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Personally, I wouldn't qualify myself as an "audiophile" but I love music. However, to be truthfully honest, I can't tell the difference with most file formats. Or rather, I can tell the difference, but things like file size, sharing and playing the content affect my decision moreso than if one file sounds a little "better" than another.

For me, and I'm sure a lot of people in my boat, without knowing how I benefit from having 24-bit music files (when it seems from reading comments that current music is already heavily compressed anyway) I wouldn't really be inclined to pay the higher price for music.

Lastly, I think the other part of it is not having the proper equipment to really get the full range of sound. Majority of people are still rocking white earbuds, and I only have a pair of Heart Beats (worst headphones, ever.) so you lose sound quality there too, right? Even if I had the most high quality file possible, I'd still be losing out due to sucky equipment... so, where is the value in it for me? I'd have to upgrade everything (including my internal hard drive it seems) just to accomodate this particular file type.

So, my question is: If you are just a regular Joe (not an audiophile or audio engineer), what exactly are the benefits of having 24-bit music vs. what is available right now on iTunes?
 
3. Anything via airtunes atm is 16/44 including the AppleTV
Not strictly true. So long as you output from an AirPort Express using an optical cable, what you transmit is just audio data.

I have transmitted both 5.1 Dolby Digital and DTS audio via AirTunes and it decodes when connected to my cinema decoder/receiver.
 
I wasn't referring to the audio quality. I was referring to the artist bashing. Music taste is subjective.
If you thought anyone here was artist-bashing, then I'm afraid you're wrong. I'm not particularly a fan of Lady Gaga, but the point we were making was about the pop industry's obsession in loudness.

I would love to hear Lady Gaga recorded and produced correctly. After all she is a talent. But sadly I fear that day will never happen!
 
If you thought anyone here was artist-bashing, then I'm afraid you're wrong. I'm not particularly a fan of Lady Gaga, but the point we were making was about the pop industry's obsession in loudness.

I would love to hear Lady Gaga recorded and produced correctly. After all she is a talent. But sadly I fear that day will never happen!

Support Dynamic Range Day - March 25!

dynamic-range-day-2011.jpg
 
I hope tis is true but as a lot of people have said extra bit depth would be mostly useless, I'd like to see higher sample rates, but the people wanting 192k are dreaming most stuff isn't recorded any higher than 96k and a basic recording would only be at 48k besides the fact that 192k is insane, unless you have studio quality gear to listen to it on it's a waste of time over 96k.

but with all that said I'd be happy with just uncompressed CD quality over what itunes have now. AND with that being said I have a hand full of SACD's that I love to death and which the format took off :(

Higher sampling rates are even more useless than more bits for dynamic range. Even a young adult will not hear beyond 20kHz and older adults typically have their hearing response greatly fall off below 18kHz or even 15kHz. Recording and playing back information beyond that will not benefit the listener in any way. Early CD player designs that used "brick wall" filters had some problems in the upper response due to the filter used to remove any quantization errors above 20kHz. However, once oversampling (and later bit-stream methods) was utilized, this problem vanished and no longer presents any sonic artifacts. In short, higher than 48kHz sampling frequencies are a waste/usless even at the professional recording level. Of course, there is so little information there to begin with, that any compression like MP3 or AAC won't need much space to accommodate it just the same. It won't make any difference, however.

The reason so many DVD-Audio and SACD recordings sound better (even if in 2-channel) than your average CD is because those recordings are made with real care along the entire chain, not because of 192kHz sampling rates or 24-bit words. There are plenty of excellent quality CD recordings out there. Basing one's opinion on a typical rock CD, for example won't prove a thing.

The "majority won't notice the difference" argument is irrelevant. This is not about the majority vs. the minority. As i stated in my earlier post, the majority does not need to care- because they are not having anything taken away. They could benefit if the expanded market due to the standardizing effect Apple would have on higher quality music and the reach of itunes if the market is expanded, giving greater incentive for artists to produce quality music.

You don't seem to get it. There is no "higher quality" music by increasing the playback bit-rate. Try reading up on the Nyquist Theorem (the foundation of all digital sampling). Saying something like 192kHz sampling rates make "better" music is like saying the foundation of that recording system is incorrect science.

