WAV samples are on the compact flash say 'wrong file'?

Greetings MPCforums!

just downloaded some gritty 8bit style samples from musicradar.com to get cracking with my new MPC1000 (JJOS2XL) but the machine says 'wrong file' and I can't audition them or load them for pads when some of the other samples from this site work fine.

Am I doing something wrong or do I need to change the file name or something?


thanks in advance

you need to search this forum for wrong + file which would tell you that you need to convert the samples to 16bit 44kHz

soundwave wrote: 8bit style samples

First, are they actual 8bit or do they sound 8 bit? Its a 16bit machine that will not read the 8 bit format. The most common file issues are:

1. Wrong file type. They must be WAV
2. Wrong bits. They must be 16
3. File names longer than 16 characters

sorry I should have made myself clear they are 16bit 44.1 WAV samples of 8bit video game style hits and noises found here http://www.musicradar.com/news/tech/sam ... 26#content

I've also checked the characters and they are only 12 in length, I can read them in the MPC but it just won't play them for some reason?

One more thing to check...not all WAV's are created equally, try to convert a sample with iTunes or whatever software you use to create WAV's that are PCM. See if that works. If so you might have to convert them all. Thats pretty much every option other than corrupted files or user error.

poundaproblem wrote:One more thing to check...not all WAV's are created equally, try to convert a sample with iTunes or whatever software you use to create WAV's that are PCM. See if that works. If so you might have to convert them all. Thats pretty much every option other than corrupted files or user error.

cheers dude will give that a try and report back

Let us know.

Are you sure they're 16 bit? Most of the samples I downloaded from music radar were 24-bit.

You can use a program like Sound Forge or Wavelab and batch process them all at once, just save as 16 bit.

I convert my samples in program called Switch. Its a simple and free program that you can find here:
http://www.fileguru.com/Switch-Audio-Converter/info

Install it, set the saving options and you're ready to convert.

Bingo!!

I'm guessing the files were 24bit even if they showed 44.1 the bit rate was double what they were when I converted them to regular 16 bit wavs using Winamp.


cheers guys

16 is the necessary BIT DEPTH not bit rate

That's completely unrelated to 44.1kHz which is the sample rate

16bit means each sample is 16 bits, 44.1kHz means there are 44100 samples per second

So the bit rate is 44100 x 16 bits per second, but talking about bit rate for WAV files isn't very useful. A 16-bit 22kHz mono WAV has the same bit rate as an 8-bit 44kHz mono WAV which has the same bit rate as an 8-bit 22kHz stereo WAV, but each has very different characteristics, so describe them in terms of bit depth and sample rate, not bit rate

so like I said in the first place, they need to be 16bit and 44kHz

Pastor-of-Muppets wrote:16 is the necessary BIT DEPTH not bit rate

That's completely unrelated to 44.1kHz which is the sample rate

16bit means each sample is 16 bits, 44.1kHz means there are 44100 samples per second

So the bit rate is 44100 x 16 bits per second, but talking about bit rate for WAV files isn't very useful. A 16-bit 22kHz mono WAV has the same bit rate as an 8-bit 44kHz mono WAV which has the same bit rate as an 8-bit 22kHz stereo WAV, but each has very different characteristics, so describe them in terms of bit depth and sample rate, not bit rate

so like I said in the first place, they need to be 16bit and 44kHz

Damn, You ARE an "expert member" I knew the difference between bit depth and sample rate, but did not know that bit depth X sample rate = bit rate.

This tool is amazing for converting audio files bit depth/sample rate/audio type.

Definitely recommend it.

https://public.msli.com/lcs/audiomove/

TribeCalledQuest wrote:bit depth X sample rate = bit rate.

it does for WAV files using the usual uncompressed formats

it's not true for all formats, e.g. MP3s or other lossy encoding formats which have a variable bitrate