Hard Light Productions Forums
Off-Topic Discussion => General Discussion => Topic started by: The E on March 06, 2012, 03:17:00 am
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So this article recently popped up on my radar:
http://people.xiph.org/~xiphmont/demo/neil-young.html
TL;DR: 24-bit, 192 kHz recordings are bull****, and you should not be paying extra for them.
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Should I ditch my $5000 speaker cables :D
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No those are entirely necessary and enhance your* audio enjoyment by at least 78%**.
*If "you" are a retailer of $5000 speaker cables
**Numbers may not be accurate to your experiences. Local variations in the earth's magnetic field, the phase of the moon, and preexisting neurological conditions may have effects.
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So this article recently popped up on my radar:
http://people.xiph.org/~xiphmont/demo/neil-young.html
TL;DR: 24-bit, 192 kHz recordings are bull****, and you should not be paying extra for them.
or wasting the extra bandwith on "hifi recordings" that sites like what.cd like to tout. FLAC, 24-bit/192kHz. f you all.
in shot, FLAC ftw, but still, nothing can save a badly mastered/overly compressed recording in any case. some **** will always sound bad.
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I agree, FLAC is still awesome, but theres no point bothering with any encode above 48kHz/16bit. Also agree on the fact that nothing can save the (in large part) horrible mastering of modern popular recordings (or the engineers that master them for that matter).
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This is a sort of thorny issue. I agree that on final recordings, such bit depth and sample rate are most likely wasted. Physiologically, humans aren't well equipped per se to distinguish between higher and lower quality recordings, IF the recordings are otherwise identical.
But.
What about editing? Digital data has a tendency to degrade when you edit it due to rounding to closest available value. This is, incidentally, the reason why professional image editing is usually done in more than 8 bits per channel depth. The final image is usually not of such bit depth (though there are exceptions where that is useful, such as height maps distributed by NASA, which are 16-bit greyscale RAW images).
Similarly in audio editing, errors will at some point start to accumulate, and using higher bit depth definitely reduces the amount of error you end up with in any specific operation.
Whether or not you would be able to hear the difference depends almost entirely on the amount of editing done to the track, but the fact remains higher quality audio will be possible to edit further without degradation becoming noticeable. Obviously, for the final result, 16-bit accuracy per sample is more than sufficient for the majority of listening purposes.
Also, in games things like spatial effects can be calculated at higher precision at higher sample rate/bit depth. Technically, this means somewhat better quality on the output audio, even if original audio samples are in 44k16b stereo standard, mixing in higher quality can lead to better results.
Then there's the issue of sample rate. Higher sample rate results in much higher accuracy with overtones, and that kind of thing is quite important in certain types of music, especially electronic (which uses a lot of non-natural waveforms such as square waves) and orchestral music. Of course, in any type of editing (post processing and such) the sample rate assists in making the resulting waveform more precise before it is downsampled to final release format. But, more crucially, there's the issue of downsampling which is probably the single most important factor in people wanting to get their hands on recordings in their original quality.
If you reduce the sample rate from 192 kHz recording to 44.1 kHz, there will likely be some trouble somewhere; whether it is audible depends on the content of the audio as well as whether or not you have the original source to compare to. Let's illustrate what happens in such a case.
I made a 192000 Hz sample rate, 12000 Hz square wave in Audacity. It sounds almost identical to a sine wave, actually, since the peaks and bottoms of the wave are equally distributed and 12000 Hz happens to be a factor of 192000 Hz, so all is fine and dandy. I saved it as 192kHz FLAC.
Then I set project quality first to 96000 Hz, then 48000 Hz and finally 44100 Hz.
Here is a visualization of the waveform degradation due to downsampling:
(http://img689.imageshack.us/img689/9932/squarewaves.png)
As you can see, there is actually a fair bit of quality reduction here if you compare the waveforms directly to each other. Well, it turns out this is not really a problem down to 48000 kHz, because the waveform remains constant and symmetric due to the frequency 12000 Hz being a factor in all three frequencies; 4x12 = 48, 8x12 = 96 and 16x12 = 192.
But then the problems start.
