No, it merely fixes the issue that a saved project that had a track or clip containing 2^31 samples or more of audio would reopen with silenced audio and all the audio block files would be reported as “orphans”.
2^31 samples equates to about 13.5 hours at 44100 Hz or about 1.5 hours at 384000 Hz (whether mono or stereo).
The project will not open correctly in 2.0.6-alpha if the AU files in the _data folders have been renamed (as appeared to be the case in the screenshot).
Gale
1.5 hours at 384000 Hz (whether mono or stereo).
So the show would have been produced with actual damaged data, or perfectly fine data if Audacity could read it? Enter: Audacity 2.0.6-Alpha.
Koz
What is the process now that the files have all been renamed?
Koz
You can’t make the recovery work in 384000 Hz. The recovery WAV will be 44100 Hz so it will be much longer than the original audio. To fix that, as already suggested, use the Track Drop-Down Menu (click in the Track Name, then choose “Set Rate”, then choose 384000 Hz).
Holy crap this actually worked. I did not understand that you wanted me to open the WAV file in audacity, I thought you wanted me to fix the HZ rate problem with the recovery program.
Now that the audio is recovered there is actually a new problem. The audio jumps around the recording every which way and the audio only stays constant for a little less than a second each time. After that it jumps to a different part of the audio, but you can clearly tell we are talking.
Either way this is a definite step in the right direction, and even if I cannot rearrange the audio I am still grateful for the help everyone has given me.
If there were no other problems with the project, the audio data in the saved project is fine. The problem is just with reading the saved project if read in 2.0.5 or earlier.
See Missing features - Audacity Support .
Gale
Once the AU files are time sorted and renamed, you use the 1.2 Recovery Utility to make a Recovery WAV (or one Recovery WAV for the left channel and one for the right channel if it was a stereo recording).
@LC3, have you so far only renamed the AU files in the “d21” folder and made the left-and right- channel recovery files for that?
Gale
So far I have renamed all of the files in the d folders and have only recovered folder “d21” which has 202 files.
BTW, the audio is mono so I’m not sure if there are different settings I should have the recovery process on.
If you mean that instead of going as it was recorded at 1 second, 2 seconds, 3 seconds, 4 seconds, it sounds something like the audio from 1 second, 4 seconds, 2 seconds, 3 seconds, then this is probably because you edited the recording. Did you edit it before you closed the project? If you edit, the AU files get moved around so that they don’t follow in timestamp order any longer. They will therefore be misordered when you time sort and rename them.
What you should have been told to do was try 2.0.6-alpha first, before you renamed the AU files. That should have worked, once you had changed the AUP file and _data folder back to their original name.
Gale
Thanks. If you have a 32-bit project (which is the default unless you change it), a complete “d” folder (256 AU files) should give you about 3 minutes of mono recording (after you have imported the recovery WAV and set the track rate to 384000 Hz).
I thought perhaps you had a stereo 16-bit project in which case each “d” folder would similarly give you about 3 minutes of audio after correcting the track rate.
Since you have a mono project you just tell the 1.2 Recovery Utility that number of channels is “1”.
Gale
I did not edit the audio before I saved it. As i said in an earlier post I just saved the audio, closed it, changed the name, and when I tried to open it back up everything went to hell.
I did look at what you said:
I recorded 3.5 hours of audio and then saved the file. Once Done I noticed I had named the project wrong, so I closed the project
It wasn’t clear if you had edited between saving and closing. You said something was “Done” between saving and closing.
So describe the problem. Is it as I said - the audio in the Recovery WAV goes forwards to some future time then back again? If you did not edit, I don’t see any obvious explanation unless you failed to time sort the files before you renamed them.
I take it you did not make a backup of the _data folder before you started renaming the AU files? Do you have Windows System Restore on for the drive you originally saved the project to? If so there is a small chance you will have a previous version of the _data folder. Right-click over the _data folder named as you originally saved it and see if there is a menu entry “Restore Previous Versions”.
Gale
Sadly I do not have the originally named files, all I have are the files I renamed. I still don’t understand how renaming them would cause them to become out of order, and more frequently than each block at that.
Each block file is only about 0.7 seconds of audio if we are talking about a 32-bit mono 384000 Hz recording.
Did you sort the files into time order before renaming them? The instructions Missing features - Audacity Support state clearly (twice) to do that. If you rename the AU files while they are sorted by name, the rename won’t be done on the timestamped order.
For example if there are three files sorted by file name:
Name Time
e0000d47.au 20:25:52
e0000d69.au 20:25:07
e0000d77.au 20:24:59
After rename, this becomes:
e0001.au 20:25:52
e0002.au 20:25:07
e0003.au 20:24:59
The renamed files are still not in timestamp order.
Gale
So I went back and re ordered the AU files in the correct order and the audio is playing in the correct order now.
Unfortunately it did not save the first hour or so and the audio is still quite choppy in between blocks. Is there any way to fix this or have I done all I can?
Good, in the recovery WAV for the “d21” folder, you mean?
What did not save the first hour?
You said you had 40 “d” folders but that would only be about 2 hours of audio. Is that what you mean? Does the folder e00d00 for that project contain the start of the recording? It should do.
Are you talking about the WAV for folder “d21” - the one that now plays in the correct order? What do you mean by “choppy between blocks”? Remember blocks are less than a second at 384000 Hz 32-bit.
If you wish, try exporting 10 seconds of WAV (16-bit) from the recovered audio and attach it so we can hear the problem. See https://forum.audacityteam.org/t/how-to-post-an-audio-sample/29851/1 for how to post a sample.
P.S. If you had exported a 16-bit WAV file as soon as you finished the recording (at 44100 Hz which is as high a rate as you probably need) then you could just have imported the WAV file to resume editing. It’s always worth exporting an immediate backup of an irreplaceable recording.
Gale
Here is a bit of the “choppy” audio as requested. This audio is not edited in any way aside from cutting it out from the rest of the track.
I’m still not entirely sure what cause this, or how to fix it for that matter.
Also the first hour or so was not saved in the _DATA folder for whatever reason, so now I only have around 2 hours of audio left.
It is not a known problem that some of the audio would be omitted from the saved _data folder if you reach or exceed the 2^31 samples limit.
The audio sounds to be in the correct order if you amplify the blocks that look like silence.
What is the source of that audio and how exactly were you recording it? If it was internet radio, did it sound like that while you were listening to it?
Gale
we recorded through a microphone and my brother and I were sitting next to each other. There was no internet access needed for this recording, as we were both in the same place using one input.
You mean a cheapo desktop mic plugged into the pink mic port of the computer? Or a USB mic? Or a condenser mic plugged into a mixer, and if so how does the mixer connect to the computer?
Were you listening to the recording in headphones as it was being made? If so, did you hear any problems?
The simple answer could be the inappropriate sample rate employed for the recording which asks the computer to push large amounts of data around and write a file every 0.7 seconds into the bargain.
Record something now with the same mic at 44100 Hz. Do you hear any problems?
Gale
My guess is the computer couldn’t keep up with the blistering high data rate chosen and missed capturing bits and pieces here and there. We have posters whose machines can’t keep up with normal data rates. There never was a whole show. Koz