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Ripping Vinyl 192khz 24bit Considerations, I have some questions to the optimal way to do this
navalverde
post May 24 2011, 23:00
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Hello,

I am currently using an EMU 1616M to rip vinyl and using Sony Sound Forge 10 to record with. I am wondering what the best settings are to use for exporting the .WAV files. I have the option to record in 24bit or 32 bit at 192 Khz and save the

.WAVs in using IEEE Float or PCM. I am wondering if it would be best to record in 32bit and bounce down to 24 when I export to .WAV, or just record in 24 if there won't be any difference. No matter what I will need a 24 bit file in the end to

convert to .flac as it is the highest bit rate the format supports.


Furthermore, I am confused as to whether to use IEEE Float or PCM. I understand the IEEE float was developed for the broadcast industry while PCM is used in the red book CD standard. I'm guessing when I convert to .flac it will end up as PCM

anyway and am not sure if I would get any benefits from utilizing IEEE Float. If anybody can shed some light on this I would be most appreciative and believe this sort of information should be included in a wiki somewhere as I have been unable to

find any useful information as to the pros can cons of IEEE Float and PCM .WAV files with regards to audiophile needs.
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greynol
post May 24 2011, 23:18
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You do know that 24-bit with vinyl as a final product is complete overkill, right? I doubt you will be able to present a single example properly mastered to make use of the more significant bits demonstrating 16-bit to be inadequate through a properly controlled double-blind test. I don't even think you'll need to dither; as a medium, vinyl simply doesn't provide enough SNR.


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Axon
post May 24 2011, 23:35
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Gosh, greynol! The OP didn't even mention audibility or sound quality. If I didn't know any better, you were putting words in his mouth.
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Axon
post May 24 2011, 23:52
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Given that: this question is still reasonable to ask, strictly from a "correctness" point of view (in the same sense that we might care about bit-perfect output even if we freely admit that it does not make an audible difference).

QUOTE (navalverde @ May 24 2011, 17:00) *
I am wondering if it would be best to record in 32bit and bounce down to 24 when I export to .WAV, or just record in 24 if there won't be any difference. No matter what I will need a 24 bit file in the end to convert to .flac as it is the highest bit rate the format supports.

Your 1616M won't have a resolution remotely close to 24 bits (to say nothing of the preamp); the noise involved is probably going to make meaningless any sort of quality improvement effected by a higher bit depth on any DSP operations you might do inside of SoundForge.

Floating-point WAVs can be nice for a ton of different reasons, but those reasons don't apply for this sort of thing. (They can be useful for encoding >0dbFS samples, but a lot of editors will just clip that.)

QUOTE
Furthermore, I am confused as to whether to use IEEE Float or PCM. I understand the IEEE float was developed for the broadcast industry while PCM is used in the red book CD standard. I'm guessing when I convert to .flac it will end up as PCM anyway and am not sure if I would get any benefits from utilizing IEEE Float. If anybody can shed some light on this I would be most appreciative and believe this sort of information should be included in a wiki somewhere as I have been unable to find any useful information as to the pros can cons of IEEE Float and PCM .WAV files with regards to audiophile needs.

I'm gonna have to nitpick you on terminology here. Virtually every possible way you could save the audio will always be PCM; saving as floating-point is still PCM; lossless/FLAC is still PCM etc. You only wouldn't use PCM if you saved as DSDIFF or with lossy codecs like MP3 or any number of other totally obscure formats you will never use.

Now, if you replace "PCM" with "fixed point" then your question makes sense.... The question becomes, should you use a fixed point (integer) format, ie 24-bit or 32-bit signed, or should you use 32- or 64-bit floating point? Basically, while there are very strong theoretical reasons why you'd want to save intermediate files in floating point, for what you are dealing with, they are pretty much guaranteed to be utterly insignificant, both from mathematical and audible points of view. The best reason why one would really need to save floating point WAVs is if either they are dealing with data that intrinsically cannot be limited to <0dbFS, or if they are dealing with purely synthetic signals, generated mathematically rather than acquired via ADC. Neither situation applies to you.

