last question though. I read somewhere that audio frequencies needed some free space/headroom. Since my sources have been broadcasted just below 12KHz (maybe at 11.3KHz, 11.8KHz, idk), does it explain why there is a deterioration when resampling at 24KHz (24KHz rate= 12KHz frequency, so "too" close to the maximum).

Re: Encoding old material at 96KHz question

Reply #49 –
IIRC it was recommended to not go over 20kHz on CDDA because some DACs had problems recreating near Nyquist frequencies.
In case of your PC probably everything gets resampled to something like 48kHz anyway so it shouldn't matter.
I would bet that there's significant amount of noise in your recordings and losing it just above 12kHz could be audible...
Also, at 32kHz you keep the NTSC horizontal refresh frequency which you may hear.
 
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