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Audio Conversion 32 bit

Sunarep

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Hey guys,

Having a sort of a problem. I have a handheld dictating machine and i am currently trying to record some discussions

the files are recorded in 32-bit - two stereo channels for front and back
I am able to play the wav files with foobar and other players but unable to open them in either audacity, adobe premiere, goldwave or any other software i have tried out.

now for future recordings i will use a different option but i would love to convert the files i have (2 hours of material) into something i can work with but i have no idea how to handle it - anyone know of good reliable conversion tools?
 
wow that was fast neg (that's what she said ;) )

program installed but when i import the audio it decodes the audio, then tells me that no video source was found and... well not really much else is happening, it doesn't really show an imported file. is there anything i need to change when importing just audio files?

interestingly enough importing other audios is no problem... damn i hate these conversions
 
I hate the name xvid4psp more than almost any program name ever.

Anywho, have you tried something like eac3to or besweet/belite or aften/aftengui/encwavtoac3 or even dBpowerAMP? Can you install 'mediainfo' and tell us how the audio is encoded? You might need a decoder such as ac3filter or 'AC3 ACM Codec'.

Also, this sentence doesn't make sense.
It will make it an MP4 but can you convert it to AC3, WAV, AAC, etc.
MP4 is a container. It likely has AAC or rarely AC3 audio inside it. Adding an extra transcoding/encoding would diminish quality, when you can just remux.

I threw a ton of names in this post, so I would google eac3to and ac3filter (and mediainfo, it's a lifesaver) first and then go from there.
 
Since you can open the file fine in foobar, why not convert with that?
 
booshman said:
Since you can open the file fine in foobar, why not convert with that?

^ I use foobar often for audio conversions. That would probably solve your problem, but doesn't answer the question of why those other programs aren't able to read it and foobar is... Use Mediainfo to verify the exact details about the file. File size and/or bit-depth (i.e. 32 bit) might be factors. I know some programs don't work with greater than 16 or 24 bit audio, and the wav files can't be more than 4GB.
 
ok so apparently it's not 32 bit, i used mediainfo and got this:

ID : 0
Format : PCM
Format settings, Endianness : Little
Codec ID : 1
Bit rate : 2 304 Kbps
Channel(s) : 2 channels
Sampling rate : 48.0 KHz
Bit depth : 24 bits
Stream size : 0.00 Byte (0%)
 
Have you tried converting it with eac3to?
 
Thanks Booshman, I just used foobar (was not aware of its conversion option) and it made me love foobar even more

:clap2::clap2:
 
Sunarep said:
Thanks Booshman, I just used foobar (was not aware of its conversion option) and it made me love foobar even more

:clap2::clap2:

No worries. There's loads you can do with foobar if you have the right components in place. Check this lot out.

http://www.foobar2000.org/components
 
In the future, if you care about quality, it might pay to read http://forum.doom9.org/showthread.php?t=125966 which is why I suggest eac3to. In this case it doesnt matter because your source is PCM. I assume foobar uses libav/ffmpeg which he says "the output is not 100% identical to the Dolby/DTS reference decoders" The libav/ffmpeg (E-)AC3 and DTS decoders are both limited to 5.1. Furthermore the DTS decoder ignores the additional DTS-HD information. The libav/ffmpeg TrueHD decoder works perfectly fine including full 7.1 decoding.

If foobar is applying dynamic range compression you may want to reencode. I am guessing it didnt because your original audio appears uncompressed.

eac3to is unique because it uses different decoders depending on the audio source.

default decoders used by eac3to:

MP1, MP2, MP3: libav/ffmpg
(E-)AC3: Nero
AAC: Nero
TrueHD: libav/ffmpeg
DTS(-HD): ArcSoft
 
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