Linear Resampling for Winamp or Foobar

[b0dhi]

Does anyone know a linear interpolation resampler for Winamp or Foobar2000?

I found one for Foobar (Secret Rabbit Code), but it crashes when I play FLACs Any help would be very much appreciated.

Before someone asks, yes, it must be a linear interpolation resampler. I'm running it at 192Khz so essentially I'm using my Juli@'s DAC as a non-oversampling type DAC. Square wave response is excellent, and the music is so much more dynamic. I can't wait to hear FLACs with it...

Thanks in advance for any suggestions.

[Honken]

The SRC plugin doesn't work? That's odd, what version of Foobar do you use?

[b0dhi]

Both Foobar and SRC are the latest versions, I downloaded them today.

It doesn't work when the resampling rate is higher than 96000 (any algorithm), and only when playing FLACs. It works fine for MP3s at any sampling rate.

At 96Khz the CPU usage is only 2% in linear mode, so it can't be lack of grunt.

EDIT: I really have to say I'm amazed at the difference it has made to the sound, I wasn't expecting anything like this. That is...when it isn't crashing

[Alleyman]

Isnt a linear interpolator the worst kind of resampler? The one with noticable artifacts?

Have you tried SSRC? http://otachan.com/foo_dsp_ssrc.html

[b0dhi]

In terms of common "distortion" measurements, yes, it's one of the worst. However, because I'm using it to oversample 44.1Khz to 192Khz, the output at the DAC is virtually exactly the same as the music data. The other methods produce only sine waves at higher (10Khz+) frequencies, and what should be square waves are not sharp and have lots of ringing. OTOH, linear interpolation gives you grainy treble because that's exactly what's in the music data, but oh well...

If anyone does find one for Winamp or Foobar please let me know.

[Hancoque]

Are you sure that this "grainy treble" really is in the original signal? I will do some research on this later and compare the three different upsampling plugins for foobar2000.

[b0dhi]

It's a limitation of the 44.1Khz sampling rate. No matter what the music sounded like originally, at high frequencies all instruments are sampled as jagged edges, and above about 10Khz, as 'triangles'. Most upsampling techniques convert these jagged edges into smooth sine waves, which sounds nice and measures well, but the lower frequencies lose dynamics, texture and timing. The ear can detect timing differences of as little as 2 microseconds.

I've been measuring the output of my soundcard's DAC (Juli@) with various upsampling algorithms at 192Khz. Linear upsampling is by far the most accurate one (in terms of being identical to the input data), but as a side-effect, you have no choice but to hear how poor 44.1Khz sampling is for higher frequencies.

My ideal algorithm would resample a 5Khz square wave with a perfectly sharp edge and no ringing, but a 15Khz wave as a smooth sine wave. None of the algorithms I tried did this.

The real problem is that 44.1Khz sampling is so crap

[Hancoque]

Why is 44.1 kHz crap? What's the problem with it?

[b0dhi]

Originally Posted by b0dhi

It's a limitation of the 44.1Khz sampling rate. No matter what the music sounded like originally, at high frequencies all instruments are sampled as jagged edges, and above about 10Khz, as 'triangles'.

Ie., it doesn't have enough resolution.

[Apocalypsee]

I found no problem with 44.1kHz, its warmer than higher sampling rate i.e 96kHz (this is as high as my soundcard DAC goes). And even worse, the Linear Interpolator sounds the worse, lacking background detail (muffled) while the Best Sinc Interpolator is a bit bright. Using Secret Rabbit Code. If it sounds OK to you its fine, but not me