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There is no point to distributing music in 24-bit/192kHz format.

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Re: There is no point to distributing music in 24-bit/192kHz format.

#111

Post by xiphmont »

yzhou wrote: post #83

The hearing of ears is a time-domain thing, not a frequency domain thing. It's the frequency response of all the frequency components added together. people might not be able to respond well to a single high frequency tone, but might respond well to a combination of tones.

No. It's both.

The basilar membrane is a loosely tuned resonator. The hair cells placed on it fire beginning on the positive zero crossing. So, to a first approximation, the ear is in fact a filterbank.

There is a time domain component in that the cochlear nucleus contains nerve cells that watch multiple hair cells at a time and correlate the firing in several different ways. Some attempt to discriminate pitch, some convolve and correlate in-phase firing energy, some look for tones to end, etc. This information is then forwarded on to the brain.

However, getting back to your point, no hair cells will fire if the basilar membrane doesn't move, and it's tuned to a frequency range.

Re: There is no point to distributing music in 24-bit/192kHz format.

#112

Post by nullc »

sliverstorm wrote: post #9

So, presuming we take this example: http://people.xiph.org/~xiphmont/demo/jaggy2.png The key to reproducing the original signal from the digital signal is a low-pass filter that rejects everything above the sampling rate, correct? That is to say, what I am getting at is while the original signal can be reproduced, it requires properly tuned, and probably reasonably high performance, hardware to remove the higher freq…

Yes, thats basically it. They do this _exceptionally_ well in fact.

Typically the technique used inside DAC is to digitally upsample the signal (by duplicating samples, often to a few MHz— also allowing them to use a low bit-depth DAC) then it applies a very sharp "perfect" digital filter to cut it right to the proper passband (half the sampling rate). The analog output then contains only a tiny amount of ultrasonic aliasing which is so far out that it's easily rolled off by simple induction in the output.

This isn't just theory. Here is a wav file I made at a 1kHz sampling rate, where every other sample is -.25/.25: http://people.xiph.org/~greg/1khz-sampled.wav (so a 500Hz tone, the highest you can represent with 1kHz sampling).

Feeding that file to a boring resampler (I used SSRC, but anything should give roughly the same result— a least when not quite so ridiculously close to nyquist, most will attenuate near-nyquist data extensively) and get this: http://people.xiph.org/~greg/1khz-sampled-to-48khz.wav

Here are the two signals plotted against each other: http://people.xiph.org/~greg/1khz-to-48khz.png

As you can see— the 500Hz sinewave is reconstructed perfectly. (Of course, a 500Hz square wave would not be (you'd get a sinewave out) but this is because a 500Hz square wave contains energy far beyond the nyquist of 1kHz sampling).

Here is a spectrograph of the same signal http://people.xiph.org/~greg/1khz-to-48khz-spec.png showing that the tone is indeed pure (the faint background noise is the dither the resampler applies when requantizing its high precision intermediate format back to 16 bits).

Re: There is no point to distributing music in 24-bit/192kHz format.

#113

Post by ef4 »

noonespecial wrote: post #84
zb wrote: post #79

Earlier quoted context omitted.

If the beat frequency is audible, it will be on the recording. Obviously.

That would suppose that the recording device precisely matched the orientation of the listener, and the recording was not created digitally in (multi-track fashion for example). There would have to be air space in order for the interference pattern to set up in. So you'd be right if your mics were head spaced and in the venue. But you'd still have secondary data, with the original lost.

> But you'd still have secondary data, with the original lost.

By that standard, the original is always lost unless you have a completely holographic recording. 192kHz doesn't help with that problem at all.

Re: There is no point to distributing music in 24-bit/192kHz format.

#114

Post by hackermom »

davesims wrote: post #95
hackermom wrote: post #39

Earlier quoted context omitted.

Same goes for 16/24 bit, however, the difference between 16 and 24 bit is actually audible No, the difference is not audible at all. At 16 bits of depth on a normal low-level audio signal (~0.3 volts), we're talking about less than 0.000005 volts per amplitude step. This difference gets lost in the THD already at the DAC in your audio output stage. Then it gets lost again in the amplifier. And again in the cable to y…

Heh, it's funny to see this late-nineties debate get re-hashed here. Also kind of fun. If it were true that there's no audible difference between 16 and 24 bit, companies like Alesis, Otari, ProTools, etc. wouldn't have spent the last 15 years ditching 16 bit like an old pair of smelly sneakers. (better metaphors welcome). Seriously, anyone who has sat down in a real listening environment for 5 minutes A/Bing 16 vs 2…

Yes, and anyone who has ever sat down infront of an LCD flatscreen watching their favorite movie on DVD/BD using gold-plated $200 HDMI cable instead of $4.99 Walmart HDMI cable see the extra sharpness immediately. This is why non-gold plated non-OFC HDMI cables are down to $4.99 a piece from their original $49.99 during introduction.

Re: There is no point to distributing music in 24-bit/192kHz format.

#115

Post by Anechoic »

For those of you who are interested in just how much of a golden ear you truly are: download Harmon's "How to Listen" software for Windows or Mac OS X http://harmanhowtolisten.blogspot.com/ (scroll down).

