Normalize after chops

Hi
i've chopped and sliced a sample, assigned them to pads
i've been stupid in not normalizin' the sample before i sliced them
i've played in my drums, and now i want to play in the several chops from the sample
but the volume of the chops of the sample are too low
is there any possible way too normalize them after i sliced them, or do i have to normalize the sample and slice it again?
thanks

trim screen-params-vol.@200. or mixer -turn down entire bank with drums then turn main outut to 6db,then adrust vol knob as needed

on 2kxl that is

and on the 1k?

if its loads of chops - import to your pc and do a bulk normalise on wavosaur or something of that ilk.

man. you could have reverted back and normalized the whole audio file and rechopped it by now!

just remember to do it!

so much easier that way!

it depends on how you plan to use the mpc... you really shouldnt normalize if you dont have to-you should use a dynamics device in a tracked out situation to make things louder. 16 bits is enough dynamic range to convey detail even if you only use half of the headroom in your sample.

sample so the meter always peaks below red. if its a record (cmon i know youre sampling a record) then you have an experts touch inherent to the recording already. you dont need to worry about peaks, they have been taken care of for you. all you need is to chop it and arrange it. leave the volume concerns for the mix down, and just make music without normalizing your samples.

what you really want is to dynamically reduce the peaks in your tracks so you can turn it up! normalizing wont do that. its tempting to crank all the levels to maximum but you often lose dynamic subtlety that way. the noise floor is not an issue with this piece of gear if its used properly.

edit your samples in the computer... this gives you the opportunity to use better tools before you load your mpc... some wav editors will let you normalize according to RMS, which is ideal. keep the RMS at around -16dB for excellent results. its also good to have calibrated monitors.

Isn't it best just to normalize everything? Then you can always turn it down in the mix.

why turn something up to turn it down? digital processes add distortion. by nature the output of a digital process is an approximation. the fewer the digital steps along the way to final the better in this digital world.

rich_biatch wrote:edit your samples in the computer... this gives you the opportunity to use better tools before you load your mpc... some wav editors will let you normalize according to RMS, which is ideal. keep the RMS at around -16dB for excellent results. its also good to have calibrated monitors.

??

Isn't that some form of limiting/compression (without any controls)?

RMS? its like the average 'energy' of a sample. sampled phrases should be at around -16dB RMS to sound good on modern equipment and take advantage of a tracking situation where you will eventually process with dynamics techniques. but of course, sometimes peaks will travel outside of the headroom and clip, depending on what youre sampling.

i can just see a buncha n00bs scratchin they heads

Just got no idea what to tell a guy who´s recommending rms normalize

rich_biatch wrote:why turn something up to turn it down? digital processes add distortion. by nature the output of a digital process is an approximation. the fewer the digital steps along the way to final the better in this digital world.

Actually this is not 100% true. Volume increases (until you hit digital clipping) do not distort the signal in any way, and technically can be reversed (assuming no digital clipping).

I have to agree with your sentiment though. Keep it simple and keep it clean.

BTW... I love to see posts like your previous post that are highly informative and then another post later with just a "no". Very elegant.

NearTao wrote:

rich_biatch wrote:why turn something up to turn it down? digital processes add distortion. by nature the output of a digital process is an approximation. the fewer the digital steps along the way to final the better in this digital world.

Actually this is not 100% true. Volume increases (until you hit digital clipping) do not distort the signal in any way, and technically can be reversed (assuming no digital clipping).

I have to agree with your sentiment though. Keep it simple and keep it clean.

BTW... I love to see posts like your previous post that are highly informative and then another post later with just a "no". Very elegant.

Not true. Assuming you have sampled and saved at 16bit then to increase or decrease a samples level it will have to interpolate where the value would fall between bit levels.

Most wave editors will work in a much higher bit rate and with oversampling (eg 32bit floating point, 192kHz sampling rate.) Once you save the wave you will loose all this extra data in the majority of editors, whether you close the program/reopen the waveform or not. To truly get the least distortion you should normalise and do all your processing the moment you record your sample or at least save it at a higher bit rate and sampling frequency that you finally want to use (24bit, 88.4kHz would make sense if you're to take it to CD quality to use on an MPC.)


Of course I seriously doubt many people on here would be able to hear the difference, even if provided with an A+B test situation with ideal monitoring. Bit comparators and distortion measurements will be able to detect the subtle differences and if you are on the school that believes everything should be done as perfectly as humanly possible, even if you can't tell the difference yourself, then it's worth thinking about. With the quality of equipment most people use, and the signal paths often involved, you can't really blame a lot of people for saying it's more effort than it's really worth...