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24/192 Music Downloads ...and why they make no sense

http://people.xiph.org/~xiphmont/demo/neil-young.html

enjoy ^)^

I already knew some of this, like the fact that vinyl in no way sounds better than cd (not that that's specifically mentioned) but it never occured to me that 192khz music can in fact sound worse.

There is one thing the guy didn't delve into though. Sure if you're having supersaws then perhaps having the 44hz and 22hz frequencies in it isn't good... but, if it's the sound of a guitar recorded, who's to say that adding those frequencies in is actually making the guitar sound worse? Just because when you only hear the 10khz artifacts from the frequencies intermodulating and that ends up sounding a bit weird/like noise doesn't mean that it is in fact going to sound worse when mixed in with the rest of the guitar sound. It never occured to me just how big the dynamic range was. I guess I had it in my head that it was perhaps 20db lower or more than what it is. The quote "cds were made to capture the full range from one orchestra to a single instrument" is wildly inaccurate. Their dynamic range is far higher than that. It's kind've depressing in a way that there really is nothing better than cd audio but oh well!

The one gap in the subject that I currently have is I do have to wonder how scaling from 44khz to 48 or visa versa doesn't make an audio signal sound worse. If it was pixels in a movie it'd make it look far worse because they don't convert between the two well. Surely converting from one to another would involve adding in additional sine waves and would just cause an in-approximate mess.

I do however stand by that 320kbps is fairly obviously distinguishable from lossless though. Especially if you've heard the source in lossless before.

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Originally posted by Psyshell The one gap in the subject that I currently have is I do have to wonder how scaling from 44khz to 48 or visa versa doesn't make an audio signal sound worse. If it was pixels in a movie it'd make it look far worse because they don't convert between the two well. Surely converting from one to another would involve adding in additional sine waves and would just cause an in-approximate mess.

Pretty much. Upsampling is always very dirty unless the target rate is a multiple of the original rate (you can make interpolation good times in that case).

Nyquist theorem is not exclusive to audio. Saying it isn't good enough for audio makes no sense. Upsampling makes a lot of sense. But given your use of a term that is pretty basic and important in digital audio, I am pretty sure the basic concept is complete gibberish to you.

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Originally posted by clay nyquist theorem is that if you have the double sample rate of what you plan to sample it should be enough to recreate it.

Wrong.

The whole point of Nyquist is merely that you don't lose the frequency entirely. No more, no less. It doesn't try to make any guarantees regarding your perception of "quality".

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Originally posted by clay however if you sample a sinus at 22kHz using a 44,1khz samplerate it wont be a sinus but trangle wave, or maybe even worse a square pulse wave.

Anything above 20khz will be high passed to avoid aliasing, so trying to sample 22khz with a 44.1khz sample rate is retarded. Why do you think we use 44.1khz instead of 40khz? The whole point is that the extra 4.1khz gives you a ~2khz transition band to cut out frequencies that can't be sampled accurately.

Nope, response to clay , not echo boy , quote function too much work

it is a math model that assumes certain things that don't occur. But n your instance, well to start, it isn't double but rather the rate sampled must be. Less than half. It doesn't state an actual rate for audio, merely a method to achieve reproduction which does work assuming certain things that don't occur in real life ie perfect band limiting.

You always seem to focus on the wrong thing. Like every fucking discussion, you focus on what isn't the issue.

If anything could be said about sound quality and it's limitations, it would be the analog playback. Speakers really haven't changed and until we I find a way that delivers sound how nature intended , it will always. Sound like a recording.

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Originally posted by echosystm Pretty much. Upsampling is always very dirty unless the target rate is a multiple of the original rate (you can make interpolation good times in that case).

While this may have been the case with naive, rudimentary implementations, my understanding is that this is no longer the case. One can always scale up to the least common multiple of the source rate and target rate and divide back down to the target rate without introducing any additional artifacts beyond those inherent to the upsampling process. This comes at additional CPU cost, granted, but that is irrelevant if quality control is your priority. I looked into this a great deal when trying to decide whether it made more sense to upsample from 44.1 to 88.2 or 96 khz. Dan Lavry discusses it some in a whitepaper on SRC if I recall correctly.

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Originally posted by MaxC While this may have been the case with naive, rudimentary implementations, my understanding is that this is no longer the case. One can always scale up to the least common multiple of the source rate and target rate and divide back down to the target rate without introducing any additional artifacts beyond those inherent to the upsampling process.

