MPC1000 OS V3.0 beta ?

cowbell wrote:

The programmer for busyness as for version upgrade we go to bed for a while.

Poor JJ. The guy is breaking apart

I hope he does wake up, as opposed to Wakai Unpro

when i use the q-link sliders in the Trim section.. wierd stuff happens.

how do i use this feature? or is it buggy?
im just not sure.

I have no idea if those are possible to implement in any future OS.


1-Velocity and then gate could be the first values when you choose an event to edit in the grid or step edit window. That would increase the speed of programming/editing.

2-A ppqn position display in grid edit.

3-a "Play" button in grid edit

4-MTC

5-a real adsr assignable to pitch/volume/filter (negative values possible)

6-The same midi note number could be assigned to multiple pads.

7-A different delay value for each pad..(up to 10000ms)

8-Tempo could go to 350bpm

Mr modnaR wrote:

medway wrote:You can always downsample/reduce bit rate regardless of the converters. From what I remember there is a grunge effect that basically does this. I doubt he could rewrite the sampling engine itself to support different rates as well as the file structure for saving/loading. Applying an effect to achieve the same sound is probably a lot easier.

sure that's true, but not the same thing as true 12bit. 12 doesn't go into 16 and leave an integer, therefore it won't be accurate, it'll merely be an effect which gets close to the sound.

But to get that nice 12bit sound you do need the converters of the old stuff. Software bit reduction is only going to sound so good regardless of whether its an effect or actually a 12bit sample. That being said you should be able to just zero out the last 4 bits of a 16bit sample to get the effect. Either that or just sample your sound 24db below 0 on the mpc and then normalize. Thats basically the same as only using 12 bits since each bit is 6db.

All this is why I still keep my old samplers around

sleepersriddle wrote:check yer math::

12 bit is 2 to the power of 12,,,
16 bit is 2 to the power of 16,,,

seems like 12bit would indeed go into 16bit evenly.
(ie if full range = full range then any 12 bit value will have exact correspondance with one of the 16 bit possibilities)
it seems clear if you think of each bit as splitting the range in 2, which is essentially what is does.


however a 12bit d/a convertor will have a different analog anti-aliasing circuit which is probably part of the magic.

16/12=1.33333333333333333333333333 etc. which is NOT a whole number (integer).

for every bit of the 12bit signal, you would use 1 1/3 of a bit of the 16bit signal, which is not possible, hence aliasing occurs which will alter the sound.

check YO math homie.

This is why I stick around....you guys are BANANAS!!

Knoxximus wrote::lol: This is why I stick around....you guys are BANANAS!!

University of Mpc, Class of '06:

medway wrote:But to get that nice 12bit sound you do need the converters of the old stuff. Software bit reduction is only going to sound so good regardless of whether its an effect or actually a 12bit sample. That being said you should be able to just zero out the last 4 bits of a 16bit sample to get the effect. Either that or just sample your sound 24db below 0 on the mpc and then normalize. Thats basically the same as only using 12 bits since each bit is 6db.

All this is why I still keep my old samplers around

i quite agree with you that you need the actual converters to get the true sound.

didn't think of the knock 4 bits off trick, but even so nomalizing is going to lose any 12bittedness the sample has anyway, so i wouldn't worry about it. having said that, if you master to 16 bit, it's going to be mucked up whatever, so i guess you are right. i think too much sometimes. grrr back to topic.....

Mr modnaR wrote:

sleepersriddle wrote:check yer math::

12 bit is 2 to the power of 12,,,
16 bit is 2 to the power of 16,,,

seems like 12bit would indeed go into 16bit evenly.
(ie if full range = full range then any 12 bit value will have exact correspondance with one of the 16 bit possibilities)
it seems clear if you think of each bit as splitting the range in 2, which is essentially what is does.


however a 12bit d/a convertor will have a different analog anti-aliasing circuit which is probably part of the magic.

16/12=1.33333333333333333333333333 etc. which is NOT a whole number (integer).

for every bit of the 12bit signal, you would use 1 1/3 of a bit of the 16bit signal, which is not possible, hence aliasing occurs which will alter the sound.

check YO math homie.

i guess you already noticed the mistake, but you forgot to include the 'power of two' as i mentioned. you cant just divide 16 by 12!

back to topic please

sleepersriddle wrote:i guess you already noticed the mistake, but you forgot to include the 'power of two' as i mentioned. you cant just divide 16 by 12!

why not? you're trying to fit 12bits in the same space as 16. for every one bit of the 12bit signal, you use 1 and a third bits of the 16bit signal. how is that wrong? i don't get it. anyway, back to topic, sorry everyone.

Because 16 bits just aint 12 bit * 1,33.

It's a logarithmic scale, so 16 bit will be 2^4 = 16 times the resolution of the 12 bit signal.

The same way that a 24 bit signal will be 2^8 = 256 times the resolution of a 16 bit signal.

Please get back to topic. We have an entire page talking about 12 bit and 16 bit sound and its not doing anything to progress this thread except watching 2 guys go back and forth about who's better in math.

3 guys. I didn't participate until now.

Mr modnaR wrote:

sleepersriddle wrote:check yer math::

12 bit is 2 to the power of 12,,,
16 bit is 2 to the power of 16,,,

seems like 12bit would indeed go into 16bit evenly.
(ie if full range = full range then any 12 bit value will have exact correspondance with one of the 16 bit possibilities)
it seems clear if you think of each bit as splitting the range in 2, which is essentially what is does.


however a 12bit d/a convertor will have a different analog anti-aliasing circuit which is probably part of the magic.

16/12=1.33333333333333333333333333 etc. which is NOT a whole number (integer).

for every bit of the 12bit signal, you would use 1 1/3 of a bit of the 16bit signal, which is not possible, hence aliasing occurs which will alter the sound.

check YO math homie.

Doesnt work like like, you either use a bit or you dont. 12 can fit in a 16 bit value with the last 4 unused. Also the aliasing filter has to do with sampling rate not bit depth so bit depths have no bearing in this matter.