dodgy dubplates

[Littlefoot]

This means WAR!!!

uh, sorry..... (I do get carried away..!

Why would sample rate effect the cut??

Just as en example, if your track is bounced at 44.1khz and you open it up in a project that is 48khz, then you track will be slowed down due to the sample rates rates difference.

which is the confusing thing seeing as Transition play the CD off a CD player when cutting, not a DAW, well when I went atleast!

[jtransition]

We generally work on the prinicpal of ''If you like what we do tell others and if you do not like what we do then then tell us'' But because this thread appeared before we were contacted it's a good oportunity to ask the following,
Who else here records at 48k, And why?

[boomstix]

i bounce my stuff at 48k/24bit

its just a bit more definition

[jtransition]

i bounce my stuff at 48k/24bit

its just a bit more definition

Have you compared it to 44.1/24bit or 88.2/24bit?
If the music is going to end up on a cd then it will have to be sample rate converted which depending on the algorithm can sound brutal so any benifit of working at 48k is lost in the translation back to 44.1k.On the other hand a sample rate conversion from 88.2k to 44.1k sounds better.

[punisha]

i bounce my stuff at 48k/24bit

its just a bit more definition

Have you compared it to 44.1/24bit or 88.2/24bit?
If the music is going to end up on a cd then it will have to be sample rate converted which depending on the algorithm can sound brutal so any benifit of working at 48k is lost in the translation back to 44.1k.On the other hand a sample rate conversion from 88.2k to 44.1k sounds better.

Agreed.

Dithering and sample rate conversion can be a bigger factor than many people realise. A good base guide is to keep in x2 factors.

I used to think that people used 48k because of using the ASIO drivers for the original SBLive card -

[dawntreader]

replacement dubplate on the way
from this i've learnt....record at 44.1 khz because it's just plain practical!
out to Jason/Leon/Transition!

[macc]

A good base guide is to keep in x2 factors.

Respectfully, this is an old wives' tale and not true in the slightest
There's a great deal more to up/downsampling than throwing away every other sample (taking your example).

The original rate is multiplied to an intermediate and ridiculously high value (35MHz I'm told by Google ), before being divided by the appropriate factor to end up at the destination rate. This applies to pretty much everything on the market, especially in computer-land. It's the filtering - ie your sample rate convertor algorithm/method - that makes the biggest difference.

I used to think that people used 48k because of using the ASIO drivers for the original SBLive card -

48k was around long before that, it's 44.1 that is the aberration 48k was the natural sample rate to choose for both numerical and auditory reasons - it just makes numerical sense, and comfortably fits the audible spectrum in without serious aliasing concerns.

48kHz would have been the standard but for the original CD manufacturer/s (Sony I think?) deciding that the new CD discs were too big. Therefore they reduced the data rate in order to make the discs physically smaller. I mean, think about it.... 44.1 - WTF? Why not 44.2, or just plain 44?

Just some semi-relevant waffle, sorry harping on.... glad the original problem got sorted out.

[paradigm_x]

i remember reading about this at uni -

The rate of 44.1 kHz was picked to be compatible with existing
50 Hz and 60 Hz video-based digital audio storage, where an
integral number of frame buffers could fit in a single
horizontal scan. Quote from Watkinson and Rumsey, "Digital
Interface Handbook" 2.7.6 Choice of Sampling Rate:

"In 60 Hz [525 line, 60 Hz vertical refresh) video there are 35
blanked lines, leaving 490 lines per frame, or 245 lines per
field for samples. If three samples were stored per line, the
sampling rate becomes 60*245*3=44.1 kHz. In 50 Hz video [625
line, 50 Hz vertical refresh), there are 37 lines of blanking,
leaving 588 active lines per frame, or 294 per field, so the
sampling rate becomes 50*294*3=44.1 kHz. The sampling rate of
44.1 kHz came to be that of the Compact Disk. Even though CD
has no video circuitry, the equipment used to make CD masters
is video based and determined the sampling rate."

The length of 74 minutes is determined by the physical nature
of the reading system. It's based on the encoding method, the
wavelength of the laser used (different wavelengths are
incompatible with current CDs) and the necessary support
information. During the development of the CD, von Karajan was
alledgedly asked how long a CD must be, to which he responded
it must be long enough to hold HIS performance of Beethoven's
9th symphony, but the parameters had pretty much already been
nailed down at that point.

[paradigm_x]

im going to do some tests of bouncing at 44/48/88 and resampling via SOX or voxengo this weekend, been going over this for a while now. ill post up results.

want to know definitively whether i (and others) can hear any difference.

Its hardly worth disk space and dsp usage working at 88k throughout, and i feel its pointless exporting at 44, up to 88, 'master', then src back down.

but ill see.

60khz is supposed to be an ideal sample rate as a trade off for space vs simple AA filters but was never adopted.

[punisha]

A good base guide is to keep in x2 factors.

Respectfully, this is an old wives' tale and not true in the slightest
There's a great deal more to up/downsampling than throwing away every other sample (taking your example).

The original rate is multiplied to an intermediate and ridiculously high value (35MHz I'm told by Google ), before being divided by the appropriate factor to end up at the destination rate. This applies to pretty much everything on the market, especially in computer-land. It's the filtering - ie your sample rate convertor algorithm/method - that makes the biggest difference.

