Hi,

I have some ASF files of a radio talk show (mainly speech) that are Windows Media Audio 2 that I wish to convert to MP3 in order to play them in a portable mp3 CD player. I have many of these files that are all about 3 hours long. I have two types of files, some are 32 Kbps @ 22 KHz files, and others are 64 Kbps @ 44 KHz sampling rate. They are from this webpage here - thankg God for ASFRecorder: http://www.triplej.abc.net.au/thissporting...oom/default.htm

I realise that the Windows Media Audio 2 codec is apparently propriety, and it took me ages to find something that would convert it to WAV or MP3. I finally found an app here called Awave that did it no problems: http://www.fmjsoft.com/awaveaudio.html Unfortunately it is shareware, does anyone know of a freeware app that does this? I don't want to use the disc writer in WinAMP because having to play through the whole file is rather time consuming at 3 hours each.

I have two questions: (1) When converting the 64/44 files to WAV, should I down sample to 32 KHz? (and when converting the 32/22 files just convert straight to 22 KHz wavs) Is it possible to make a compliant MP3 from a 22KHz file? Or is 32 KHz the minimum?

(2) What are the best settings for LAME when encoding both types of resultant WAV files. I realise it is bad re-encoding already compressed files, but my cd player can't play WMAs. (as an aside, do CD players that say they play Windows Media play WMA 2 files? Or only WMA7/8/9?

I did a quick convert of one of the 64/44 files to a 64/44 MP3 using LAME_ENC.DLL which sounded pretty close to the original WMA (I wouldn't say it sounds "worse" just different). Should I just stick to 64 Kbps files considering the recompression?

Thanks for any advice,

Simon Howson

dBpowerAMP Music Converter

The best freeware-app for converting is ffmpeg, you can get win32 builds here: http://www.geocities.com/hstink/
It contains it own open-source wma-decoder, so you don´t need to intall anything, like hacked drivers etc.
It is a command-line application, use this to decode wma to wav:

ffmpeg -i "yourwma.wma" "decodedwma.wav"

Unfortunately mp3 in ffmpeg is buggy, so decode to wav first and then encode to mp3 using Lame. If you would want to encode to mp2 or ac3 you could do it without any problems.

WMA isn´t very high quality, so I thing alt-preset standard is a waste of space, use a lower bitrate recommend setting from this forum.
If you like to resample, you can do so directly in ffmpeg: ffmpeg -i "yourwma.wma" -ar 22050 "decodedwma.wav"
mp3 has differnet versions, supporting differnt samplerates:
MPEG 1 Layer 3: 44,1kHz; 48kHz; 32kHz
MPEG 2 Layer 3: 22,05kHz; 24kHz; 16kHz
MPEG 2.5 Layer 3: 11,025kHz; 12kHz; 8kHz (also this is invented by FhG, it is not official standard)

Always remember, only transcode lossy->lossy if there is no other option (portables etc.). You´ll always lose quality!

Thanks for the tips

Regards dbpowerAmp, I have installed the Windows Media Audio 2 plugin, but WMA doesn't appear as an option to convert FROM, only to convert to.

I note on this page: C:\Program Files\Illustrate\dBpowerAMP\Help\Codec\WMAV2\help.htm

It says "no WMA files can be read (CONVERTED FROM, or played in dAP) with this Codec". Is that the case, or have I installed something wrong?

I tried ffmpeg from the download link suggested, but when I decode (entering the command line exactly as you suggest) I get "error while decoding stream" messages. This is getting frustrating, the only app I have found that works is shareware - US$50 too!

I've been using Awave to take the 64/44 files and convert them to 32000 KHz WAVs. Then I used LAME with the following settings:
-b 32 -m m -h --abr 40 -B 48 --resample 32 --lowpass 12 --lowpass-width 0 --noshort --strictly-enforce-ISO

Is any of this over kill? Is there a forum that discusses the best settings for voice recordings? I don't know much about filtering, is the "-lowpass 12" switch suitable?

