Winamp & Shoutcast Forums - Plugin to upsample WAV files to 24/96

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Plugin to upsample WAV files to 24/96

Hi

I am building a PC based jukebox as part of a new audiophile sound system. I already have a dCs Elgar 24/192 A/D converter that is currently driven by a dCs Purcell upsampler to convert 16/44.1 SPDIF data from the CD transport to 24/96 or 24/192 input for the Elgar.

Does anyone know of a plugin for Winamp 2 or 3 that will perform the same upsampling function as the Purcell from WAV files stored as 16/44.1 on disk. I plan to use the M-Audio revolution 7.1 card which gives the required output, but does not perform the upsampling in hardware.

Many thanks

http://www.blorp.com/~peter/zips/out_ds.exe
http://www.blorp.com/~peter/zips/out_wave.zip

the ssrc versions of the above plugins can resample, but the resampling process is lossy - why do you want to do that on an audiophile sound system?

Lion King - thanks for the suggestions.

I don't understand what you mean when you say that the reampling process is lossy. Are you refereing to the way these particular plugins work or the process in general?

I know that the question of upsampling is open to debate. After all you are not adding any more information to the CD datastrem, just interpolating between the samples to increase the frequency and adding filtered noise to increase the bit depth. With the dCs Purcell, you can change the frequency and bit depth of the upsampling using the romote control while listening to the music and I can assure you that the sound just opens out and becomes more natural when you go from standard 16/44.1 to 24/196.

The technical explaination from dCs seems to be related to the brickwall filter in the D/A. With the higher frequency, this is nowhere near as steep and is far higher in frequency anyway.

I would have been happy with my existing system, except for the fact that it was all stolen last week while I was away on holiday and so I have to find a replacement. I will replace the Elgar DAC, but before spending $6000 on a new dCs Purcel, I would like to see if the same effect can be acheived with a $99 sound card and some software!

lost parts from the out_ds faq:

Quote:

resampling stuff (for out_ds_ssrc)

Q: Can i improve audio quality with resampling?
A: No. You can't "improve" audio quality with resampling. Resampling is a lossy process. Think of it as of scaling a bitmap image to another size - output image looks very similar, but if you zoom it in, it's no longer the same. You can only avoid greater data loss with certain sorts of sound hardware by using high-quality software resamplers instead of letting your drivers/hardware handle the resampling process. I know that all the placebo kiddies around already use it to get "brighter sound" and happily upsample to 96/24 on their sb16 (then windows kernel mixer downsamples back to 44/16); don't listen to their BS.

Q: When should i use resampling and what parameters to set?
A: First, refer to your sound hardware specs. Some cards (eg. all old soundblasters) don't resample, meaning that you will always get the optimal quality ( = no resampling-related quality loss) when you don't put extra resampling. All the recent sblive/audigy/ac97 junk has fixed 48khz output sample rate and whatever you play gets resampled to 48khz/16bit; resampling quality varies between drivers / hardware models / windows mixer settings / etc. In most of cases, the SSRC resampler is superior to the one used by your hardware/drivers; you should set resampling to 48khz/16bit to prevent your hardware/drivers from performing their own resampling.

Q: But i can resample to 96khz/24bit on my soundblaster...
A: Sure you can, because windows mixer is downsampling back to format supported your hardware (eg. 48khz/16bit). Another useless data loss. *DO*NOT* upsample above your hardware specs.

Q: How do i configure input plugins for best audio quality when using resampling?
A: First. Avoid resampling at all when only possible. Some input plugins (eg. mod/spc players) have variable output sample rate; set it to whatever you resample to and you are done. If you can't avoid resampling with particular input format, set bits-per-sample to 24bit (32bit is not supported at the moment), eg. with in_vorbis, in_mad; this reduces data loss between the input plugin and the resampler (most of lossy format decoding algorithms operate on real numbers, which are converted to 8bit/16bit/24bit integers for hardware, but again, the resampler operates on real numbers too; using biggest sample size possible reduces data loss caused by real->integer->real conversion).

if your hardware works better with upsampled stuff: fine