I used to think that people used 48k because of using the ASIO drivers for the original SBLive card -

48k was around long before that, it's 44.1 that is the aberration 48k was the natural sample rate to choose for both numerical and auditory reasons - it just makes numerical sense, and comfortably fits the audible spectrum in without serious aliasing concerns.

48kHz would have been the standard but for the original CD manufacturer/s (Sony I think?) deciding that the new CD discs were too big. Therefore they reduced the data rate in order to make the discs physically smaller. I mean, think about it.... 44.1 - WTF? Why not 44.2, or just plain 44?

Just some semi-relevant waffle, sorry harping on.... glad the original problem got sorted out.

44.1 was used because of nyquist theory.

44.2 would probably be ok as the 200hz 'extra' over nyquist is just for 'safety'. 44 would not offer anything and therefore probably scare people into thinking its not enough!!! 48k is certainly wasted on me as I don't think I can hear much above 12k other than my tinnitus

I still think the x2 is a good guide. Your indeed right that its down to the algo used by your daw/convertor, but I was led to beleive that its easier for the algo's in multiples of 2 and therefore beneficial in sound.

Also x2 would have hopefully stopped the original posters problem from happening as 88.2 would have sounded significantly slower and maybe raised the alarm - lol.

I am aware that 48k was about before SBLives but this was my first dealings with this SR and remember the pitch and speed issues I had for a while until I got my head round it. I later had a DAT machine and this was 48k too. As of now I use 24bit/44.1khz as to be honest I get no issues and I can sleep safer with my math head happy with the comfort of x2 being easy for it to calculate

A lot of my plugins upsample anyway so I get the benefit imo of what higher rates actually achieve and that is when processing audio with effects.

Just MHO

[macc]

I still think the x2 is a good guide. Your indeed right that its down to the algo used by your daw/convertor, but I was led to beleive that its easier for the algo's in multiples of 2 and therefore beneficial in sound.

It's not

Also x2 would have hopefully stopped the original posters problem from happening as 88.2 would have sounded significantly slower and maybe raised the alarm - lol.

I thought all dubstep was made that way for that really stoned vibe

As of now I use 24bit/44.1khz as to be honest I get no issues and I can sleep safer with my math head happy with the comfort of x2 being easy for it to calculate

But it's not. It might make more sense in a common sense way, and is the simplest way to do it but in actuality it isn't implemented like that in any of the software we use. Using the lowest-common-multiple approach allows the manufacturers to save themsleves work at no real cost to the audio. They can use one filtering approach rather than the two sets of filters they'd have to design/use if they used one method for integer ratios (eg 88.2 > 44.1), and one method for non-integer ratios (eg 96 > 44.1).

So, just because it seems (well, is) simpler, it doesn't mean it is implemented that way. The maths isn't any easier for your computer because it doesn't take that route to get there.

A lot of my plugins upsample anyway so I get the benefit imo of what higher rates actually achieve and that is when processing audio with effects.

It's not quite the same (it still has to come back to the host sample rate, ie extra SRC) but I totally agree, especially in eqs up top!

Not bitching or anything here - I know I get a bit full-on like But this SRC thing has been floating about for years and comes up every now and then... it's one of those little things the geek in me won't shut it's mouth about

[jtransition]

Punisha wrote:

I still think the x2 is a good guide. Your indeed right that its down to the algo used by your daw/convertor, but I was led to beleive that its easier for the algo's in multiples of 2 and therefore beneficial in sound.


It's not Smile

Quote:
Also x2 would have hopefully stopped the original posters problem from happening as 88.2 would have sounded significantly slower and maybe raised the alarm - lol.


Laughing I thought all dubstep was made that way for that really stoned vibe

Quote:
As of now I use 24bit/44.1khz as to be honest I get no issues and I can sleep safer with my math head happy with the comfort of x2 being easy for it to calculate Wink


But it's not. It might make more sense in a common sense way, and is the simplest way to do it but in actuality it isn't implemented like that in any of the software we use. Using the lowest-common-multiple approach allows the manufacturers to save themsleves work at no real cost to the audio. They can use one filtering approach rather than the two sets of filters they'd have to design/use if they used one method for integer ratios (eg 88.2 > 44.1), and one method for non-integer ratios (eg 96 > 44.1).

So, just because it seems (well, is) simpler, it doesn't mean it is implemented that way. The maths isn't any easier for your computer because it doesn't take that route to get there.

Smile

Quote:
A lot of my plugins upsample anyway so I get the benefit imo of what higher rates actually achieve and that is when processing audio with effects.


It's not quite the same (it still has to come back to the host sample rate, ie extra SRC) but I totally agree, especially in eqs up top!

Not bitching or anything here - I know I get a bit full-on like Laughing But this SRC thing has been floating about for years and comes up every now and then... it's one of those little things the geek in me won't shut it's mouth about Embarassed

Just to clear a few things up,
The original problem was caused by a mixture of the folowing;Human error,Clocking from outside the PC,Using External converters,Non labeling of audio sent over the internet etc.
Most of the tracks that we master come in at 44.1k with varying bit rates so i asked the question about who else was recording at 48k to guage if there were more people doing so and why.That way we can learn from this.

[macc]

Understood Jason - I've basically just gone off on one about SRC like a knob

Sorry for derailing it!

[abZ]

One time I got a mp3 off of a friend that I had to play it at plus 8. I asked him why did you have to make it so slow?!?! He said 140 is that slow? No that is perfect. I still have no idea why it played slow like that.