Has anyone come across the problem of audio quality being inferior after uploading a perfectly good audio file onto archive.org?
Most of the files I upload, seem to lose millisecond bits from the audio here and there, throughout the file, as though it wasn't recorded properly to begin with, even though the original file is perfect and has no issues.
An example of what I mean is the audio here, listen to only the first 3 seconds, which has three omissions, even though the original file doesn't have any such problem:
If it helps or is relevant, it's 128kbps, and is 1.29MB.
Any ideas, or are there any other good audio hosting sites?
I had an account on 4shared, but it deleted everything without prior warning, as I hadn't accessed it for a certain time.
I don't remember coming across such problem but I think it is related to embedding the file here. If we try to download the file and then listen, or go to the actual folder on the archive.org site and listen there directly, there doesn't seem to be any problem.
Maybe the time taken for loading the audio or the buffering is the cause of the problem. Did you try the new audio/video player on archive.org? Using the new player plays the file without any errors.
Try lowering the bitrate to 96 or 48 kbps and check the difference. Maybe lowering the bitrate helps in loading the file properly while playing.
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_Y168
11-04-2012, 09:27 PM
format_quote Originally Posted by Insaanah
Has anyone come across the problem of audio quality being inferior after uploading a perfectly good audio file onto archive.org?
Maybe the time taken for loading the audio or the buffering is the cause of the problem. Did you try the new audio/video player on archive.org? Using the new player plays the file without any errors.
Yes, it was using the new one. And even trying the bitrates below, they were all bad on the new player, apart from 16kbps.
format_quote Originally Posted by ibṉĀdam
Try lowering the bitrate to 96 or 48 kbps and check the difference. Maybe lowering the bitrate helps in loading the file properly while playing.
Tried 96, 64, 48, 32, 24 and only works properly at 16, which lessens the sound quality and clarity, but at least it plays without omissions and missing bits, so that's good, alhamdulillah.
The audio in my signature has a bit rate of 64 kbps and it works perfectly fine. Most of the audio files that I have uploaded are in the range of 96-32 kbps and I don't remember encountering such problem. Still not sure what exactly is causing this problem.
^ Jazaakallah khayran katheera for looking into it, and for taking the time and trouble to make those files. Much appreciated. I've never used variable bitrates before, only constant, so will need to try them out further. I tried a 32-96 with the original file, but that didn't work, though haven't had a chance yet to adjust it a bit more.
I think I've worked a bit of it out, but only for one that I uploaded as .wma rather than as .mp3. This is the resulting mp3 file as created by archive.org (192kbps on my computer):
A recording I uploaded and marked 'no lossy formats' had them created (mp3, ogg, m3u, etc...) . How can I remove them?
If you come across this situation and you are the uploader, click [edit], select the derivation option you prefer, and then 'Update'. You should see the message "Format Options Updated Successfully". Within 10 minutes the system will create a "_rules.conf" file in the recording's folder. Then, the next time the system performs an automatic sweep looking for changes, it will notice the new rules file and remove the lossy files automatically. The sweep occurs approximately twice a day, so you should see the files removed within 12-24 hours.
http://archive.org/about/faqs.php under the section called "Live Music Archive". The section just above it called "Audio" is useful too:
We prefer that you submit the highest quality file that you have available, and then we will attempt to create smaller file sizes and formats automatically with our deriver program. We recommend that you do not attempt to do any special encoding of your files - the more settings you mess around with, the less likely our deriver code will be able to process the file.
The way I did it, was to go to the page for the .wma file I uploaded, then clicked "Edit item" on the top. On the next page, click "change the files". On the next page, under "Manage your directory tree", keep the yellow tree root folder, but delete every other derivative apart from MP3, by right-clicking. Then click update item below. It will then create a new page for the item, and the mp3 will play fine. That's what I understood from the first quote, and it seems to work, though doesn't make any difference to the first mp3 file I originally uploaded in mp3 format, only to the one I originally uploaded as .wma as created by Windows Movie Maker which I use for audio splitting/joining.
But this cutting out of bits happens on some files, and not others, e.g. this 128kbps audio is fine:
This was originally mp3 and not converted from wma, I wonder if that has something to do with it. I might try uploading future ones (files I make myself) as the original wma files without converting, and let archive do the conversion. Must be something to do with the encoding...
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