thanks for the replies. i knew that u can re-sample it in the mpc but it takes hours when u do that. alright, than i have to try this with the 12 bit and with the khz i have to figure out.
can someone explain me what this actually means 44,1 khz and 16 bit. it sounds stupid i should have asked before this thread i mean khz and bit. what does it mean excactly? thanks man.
Bit Depth refers to the number of bits you have to capture audio. The easiest way to envision this is as a series of levels, that audio energy can be sliced at any given moment in time. With 16 bit audio, there are 65, 536 possible levels. With every bit of greater resolution, the number of levels double. By the time we get to 24 bit, we actually have 16,777,216 levels. Remember we are talking about a slice of audio frozen in a single moment of time.
Now lets add our friend Time into the picture. That's where we get into the Sample Rate.
The sample rate is the number of times your audio is measured (sampled) per second. So at the red book standard for CDs, the sample rate is 44.1 kHz or 44,100 slices every second.
So lets put it all together now. This brings us to the Bit Rate, or how much data per second is required to transmit the file, which can then be translated into how big the file is. Your CD is 16bit, 44.1 so that is 44,100 slices, each having 65,536 levels
and no I'm not that smart, i got this from another site
Sample rate also deals with "Frequency" range, so with a 44.1k sample rate you can sample up to 22.05k (just above the average hearing range); with 48k it's up to 24k an so on.
And the higher the sample rate the higher the Harmonic content.
It's called the "Nyquist theory" here's a good link to explain it better.
You can indeed load 12 bit samples into the mpc. The sample rate however must be 44.1 kHz or else it will not play back at the right speed. You never mentioned what media you were loading from? I would suggest using an audio editing program to convert these samples before loading into MPC. I like Audacity for this and it's a free download. The conversion will not make your samples sound any better or worse, it just changes the file format. Hope this helps.
^ are you sure about that? I'm almost positive the 2k requires 16 bit files, but the frequency rate can be variable (if the speed is wrong, you can adjust the tune) If you resample the sound to 12 bit on the mpc, its still saved as a 16 bit file. Variable frequency rates are important since the sampler needs to be able to pitch up and down sounds. I did some tests a while back and was able to load a file at 5000hz all the way to 65000hz. Bottom line is, it will always be output at 16/44k.
If you can't load your 12bit/26khz sample, try resaving it as a 16 bit file on a computer (it will still sound like 12 bit) and it should load up on the MPC.
According to the Nyquist theory we should not record in 24 96? If 16 44.1 is the best option for CD's why the need for higher Sample rates ? I know it is supposedly better for mixing to have higher rates but if you are bouncing down and degading what was recorded wouldn't it make sense to do 16 44.1 the whole way...or am I missing something...?
i only limited myself to 16-44 when working with the mpc, theres is a sh1t load of apps around for doing batch conversions, most of the samples i record into my pc these days are 24-96, and some sample cd i get all also higher quality than what the mpc can work with.
The real reson to record at 96 is for filter implementation.
According to Nyquist you need to sample at the twice the rate of a BANDLIMITED signal to be able to reproduce the waveform accurately.
The operative word is Bandlimited. This means that you must limit the signal to a given frequency before you sample. A low pass filter is used to get rid of any frequencies above a given point. It's called an antialiasing filter.
The problem is that it is impossible to build a filter that is ruler flat so in the case 44.1k the filter starts around 20k and curves down to 22.05k. 44.1k/2=22.05.
This can affect frequencies in the audible range. That ringy, brittle sound in early digital recordings was caused by bad filters amongst other things.
Most samplers use oversampling nowadays and the Low pass filters are digital.(ther is also an analogue LP filter before you sample) The original sample rate is multiplied by a number so the filter design is easier to achieve at really high rate. The sample is then downsampled (also called decimation) and passed to the DA convertor.
Converter designers still have problems building a filter at 44.1k even with all these tricks. Using a higher sample rate like 96k relaxes the difficulties in designing the filter. And therefore the sound is reportedly more open.
Higher sample rates might be useful when using effects like compression or distortion as they contain a ot of high end harmonic information that can cause difficulty at 44.1.
Also if you're pitching down a sample there is more information sampled at 96k so that too is an advantage.
I can never tell the difference between 44.1,48 or 96k. As my ears top out at around 15k. I have no use for 48k audio(at 96k).I use 44.1/24 bit. It allows more processing power and takes up half as much space and I also believe if the project is destined for CD then it should avoid downsampling.
I know this shyt sounds complicated but the bottom line IMHO is that there is no magic at 96k for humans. Maybe for dogs and bats that can hear far higher. But we aint makin music for dogs....