The real reason you don't get better quality music more often from artists is that there is no incentive within the sphere of the industry to create it. If anything there is a real push to make BAD music quality, namely extremely limited dynamic range because it sounds "louder" on the TV and Radio. "Louder" is a perception that creates "better" in the mind of an average, non-discerning listener. This has lead to many many bad BAD albums in terms of sound quality. One example is the Red Hot Chili Peppers "Californication" album. It's so compressed (or rather POORLY compressed) that it CLIPS regularly, badly distortion the sound. It's not just loud, it's tainted with clipping distortion noise (not pleasant). I imagine the master tapes do not have this issue, but if the band doesn't care (assuming they even have control over the issue in terms of their recording contract), bad things like this can happen. It shouldn't be that bad anyway, really because digital filtering can easily heavily compress music without clipping. Someone appeared to be deaf when they mastered this album and that's not a good thing to have in a recording engineer.

Now to be fair about recording engineers in general, they are generally not the last word in the sound quality you get. In other words, they still have to do what their bosses tell them to do or lose their jobs. If the boss says we need to make a louder recording with virtually no dynamic range, that's what they have to do.

OTOH, a lot of synthesized music really has no meaning for "dynamic range" to begin with so the idea that just because you CAN have 96dB (or more with longer words) dynamic range doesn't mean you NEED to. A good sounding recording is often the result of care in recording and mixing the sound (both in terms of relative volume, care in microphone placement, etc. and phase/space placement of the stereo image in space. Often these things are vastly more important to the "perceived" quality than a given amount of dynamic range or frequency response. Any given instrument or track in the recording can affect the overall result as well. I can have the best sounding drums on the planet and if my saxophone is sibilant and harsh sounding, it won't matter much to my perception at that point.

A great system really does help. In school we did listening tests with CD, DVD-A and SACD and you could immediately tell the difference because the speakers were 20,000K+ each and sounded like it.

The source material also matters. Some mixes are better than others.

It's highly doubtful that the albums used were simply mix-downs of the SACD master. For instance, buying Pink Floyd's "Dark Side Of The Moon" on SACD will not be the same mastering as the current CD release. It was remastered specifically for surround sound and therefore has no bearing on the 2-channel master used originally for CD, the 25th anniversary edition digital remaster or the Mobile Fidelity mix (which uses a different master tape). If you get the DVD-Audio "Alan Parsons Quad" mix off the Internet, it uses yet a different master and unlike many of the studio releases isn't compressed or otherwise "mastered" for commercial release (and thus sounds the best quality of them all, IMO, assuming you play it on a quality system). Mix is everything. OTOH, it's not that hard to make different bit/sampling versions if you own/control the master recording. I write my own music and record in 24-bit/96kHz. I can easily generate 16-bit/44 mixes or 24-bit/48 or 16/96, etc. They all sound the same.

96KHz does a lot more than increase dynamic range. Look at the encoding of high frequencies. In 16-bit PCM you will see that only a few discrete volumes are available. You might see, for example, that only 96db, 90db, 84db, etc volumes are encoded at a given frequency even though volume is clearly a continuous function. That's why CDs are often said to have harsh high frequency sound, because 16-bit encoding loses too much information. 24-bit encoding restores the missing information. Also, 96kHz sampling isn't pointless even if few people can hear 20kHz, because it eliminates the need for an analog lowpass filter and its associated harmonics, which are audible at frequencies most people can hear.

I don't know what magazines (or other sources of information) you are reading, but you should throw them out. What you just said is pure scientific NONSENSE and is the epitome of the "high-end myth" about digital audio (I remember reading a lot of nonsense from "Stereophile" and even more from "Stereo Review" back in the day (who always appeased their advertisers with good reviews from what I could see). It demonstrates a total non-understanding of how digital audio works. There is no 'missing information' in a digital signal within the bounds of its dynamic range and frequency limits. Furthermore, sampling rate has NOTHING to do with 'volume'! The word length controls dynamic range (which is still not "volume" which is a function of the amplifier and speaker output for a given playback setting).

Try looking up Nyquist Sampling Theorem and do a lot of reading on the subject.
 
I'm sure a lot of people won't mind the improved quality, even if most can't differentiate between the two.
 