Where did you hear that the broadcast industry cares about floating point formats? That's somewhat surprising to me, but I am not knowledgeable about their practices.

This post has been edited by Axon: May 24 2011, 23:56
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greynol
post May 25 2011, 00:48
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QUOTE (Axon @ May 24 2011, 15:35) *
Gosh, greynol! The OP didn't even mention audibility or sound quality. If I didn't know any better, you were putting words in his mouth.

The comment, "No matter what I will need a 24 bit file in the end to convert to .flac as it is the highest bit rate the format supports," raised a flag. If the OP already knows the information, then I don't see any harm as opposed to the potential risk in validating a possible and all-to-common misconception that throwing more of those pesky 1s and 0s at an analog source will better preserve the quality of the audio.


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DVDdoug
post May 25 2011, 01:09
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QUOTE
I have the option to record in 24bit or 32 bit at 192 Khz and save the .WAVs in using IEEE Float or PCM. I am wondering if it would be best to record in 32bit and bounce down to 24 when I export to .WAV, or just record in 24 if there won't be any difference.
There are no 32-bit or floating-point analog-to-digital converters or digital-to-analog converters. Your interface is 24-bits. (Most "consumer" soundcards are 16-bits.) If you "record to" higher resolution than you can capture, you are just wasting disk space.

Most audio editors work in floating-point (32 or even 64 bits) internally, no matter what you load-in. The sample rate (kHz) stays the same. As Axon suggested, you may have reasons for saving an intermediate file in floating-point. But, for your original recording and your "final release", there is no benefit.

If you listen to a vinyl record between tracks, and to a CD (16-bits) between tracks, (or very-quiet passages) it's pretty clear that 16-bits has far more dynamic range than analog vinyl... In other words, 16-bits is plenty!

It turns-out that 16/44 is better than human hearing... Don't believe the audiophile hype! Most of these guys making claims about "high-resolution audio" have never done proper blind ABX tests, and they are fooling themselves. And/or they make excuses about why blind testing doesn't work... Maybe there are a few "golden ear" audiophiles that can hear a difference between CD and a higher-resolution recording, but I remain skeptical. And, some of these guys actually prefer the "warm-crackly sound" of vinyl! biggrin.gif Anyone who "likes noise" has different goals than me and his/her opinions & advice are not of much use to me.

QUOTE
convert to .flac as it is the highest bit rate the format supports.
FLAC is lossless, period. When you decode/playback a FLAC, you always get-back the exact-original PCM data. When you encode FLAC, you don't "choose" a bitrate or quality setting. The different settings determine how much "work" goes into squeezing the file. A FLAC made from a 24/192 file will have a much higher bitrate (and larger file size) than a 16/44 file (just like the uncompressed WAVs).
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DVDdoug
post May 25 2011, 01:37
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P.S.
Speaking of noise..... I don't use Sound Forge, and I'm not sure what tools it has for removing "snap", crackle", and "pop". For me, noise reduction is the most important part of digitizing vinyl. (Far more important than bit-depth & sample rate! wink.gif )

If Sound Forge doesn't do an adequate job (this is special type of noise reduction) here are some software suggestions (and a ton of other helpful-related information). I use Wave Repair ($30 USD). It does an amazing job with most clicks & pops in the manual mode, and it only "touches" the few-milliseconds of audio where you mark a defect. The downside is that it usually takes me a full weekend to clean-up a vinyl transfer. sad.gif (I always buy the CD if it's available!)

This post has been edited by DVDdoug: May 25 2011, 01:42
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Audible!
post May 25 2011, 04:20
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Audacity is a free program analogous to SF/CoolEdit(Audition) that does click removal and NR out of the box. I hope SoundForge does a better job, since Sony charges for it, but Audacity will give you a free point of comparison. I've heard good things about the ClickRepair Programs, which have a three week trial period, but no direct experience. Goldwave has a trial version as well IIRC.

