.asf to .wav conversion; poor results

[Saratoga Sam]

Hi -

I have a 3Kb .asf file that I would like to convert to .wav. The file is a mono recording of Curly of the Three Stooges doing his signature "woo-woo-woo" sound.

I'm using your Music Converter and comparing the results to the original .asf file using the latest version of Windows Media Player; version 9.

I cannot get the resultant .wav file to sound like the original .asf file! The closest result I get is when I choose an MP3 format at 39Kb/sec. The resultant file is 29Kb in size, however the sound is truncated at the end. Other attempts resulted in a file that sounds like it's played considerably "faster" with less fidelity.

I'm new to the world of audio conversions so I'm probably doing something basically wrong.

Any ideas?

~Saratoga Sam

[Joseph]

What is transcoding and why is it bad?

[Saratoga Sam]

ok -

If transcoding is the problem I am encountering, then how can I convert a .asf file to a .wav file without loss?

Thanks

[RossRoy]

In theory, converting to WAV from a compressed format like MP3 or WMA shouldn't show any loss of quality. WAV being a lossless uncompressed format (usually that is) there shouldn't be any quality loss.

What happens is you probably have the WAV output module configured to embed a compressed format into a WAV. When you go to CONVERT TO, choose the WAV format, and make sure the radio button on top is NOT next to the compressed format, and for the three drop list, place them all at the option that have the (CD) mention at the end.

Let me know how it turns out.

[Saratoga Sam]

Odd. What you're describing is not what I'm seeing in the user interface of the Music Converter.

This is what I get:

Step1: I click on dBPower Amp Music Converter.
Step2: I select a file to convert: curly.asf. I click OPEN
Step3: I select WAVE
Step4: Now I'm in the Music Converter window. It tells me I am converting 1 file to WAVE. Attributes: CD quality WAVE [PCM]
Format: 44.100Khz, 16 bit, stereo

I can click to "change format" or click "To CD Quality" I do neither.
Output to: volume normalize is checked and so is Preserve ID tags.

and that's it. I still get a poor quality conversion.

any ideas?

Thanks for your help so far!

[Joseph]

e-mail the file to me. Let me see what I can do with it.

[RossRoy]

You could try getting latest Music Converter beta (found in the beta forum)

I'd be surious to see this file too. You can attach it to the forum or email it to me at slacs@yahoo.com

[Saratoga Sam]

Wow - thanks for the help. This far exceeds anything I was expecting. I have emailed the .asf file to both of you.

Good Luck. I hope this is a simple problem.

[Spoons]

I'd be intersted in this big 3KB :D file too : music @ optusnet.com.au

thanks.

[Saratoga Sam]

ok - is it the academic challenge or the desire to have the Curly woo-woo-woo sound driving your interest?! ;-)

(I'll send you the file too. I have a whole bunch of Three Stooges .asf files that I downloaded off the the official Three Stooges webpage. If this kind of stuff interests you, here's the URL: http://www.threestooges.com/sounds.htm )

Have fun - and, again, thanks for the help.

[RossRoy]

I think I got your problem. It's a frequency conversion problem.

Looks like the ASF file is not encoded at 44100Hz, it's actually at 8000Hz

So, since any audio file is frequency dependent, you need to make sure you destination frequency matches the original freuqency, in our case 8000 Hz whereas most codec defaults to the CD quality of 44100 Hz.

So, if you want to make a WAV file out of your ASF, just make sure to set the frequency to 8000 Hz or "Freq as source".

Oh and it was the academic challenge of course! :p

To Razgo or Spoon,
isn't the professional frequency conversion thingy supposed to handled that kind of problem? I tried it activated and not activated and I get problems both ways. The converted sound file sounds distorted and faster if I leave it at 44100 Hz freq. and that goes for MP3 and WAV

[Saratoga Sam]

Yup - that "soitenly" did the trick! Thanks!

So, educate me; how can you tell that "curly.asf" was encoded at 8KHz? And when you say encoded, do you mean this is the sampling rate used to digitize the audio?

[RossRoy]

Yup - that "soitenly" did the trick! Thanks!

So, educate me; how can you tell that "curly.asf" was encoded at 8KHz? And when you say encoded, do you mean this is the sampling rate used to digitize the audio?

I think if you just play it in Windows Media Player, it shows the technical specs in the bottom, or you can right-click the file in the playlist to see statiscal information. But, what I actually did is convert the file to a WAV file and selecting the "freq as source" option. Because when I heard the result using the standard conversion, I knew right away that it was a frequency related problem. THen, if you right-click on a WAV file and choose properties, Windows XP shows the information about the file: freq, bits and channels. That's how I found it.

Anytime audio information is digitized, it is referred to as being "encoded". So frequency, bitrate, bits and channels are all encoding parameters that define the final quality of the audio file, there are more parameters that can affect audio quality of course, but these four parameters are the most mentioned.

[Unregistered]

Hi -

I have a 3Kb .asf file that I would like to convert to .wav. The file is a mono recording of Curly of the Three Stooges doing his signature "woo-woo-woo" sound.

I cannot get the resultant .wav file to sound like the original .asf file! The closest result I get is when I choose an MP3 format at 39Kb/sec. The resultant file is 29Kb in size, however the sound is truncated at the end. Other attempts resulted in a file that sounds like it's played considerably "faster" with less fidelity.

~Saratoga Sam

I have done something similar recently. Started with an .asf file and went "directly" to MP3. Results were not very good. Music sounded as if it was played in fast forward. I was using Lame to encode.

After that, I decoded to wav. Then encoded to MP3. Used another app to encode. Results were much better.

As to the truncation, if you are referring to the last second or two, and are listening with WinAmp (Gapless plugin), that's where your problem may be. If the truncation is much longer - then don't know.

[Spoon]

Changing frequencies is not a good idea period, only do it if you are forced to.

For the latest dMC I have a new wave compression page with 'Frequency same as source'