Also, 96kHz sampling isn't pointless even if few people can hear 20kHz, because it eliminates the need for an analog lowpass filter and its associated harmonics, which are audible at frequencies most people can hear.

This is a good reason to RECORD at 96 or 192 - since there IS an analogue lowpass filter employed in front of the A to D at the studio.

Once mixed, digital brick wall filters are used during the down conversion to 44.1, so that works fine...

On playback, most D to As oversample the 44.1 back up to a much higher sampling rate, so the analogue lowpass filter can be a pretty gentle one, rather than a phase-mangling brick-wall filter.

So, it's not necessary to distribute in 96 or 192, and due to oversampling, talking about analogue filters in playback is definitely not a justification.
 
What you just said is pure scientific NONSENSE and is the epitome of the "high-end myth" about digital audio (I remember reading a lot of nonsense from "Stereophile" and even more from "Stereo Review" back in the day (who always appeased their advertisers with good reviews from what I could see). It demonstrates a total non-understanding of how digital audio works. There is no 'missing information' in a digital signal within the bounds of its dynamic range and frequency limits. Furthermore, sampling rate has NOTHING to do with 'volume'! The word length controls dynamic range (which is still not "volume" which is a function of the amplifier and speaker output for a given playback setting).
You need to listen better. I wasn't talking about sampling rate, I was talking about encoding. Sampling rate in this case is irrelevant. 16-bit LPCM has more quantization noise at high frequencies than low, that's why CDs have a reputation for harsh highs. 24-bit systems basically eliminate this problem, that's why 24-bit systems have a reputation for sounding better.
 
You need to listen better. I wasn't talking about sampling rate, I was talking about encoding. Sampling rate in this case is irrelevant. 16-bit LPCM has more quantization noise at high frequencies than low, that's why CDs have a reputation for harsh highs. 24-bit systems basically eliminate this problem, that's why 24-bit systems have a reputation for sounding better.
To be fair, you were talking about 96kHz increasing the dynamic range. 96kHz refers to sampling rate, and has nothing to do with dynamic range as you stated.

Also, if you read up on Nyquist you'll see that high frequencies may have lower resolution, but the harmonic noise produced from the quantisation is out of human hearing range. Professional audio is recorded at 96k/24-bit because then there is enough range for tweaking and mastering for CD... In much the same way that photographers and graphic artists use RAW files and 16-bits-per-channel in Photoshop.

At the consumer end, it is physically not possible to notice a higher sample rate, unless you are a dog/cat/superhuman.
 
Great News, I'd definately buy HD audio 24/96 content, iTunes already has no problem playing this just need the iOs devices enabled.
The entire iTunes catalog is already lossless 16bit has been since day one, so Apple could easily flick the switch and sell us the Lossless ALAC or WAV versions of what they have now, easier for them too because at the moment when we buy a tune it actually is encoding from WAV/ALAC to the 256 we buy.
All labels have been doing since day one of iTunes is using Apple "iTunes Producer" software to rip their CDS to WAVS (up until 2007) and upload them to territories iTunes stores and from 2007 onwards uploads have been in ALAC, shows you what Apple thinks of ALAC, it's pretty darn great imo, FLAC is good but there is no difference between the two, there isn't that's FACT and since iTunes is the best player now and possibly for the rest of our lives makes sense to use ALAC.
Bring on 24bit 96kHz
 
OK I have just read through this thread and all of the arguments being made by people who seem to be educated on the subject, as well as those who think they are are contradictory. I just got done reading another idiotic article over at Gizmodo on this subject and I am tired of the mis-information. Here is the thing- I do appreciate the "audiophile" quality of a superior recording on a good system. The best example of this is Pink Floyd's Dark Side of the Moon on SACD on a really really good 2 channel system. So I want to start with something:

SACD has a superior sound quality to CD. I have heard it, I have noticed it, I can and do appreciate it.

Can someone please explain to me what- in terms of specific digital audio properties (dynamic range, sampling rates, etc.) the differences are between what the rumored "24-bit audio" files coming from iTunes and the sound quality of SACD?

i am sorry, I just don't understand why this new audio quality format would not be a GIGANTIC step forward for sound quality and why anyone would not like that, provided iTunes does not instantly kill all of their existing format files for those that want them...