As everyone else has pointed out, 24 bit recordings (of an analog medium as inherently limited as a vinyl LP) are a massive waste of hard drive space with no value other than a psychological one. Perhaps if you had some good studio master tapes (with dolby studio or whatever) then it might be worth considering.

Even assuming never played, perfectly stored, new-in-sleeve 180/200 gram virgin vinyl records perfectly cut directly from the master tapes at half speed , 24 bit is a massive waste of space (and time, the larger the file the longer it takes to open, save, manipulate, etc.). The SNR of records in real life is crap. To be precise, warmed over, day-old refried dog crap of the most offensively odoriferous variety.

I'm going to assume that your target records have been previously played (and hence have notably decreased HF content), are imperfect to begin with (record masters being what they were), and are being captured from a good quality record player that is not calibrated daily.

Provided the EMU 1616 ADCs perform comparably well at all available resolutions and word lengths, 16 bit is what you want. I've seen a few computer audio solutions that perform rather differently (at least judged by the fallible RMAA) depending on setting, but it seems nearly impossible that an EMU semipro rig would have such issues.

If you're enamored of hand-waving and demonstrably useless audiophilic woo-woo, record at 16bit/96kHz, process the recording for noise and then resample down to 16/44.1. That way people with $100k tube amps won't point and laugh (especially when the roles should be reversed).

If for some inexplicable reason you do decide to capture at 24 bit, dither down to 16 bit for storage & playback anyway. Your dog, your hard drive, your portable player, and the universe itself will thank you.
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Cavaille
post May 25 2011, 07:51
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For editing Iīd leave it at 32 bit floating point, thatīs what I do. The only consideration is that it takes slightly longer than editing in 16 Bit - as one poster above stated most programs work with 32 bit floating point internally anyway (Sound Forge: 64). Iīd even leave it at 24 bit after youīve finished your de-noising, de-crackling, equalization etc. If you have the space why not leave it at that? Your PC wonīt get slower, most hardware can play it. Folks around here will tell you that itīs overkill, which is true for vinyl - but have you considered that you might have the desire to re-edit your recording, for example when you encounter a problem you didnīt see before? For such cases itīs best to leave it at such bit depths.

And if you rip with 192 you might want to downsample to 48 kHz afterwards. Vinyl rarely has frequencies beyond 16 kHz. Most of the things you can see with a spectrum analyzer are distortions.


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DonP
post May 25 2011, 11:25
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QUOTE (DVDdoug @ May 24 2011, 19:09) *
QUOTE
convert to .flac as it is the highest bit rate the format supports.
FLAC is lossless, period. When you decode/playback a FLAC, you always get-back the exact-original PCM data. When you encode FLAC, you don't "choose" a bitrate or quality setting.


From the context it's clear he meant the highest bits/sample that FLAC supports. Or bit rate of the uncompressed file.

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navalverde
post May 25 2011, 13:35
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Thankyou all for the information. I quickly realized after posting the 32 bit conundrum considering as you said there are no 32 bit dacs. I also appreciate the information regarding integer vs floating point as I was merely regurgitating program parameters. As far as the broadcast industry is concerned I can't remember where I cobbled up that information. I had orignially assumed that a higher sampling rate would result in a more accurate copy of an analog signal given the increase in samples per second, yet it seems as you say 44.1khz is more than enough to replicate everything in the 20hz-20khz bandwidth according to Nyquist. This then makes me wonder why we even have 24/192 dacs in the first place except to sell stuff or make use of some psychoacoustical effect of higher frequencies or if there is a secret audiophile community of dogs somewhere................. ...................................... While certainly noise will always be an issue with vinyl to an extent, I will not take sides in the vinyl vs cd debate as I have heard examples playing to the strengths and weaknesses of both formats and putting all the technical aspects aside preference for one or the other is merely subjective in the end.