Harmon requires its trained listeners to pass tests based on this software before participating in juries to evaluate Harmon products. It doesn't directly address the sample rate/bit depth issues discussed in the linked article, but it does address a lot of the issues brought up in the HN discussion, so you can have a chance to see how much those characteristics really matter.

You may be surprised.

Re: There is no point to distributing music in 24-bit/192kHz format.

#116

Post by sjwright »

davesims wrote: post #71
leephillips wrote: post #17

Earlier quoted context omitted.

The linked article was accurate. You are confused. "I'm an ex-audio engineer" Hard to believe. "The distinct sound of the triangle constitutes of a high fundamental frequency, ballpark 10kHz" That's a pretty high note - higher than the top key on the piano. But an "audio engineer" would know that. "many very high-pitch harmonics" Since the next harmonic after the fundamental would be at 20khz, which only young people…

Another "ex audio engineer" here, you can believe or not at your leisure. Many hours spent in high-end recording and mastering environments. I'm not sure what your background in audio is, but everything he says is correct. High end frequencies well past 15k and up (22.1k actually) are widely acknowledged to influence the lower frequencies and play a huge role in the perception of the quality of a recording. This is a…

I have also spent many hours spent in high-end recording and mastering environments, and it's my observation that most engineers suffer from confirmation bias just like everyone else on the planet.

I've caught engineers using L1-Ultramaximizer (or similar) to bounce a recording down to 16-bit/44.1khz as part of the mastering process, and they're always surprised when they're completely unable to hear the difference even in the most simple cover-the-screen-and-toggle-bypass test.

Re: There is no point to distributing music in 24-bit/192kHz format.

#117

Post by cdvonstinkpot »

davesims wrote: post #95
hackermom wrote: post #39

Earlier quoted context omitted.

Same goes for 16/24 bit, however, the difference between 16 and 24 bit is actually audible No, the difference is not audible at all. At 16 bits of depth on a normal low-level audio signal (~0.3 volts), we're talking about less than 0.000005 volts per amplitude step. This difference gets lost in the THD already at the DAC in your audio output stage. Then it gets lost again in the amplifier. And again in the cable to y…

Heh, it's funny to see this late-nineties debate get re-hashed here. Also kind of fun. If it were true that there's no audible difference between 16 and 24 bit, companies like Alesis, Otari, ProTools, etc. wouldn't have spent the last 15 years ditching 16 bit like an old pair of smelly sneakers. (better metaphors welcome). Seriously, anyone who has sat down in a real listening environment for 5 minutes A/Bing 16 vs 2…

...And the new Pro Tools 10 just added the ability to record in 32-bit floating point. http://www.avid.com/US/products/Pro-Tools-Software

Re: There is no point to distributing music in 24-bit/192kHz format.

#118

Post by _zeo8 »

In normal listening conditions and for most people the difference between 16/44 and 24/192 is inaudible.

Given a 5 minute song, if I have the choice to download a 11MB file (320kpbs MP3) or a 330MB file (24/192) I would of course choose the 11MB file. The sound quality is perfectly acceptable and the file size much more convenient to manage (storage, backups, etc.).

In terms of the convenience of managing the file size and sound quality I think 320kbps MP3 is the best compromise.

Here's a file size comparision of a 5 minute stereo song:

MP3 128kbps > 5 MB

MP3 320kbps > 11 MB

Uncompressed 16/44 > 50 MB

Uncompressed 24/192 > 330 MB

When talking about sound quality there is a much more relevant issue: the amplitude compression (distortion) abuse used by mastering engineers and producers that totally destroys the dynamic and life of the sound. That is a real issue. When buying a song there should be two versions to choose from:

A) "Loud", dynamically destroyed / distorted version.

B) Normal, dynamic, non-distorted version.

Today only version A is available to buy.

Re: There is no point to distributing music in 24-bit/192kHz format.

#119

Post by brigade »

sjwright wrote: post #78

I must say I get rather irritated when people spend time worrying about dubious 'tweaker' methods to improve their audio, when the most under-performing component of most people's sound equipment also has the lowest-hanging fruit: The room itself. When people ask me where they should spend money to improve the quality of their hi-fi or home theater system, in nearly every case my response will be something like "get…

I'd disagree, at least among the people I know: they all have cheap HTIB systems and the single biggest, most cost-effective improvement you can make from there is to buy better speakers.

Then you can worry about your room.

Re: There is no point to distributing music in 24-bit/192kHz format.

#120

Post by sjwright »

xiphmont wrote: post #96
davesims wrote: post #71

Earlier quoted context omitted.

Another "ex audio engineer" here, you can believe or not at your leisure. Many hours spent in high-end recording and mastering environments. I'm not sure what your background in audio is, but everything he says is correct. High end frequencies well past 15k and up (22.1k actually) are widely acknowledged to influence the lower frequencies and play a huge role in the perception of the quality of a recording. This is a…

As much respect as I have for Bob Ludwig's hard won mastering skills, he also strongly believes in $n,000/foot speaker cable, which is what he has installed at Gateway. So by all means give him well deserved props, but don't assume he's an expert on all aspects of audio theory or practice.

I've always thought the most expensive speaker cable sounds a lot better... to the wallet of the salesperson.
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