If iTunes is preparing to offer a sound quality that is technically superior to CD as an option for music download I am really happy. If it is at the same level as SACD- technically speaking- I am ecstatic!

Someone please clear this up- this is an invitation to argue your opinion! :)
 
FLAC is good but there is no difference between the two, there isn't that's FACT


Well, apart from the fact that Apple haven't released any public documentation on ALAC and that any direct conversions from ALAC to FLAC have to be done through reverse-engineered solutions... and that ALAC doesn't include the error-checking that FLAC does, you could say they're identical. Which they're not.

Don't get me wrong, I use ALAC at home and am slowly working my way through re-ripping all my CDs with XLD... but ALAC, in my opinion, was invented for two reasons:

1. Vendor lock-in
2. DRM

So, no. It's not ideal.
 
I've got lots of 24 bit lossless (Beatles FLAC etc.) and have never been able to tell the difference from CD.

I also listen to some music from Linn Records that is up to the 24-bit/192 kHz format. That said, I heard straight from Dr. Karlheinz Brandenberg, one of the inventors of the MP3 format, that this format's dynamic range isn't compromised compared to the CD format. However, the waveform of MP3s are deformed and highly-compressed versions (i.e. 32 kbps) of this format can sound bad to most people.

This CD demonstrates various perceptual effects of compressed audio:


I have at least normal hearing (i.e. fairly linear up to 8 kHz), which has been tested by the House Ear Institute in Los Angeles, but could not hear the difference between 128 kbps AAC, 128 kbps MP3 and the CD original versions of a musical excerpt using Etymotic ER-4P headphones.

My concern about 24-bit downloads on iTunes is that the 24-bit/192 kHz format has a bit rate of 9,216 kbps which is about 650 MB for a 10-minute AIFF file. Unless we live in South Korea with access to Gigabit internet, I don't see how downloading these kinds of tracks can be done conveniently. And if many of us can't hear a difference, perhaps 24-bit tracks aren't necessary.
 
You need to listen better. I wasn't talking about sampling rate, I was talking about encoding.

This is what you said:

96KHz does a lot more than increase dynamic range. Look at the encoding of high frequencies. In 16-bit PCM you will see that only a few discrete volumes are available. You might see, for example, that only 96db, 90db, 84db, etc volumes are encoded at a given frequency even though volume is clearly a continuous function. That's why CDs are often said to have harsh high frequency sound, because 16-bit encoding loses too much information. 24-bit encoding restores the missing information. Also, 96kHz sampling isn't pointless even if few people can hear 20kHz, because it eliminates the need for an analog lowpass filter and its associated harmonics, which are audible at frequencies most people can hear.

You were talking about 96kHz (sampling rate) and then going on about distinct volume levels available at 16-bit. 96kHz is a function of frequency response and has NOTHING to do with volume levels period. I don't care if you're talking about cooking french fries. 96kHz sampling has nothing to do with volume and there is no other way on earth to interpret what you said so spare me the "listen better" bologna.

Furthermore, there are no "distinct" volume levels with 16-bit sound at "96dB,90dB,84dB, etc" as you indicate. Volume is quite linear within the confines of the dynamic range allowed. 16-bit audio has 96dB of dynamic range. If your maximum volume at a given amplifier/speaker combination in the room is 100dB that would mean that the most quiet sound produced without dither would be 4dB loud and the maximum 100dB. There are no jumps in sound level at 6dB intervals in between those levels.

Assuming oversampling is used on playback, high frequency sounds right up to the maximum are accurately reproduced. "Harsh high frequency sound" is a misnomer in high-end circles. The LP record has usable high frequency response to perhaps 14-15kHz or so and even there it's heavily distorted. In other words, if CDs sound "harsh" at really high frequencies (especially with older recordings), it's probably because the LP version doesn't have any information at those frequencies to begin with! You can't have harsh frequencies if none are recorded. A proper/quality 16-bit recording played back on quality equipment will not have "harsh" anything.

Sampling rate in this case is irrelevant.

Then why did you mention it? :rolleyes:

16-bit LPCM has more quantization noise at high frequencies than low, that's why CDs have a reputation for harsh highs.