12000Hz is not a factor of 44100 Hz. The samples are mis-aligned. So, in the process of downsampling to 44.1 kHz, the audio signal that you actually hear will change. It will no longer be a sine wave. Here's a zoomed out image that shows the problem:
(http://img525.imageshack.us/img525/5475/squarewaves2.png)
As you see, the mis-alignment of samples creates sub-tones to the signal, oscillations that you actually sound as multiple tones being played at the same time.
Here are all four signals in FLAC, so you can try it out yourself.
12000Hz-Sqare-Wave-Downsampling.7z (http://www.mediafire.com/?5h8gzv454zw80gy)
You'll notice a pretty clear and distinct change in the 44.1 kHz version compared to other three.
And now the reason for the need for having recordings in their original quality makes a little bit more sense, I hope. CD Audio is still the industry standard for most audio that is put on the market. CD Audio is 44.1 kHz by definition, while the recordings are usually done in another sampling rate.
At the very least, it would be a welcome change for most music to be marketed in a sample rate that would nicely factor into the original sample rate. 48kHz would be good. 44.1 kHz is, as demonstrated, technically problematic and mostly used for legacy reasons.
So, is the 44.1 kHz sampling rate good or bad? It depends. It is certainly sufficient in delivering good audio, especially if the sound is converted directly from analog signal to 44kHz digital signal (bit depth notwithstanding). Problems start when you start messing around with digital signal and downsampling them from one to another, especially if your resultant sample rate is not a nice factor of the original.
So is 44.1k audio better or worse than original audio?
It's never better in quality, and can be worse. But it depends so much on the signal content and downsampling algorithm that it's very hard to give any definite answer. Personally, I think if original recordings are made in digital format, for goodness' sake at least use a sensible downsampling scheme that doesn't result in atrocious bit crushing. If original audio recording is in analog format, then 44.1 kHz sample rate is probably very much sufficient.
EDIT: The article addresses many of these issues mentioned, and especially points out that higher bit depth and sample rate are used in editing and post-processing for exactly these reasons. Teaches me to read an article rather than just skim over it... :p
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it doesnt sound kvlt!
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not that ive ever enjoyed the sound of a 555 timer, a square wave will have a very hard time staying square in the analog portions of your sound card/stereo equipment. hook your line out to an oscilloscope and look at the waveform after being passed through the analog parts of the sound card. the signal gets smoothed out and filtered. the square wave example probably wont look very square at this point. and then when it goes through your amp, crossovers, and then to your speakers, it will have gone through several more levels of filtering.
ive come to the conclusion that better speakers (larger speakers in the woofer, midrange, tweeter configuration) tend to result in better sound. now i got my speakers at a thrift store, for $2 each, and they sound way better than any set of computer speakers ive had (especially 5.1 systems where its all midranges and a single sub, and not a tweeter anywhere). the reason is that the input frequency is divided up into frequency bands appropriate for the speaker types to which they are routed. of course its not like i really care. my stereo looks like it was made in the 80s. the tape deck doesnt work, the radio doesnt work (probibly due to lack of radio stations in this part of the world), the turntable works but the only record i have is an old hank williams album i stole from my grandma's collection. the amp likes to cut out the right channel if it turn it up too high. and half the equilizer sliders are stuck. i cant wait for it to fail so i can salvage its parts. but its what i have and what i use, and it sounds way better than the last set of 5.1s i had.
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not that ive ever enjoyed the sound of a 555 timer, a square wave will have a very hard time staying square in the analog portions of your sound card/stereo equipment. hook your line out to an oscilloscope and look at the waveform after being passed through the analog parts of the sound card. the signal gets smoothed out and filtered. the square wave example probably wont look very square at this point. and then when it goes through your amp, crossovers, and then to your speakers, it will have gone through several more levels of filtering.
ive come to the conclusion that better speakers (larger speakers in the woofer, midrange, tweeter configuration) tend to result in better sound. now i got my speakers at a thrift store, for $2 each, and they sound way better than any set of computer speakers ive had (especially 5.1 systems where its all midranges and a single sub, and not a tweeter anywhere). the reason is that the input frequency is divided up into frequency bands appropriate for the speaker types to which they are routed. of course its not like i really care. my stereo looks like it was made in the 80s. the tape deck doesnt work, the radio doesnt work (probibly due to lack of radio stations in this part of the world), the turntable works but the only record i have is an old hank williams album i stole from my grandma's collection. the amp likes to cut out the right channel if it turn it up too high. and half the equilizer sliders are stuck. i cant wait for it to fail so i can salvage its parts. but its what i have and what i use, and it sounds way better than the last set of 5.1s i had.