That's like saying "i'll punch you in the face, but it's ok because you won't feel any pain other than being punched in the face".

I'm happy to be proven wrong, but all you've done is outline one of the two approaches to sample rate conversion (the other is interpolation as a time series). In either case, rounding is going to happen unless the target rate is a multiple of the original rate.

It's just math. They both result in the same end. One requires extra operations.

There's a lot of bullshit in this post and although he's theoretically correct in the argument that the extra headroom is of no consequence due to our threshold of hearing.

However, what he doesn't mention is that in a professional environment and with pro level equipment, the actual real world dynamic range of 16bit systems is 93db and for 24bit it's around 116db - that extra 21db of headroom is actually useful on a calibrated system, especially when doing any recordings.

It's also useful due to the fact the noise floor is inherently proportional to the signal - with a higher bit depth and therefore usable headroom you can jack much quieter sounds up without that noise floor being so much of a problem.

One of the people in the thread state some interesting things such as there is no such thing as 24bit DAC output, only 20bit as resistors and capacitors are a bottle neck to the overall output range, and if this is true, only using a 16bit output stage would result in only about 12bits of headroom which is now getting dangerously low in terms of not having a enough dynamic range for the human thresholds of hearing as they relate to headroom.

I've stated on here plenty of times that analogue tape only has a usable bit depth of 60db but people love it because of how it sounds, and that is also the main flaw with the CD vs Vinyl debate.

Looking purely at bit depth, CD wins, but when you have to look at sample rate with the same argument, vinyl has no sample truncation or aliasing so it is the clear winner in terms of audio reproduction.

He also glosses over three other major points:

1, Why did Steve jobs NOT want 24bit recordings? It was not a technical discussion of bit depths. It was purely the ratio of Money per Megabyte. an Apple lossless 24bit file of 5 mins = 100mb. A 320kbs MP3 of the same length? 10mb at most.

For the great unwashed who will never use anything more than ipod or beats by dre headphones, MP3 Is all they will ever know so there's no point apple using 10 times the storage space for billions of downloads when it won't make a shade of difference to 99.9999% of their users.

2, I'm calling bullshit on the author regarding his claim to be able to identify each MP3 encoder based on it's sonic qualities. While I'll concede back in the early 90's there were some very rudimentary encoders which at low bit depths could easily be told apart due to how badly they introduced articles and coloured the sound, there is no fucking way you could do this on even slightly higher bit depths with anything more recent than say 1998 technology. I've worked with some of the best pro engineers in the world and while some of them can tell you that something sounds different to another identical recording, they can't blindly identify encoders to any accuracy, even though they work day in, day out with this technology and have done for 30 odd years.

This guy, quite literally would have to be the only guy in the world who has ever been able to do this.

3, The final problem he glosses over is to work, dither introduces random white noise to essentially spread the errors so we don't hear them. Whether you like it or not, it's a form of pyschoacoustic masking. We may not perceive it outright but it's there, and within the human threshold of hearing. There is again an advantage (that someone points out later in the thread and the author has to concede it) that having the lower relative noise floor in 24bit (over 16bit) is an advantage of of the higher bit depth.

Will any of us ever be able to hear that difference? Doubtful, but it's still there.

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Originally posted by clay I still am unsure due to the nyquist theorem possible faults.

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Originally posted by clay interestingly enough it seems that 44,1kHz was not chosen due to the space required for a transition band

You missed the point entirely. The first point was that your example of trying to sample 22khz is retarded, because it will get filtered anyway. The second point was that we use rates higher than 40khz (eg. 44.1khz), because they provide space to do the aforementioned filtering. There's no "fault" with Nyquist. Fuck.

This thread has been palm'd.

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Originally posted by clay my point was that by using 44,1khz you cannot really recreate a sinus at 22kHz, 20kHz or even 15kHz, at best maybe at 11kHz (four samples) however i see now that due to stuff like dithering and difficulties in the dacs analog parts will not be able to recreate it perfectly anyway, and also the fact that humans cannot hear the difference, i understand that i made a fool out of myself. please forgive my stupidity, please please. my entire day depends on it. however i think its interesting to read about it and it seems to me that our problems with soundquality today does not lay in the bit depth or sample rate, its in the music, bad mixing, the mastering (loudness war), and in the gear used both in monitoring while creating the music, and reproduction in the consumer market, and finally also mp3/streaming codecs off course.... when i think about it stuff sounded perfectly fine during the early days of 80s throughout most of the 90s.