More nonsense. Oversampling eliminated all these problems in the mid-80s. Most CD recordings back then that sucked were a result of using LP masters that were never optimized for CD playback instead of remastering. Most album re-releases in the late '80s and '90s were much better sounding. It was still 16-bit audio.

24-bit systems basically eliminate this problem, that's why 24-bit systems have a reputation for sounding better.

Basically? 24-bit is about headroom for recording. A bad recording will sound bad in 24-bit or 16-bit. 24-bit isn't needed for playback in any Universe I know of. 18-bit would more than sufficient for the limits of human hearing and no one listens to music at 120dB without destroying their hearing in a short manner. For most recordings, 16-bit is more than sufficient to convey 100% of the recording's audible material.

Once again, I suggest you go study digital sampling because your ignorant posts on it tell me you know nothing about it. A little information is clearly dangerous because you're all over the map. Back-peddling with more incorrect information doesn't help your case any.

SACD has a superior sound quality to CD. I have heard it, I have noticed it, I can and do appreciate it.

Don't confuse a good recording or mastering with a format. Most SACD recordings sound better because there is more care in their recording and mastering with focus on sound quality whereas most recordings (especially pop/rock) made for CD are focused on radio play and therefore colored/loud playback. If you take a 2-channel SACD master and output a 16-bit CD without touching the mix, it will sound virtually identical in most cases (barring the odd recording with slightly higher dynamic range). Everything else but the dynamic range is a result of the mastering/recording, not the SACD format. The only other good option with SACD is the distinct surround channels (those can make a huge difference in perceived sound imaging).
 
OK I have just read through this thread and all of the arguments being made by people who seem to be educated on the subject, as well as those who think they are are contradictory. I just got done reading another idiotic article over at Gizmodo on this subject and I am tired of the mis-information. Here is the thing- I do appreciate the "audiophile" quality of a superior recording on a good system. The best example of this is Pink Floyd's Dark Side of the Moon on SACD on a really really good 2 channel system. So I want to start with something:

SACD has a superior sound quality to CD. I have heard it, I have noticed it, I can and do appreciate it.

Can someone please explain to me what- in terms of specific digital audio properties (dynamic range, sampling rates, etc.) the differences are between what the rumored "24-bit audio" files coming from iTunes and the sound quality of SACD?

i am sorry, I just don't understand why this new audio quality format would not be a GIGANTIC step forward for sound quality and why anyone would not like that, provided iTunes does not instantly kill all of their existing format files for those that want them...

If iTunes is preparing to offer a sound quality that is technically superior to CD as an option for music download I am really happy. If it is at the same level as SACD- technically speaking- I am ecstatic!

Someone please clear this up- this is an invitation to argue your opinion! :)

I'll have go....

Not knowing the technical details of the rumoured 24 bit itunes format I can't compare it to SACD, but I can explain why high sample rates and bit depths in general make no sense for human hearing whatever format they arrive on.

24bit audio and higher sampling rates ARE very different from normal 16 bit CD audio at 44.1KHz but our poor human hearing is not equipped to notice the difference. The practical uses for these superduper recording rates include giving extra bit depth for studios to apply loads of processing without worrying about introducing noise, and in the case of higher sampling rates to allow such things as the recording of ultrasonic bat sonar calls. The practical uses do not include normal human beings listening to music.

Heres' the science bit.

There are two main things to look at in digital audio :

Sampling rate sets the highest frequency you can reproduce from your samples - that frequency is basically half your sampling rate - so for 20KHz top frequency you sample at 40KHz plus a bit faster to compensate for limitations of the filters used to limit the signal bandwidth and prevent aliasing. That's where 44.1KHz comes in. 96KHz and all those higher sampling rates give you higher frequencies (up to 48KHz for 96KHz sampling rate) but you won't hear them because humans can't hear that high - animals like dogs, cats and bats can though.

There seems to be a common misunderstanding with sampling that more samples (faster sampling rates) somehow equals 'better' beyond the twice as fast ('nyquist') limit. This seems to be because people assume that reconstructing from samples is somehow about joining the dots - so more dots must be better right? Turns out this is not at all how it works. It's about summing sinc (sinx/x) functions (the Dave Lavry paper shows how this works with illustrations!) and the nyquist rate is the necessary condition to uniquely describe the signal you are trying to reproduce perfectly.