Soundforge is a terrific program with very powerful dsp features, although I fear none of them are very helpful in dealing with pops and whatnot, however its editing interface is fantastic. I am somewhat curious how well something like Izotope RX2 works as it is a professional grade VSTi

Thanks
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LocrianGroove
post May 25 2011, 14:32
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The argument that 16 bits is overkill for vinyl is that the SNR of vinyl playback is less than what 16 bits can provide (96 dB or so). That sounds completely reasonable. I have a question though, about the nature of noise in an audio recording. Can we hear into the noise? In other words, can we hear things below the noise floor? If so, would we need to use some figure in addition to SNR to describe the amount of audible information that could potentially be captured in the playback of a vinyl recording?
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pdq
post May 25 2011, 15:10
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We can certainly hear single tones well below the rms noise, but people don't generally listen to single tones. Speech, music, etc. are not heard nearly so well below noise level.

I don't know if anyone has mentioned this, but while 16 bits is surely more than adequate for recording vinyl, the same may not be true for 44.1 kHz. Clicks and pops can have frequency content well above 22 kHz, and including the higher frequency content can be useful for their removal. This is not to say that one needs anywhere near 192 kHz, however.


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Juha
post May 25 2011, 15:11
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For post-processing sake, wouldn't it be better to have 192k samples/s than 44.1k samples/s and bit-resolution as many bits above 16 than it's possible (meaning use of ADC max bit-resolution))?

Juha

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2Bdecided
post May 25 2011, 15:16
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QUOTE (LocrianGroove @ May 25 2011, 14:32) *
The argument that 16 bits is overkill for vinyl is that the SNR of vinyl playback is less than what 16 bits can provide (96 dB or so). That sounds completely reasonable. I have a question though, about the nature of noise in an audio recording. Can we hear into the noise? In other words, can we hear things below the noise floor? If so, would we need to use some figure in addition to SNR to describe the amount of audible information that could potentially be captured in the playback of a vinyl recording?
The 16-bit digital quantisation noise is way below the vinyl noise - and it's possible to hear "into" either (but not at the same time, since vinyl noise completely hides digital noise, which is already inaudible at reasonable volume settings anyway).

Plenty of threads on this. Some even in the FAQ wink.gif

Cheers,
David.

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xnor
post May 25 2011, 16:16
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QUOTE (LocrianGroove @ May 25 2011, 15:32) *
The argument that 16 bits is overkill for vinyl is that the SNR of vinyl playback is less than what 16 bits can provide (96 dB or so). That sounds completely reasonable. I have a question though, about the nature of noise in an audio recording. Can we hear into the noise? In other words, can we hear things below the noise floor? If so, would we need to use some figure in addition to SNR to describe the amount of audible information that could potentially be captured in the playback of a vinyl recording?

Things you could hear in/below the vinyl noise floor would be recorded with 16 bits as well. Guess you mean 'dynamic range' which 16 bit offers plenty (above 140 dB with dithering iirc).

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Juha
post May 25 2011, 16:24
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IIRC, I read long ago on some site that using high samplerate (192kHz) in recording improves the AD conversion quality at higher frequencies (inside Nyqvist range) because of there are more samples to present the analog data as digital then (i.e. how many samples represents lets say 12kHz tone if I have understood it correctly).

It wasn't this site - http://electronics.howstuffworks.com/analog-digital3.htm but, maybe this site supports what I tried to remember.


Juha

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pdq
post May 25 2011, 16:37
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It's very simple, really.

If the analog signal contains NO content above Nyquist, then it can be reproduced EXACTLY by sampling at 2x and then converting back to analog. End of story!
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xnor
post May 25 2011, 16:46
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QUOTE (pdq @ May 25 2011, 17:37) *
It's very simple, really.

If the analog signal contains NO content above Nyquist, then it can be reproduced EXACTLY by sampling at 2x and then converting back to analog. End of story!


In reality however it's not that simple. You'd need perfect brickwall filters ... so in reality you can expect to see the effects of a low pass filter down to something like 0.9*nyquist.

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Juha
post May 25 2011, 16:50
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Doesen't the analog (vinyl) signal always contain content above Nyqvist (when recorded > 44.1kHz)?