90s and 80s were definitely not considered a period with great mixing. Those early digital interfaces compared intro level interfaces were rough. Unfortunstely most music made now only has production and mixing to showcase. Sounding ok wasn't so bad when the performances were new different and well ya ....

Most of that aliasing is masked by the louder main signal that it is a non issue with today's music. And you can always use a higher sample rate.

Thus should keep you busy
http://www.researchgate.net/publica...d0a1b80ccef.pdf

quote:

Originally posted by MSZ This thread has been palm'd.

I admit some ignorance regarding the topic, but there were two things I didn't easily find in the article:

1. Acoustics

2. Harmonics

I get the premise (though I wish something like infrared was not brought up in this case, not a topic I've seen here before), but the physics involved perhaps do make differences that are imperceivable still possible.

I'm not saying that's true, just asking?

The message I kept taking away from each part was the fact that it was not capable to hear. That's true, but when you push several different sound sources, or even one in an odd room, things start acting a bit differently.

I don't question that human capabilities are different, especially from the golden-standard mentioned, but in certain playback situations, perhaps these work in different ways? A square room will boost the shit out of a bass, but the acoustic and harmonic responses will shape it completely differently (not just talking playback but other instruments as well). I may have missed the point, I'm asking?

But. there is some laziness:

Better headphones

The easiest fix isn't digital. The most dramatic possible fidelity improvement for the cost comes from a good pair of headphones. Over-ear, in ear, open or closed, it doesn't much matter. They don't even need to be expensive, though expensive headphones can be worth the money.

Lith again: I ask you to compare in-ear vs. over-ear and ask yourself if that makes a difference. a 1/4" driver does make a difference vs. a 1 1/2" driver, cmon now.

Anyway, really good thread. I'm interested to hear more opinions!

and after you spent the 200 bucks. The safest thing to do if you fall into the ocean is to keep your head above the water and not drown. Thanks.

My headphones cost $75. I would put up a good argument for them against Apple's or Samsung's.

Help me out here guys:


I actually think that Clay and Richie are making very valid points here and if you go on to the net, and look up higher bit depths, you'll find blog upon blog and thread upon thread suggesting more is clearly better.

As the info in this thread has demonstrated, that's not quite a clear definition and really thinking about this and the maths involved has convoluted the situation for me.

The bit that really fucks me off about the original article, is that his whole argument is really just about dynamic range.

Yes, agreed in real working practices (i.e. real world usable dynamic range, ambient noise interference etc) the higher bit depth is of no advantage but the massive bit he keeps glossing over, is that 24 bit audio provides more steps of resolution (i.e. is samplerate is resolution on a horizontal plain, bit depth is resolution on a vertical plain).

These are the only passages where he addresses it:

quote:

It is also worth mentioning that increasing the bit depth of the audio representation from 16 to 24 bits does not increase the perceptible resolution or 'fineness' of the audio. It only increases the dynamic range, the range between the softest possible and the loudest possible sound, by lowering the noise floor. However, a 16-bit noise floor is already below what we can hear.

quote:

None of that is relevant to playback; here 24 bit audio is as useless as 192kHz sampling. The good news is that at least 24 bit depth doesn't harm fidelity. It just doesn't help, and also wastes space.

Can anyone see anything part in there where addresses higher bit depth giving more steps/finer resolution/etc?

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Originally posted by DJ RANN I actually think that Clay and Richie are making very valid points here

Clay said "Nyquist theory isn't good enough for audio". That's like saying gravity isn't good enough for human flight. Gravity doesn't give a fuck and neither does Nyquist, which is why we build aeroplanes and use sample rates greater than double the highest intended frequency. Not sure what's valid about that point at all?

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Originally posted by DJ RANN Help me out here guys: I actually think that Clay and Richie are making very valid points here and if you go on to the net, and look up higher bit depths, you'll find blog upon blog and thread upon thread suggesting more is clearly better. As the info in this thread has demonstrated, that's not quite a clear definition and really thinking about this and the maths involved has convoluted the situation for me. The bit that really fucks me off about the original article, is that his whole argument is really just about dynamic range. Yes, agreed in real working practices (i.e. real world usable dynamic range, ambient noise interference etc) the higher bit depth is of no advantage but the massive bit he keeps glossing over, is that 24 bit audio provides more steps of resolution (i.e. is samplerate is resolution on a horizontal plain, bit depth is resolution on a vertical plain). These are the only passages where he addresses it: Can anyone see anything part in there where addresses higher bit depth giving more steps/finer resolution/etc?

