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how much can e6600 take ? (pg. 4)
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| DigiNut |
| quote: | Originally posted by camsr
...at the 192 sample rate things become more fluid and continuous. And it STAYS that way even when it's downsampled to 48 or 44 or whatever. |
Highly unlikely, unless it's specifically programmed to behave differently at the higher sample rate (rarely the case). You can't really dither to a lower sample rate the same way you'd dither to a lower bit rate, so you're always just cutting out half of the data points and as a result you'll end up with the exact same thing you would have had if you'd just set the SR to 44 kHz in the beginning.
What a lot of people miss is that when you work with a 44 kHz signal, you're not jamming the digital signal through an analog speaker. It's actually getting interpolated on the way out, using algorithms that recreate the analog signal nearly perfectly. So 192 kHz may seem more "continuous", but the eventual output signal is infinitely continuous no matter what sample rate was used.
The only reason you'd see a difference in the analog signal is if some really wacky stuff went on between the data points, and in order for that to happen, there'd need to be frequencies much higher than 22 kHz in the track. If you've got frequencies higher than that, rest assured that none of your listeners can hear them. If you're making music for dogs, then you can start to worry about those.
I suppose if you were working with a really unbelievably ty DAC, like one with some naive straight-line interpolation, you might get better results with a higher sample rate. But just about any board that supports 192 kHz ought to have a halfway decent interpolator. |
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| -_1_--Ben--_1_- |
| quote: | Originally posted by dj_palm
exactly what i tryed to say to but it ended up in the conclusion that my ears are bad. |
it was a question!!! :) crazy
or, you mean your own conclusion ?
but, derail is right about what he says |
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| DJ RANN |
Eldrtich and Diginut you're wrong, as many top studio engineers will testify. Well done for regurgitating the nyquist theorum, I too learnt it in the first two hours of of the first class of audio engineering school.
Just because the frequencies being recorded are lower than the upper limit of a given sample rate, doesn't mean there will be no benefit in using a higher sample rate. Becuase if various factors (such as more harmonics being captured, greater resolution on a horizontal plain i.e. "snapshots" etc.) artifacts will remain causing a perceievable difference even when rendered down to lower sample rate.
And rendering down sample rates from 192 to 44.1 doesn't just work by dropping frames out on a sequencial basis (i.e dropping every other frame to go from 96 to 48). The algorythms are far more complicated than this.
This is all theory though and I seriously doubt that anyone on this forum (myself included!) has both the equipment, ear and knowledge to allow all the other necessary elements facilitate the potential benefits of using a higher sample rate. Unless anyone's got a two inch studer tape machine that they use for recording the final mixdown.............. |
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| derail |
Well, the question of how something is rendered from 192 kHz down to 44.1 kHz is a separate issue. It doesn't really resolve whether or not the audio quality of the resulting audio CD is better or not.
People's ears just aren't as finely developed as their eyes. Put a 1366*768 TV next to a 1920*1080 screen and generally the higher resolution screen is noticeably more detailed (it also depends on the manufacturer of the TV as well as the show or movie that's playing)
With audio it seems to be much more about the skill of the engineer. I haven't been exposed to 96 kHz or 192 kHz often, but it never hit me as noticeably better. I could very easily listen to 44.1 kHz afterwards and have no problems with it.
Right now I don't see a major consumer push for higher audio quality. DVDs and blu-ray, HD-DVD have gone to 5.1, 6.1 etc surround sound, that's where the push is currently - for movie audio rather than music audio. Surround sound music is currently a niche market. For dance music, which is going to be played in arenas and clubs of all different shapes and sizes, you can't fit the whole crowd into the centre of the room so they can hear all the speakers at once. Some of them are going to be directly in front of one speaker. So you have to make sure that the whole mix is coming through every speaker. The mixes have to work in mono, otherwise different people are going to hear drastically different things.
Also, portable music is much more about mp3 players than high end CD players (hmm..you can probably get super audio cd (SACD) players, though I'm not sure how much trance is released in the SACD format). People are happily listening to music where everything above 16 kHz has just been chopped off.
The higher sample rates, from my viewpoint, seem to be a good thing to have on a soundcard's specification overview ("now goes up to 384 kHz!") and appeals to buyers of the hardware, but the consumers don't seem very interested.
I'm totally open to seeing things differently if affordable 192 kHz portable audio players become available and most music is released in that format! |
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| Eldritch |
| quote: | Originally posted by DJ RANN
And rendering down sample rates from 192 to 44.1 doesn't just work by dropping frames out on a sequencial basis (i.e dropping every other frame to go from 96 to 48). The algorythms are far more complicated than this.
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Of course it doesn't. That would introduce a whole lot of aliasing.
It's funny that you can't provide any facts supporting your claim.
Even if there is different harmonics in the 192kHz recording, they will be destroyed by the antialiasing filter. |
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| camsr |
This argument is just as pointless as the bit depth one -- it only matters during production. Listening with 16-bit, 44100hz media is enough. But when it comes time to do discrete mathematics on sets of numbers, the more the merrier.
http://en.wikipedia.org/wiki/Rounding_error
Use as many bits or samples or whatever it takes to get the high quality media to a large enough precision so it can be "chopped" to a lower precision, THAT IS STILL HIGH ENOUGH TO GO UNNOTICED. |
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| echosystm |
Guys... seriously...
Carrots. |
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| Eldritch |
| quote: | Originally posted by camsr
This argument is just as pointless as the bit depth one -- it only matters during production. Listening with 16-bit, 44100hz media is enough. But when it comes time to do discrete mathematics on sets of numbers, the more the merrier.
http://en.wikipedia.org/wiki/Rounding_error
Use as many bits or samples or whatever it takes to get the high quality media to a large enough precision so it can be "chopped" to a lower precision, THAT IS STILL HIGH ENOUGH TO GO UNNOTICED. |
True, almost all effects and softsynths oversampled anyway, there's no need to oversample the actual project too.
Anyhows, this is going way off topic. |
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| echosystm |
| Eldritch, my feelings for you are potato mash |
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| camsr |
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| echosystm |
TA must be so scary for noobs :haha:
its their own fault for not listening though. |
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| DJ RANN |
| quote: | Originally posted by Eldritch
Of course it doesn't. That would introduce a whole lot of aliasing.
It's funny that you can't provide any facts supporting your claim.
Even if there is different harmonics in the 192kHz recording, they will be destroyed by the antialiasing filter. |
Well this is what diginut was implying unless I read his post wrong
| quote: | Originally posted by Digitnut
.....you're always just cutting out half of the data points and as a result you'll end up with the exact same thing you would have had if you'd just set the SR to 44 kHz in the beginning.
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Also, I don't know why you mention aliasing? - true aliasing and the adding of these components only ever occurs when the sample rate is below twice the relative nyquist frequency. We are only talking about 44.1 and above. Anti aliasing filters only attenuate frequencies above the relative nyquist frequncy in these situations.
People need to stop quoting wiki - it's really not a reliable source as it's editable by anyone. duh.
I go back to my original point that 16b 44.1 is really all we need as the general level of equipment and knowledge does not warrant hihger standards.
There is an argument to say that as moores law progresses in relation to memory and media especially, the enthusiast (followed by the consumer later) will expect higher sample rates to become the norm.
Echo, how come your oar has managed to stay in the boat on this one? Apart from the vegetables obviously.....:D |
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