Recorded at wrong sample rate, can I fix it?

[melbendi]

I recorded many files at the wrong sample rate.  It was set to 44100 Hz in the Audition Preferences Audio Hardware panel, and set to 96000 Hz everywhere else, so all my audio sounds very low and slow.  I tried Edit>Interpret Sample Rate to see if I could change it 44100 and if that would fix it, but it didn't. I also tried Edit>Convert Sample Type and nothing there seems to help, either.  Could anyone please tell me if there a way to save my recordings?  Currently they're 24bit 96000, but will be happy if I can get them to just sound correct at any sample rate or bit depth.  Thank you for your help!

[SteveG_AudioMasters_]

You have to do both processes. First you have to interpret the sample rate as being what it was recorded at - so if it was recorded as 96k and you're playing it as 44.1k, then that's what you put in the 'interpret sample rate as' box - 96k. When you've got it playing at the correct speed, then you do a Convert Sample Type, and you put 44.1k in the box at the top. Then you save the result. You should end up with a 44.1k file that plays correctly. You have to get the order of processing right, and you need both steps!

I have to say that I haven't done this for a while, but it worked fine last time I did it, and I'm reasonably confident that these are the correct steps in the correct order. If for any reason it doesn't work, let me know.

[melbendi]

HI Steve, thank you for your reply.  Here are the steps I took:  First, I open Interpret Sample Rate, and the Interpret Sample Rate box reports the current sample rate is 96000 Hz, and 96000 Hz is selected in the drop down menu, so I say OK, then open the Convert Sample Type and set it to 44100. After it processes, it's still low and slow, so I go back to Interpret Sample Rate, and set it to 96000 Hz, then the speed is correct, but the audio is still jumbled.  I can make out the pieces enough to know the speed is correct, but it sounds like an odd mix of forward and backward sounds or something.  Does this mean we're getting somewhere?     I appreciate your help, Steve!

[SteveG_AudioMasters_]

kalibahlu  wrote

It was set to 44100 Hz in the Audition Preferences Audio Hardware panel, and set to 96000 Hz everywhere else, so all my audio sounds very low and slow. And... First, I open Interpret Sample Rate, and the Interpret Sample Rate box reports the current sample rate is 96000 Hz, and 96000 Hz is selected in the drop down menu,

I wondered about this, as I had a feeling somehow that it wasn't going to be that simple...

The initial problem is that it's reporting that this is, in fact, a 96k file when you think it's a 44.1k one. When you say that it was 'set to 96k everywhere else', what does that comprise? Audition is remarkably good at recording exactly what it's been sent, but if it's set to 44.1k and that's what the OS thinks it's getting, then it is liable to re-sample this on the way in - and at that point, all bets are off. What hardware are you using?

It's rapidly got to the point where we'd need to see at least a sample of one of these files in its original format to work out what's possible to do to fix it, I feel. You'd need to post it somewhere like Dropbox and put a public link to it. It doesn't need to be large - even 30 seconds of it will do, as long as the content makes it reasonably clear as to what the correct speed is supposed to be.

[melbendi]

Hi Steve, I have that same bad feeling, I can't believe I went that far without testing. 
The Audio Hardware prefs were set to 41000Hz (which is the only place I had the setting wrong, should have been 96000Hz).  When I said "everywhere else", I was referring to the New Audio File dialogue box, which was set to 24-bit 96000Hz, and the Save As dialogue box with was set to 24-bit 96000Hz WAV.  Here's a sample of one of the bad recordings:  Dropbox - Sample_Recording.wav   I searched online, but so far, I haven't been able to find any samples for you to compare it to the way it should sound.  Honestly, if you're able to even get the vocal to sound like a human voice speaking proper English, then it will be correct.   Will this help, or should we wait until I get a chance to record one of the tracks correctly?  Thank you again for all your help!

[ryclark]

As part of the diagnostic process can you tell us how long the track should have played for in real time? Interpreting the sample rate as 176400 Hz ie. 4 x 44.1K, the pitch sounds to be more or less correct although the speed is still wrong. However there are a lot of missing samples which is what is causing the crackly sound on the audio, particularly the bass.

Your music sounds as if you were trying to digitize a vinyl record of some sort. How was the audio getting from the player into Audition via your computer?

[melbendi]

Thank you for your reply, ryclark.  Yes, I was digitizing a vinyl record using a McIntosh MP-100 Phono Preamp.  It connects to my MacBook via USB2.  Files I've recorded previously, when I've remembered to set the Audio Hardware prefs correctly, came out perfect with this hardware set up, so I think the problem was just the Audio Hardware setting being different than my New Audio File and Save As settings.  I'm not sure of the exact length of the track, but I also thought 176400 sounded close in that the pitch of the singer's voice sounded correct.  In other words, it sounds like his voice.  However, the pacing of the music is too fast.  So just to see what happens, I tried 132300Hz (3x44.1k), and it seemed the pacing was almost right, just a little slow, but now the voice was too low again.  Do you think there's any hope?   It may just be that I have to bite the bullet and start all over again from scratch, but if you know of any possibilities, that would be great   I appreciate your help!

[SteveG_AudioMasters_ · Accepted solution]

Sorry about the delay - I'm in a somewhat different location now!

kalibahlu  wrote

The Audio Hardware prefs were set to 41000Hz (which is the only place I had the setting wrong, should have been 96000Hz).  When I said "everywhere else", I was referring to the New Audio File dialogue box, which was set to 24-bit 96000Hz, and the Save As dialogue box with was set to 24-bit 96000Hz WAV. 

What you've done is to record a signal that's only got 44,100 samples to fit when the software is expecting 96,000 of them. So it's not going to fill up all the sample points, and since that's not an even division ratio, there are inevitably going to be places where there will be no data - which explains the dropouts; any samples that don't align sufficiently with the 96k rate get dropped.

There's no way to soften this, I'm afraid - you're stuffed. There's absolutely no way you can recover from that, as the missing data simply isn't on the original recording. If it's vinyl, best set everything to 44.1k (unless you like recording a lot of noise unnecessarily) and stick with that; there is no advantage whatsoever to recording at 96k unless you are trying to sample bats.

[melbendi]

OK, thank you Steve, that's what i was afraid of.  I wish those settings were somehow linked in Audacity.  Well, at least I know I won't make that mistake again.    Back to the drawing board!