Sounds a lot like my setup; a bunch of parts either given to me by family or friends that they were going to throw out, or purchased at yard sales and thrift stores. I made my own surround system that way (the 42" Hitachi widescreen rear projection TV I found for free lets me use it's 40w set as the front pair), and it sounds excellent.
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EDIT: The article addresses many of these issues mentioned, and especially points out that higher bit depth and sample rate are used in editing and post-processing for exactly these reasons. Teaches me to read an article rather than just skim over it... :p
I don't blame you. I didn't notice the part where the article mentioned any use for 192KHz until you posted this.
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I read somewhere that more speakers (because of the chorus effect--the reason why pianos use more than one string on the majority of its keys) produce a better presentation of sound than larger speakers do, which explains why the Marshall half-stack guitar amplifier I have that has four twelve-inch speakers in its speaker cabinet produces a better sound than one big speaker.
Though it probably depends entirely on the signal being sent to it, which is probably the whole point of this thread :nervous:
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I seem to recall that 44Khz 16Bit provides a wider range of audio than the human ear can detect anyway, but yes, I'll agree that from an editing point, the higher the resolution of the sound, the better, because you have a lot more control before the quality starts to degrade.
The problem with most audio recently is the 'loudness wars' bought about by record labels, rather than allowing the mixer to provide the best quality of sound, they are told instead to focus on making it as loud as possible when played over the radio etc in order to fix it in people's minds. Listen to Derezzed from Tron 2 for an excellent example of depth being destroyed by compressing the living **** out of the audio...
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I read somewhere that more speakers (because of the chorus effect--the reason why pianos use more than one string on the majority of its keys) produce a better presentation of sound than larger speakers do, which explains why the Marshall half-stack guitar amplifier I have that has four twelve-inch speakers in its speaker cabinet produces a better sound than one big speaker.
Though it probably depends entirely on the signal being sent to it, which is probably the whole point of this thread :nervous:
while my sound system is essentially stereo with 2 large speakers there are technically 6 speakers in the mix (tweeter, mid, sub for each box). a lot of the computer speaker setups (especially the 5.1s) have a lot of identical mid-range speakers and usually a single sub. ive never found these systems to sound very good, especially for music, which needs to be upmixed from stereo to 5.1. id actually lean toward a quadrophonic setup instead with another pair of similar speakers. speaker arrays are nice provided that the there are speakers that cover different frequency ranges.
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I read somewhere that more speakers (because of the chorus effect--the reason why pianos use more than one string on the majority of its keys) produce a better presentation of sound than larger speakers do, which explains why the Marshall half-stack guitar amplifier I have that has four twelve-inch speakers in its speaker cabinet produces a better sound than one big speaker.
Hehe, guitar amplification and audiophile stuff are quite different issues, though they do both inspire otherwise rational people to spend ungodly amounts of money in search of their own personal holy grail of sound. The four very similar speakers in your cabinet will produce some phasing effects (which some people think helps to round out the sound), though it's hard to say that it's objectively "better" than the sound produced by a single-speaker cab. It can certainly be louder though, and it gives the manufacturer the option to wire things up differently.
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no number of speakers will make my guitar playing any better. the least it would do is piss off the neighbors and the most it can do is get us evicted.
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Best/worst guitar playing advice I ever got: "If you're loud enough, you don't have to tune". That guy was awesome.
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Huh. Good article. Glad to know my recording classes taught me correctly. (My professor was adamantly against 24/192 as a distributed recording for essentially these reasons. Though he never really went into the science of it all.)
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ive always figured 24 bit would work just fine to increase the resolution, you just have to decrease the increase in voltage per increase per value so that the maximum signal voltage is no greater that it would be with 16 bit. actually thats how dacs work. dacs map values to a range set by a pair of reference voltages. of course that said i really dont care, cd quality is better than what i usually listen to anyway.