Potentially corrupt batch processor file - Guidance requested

Hi PG

I had a new deliverable request for a set of mp3s at 48k

This is for a major client that you just deliver what they ask for … not question by the way.

While I can do this easily in ProCodec, I have been using the WL batch processor to generate mp3s etc for some time … to bring everything under the one hood.

For reasons I do not understand (but suspect a batch file corruption), last night the batch processors refused to generate an mp3 48kHz.

The resultant mp3 always seems to be 44.1k

This was the case no matter whether I feed it 24/48 or 32f/48.

I also tried to change the output from match input stream to 48k. No difference.

I closed the .wpr Project and opened a new one … then went back to the wpr Project I was working in. Still the same … 44.1 kHz only… On reboot the issue remained.

On the suggestion of someone on the WL facebook group, I initiated a new batch processor and wunderbar it generated a 48 kHz mp3. Settings were identical to my usual batch processor.

This suggests to me I have a corruption somewhere but I do not know what file to delete.

PG, do you have any guide to which files could be corrupt that I should delete.

I am on Windows 10 current build and current version of WL 11

Thanks

Hi!

Have you tried if using Fraunhofer switch and use Lame instead !?
but it should work anyway…

regards S-EH

Thanks but I want to know why it is suddenly misbehaving and it looks like it’s a corruption of a bat file or something in my system.

I also have the Sonn*x ProCodec Manager and just rendered them from that so the job is done.

The Fraunhofer mp3 codec reduces the sample rate by itself if the selected bit rate is too small. Indeed, the codec estimates (rightly), that there is no audio benefit of using a high sample rate for some low bit rate settings.

Two ways to change this:

  • augment the bit rate
  • try the Lame encoder (available in WaveLab), which has now the same logic as Fraunhofer.
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Ahhh … thank you. I was not aware of the depth of the codec. This explains a few things.

I need to read Principles of Digital Audio again I think (it’s been a while).

Appreciate your time as always.