Twenty Four / 96?
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- KVRian
- 1023 posts since 8 Apr, 2003 from Östersund
256 samples at 96 kHz is not 6ms, it's less than half of that, around 2.5ms or so. Meaning you should raise it to 512 samples when running 96k.
/Majken
/Majken
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- KVRAF
- 4266 posts since 21 Oct, 2001 from my bolthole in the south pacific
It makes sense to increase your buffer size if you are increasing the sampling rate. Go up to 512 if you are playing/recording at 96kHz - you will get the same latency as you do playing with 256 at 48 kHz. Your machine will be much less likely to stumble if it is working with larger buffers.munchkin wrote:I've gone for 24/48 at 256 (6ms) for now. It seems to work well with the Multiface and probably won't use as much disk space as 24/96. I think 24/96 was pushing the Multiface to its limits.
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- KVRAF
- 42529 posts since 21 Dec, 2005
yes yes YES YES YES!egbert wrote:It makes sense to increase your buffer size if you are increasing the sampling rate. Go up to 512 if you are playing/recording at 96kHz - you will get the same latency as you do playing with 256 at 48 kHz. Your machine will be much less likely to stumble if it is working with larger buffers.munchkin wrote:I've gone for 24/48 at 256 (6ms) for now. It seems to work well with the Multiface and probably won't use as much disk space as 24/96. I think 24/96 was pushing the Multiface to its limits.
You won't be f......'n around with the buffer so much in 24/96...........
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- KVRAF
- Topic Starter
- 2336 posts since 13 Oct, 2002 from Terra Firma
Why is does the buffer size double but the latency remains the same when the frequency doubles?
I've set the buffer size to 512 samples at 24/96 which is 5ms. I think that's quite a low latency. Is this a low latency for 24/96 or an average setting?
I've set the buffer size to 512 samples at 24/96 which is 5ms. I think that's quite a low latency. Is this a low latency for 24/96 or an average setting?
- KVRAF
- 16891 posts since 8 Mar, 2005 from Utrecht, Holland
because Time = NrOfSamples / SamplingFreq.
If you double the sampling frequency the time halves. So you can double the buffer size and keep the same latency.
5ms latency is average and very workable I'd say.
If you double the sampling frequency the time halves. So you can double the buffer size and keep the same latency.
5ms latency is average and very workable I'd say.
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- Suspended
- 17889 posts since 14 Jun, 2001 from Somewhere you're not!
Standard for what? Sound-cards? Maybe that has to do with bus-speed?hibidy wrote:24/96 is the standard now...
No it doesn't. Well, maybe, but no more "automatic" than lowering the buffer size yourself.The first and probably the most important reason why 96 sampling rate is highly sought is because it gives you automatic improved latency.
Wow! You know what else you can do? Change the buffer size from 512/256/128/64 etc and see how the latency decreases immediately. Its amazing!Switch any host/card from 44.1/48/88.2/96....etc with a 512 buffer setting, and you will always see the latency increase immdiatly. For example, I run 24/96 with a 256 buffer and it's just sick how low the latency is. It puts a bit extra on my cpu's but it worth it.
In deciding how to preceed, you need to look at your goal. e.g. I only record/render stuff to make a CD from it. So for CD audio, I only ever use 44.1kHz as both 48 and 96 would require some interpolation when converting to 44.1 which could affect the sound quality. I record at 24 bit and render at 32 [ORION doesn't render at 24] because it allows me to record something a little quiet and not have to worry about any loss of quality. Its always easier to normalise audio than it is to remove clipping.
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- KVRist
- 34 posts since 15 May, 2006
I also only record at 44.1 kHz for the same reasons as you, plus space concerns and processing power. But barring those factors, recording at 88.2 and downsampling to 44.1 would make for far easier math than from 48 or 96. I wonder what, if any, sonic benefits could be gained from recording at a higher resolution. I'm of the mind that there are more important variables than sampling rate.BONES wrote:I only record/render stuff to make a CD from it. So for CD audio, I only ever use 44.1kHz as both 48 and 96 would require some interpolation when converting to 44.1 which could affect the sound quality.
