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Audio encoding problems en masse

By Vigilante
Oct 3, 2006
  1. Hey, I think I might have talked about this problem before. But it is needed anew.

    I can't convert my WAVs into ANYTHING!

    At the basic level, I try to use Winamp's disk writer, I set the options and go. This is actually the only thing that works. However, the resulting MP3 is either playing to fast or to slow. It doesn't encode right or something.

    So next I try to use LAME in Audacity, exports 0bit files no matter what options I set. If not a 0bit file, it throws an error instead.

    Next I try to download an encoding program, found this one called "BonkEnc" which is freeware and looked pretty nice. It supports (after install) OGG, MP3, FAAC, and FLAC. NONE work, not even it's native WAV or BONK encode. Either I get an obscure error, a tiny file that doesn't even squeek when played, or it goes "through" the process but creates no file, and gives no error. None of it's methods encoded my WAV.

    I also download the stand-alone FLAC encoder, no beans.

    I don't understand why no program is able to encode my WAVs, it's really upsettings.

    For reference, my source file is a WAV created with bit rate 705kbps, 16bit, mono, 44kHz, PCM.

    I guess the logical question is, is there something wrong with the original bitrates or quality, preventing converters from working right? Should I change bit rate or kHz? I do want to record at high quality, obviously. So I don't want to lower the quality of my WAVs to much.

    Any ideas? Thanks.
     
  2. Nodsu

    Nodsu TS Rookie Posts: 5,837   +6

    What program did you use to create the WAVs? Maybe that one's buggy and not all the encoders you tried?
     
  3. Vigilante

    Vigilante TechSpot Paladin Topic Starter Posts: 1,666

    The program is Cubase LE, came with my Tascam USB device.

    But I've also had trouble converting WAVs that came from Audacity as well.

    At one point I was able to create FLAC files, but the person who I created the files for, could not listen to them, even after installing Winamp, and the FLAC codec. So I'd rather use MP3, or even WMA if I could.

    I'm wondering if there is a tool that can more fully tell me the details of my WAV and what might be required to convert it? Or if I created the WAV at some silly setting that converting tools can't deal with?

    Thx
     
  4. N3051M

    N3051M TS Evangelist Posts: 2,115

    wav should be universal... how about converting/importing other format like ACC, AIFF, MP3 etc? and vice versa to wav?
    have you tried with different bit rates? (192kbps etc)?

    maybe your audio codecs are stuffed up - device manager>sounds, games etc>Audio codecs/devices [uninstall]>reboot>reinstall...
     
  5. Vigilante

    Vigilante TechSpot Paladin Topic Starter Posts: 1,666

    Ok, so I took my WAV file and copied it to my work PC. I installed Audacity so i could try to resample if needed.
    But here's the thing, my original WAV specs are:
    Bit Rate 1536kbps
    Sample size: 16 bit
    Channel: 1 (mono)
    Sample rate: 96 kHz
    Audio format: PCM

    Now when I went in Audacity with this file, I tried changing to every bit rate and sample rate I could change, but every time I exported to WAV, all those rates were the SAME!
    Next I created an all new file and recorded a few seconds and exported, still the same rates!?!
    So I don't know why Audacity won't export to the bit rate and sample rate I set. This is just getting weirder and weirder.

    I also downloaded this BonkEnc program on this PC and tried to export to MP3. This time it says "Bad sampling rate! The selected sampling rate is not supported."
    I don't know if that's my original rate, or the rate I set it to export at. But it doesn't accept any setting anyway.

    Errrrg. Why is it so dang hard to create a recording and convert to MP3? Oh and by the way, I couldn't convert the WAV, or save a new one, to any of the other supported types either.
     
  6. Vigilante

    Vigilante TechSpot Paladin Topic Starter Posts: 1,666

    I just came across a random post on the Internet where someone said that uncompressed PCM is ALWAYS 1536kbps Bitrate.

    Can anyone validate that claim? If that's the case, my audio is perfectly normal. 16bit, 96khz PCM, there shouldn't be any reason why it won't convert.

    Interrestingly, my Cubase LE can export to MP3 just fine. I just can't take it's exported WAV and convert to anything else.

    Also can anyone confirm if PCM (1536kbps) NEEDS to be set at 48khz? Because I saw a comparrison of formats and that is what it said. So I wonder if creating at 44khz or 96khz is not correct to do?

    This is so confusing.
     
  7. N3051M

    N3051M TS Evangelist Posts: 2,115

    go to audacity's preference and set everything back to default? maybe you are recording to high quality for audacity to follow.. try downloading a sample off the net and playing around with that..

    By the way, this may/may not help but my settings on Audacity is:
    Qualtiy Tab
    Default Sample Rate: 48000 Hz (original was 44.1k, of course)
    Default Sample Format: 32-bit float
    Fast sync
    high quality sync
    no real time dither
    triangle high quality dither

    File format
    Uncompressed Export Format: WAV (microsoft 16 bit PCM)
    MP3 bit rate: 128

    Edit: I don't know for sure the bitrate for 16bit PCM sorry.. however i do think that some software/hardware dont support 96k all to well (and from how much disk space it uses)
     
  8. Mictlantecuhtli

    Mictlantecuhtli TS Evangelist Posts: 4,345   +11

    No. The bitrate depends on the number of channels, sample rate and sample resolution (channels * sample rate * resolution = bitrate).

    For CD quality, that's 2 * 44100 * 16 = 1411200 bps.

    There is a reason - MP3 doesn't support sample rates higher than 48 kHz.
     
  9. Vigilante

    Vigilante TechSpot Paladin Topic Starter Posts: 1,666

    Interresting theory, amazing such simple specs are not on the front page of any audio article you read about. I didn't not see that on any page I've found yet.

    So no higher than 48khz. Funny no other site mentioned that. So I went in Audacity, and yes my settings in Preferences are fine. I changed the sample rate from the project bar itself, but it was still exporting as 96khz. Then I saw in the far left bottom corner yet another place to change rate, which was still 96. So I changed that to 44 and exported. BINGO, my wav is finally 44khz.

    So next I went back in my BoncEnc program and exported to MP3 again and.............new problem!
    Well instead of errors, it actually DID export to MP3, with a proper bit whatevers. But the song is twice as long and twice as slow! lol

    Apparently Audacity couldn't resample or whatever, without slowing down the audio. So now I have to resample the files without making them slower and longer. And THEN export to MP3.

    Still a mess, but getting there. Thanks for the info, you could write a book with just that info, and it would probably sell because I couldn't find that anywhere else. lol
     
  10. Vigilante

    Vigilante TechSpot Paladin Topic Starter Posts: 1,666

    Oh boy, getting there now!

    Thanks for all the help! It slowed down because I had converted the bit rate of the project itself, BEFORE exporting. It was simply converted in Audacity, which screwed it up from the get go. But I started over and just changed the export bit rate or whatever, and it worked. And BoncEnc worked, I have a working MP3!

    Now if I can just get LAME to work with Audacity, I wouldn't have to use a 3rd party app to convert, *sigh*
     
  11. N3051M

    N3051M TS Evangelist Posts: 2,115

    lol.. whats up with LAME? didn't you point Audacity to the DLL file?
    (Prefrence>file format tab>MP3 [Find Library]>%directory%lame_enc.dll)
    or you just missing the file?

    hmm.. slow songs.. good thing theres time compression :D
     
  12. Vigilante

    Vigilante TechSpot Paladin Topic Starter Posts: 1,666

    Ya I did all that right, it's just that it exports 0kb MP3s.

    Have to see if it'll work now with the lower bandwidth WAVs, probably will work. I hope so!
     
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