Bit depth of each sample sets the swing from soft to loud sound that you can represent - this swing is called dynamic range. Turns out that in the maths this is about 6dB for each bit you have in your sample. 16 bits gives you 96dB (CD) and 24bits gives you 144dB (DVDA etc). To use all this dynamic range you have to rise above the noise floor in your environment, and 30dB is a typical figure in quiet rooms. So that gives us 126dB and 174dB respectively. 126dB is staggeringly loud - like the front row at a big concert. 174dB is staggeringly, staggeringly loud - close to a full scale reproduction of the Krakatoa Volcano eruption of 1883. Assuming (and this is a big ask) you could find a way of reproducing sound this loud you'd likely go deaf, bleed from your ears and/or be arrested.

Here's the science bit about why you may well hear a difference but it probably has nothing to do with sample rate or bit depth.

Unless you can pick the difference between SACD and CD or indeed any such comparison in double blind trials using volume matched material from identical masters you can't say that any difference is down to things like bit depth or sample rate. Given the science of hearing and digital audio says that the effects of these high bit depths and sampling rates are imperceptible without bionic ears (see above), it is much more likely to be down to :

* The masters are different - using better equipment, personnel or more time the record company could make a nicer sounding master for your SACD, but a CD of that master would sound just as good if they made one.

* The volume is not matched - experiments have show that differences in levels even below 1dB cause people to exhibit marked preferences for the louder music.

* Expectation bias - the fact that you know you are listening to SACD and that is 'better' is an example of an extremely common bias that double blind trials are designed to remove as you don't know which source you are listening too (eg. by hiding the players behind a curtain and having another person do the selection of players).

* Single blind trials. Suppose you do get a friend to play you identical music from a CD and SACD with the players behind a curtain so you can't see. Turns out that your friend may well unintentionally send you subtle clues in terms of body language, voice intonation etc. as to which is playing. The double blind part is that your friend shouldn't know either which is playing. Using a computer programme to randomly switch sources and only show you afterwards which was playing when, is a good way of doing this.

My overall thought - It's fantastic that we all enjoy music and care about it so much. My concern is that as music lovers surely we are better spending our money on things like more music, going to concerts or taking up a instrument ourselves than on more expensive & seemingly impressive audio formats which in reality are of no practical benefit.
 
I'll have go....

24bit audio and higher sampling rates ARE very different from normal 16 bit CD audio at 44.1KHz but our poor human hearing is not equipped to notice the difference. ...

Your argument is based on pure sample theory. But misses a few things.

1) Dithering. It's common whaen matering a CD to record a series of samples like "11, 12, 11, 12, 11, 12" so as to have the effect of doing "11.5, 11.5 11.5" with a higher sample rate ditering works better. So the high sample rate can actually improve the effective bit depth

2) No one listens to pure samples. In most cases (except for a few well informed audiophiles) they will apply a digital volume control, basically a multiplier or shifter. 24 bits of samples in a way makes up for the bits "lost" in processing due to integer math round off error that occurs between most CDs and thr D/A converter. This is the same reason engineers like to record in 24-bit, it allows for loss in mixing

3) You were talkig only about frequency response and dynamic range but said nothing about signal to noise ratio. the higher sample rate in theory gies a better S/N by a factor of about 1.4

But I agree, over all few people will notice the improvement, CDs can be very good. Noticing the different requires both good equipment (that most don't own) and years of experience with listening carefully. Man musicians can hear this well some in the recording industry can but must consumers lack interest in developing such skill and it absolutely does not come without effort

Oh and then there are experiments where people have been found to be able to hear 40K and higher. Not directly but lets say there are two instruments and one has an overtone (harmonic) at 40K and the other at 42K. You should expect to hear a beat frequency at 2K. I think this is one of the ways we locate the source of sounds by hearing the beats between ultrasonic harmonics. If so it is best to record these sounds. try it yourself with a pair of audio signal generators. Not psuedo science there is math to back it up
 
Haven't read the last few pages, but just very interested to get an email from hdtracks.com with this header... just wondering if there could be a link...?
 

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Haven't read the last few pages, but just very interested to get an email from hdtracks.com with this header... just wondering if there could be a link...?

I can see it now: "Mozart! Now with twice the bitrate!"

