Frequency Analysis: Sub 20Hz Frequencies Shown Across All Files

Eric Holtz wrote: How would I alleviate a zero centering issue?

That's easy. Go to Effects>Amplitude and Compression>Normalize, and you have an option there.

I can see a very small offset on the file with the 0-5Hz signal whereas I can't see one in the other file. Whether that rather small offset would make that much difference to the analysis isn't clear to me, but it might, possibly. There are a couple of other things to note about the whole business though. Firstly that with the best will in the world, an FFT analysis is never going to be perfect across the board - that first F stands for 'Fast' and that invariably means a compromise in the way that the window is analysed. The greater the FFT number, the smaller the sampling 'Window' is, and rather perversely, the less accurate the extreme bass analysis is; indeed any analysis where the window size is smaller than the wavelength of a frequency being analysed is potentially inaccurate, and that's where the problems with having DC in your waveforms comes into play.

On its own, all the DC would do is offset your speaker cone very slightly - assuming that everything in your system up to that point is DC-coupled. But that does lead me onto the other thing, which is in effect a sanity check. If there really is the level of signal between 0 and 5Hz shown, it's going to lead very definitely to bass unit speaker-cone flap - pretty severe cone flap at that. So if you can't see any, then it's almost definitely an analysis anomaly. If you have cone flap, then you should definitely get rid of that part of the signal - it saps power from your amplifier, and restricts the amount of cone movement available for 'real' sounds higher up the frequency range; it's not healthy!

Thank you for the information. I have a question regarding Audition's internal 32 bit (float) processing that hopefully I can get answered here. Most of my work involves the transfer and archiving of analog sources, primarily LP (and 78) records, cassettes, and reel-to-reel. The original sources were recorded as 16bit/44.1k, 24 bit/88.2k, and some 32bit(float)/88.2k files. I primarily apply effects (noise reduction, light click repair, EQ, etc.) to copies of the files themselves using the Waveform editor window, and may save over the same edited file several times. How does Audition's 32bit float environment handle re-saving a 16bit, 24bit, or 32bit(float) file? I understand that Audition performs all calculations in 32 bit float, so does it automatically apply dither each time I save the file, or does it simply truncate the file back to the original bit depth?

I've read many threads about how integer files are handled by 32bit (float) DAWs, but they almost all relate to mixing in a Multitrack window, and not editing the files themselves. If this question should be posted as another thread, please let me know and I will follow accordingly.