I believe simply setting it to 48k without re-sampling would change the pitch. At 48k, it sounds exactly the same as the 44.1 version, provided that I actually tell Sound Forge to re-sample at 48k and not simply SET the sample rate at 48k without re-sampling it. I then use Sony Sound Forge to resample the song as a 48k WAV for video work.
I usually mix my song at 44.1 Hz which I use for WAVs, mp3s, etc. Thanks for asking this question! It's inspired me to find out some stuff.
(I haven't tried it so I have no idea how effective it is.) Link to "Music Editor", a freeware re-sampling program.
#WAVE EDITOR FOR MAC UPSAMPLING SOFTWARE#
There are software programs to do this and it might be that your video software has this ability.Īs a consequence of my net searches, I've just stumbled across this. In going from one sampling frequency to another one, the best method is to use the original data to calculate new values for new rate. If the sample rate is 48,000 Hz, then the variation of the sound wave's movement during one second is captured as 48,000 individual snippets of information. If a sample rate is 44,100 Hz, the above one-second variation in movement of this sound wave is captured as 44,100 individual snippets of information.Ĥ. It's this horizontal variation in sound that we hear as pitch the vertical variation we hear as volume.)ģ. (I've marked this movement by the red double-ended arrow. One second later it has traveled up to a maximum value, gone back down past the horizontal axis to a minimum value and then returned back to the horizontal starting line. Below is a simplified image of a sound wave.Ģ. Now of course with SRC this is all done in the digital domain, and usually implemented with a clever filter that deals with the resampling at the same time as anti-aliasing, but that's an implementation detail.Īnd so in effect the same rules and constraints still apply, so the quality of SRC largely depends on the quality of the filter (just as it would in the DAC->ADC situation) and also on how accurately in "time" new samples are "taken" (they're calculated rather than taken, but errors in the calculation can be viewed as pretty much the same as jitter).The 44,100 Hz and 48,000 Hz is the rate at which an audio signal is processed in relation to the signal's variation with time. You have the source levels generated at the old sample rate, you have a filter which takes out everything above either Nyquist (in the DAC->ADC situation you would normally have two, an anti-imaging filter on the DAC and an anti-aliasing filter on the ADC, but in this situation you only actually need one, with a cutoff at the lower of the two Nyquist frequencies), and then you have sampling of the result at the new sample rate. The tech of file formats and conversion algorithms is beyond my expertiseMP3 and the like are rather complicated, but conceptually SRC is no different to plugging a DAC running at your source sample rate into an ADC running at your destinations sample rate. I know this is spitting hairs, but that's the point of the question. The tech of file formats and conversion algorithms is beyond my expertise so would appreciate anything you guys may have tested over the years. Some aren't supported on Mac so I use 3rd party components and FOSS converters. Older QuickTime videos I typically export wav from QuickTime at the desired rate and depth at best quality. This would be like converting 44.1/16 to 96/24 for example using wav, flac, ogg, MP*, audio embedded in video. Was just curious if anyone has tested using other software for up-sampling rather than their DAW-like RX or Saracon-and if there was a noticeable difference in the mix or if the way up-sampling works and converting formats it's a non-issue. I've had numerous clients (even big corporations) request I rip their own DVDs because they lost track of their originals or never had them to begin with While I try and always start with the best source material, sometimes it's not possible. Not sure what algorithm it uses for this (their own or an open or licensed one). I typically just drag audio into DP and let it auto-convert to my project and never give it a second thought. I never gave much thought to this, but all this talk about sampling and bit depth lately has me curious.
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