Adobe Audition CC 2015 32Bit Crack
Laquanda Gede
I'm having this issue with recording audio in 32-bit where the peaks of the signal that exceeded 0dB are still being cut off when I reduce the gain after recording. I'm not being able to get the headroom that is expected when recording in 32-bit.
The reason why I'm having high +0dB peaks is because I'm transferring audio from a Tascam 388 and some of that audio on the reels was recorded a bit in the red. I'm running from the Tape Outs on the 388 and have the input gain at minimum level on the interface that I'm using for transfer (Tascam 16x08) so I can't reduce the signal input volume any further.
You need to attenuate the input signal more. 32-bit FP is very essentially a software fix, and doesn't have any effect on the digitising part of an interface at all; if you overload that, it distorts at this point and there's nothing you can do subsequently to fix it. Your digitiser works as a 24-bit integer device - they all do. So if your input level goes above +8dBv, it's going to distort. What you need is in-line attenuators for the feed into the Tascam. I have to say that most devices with a line input will handle rather more signal than the Tascam does - I would have said that to deal with line level signals from any sort of pro gear, it should really be able to handle +24dBv, not just 8! If you can find attenuators that reduce the signal level going in by 10dB, you should be fine.
I'm having the same problem. I recorded some test footage on an iphone 12 that was intentionally clipped. Then I imported the footage to adobe premiere pro and used the properties feature to confirm that it IS a 32-bit file and so is the sequence and project. I then right clicked the audio to open it in adobe audition. I lowered the volume of the clip and it remained clipped and distorted with a flat wabeform, just lower in volume. Am I doing something wrong or is there a reason I cannot recover the peaks? Thanks
You haven't got the same problem at all. Your original was overloaded and clipped on purpose. Whatever you do, you can't get over that - this is nothing to do with the bit depth. Distortion resulting from significant clipping cannot be fixed, because there is no accurate reference to what it should be like. The only clipping that can be partially repaired is the odd single clip that isn't too large, and even that repair won't necessarily be accurate. The OP didn't have a distorted original, just an output that couldn't be turned down enough, which is why attenuation fixed it.
Thanks for the reply. I thought 32-bit float can record up to 700 db over 0. And you can just bring it back down in a daw. I've seen videos of people doing it. I have concert footage that I would really like to save but having same problem with that.
You need to understand that recording bit depth has nothing whatsoever to do with distortion. You can't fix distortion. Nothing can fix distortion. And Floating Point processing is only a software trick - real hardware produces integer-based outputs and they have an absolute ceiling of 0dB before they overload. Converting them to Floating point after this (which is the only way you can do it) makes not a shred of difference to that. Floating Point is not a recording format.
Yes you can process material, once it's in Floating Point, so that it massively overloads and can subsequently be bought back down without loss. But if that signal is distorted in the first place, then it will be just as distorted as it was when you return it to a more sensible level. So if you overload a mic stage and it distorts, you're stuck with it.
Thank you for taking the time to explain this to me. I do understand that once the audio is distorted, it cannot be fixed. But I also believe that with 32-bit float, if clipping was the only thing causing the distortion, the volume can be lowered below 0 and recovered. So I guess I don't understand what actually caused the distortion and how I can avoid it in the future. Thanks
im pretty sure an iphone can't record true float. some devices record multiple tracks and combine them. the more bits, the more dynamic range you can store. check out some of the new 32 bit recorders. they don't even have a gain knob!
Please read my other reply; I don't think you've really got a handle on this at all. Preamps have nothing whatsoever to do with 32-bit values. You can turn any digitised signal into a 32-bit value. What the Floating Point system does is to rescale the integer bits so that they are represented by a 23-bit digitised number between 0 and 1, and then there's a sign bit, and eight scaling bits. That's what makes up your 32-bit signal. So it doesn't matter what sort of integer values you have - they can all be stored as Floating Point signals. All you do when you scale the signal is move its range up and down; it doesn't alter the 23-bit number or sign in the slightest, which is why you can't make it clip.
