I purchased a new sound card (EVGA NU Audio) that has the ability to record in 32-Bit 384000 although I keep getting a error every time I try to record. It works in Audacity just not SoundForge 13. Anyone have any ideas?
Error message: IN 1/IN 2 does not support 32-bit floating point input.
I am not a Sound Forge user but I have to ask, why would you want to record audio at a sample rate of 384kHz?? That gives a frequency range up to 192kHz. But human hearing is limited, at best, to 20kHz. IOW, "most" of the audio your computer is recording is inaudible to you, even assuming that your headphones/speakers can reproduce such frequencies. I am not aware of any such headphones or speakers.
In addition, I know of no microphone that can actually handle such an extreme frequency to pass it to your interface to record.
The "bottom line" of this is that you will be producing huge audio files, the vast majority of whose content will be digital "noise" and hence, useless because inaudible. 😀
Sound Forge internally runs at floating point 32 bit, but saves the file as to what the user specifies (at least up to supported max). AFAIK, the max sample rate in SF is 192kHz. Most pro music studios and S/FX recordists record at 88.2k or 96k, in extreme circumstances,192kHz. 24 bit is the norm and very few A/D converters go beyond that.. in fact many are 16 bit, though a PCM file's bit depth will be up-converted if the files save parameters are set higher... (kind of like digitizing a VHS videotape to an HD format).
Thank you for some insight into this topic. I will explain why it's necessary in my situation. I understand that human hearing cannot hear anywhere close to this frequency. Although, here is where this high bit rate recording process comes into play. First off this new sound card that came out can actually record up to 32-Bit 384KHZ so it's possible. I want to record with the highest quality possible, because I will end up converting the file or files to MP3.
In my situation I am mainly transferring vinyl to digital . Yes, when converting to MP3 it will lose much quality. Although, from comparisons and a lot of testing. When I recorded in 16 or 24-Bit between 44100-192000 KHZ range then converting to MP3 from WAV it sounded pretty good but at the higher spectrum.
Therefore, I wanted it to sound even better and this sound card has allowed me to do so. After some rigorous testing last night with previous comparisons from lower quality recordings of WAV files that I recorded in the past.
I have come to the conclusion that when I recorded in 32-Bit 384 KHZ then saved the file as WAV with those same settings > finally then converting WAV to MP3 the sound was very high quality, since it was able to carry over more data then recording from 16\24 Bit. You would have to hear it to believe it when compared to converting from a lower bit rate etc. I still have not been able to get Sound Forge 13 to work but I managed to make it work in Sound Forge 12. :)
since it was able to carry over more data then recording from 16\24 Bit.
Agreed, though you do need to realise that 32 bit Floating Point is essentially 24 bit with "extras", i.e. no real additional data!
However, that fact has nothing to do with the matter of sample rate, which is what your question is about!
"Sample rate" determines the highest frequency that can be "recorded"; that frequency is always half the sample rate, so 384kHz sample rate (384,000 samples per second) can record up to 192kHz frequency.
"Bit depth", (16, 24 or 32 etc.) determines the amount of data each individual sample contains; the greater the bit depth, the more "detail". I would fully support your intention to record at 32 bit FP, that allows for occasional digital "overs" without the danger of digital distortion, amongst other advantages.
However, your intended destination for your recordings is mp3, designed as a "distribution" format. By definition, its upper frequency limit is around 16kHz, whatever the bit depth. So you need to understand that the overwhelming majority of the "sounds" you recorded between the upper limits of human hearing (about 20kHz at most) and your 192kHz sampling limit, is going to be "thrown away" by the mp3 encoding. And notice, "thrown away", not merely "hidden". There is a misconception that the audio "lost" in mp3 conversion can be "restored" if that same mp3 is reconverted to a .wav file; it can't!
I cannot comment on the differences you perceive in your final mp3s since that is an entirely personal perception. What I can say is that a very large number of "blind test" experiments have completely failed to support the theory that recording at extreme sampling rates has any audible benefit!