As much as I would like it to say "Big mainstream Label now available!"
 
Hey, if Apple starts offering Apple Lossless audio for their digital music downloads, I'm all for it!

Only one downside, though: ALAC files tend to be huge on a per minute basis, and I wouldn't recommend them for any iPod that stores under 20 GB of media storage.
 
Your argument is based on pure sample theory. But misses a few things.

Actually, he's pretty much spot on, IMO.

1) Dithering. It's common whaen matering a CD to record a series of samples like "11, 12, 11, 12, 11, 12" so as to have the effect of doing "11.5, 11.5 11.5" with a higher sample rate ditering works better. So the high sample rate can actually improve the effective bit depth

Any format that offers 96kHz or 192kHz sampling rates also typically offers greater than 16-bit word lengths. Achieving real-world dynamic range much beyond 20-bit at volume levels the human ear can tolerate is pretty much impossible, so why would you ever need greater "effective bit depth" than what's already possible using word length? Dithering does need time to affect perceptible changes in the noise floor, but I don't see how increasing the sampling rate beyond 44.1kHz even at 16-bits would be any more effective given the white noise itself is also encoded at 16-bits. Sony created Super-Bit Mapping (noise-shaping dither) many years ago and I've seen evidence of real-world increases in apparent reduction of the noise floor in critical human perception areas of perhaps 3dB or so maximum (going by memory), which isn't going to replace 20-bit words any time soon, but might be worthwhile in some circumstances (it didn't use higher sampling rates, though). It also shifted the noise to another frequency range (i.e it didn't actually increase overall dynamic range, but relied on human perception/audibility at one frequency range versus another to given apparent dynamic range improvements.


2) No one listens to pure samples.

WTF is a "pure sample" ??? :confused:

A sample is a sample, regardless of whether it contains a square wave or a sin wave or a tiny sample/part of a much more complex signal.

In most cases (except for a few well informed audiophiles) they will apply a digital volume control, basically a multiplier or shifter. 24 bits of samples in a way makes up for the bits "lost" in processing due to integer math round off error that occurs between most CDs and thr D/A converter. This is the same reason engineers like to record in 24-bit, it allows for loss in mixing

Who are 'they'? If you're worried about the effects of a digital volume control on the lowest bit, use an analog pot instead. But as you turn down the volume, you need less dynamic range to represent the signal in question so it's pretty much tit-for-tat other than the noise floor of the DAC itself.

In other words, digital volume controls aren't nearly as evil as some have made them out to be in certain "high-end" circles. In any case, my 2-channel listening room has a straight analog pre-amp (with motorized volume control), a custom active crossover and two amplifiers connected to Carver Ribbon speakers (AL-III), which are the same ribbons used in the $50,000 per pair Genesis II speakers (albeit additional drivers and different woofer). So it's not like I'm listening to cheap speakers from Radio Shack here.

But I agree, over all few people will notice the improvement, CDs can be very good. Noticing the different requires both good equipment (that most don't own) and years of experience with listening carefully. Man musicians can hear this well some in the recording industry can but must consumers lack interest in developing such skill and it absolutely does not come without effort

You can listen carefully all you want or even delude yourself all you want (an all-too-common thing in the high-end audio circles, IMO), but it won't change an opinion into a fact. Double blind tests (the only real proof of perception differences) always seem to expose snake-oil and high-end magazine claims (that seem to benefit valuable advertisers more than anyone else).

However, quite a few people made good money selling overpriced (I think they went for $25-30 each) green marker pens to outline the outer edge of their CDs, which somehow magically made things sound better (supposed jitter reduction despite the fact it could not be measured in any way) among many other devices of dubious value. In other words, there's all kinds of "snake-oil" in the high-end audio arena. I haven't read magazines like Stereophile (and worse yet, Stereo Review, which I think was changed to a home theater rag) in a decade or so, but they perpetuated it by giving credence to nonsense and junk science in order to sell advertising to the people that make such crap. They didn't do the readers any real favors in the process. People that could have put their money into better speakers or room treatments put it into multi-thousand dollar DACs and transports instead which usually had less than 0.01dB in difference from a $25 DAC (i.e. virtually inaudible and certainly at least a magnitude less than a typical loudspeaker would create over a given frequency range).