Thanks for any more advice

With the next build of ffdshow (based on ffmpeg), you could do the conversion in GraphEdit. Look for it.

I would try "--alt-preset 32" with Lame 3.93.1.

I haven't tested, but it should do the job very reasonably.

For the conversion to WAV... no suggestions right now, sorry.

As KBlood says --alt-preset xx is the way to go. It will choose the most appropriate sampling frequency and low-pass for you. So there is really no need to downsample prior to encoding. The problem is to determine what bitrate(=xx) to use. I would simply try different bitrates and see how high bitrate is needed to not lose to much quality.

You need the WMA v9 Codec (both encode and decode).

The help is working out great, I have got dBpowerAmp Music Converter working nicely converting WMA2 -> WAV. However, it is much slower compared to Awave, it takes about twice as long for any given file, and I've compared the output WAVs and I can hear a difference between them. The upside is that it is free :-) I have some further questions about the best way to do this with regard for retaining the best quality and not introducing artifacts.

I need to create MP3 compliant files, so I need 32 KHz sampling rate files minimum - my player doesn't seem to play MPEG 1 layer II, or 2.5. My questions are:

(1) For the 32Kbps/22KHz files, is it best to extract to 32 KHz WAV, then go to MP3 using LAME. Or is it best to go straight from 32/22 to 32 KHz MP3 using LAME's "resample" featre? i.e - Is it best to skip the intermediate step of making the WAV? Am I basically testing what app has the better resampling feature? Is LAME_ENC.DLL worse at this than LAME.EXE? Or should they be identical?

I was using "--preset 32 -resample 32". Is it OK using the resample feature inconjunction with the "--preset" settings? Or are the settings tuned only for 44.1 KHz files?

(2) For the 64/44 files is it best to go to go to a 32KHz wav, or to use LAME's resample feature during compression?

(3) Is it advised at such low bitrates to convert the files to MONO, does stereo mean bigger file sizes? At such low bitrates does stereo reduce quality?

Thanks in advance for any suggestions.

If you want the best quality at 32KHz then make sure the option 'Professional Frequency Conversion' is switched on (dMC Configuration) and write the wave files (or even mp3 Lame files direct from dBpowerAMP) at 32KHz. SSRC will be used to convert the frequency, it is a little processor intensive.

A few more things (I realise I am getting annoying now).

What does SSRC mean? I do have Pro Frequency Conversion on. The other app I was trying Awave uses "F.I.R. filter" conversion, and says "65 TAP" next to it. At that setting it was taking about 20 minutes to decode each 3 hour file (Athlon XP 2100), but the quality was margainably better than the dBpowerAmp converted files.

I have been using this setting to make the MP3 files:
LAME 3.93 -b 32 -m m -h --abr 40 -B 48 --lowpass 12 --lowpass-width 0 --noshort

I tried "--preset 40" however it seems that is set to automatically resample the files to 16KHz which degrades the quality (Considering they have gone from 22 -> 32 then back to 16, my player can't play these files anyway - 32KHz sample rate minimum). Is their a way to use "--preset 40" but without the resampling?

I came with the string I did by using the "VOICE" profile in RazorLame (is this profile obsolete?), but then removing the resampling, and reducing the bit rate to 32 - 48 with 40 as the target bitrate. The files sound CLOSE to the original WMAs but not perfect, there is a bit of that "tin can" distortion, sorry for not being any mores specific than that. Any ideas of any settings I should change? is "--noshort" desireable at such low bitrates?

Shibatch Sampling Rate Converter, its a freeware sampling rate converter. Considered to be the best free sampling rate converter.


3.93 is buggy, you should use 3.93.1 or 3.90.3


The best way is of course to increase the bitrate. For a specific bitrate Lame chooses the most appropriate lowpass frequency and sampling rate to achieve the best quality for that bitrate. If you must stay at 40 and use 32 kHz I think you can use the --resample switch, but I am not sure.