I've got a bunch of files encoded at 44.1k that I'm converting to 48k for use on a DVD. They all peak at -.3 in the original files. But after converting (in "very high" mode) to 48k, a lot if not all of the files now peak over .0. Why does converting the sample rate affect the output level, and can this be avoided/fixed somehow?
Thanks for the response. So, how low does my initial output level have to be not to exceed .0 on resample? Can you give me an estimate?
Does r8brain pro include any kind of limiter to avoid this issue? I wish there was a VST version of this, so I could insert a limiter in the chain of, e.g., WaveLab, so the processed file would be at the right output level.
I can't readily tell what is the limit. I guess 3dB is a good estimate for linear-phase conversion. For min-phase 6dB is a better estimate. It really depends on how loud the higher frequencies are.
There is no limiter included since a better workflow would be loading the converted file into the audio editor and processing with your favorite limiter.
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