Oh and then there are experiments where people have been found to be able to hear 40K and higher.

Care to quote some links to these experiments? I've found some evidence of the inner ear being able to hear ultrasonic frequencies, but not the outer ear.

Not directly but lets say there are two instruments and one has an overtone (harmonic) at 40K and the other at 42K. You should expect to hear a beat frequency at 2K. I think this is one of the ways we locate the source of sounds by hearing the beats between ultrasonic harmonics. If so it is best to record these sounds. try it yourself with a pair of audio signal generators. Not psuedo science there is math to back it up

A beat frequency requires a non-linear medium to create it. If that medium is in the room where the sound is being recorded, it will be picked up within the bandwidth constraints of audible hearing (and it might not be a good thing in some cases, such as unwanted distortion off some kind of treated walls of the room that weren't meant to be recorded in the first place).

Now if that non-linear medium is supposedly the ear drum itself, it seems at least theoretically possible that a beat frequency might be missed by the brain with the recording that might exist in the real world (with say violins that would produce many such harmonics). I've never seen any really convincing evidence that such a thing is audible in practical situations. At best, it would be hard to discern from other distortions in the playback chain compared to a real event in the room. Most microphones will not even record linear above 25kHz, so getting a signal source to even compare might prove very difficult. I've certainly never read of any actual acoustic content above 40kHz on something like an SACD recording. If the source of sound is artificial, there is no basis at all for comparison to a real world event.

What I'm saying is that evidence for any reproduction benefit of information above 20kHz is controversial at best and considered non-existent at worst. My Carver AL-III Ribbons begin with roll-off above 17kHz and pretty much don't output anything significant above 22kHz anyway so it wouldn't do me much good to test any purported material anyway. My PSB speakers in my home theater room are good to perhaps 27kHz. Short of adding an ultra-sonic tweeter, they're not going to produce much ultrasonic energy to even test it. Most speakers have similar limits (give or take) and thus the question of high sampling rates becomes moot once again. At the very least, I can be reasonably certain that the industry's use of such rates had nothing to do with the average reproduction chain (i.e. what's considered when mastering a typical audio recording). Certainly, I don't think ultrasonic reproduction is the first thing even high-end audio magazines (snake-oil orientated or not) look at when considering what is a great sounding speaker and yet I've seen many put emphasis on the 192kHz sampling rate even though it's pointless compared to 96kHz (which is pretty pointless compared to 48kHz as well, but 192 is just absurd; show me ANY speaker of merit that can reproduce 96kHz and then show me a signal that has any musical information at that frequency to reproduce in the first place).

In any case, I'm not worried about such purported content given the lack of sources, the lack of playback equipment (that I own) and the fact I'm not missing anything.

Ultimately, I still maintain that if Apple is considering 20-bit or 24-bit encoded files in the future that their reason is to re-sell the files based on the ignorance of most people as to the supposed benefit of such a change, not any actual sound quality differences. In fact, I can imagine them re-encoding 16-bit sources (i.e. no content above 16-bit present in the final file) and still giving it a 24-bit label. The key would be in the wording of the sale.

Lossless encoding (regardless of bit/rate) would be a far more substantial offering, IMO, especially given many CDs cost about the same as iTunes compressed offerings and also give you a re-saleable medium (no signature watermarks, regardless of DRM), artwork, etc.
 
I consider myself to be an audiophile, in the sense that I appreciate high quality sound reproduction, but unlike most others, I consider the equipment involved, not a hobby, but just a means to an end.

I have a little challenge for all those people who glibly state how easily they can fear the superiority of SACD over CD, or 16 bit vs 24 bit, 44.1khz vs 192khz and 256 kbps compressed iTunes tracks vs the uncompressed source. I have made a composite track which contains segments that are 223 kbps AAC and others which are uncompressed.

Download and listen to the track and tell me the time codes for the different sections. Bet you can't. :cool:

http://hotfile.com/dl/106252620/faa0282/Srceplus223aac.rar.html
 
Download and listen to the track and tell me the time codes for the different sections. Bet you can't. :cool:

I'll give it a try. I do have semi high-end components in my rig (DAC/headphone-amp/headphones). Gonna be interesting :)
 
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