If it makes any sense, that is, the input stream to the analyzer is 64bit floats internally, would it be possible to add the raw double file format, as known by other software like Adobe Audition and AudioVero Acourate? This is also the file fomat I use with my own software, both for generators and a block averaging tool... because it is so simple to handle.
That would require a third field "sample rate" in the input dialog, which should allow arbitrary values as is it just a scale factor for frequency display anyway. Also the dBFS display range should extend to lower than -237dB where the display values are cropped currently. A workaround is to apply gain to the file prior to loading but then I would run into the +20dB top limit.
That would require a third field "sample rate" in the input dialog, which should allow arbitrary values as is it just a scale factor for frequency display anyway. Also the dBFS display range should extend to lower than -237dB where the display values are cropped currently. A workaround is to apply gain to the file prior to loading but then I would run into the +20dB top limit.