32-bit Floating Point digitising is essentially 24-bit audio stored with scaling bits. Unfortunately there are one or two sites around that don't give a clear explanation of what this means as far as recorded signals are concerned. For instance, the Sound Devices site's opening paragraph is well less than clear on the subject; despite their claims, it's still perfectly possible to overload their system front end.
Floating Point format and clipping are not in the slightest bit related. Any overload causing clipping has to occur before digitisation. Nothing after that makes any difference; it's clipped. Because of the way it works, you can't cause any additional clipping with a Floating Point format - the scaling digits just stop getting any larger.
You prevent distortion causing an overload by using a gain control on whatever input device you are using. The whole idea is to stop the analog signal hiting the highest number the A-D converter can output. This is where the concept of 'headroom' comes in; that's the amount of room you leave for any accidental peaks that go higher than your intended maximum. With digital recording, that's normally about 15dB. In other words, the loudest thing you think you'll be recording shouldn't go above -15dB. Depending upon where you look, you might find slight variations in this figure, but I've never heard of anybody using less than 12dB of headroom, certainly for any acoustic-based recordings. I always use -15dB.
Like I said, they don't explain it very well, and the last five words are very significant, especially 'additional'. If you have an analogue signal fed into analog electronics (ie, a mic preamp) it will have a performance ceiling at which point it will distort - and if you digitise that, you are digitising distortion. The best input stages you can build are limited by the Laws of Physics to have a dynamic range of about 120dB, and that is easily digitised in its entirety by a 20-bit ADC, never mind a 24-bit or 32-bit Floating Point one. That's not the issue. The issue is that the input is still overloadable by a mic if the gain is set too high - because the mic output from a loud source will, unless the gain is adjusted accordingly, overload the input and cause distortion. There's no such thing as a 'native maximum spl' either - all mics are different and there are many professional ones around that will produce undistorted outputs of levels that your ears could never stand. It's not the limits of the mic you will hit, but the limits of the input stage it's connected to. And that is what Sound Devices don't describe adequately. Their mic preamps are good - I use them - but they aren't perfect and they very much are oveloadable.
I think the idea here is file format. you can store a special kind of data file format("audio signals above 0 dBFS are preserved in the file"), that is compatible with your computer to recover. if you record at 24 bit, it simply clips...
A Sennheiser MD 421 can easily cope with way more than 142dB. 150dB with effectively no distortion when Sennheiser tested it, and they say that there is effectively no upper limit. And it's by no means the only one.
And as for the idea of you convincing me of anything, especially by trying to put words I didn't use into my mouth... well, let's just say that it ain't gonna happen. In just the same way that I can't convince you of why, I expect.
hi,
when i make a .Wav file recording on my pc and compress it, it has a ratio of about 50%. But when my friend makes a recording the same way when he compresses it with the same settings he only gets a ratio of about 80%, even if he sends me the file and i compress it i also only get 80%.
Maybe different resolution.
Also wav data can be stored in compressed and uncompressed format.
Try same settings and same recording time, аor example, 20 seconds, And compare file size before compression and archive size after compression. Look also properties of wav data with some sound software.
the file on the left was created on my pc, the one on the right my friends.
as you can see the sizes before compression are similar, one is slightly larger but not enough to make much difference.
left file before compression is 1.34GB after is 683MB almost half the size smaller but,
right file before compression is 1.33GB and after is 1.15GB no where near comparable.
both files were transferred to the same hdd and compressed with the exact same settings in 7zip
so i opened the file with the highest compression ratio of 80% and silenced a chunk of the audio in the middle, saved the file and tried again but this time a ratio of 20%.
so is it possible that because our voices are different for eg. My voice is more consistent in the recording less variations between quite and loud and my friends recording has louder parts and quieter parts. Dynamic range, could that alter the ratio?
ok, I've attached screenshots of both the info and the hex info.
again file created on my pc left, and file created on friends pc right. finally both compressed using 7zip on my pc same settings.