Not exactly sure what you're asking here?

Are you suggesting that any bit-depth higher than 16 bit is of no advantage?

I mean you're right in saying that the sample rate doesn't have much bearing to dynamic range. I agree 16 bit at 44.1 is adequate for playback, but I was also under the belief that a higher bit depth is desirable for recording due to being able to more accurately represent the dynamic range of a waveform.

I'm no expert but definitely interested in learning more about this stuff.

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Originally posted by Teezdalien Not exactly sure what you're asking here? Are you suggesting that any bit-depth higher than 16 bit is of no advantage? I mean you're right in saying that the sample rate doesn't have much bearing to dynamic range. I agree 16 bit at 44.1 is adequate for playback, but I was also under the belief that a higher bit depth is desirable for recording due to being able to more accurately represent the dynamic range of a waveform. I'm no expert but definitely interested in learning more about this stuff.

You did actually read the thread and the first link right? The entire thing is about higher bit depths NOT being of any advantage at for playback.

Palm - you raise some good points, and I've always believed (and now it seems rightly so) that for producing and recording higher bit depth is better.

As you point out though, the content doesn't seem to address pro audio uses especially with very serious pro audio interfaces/DAC etc. I think his main argument is for consumer audio devices and that makes sense.

The bit I'm still struggling with (and it's more theoretical that practical at this point) is truly whether or not there is an advantage in resolution of the higher bit depth.

Even with the real world usable dynamic range 93db (or 120, however you want to look at it) that doesn't affect the resolution, in the same way, you don't get more pixels if you're only look at one part of the screen (if that makes sense); The resolution stays the same regardless of how much of the range you are using.

I can't seem to figure a straight answer on this, and whether that extra resolution con be perceived as a better sounding?

It is more forgiving. If you are a technical mixer with impeccable gain staging , then sure 16 bit is fine. If you run a fledgling mastering house and your wife is overweight and her vagina has too much headroom , 24 bit.


That is how we do

quote:

Originally posted by DJ RANN You did actually read the thread and the first link right? The entire thing is about higher bit depths NOT being of any advantage at for playback. The bit I'm still struggling with (and it's more theoretical that practical at this point) is truly whether or not there is an advantage in resolution of the higher bit depth. Even with the real world usable dynamic range 93db (or 120, however you want to look at it) that doesn't affect the resolution, in the same way, you don't get more pixels if you're only look at one part of the screen (if that makes sense); The resolution stays the same regardless of how much of the range you are using. I can't seem to figure a straight answer on this, and whether that extra resolution con be perceived as a better sounding?

Sorry for the confusion. Of course I read the article, I was just also left a bit puzzled with your previous post, this comment in particular as you didn't mention playback.

quote:

Yes, agreed in real working practices (i.e. real world usable dynamic range, ambient noise interference etc) the higher bit depth is of no advantage but the massive bit he keeps glossing over, is that 24 bit audio provides more steps of resolution (i.e. is samplerate is resolution on a horizontal plain, bit depth is resolution on a vertical plain).

Now these unanswered questions you raise. It just goes against the things I've been taught and understood. I'm not saying you're wrong, I'd just like to get to the bottom of this too.

I dunno, I just thought that it was ideal to use the higher resolution bit depths because they are more accurate when capturing or rendering digital audio. Personally I don't hear any improvement when listening back when compared to a 16 bit file, and I don't really think there is any either.

Then I think you either didn't actually read the link and thread, or at least, didn't understand it.

The whole crux of the argument in the link is that due to physical limitations (hardware, DAC circuit design, ambient noise floor, etc) and our own physiological limitations (SPL sensitivity etc) higher bit depth does not increase the quality of what we hear.

All it essentially does is increase the available dynamic range from more than enough, to even more than enough.

For me as well, this does go against everything that I was taught (more = better, to a point) and if you google it, you'll find blog after blog and post after post form some pretty well read people suggesting it does, but in actual fact, at least for consumer grade audio they all wrong and it makes no real world difference.

I fully understand that the biggest difference between 16 and 24 bit is dynamic range (but it's essentially useless as true 144db SPL will cause incredible pain and most likely permanent eardrum damage), but the bit that is in question is a straight answer as to whether the resolution or "fineness" of the audio increases with a higher bit depth, and furthermore, if we can realize that increase perceptively.

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