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- KVRAF
- 4222 posts since 23 Feb, 2004 from Tucson Arizona USA
I use 48kHz for the simple reason that one of my devices has a very accurate master clock at that frequency, and the related reason that any media I write it to will also be 48kHz. (That's sort of the same reason people who still use CD-Audio format argue in favor of 44.1kHz, but who bothers with that format anymore?)rp314 wrote: I know of no one who has the choice yet continues to record at 16 bits but more and more folks are using 44.1 or 48K for the practical as well as possibly "objective" reasons.
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VladimirDimitrievich VladimirDimitrievich https://www.kvraudio.com/forum/memberlist.php?mode=viewprofile&u=90429
- Banned
- 525 posts since 8 Dec, 2005
Filters tend to exhibit weird behaviours close to Nyquist but its really a moot point since most of them are oversampled internally for that exact reason anyway.noetic wrote:I also only record at 44.1 kHz for the same reasons as you, plus space concerns and processing power. But barring those factors, recording at 88.2 and downsampling to 44.1 would make for far easier math than from 48 or 96. I wonder what, if any, sonic benefits could be gained from recording at a higher resolution. I'm of the mind that there are more important variables than sampling rate.BONES wrote:I only record/render stuff to make a CD from it. So for CD audio, I only ever use 44.1kHz as both 48 and 96 would require some interpolation when converting to 44.1 which could affect the sound quality.
- KVRian
- 1325 posts since 6 Mar, 2001 from London, UK
What the man said.Jason Brian Merrill wrote:if you do a search on the subject i feel the general consensus is that 24 bit is worth it, 96k is debateable. However, running your vstis at 96k could prevent aliasing....
I use 24/48 (because my s/pdif runs at that across the rack) and 24/96 for recoding soft synths and r8brain down to 48.
- KVRian
- 1325 posts since 6 Mar, 2001 from London, UK
That's a bit of a myth. Converting across exact multiples of the sample rate is an easier calculation, but it is not intrinsically any more or less accurate than non-multiples.BONES wrote:I only ever use 44.1kHz as both 48 and 96 would require some interpolation when converting to 44.1 which could affect the sound quality.
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- KVRAF
- 42529 posts since 21 Dec, 2005
I'd say valid points and raise you a couple of "extra" food for thought "thoughts"BONES wrote:Standard for what? Sound-cards? Maybe that has to do with bus-speed?hibidy wrote:24/96 is the standard now...No it doesn't. Well, maybe, but no more "automatic" than lowering the buffer size yourself.The first and probably the most important reason why 96 sampling rate is highly sought is because it gives you automatic improved latency.Wow! You know what else you can do? Change the buffer size from 512/256/128/64 etc and see how the latency decreases immediately. Its amazing!Switch any host/card from 44.1/48/88.2/96....etc with a 512 buffer setting, and you will always see the latency increase immdiatly. For example, I run 24/96 with a 256 buffer and it's just sick how low the latency is. It puts a bit extra on my cpu's but it worth it.
In deciding how to preceed, you need to look at your goal. e.g. I only record/render stuff to make a CD from it. So for CD audio, I only ever use 44.1kHz as both 48 and 96 would require some interpolation when converting to 44.1 which could affect the sound quality. I record at 24 bit and render at 32 [ORION doesn't render at 24] because it allows me to record something a little quiet and not have to worry about any loss of quality. Its always easier to normalise audio than it is to remove clipping.
If you can hear the difference/or have the equipment (mutually exclusive) 96k is going to give you better sound results.....especially in mixes and bouncing...(and now my point) it would be best to do 96k and don't worry about the buffer. Of course it does depend on your equipment......and ears.
96k is the standard right now. And I'm talking about studio's, "swaping" or out of house mixing/mastering. It is now preferred to have the tracks at 24 bit/96ksr then the engineers have more to work with. It's kinda like having a big piece of clay to work with vs a just big-enough piece..........it's more flexible.