Juha

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2Bdecided
post May 25 2011, 17:00
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QUOTE (Juha @ May 25 2011, 16:50) *
Doesen't the analog (vinyl) signal always contain content above Nyqvist (when recorded > 44.1kHz)?

Juha

"content"? No. Not always. Not even usually. Very occasionally.

Noise and distortion: yes.

Anyway, how well do your ears work above 22kHz?!

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David.
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Juha
post May 25 2011, 17:40
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QUOTE
"content"? No. Not always. Not even usually. Very occasionally.

Noise and distortion: yes.

Anyway, how well do your ears work above 22kHz?!


Hmm... least 2nd - nth harmonies of tone can be seen above 22kHz in spectrum analysis (aren't those part of the base tone least when tone is from acoustic instrument)? In my understanding, if harmonies are removed from tone the original 'sound' changes a bit (another question is can this difference be heard if the removed harmony is outside human hearing range)?

My ears works well even @ 192kHz but I can't hear/feel those high frequencies if levels are that low (I'm over 50 so maybe I can hear somewhere upto around 18kHz nowadays).

Juha

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xnor
post May 25 2011, 18:51
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QUOTE (Juha @ May 25 2011, 17:50) *
Doesen't the analog (vinyl) signal always contain content above Nyqvist (when recorded > 44.1kHz)?


'Nyquist frequency' refers to the frequency that is half the sampling rate, i.e. if the sampling rate is 1000 kHz the nyquist frequency is 500 kHz, so this question doesn't make much sense.

A vinyl record can contain frequencies of 96 kHz, probably even higher.

QUOTE (Juha @ May 25 2011, 18:40) *
Hmm... least 2nd - nth harmonies of tone can be seen above 22kHz in spectrum analysis (aren't those part of the base tone least when tone is from acoustic instrument)?
[...]

My ears works well even @ 192kHz but I can't hear/feel those high frequencies if levels are that low (I'm over 50 so maybe I can hear somewhere upto around 18kHz nowadays).

Guess you're talking about fundamental frequencies and harmonics here.
see frequency instruments chart

Which instrument produces fundamentals above 10 kHz?

I don't understand how your 'ear works well @ 192 kHz' but cannot hear above 18 kHz.

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Juha
post May 25 2011, 19:18
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QUOTE
Guess you're talking about fundamental frequencies and harmonics here.


Yes harmonics.

Juha

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Notat
post May 25 2011, 19:43
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QUOTE (Audible! @ May 24 2011, 21:20) *
As everyone else has pointed out, 24 bit recordings (of an analog medium as inherently limited as a vinyl LP) are a massive waste of hard drive space with no value other than a psychological one.

24-bit files use 50% more space than 16 bit. With storage at less than 10 cents per gigabyte, that certainly is not a massive concern.

I don't think we'll get anywhere trying to determine the inherent resolution of an analog medium. What we do know is that usable resolution of a decent ADC is greater than 16 bits and no more than 24 bits. To ensure no loss of data, 24-bit is a reasonable choice.

QUOTE (Juha @ May 25 2011, 08:11) *
For post-processing sake, wouldn't it be better to have 192k samples/s than 44.1k samples/s and bit-resolution as many bits above 16 than it's possible (meaning use of ADC max bit-resolution))?

Some ADCs have better S/N performance at lower sample rates. Switch to 192 and you get more bandwidth but reduced S/N. It is not safe to assume that highest sample rate gets you the best performance.

QUOTE (navalverde @ May 25 2011, 06:35) *
This then makes me wonder why we even have 24/192 dacs in the first place except to sell stuff or make use of some psychoacoustical effect of higher frequencies or if there is a secret audiophile community of dogs somewhere.

In some situations, it is possible to hear subtle difference between 48 kHz and 192 kHz recordings. As far as I understand this is because of differences in the anti-aliasing filters, not due ultrasonic hearing or anything exotic like that. The differences, when they exist, are probably too subtle to accurately compare quality but listeners and sales people will uniformly assume that 192 is better.
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