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[thezone]

Hi All,

I have a few 5.1 FLAC files that are encoded at 352,800Khz. My receiver can not actually handle this particular sampling freq so I have set DSP to output as 44,100. This worked fine when bitstreaming however, since changing my set up to non-bitstreaming over hdmi to take advantage of video clock a strange behavior is occurring.

The file will play for a random amount of time (could be 10 seconds could be 40 seconds) and then wham....garbled loud noise!

Can anyone help?

Thanks

[Vocalpoint]

Can anyone help?Thanks

Downsample to 24/96 (at best) - problem solved.

VP

[thezone]

Downsample to 24/96 (at best) - problem solved.

VP

Thanks for the reply,

Sorry how do you do this? Do you mean use a audio software program to permanently change the files or set MC to handle this on the fly?

Thanks

[Vocalpoint]

Sorry how do you do this? Do you mean use a audio software program to permanently change the files or set MC to handle this on the fly?

If it were me - and they were my files - I would instantly change them to 24/96 and carry on. I cannot see the point to a file at that sample rate - since 99.9% of humans could not tell the difference between it and a normal 44.1 file - but that's just me

If you want to keep things as they are - try and see if MC could swap it on the fly down to 24/96 or something more reasonable. I do not use any of the MC conversion protocals offered as I downsample everything to 24/96 at the max using iZoTope RX.

I can't justify the file storage and cannot hear any difference anyway

VP

[mark_h]

You could set the (Ctrl+D) DSP->Output Options to convert on-the-fly for playback

Or you could use the same setting in the Convert Format dialogue and permanently downsample the file to 96/24..