Windows 7's resampling sucks, How can it be improved? Options

[googlebot]

Playback over WASAPI must match the sample rate of the output device. Thus applications must supply their own resampling if the playback rate does not match. Most applications don't do this but just employ MME or DirectSound, which provide transparent resampling for non-matching rates.

Since some of my content is 44.1 kHz and some 48 kHz I used to set the output rate to 192 kHz so that all audio had to be upsampled. With good software resampling and a cheap onboard codec this could, in theory, even improve the overall result. But the opposite was true. Today I couldn't longer ignore the impression that the resampled output sounded somewhat muffled and did some measurements.

You can view the result here. The upsampled output (44.1 -> 192 kHz) suffers from quite a HF roll-off in comparison to pure 44.1 kHz output.

In Windows XP the SRC quality could be adjusted. I can't find anything comparable in Windows 7. Does anyone know more?

This post has been edited by googlebot: Feb 9 2011, 20:33

[googlebot]

The problem is mostly solved. It seems to be a really obscure bug. All analog 192 kHz output gets low-passed at 20 kHz unless you are concurrently recording at 192 kHz!!! That's right, pure 192 kHz playback - whatever you have locally set as recording sample rate - and captured by another computer will get a 20 kHz low-pass, but a local RMAA test will show a super-flat bandwidth of over 50 kHz. Since I almost exclusively used 44.1 kHz source signals in RMAA this almost drove me . Sending a 192 kHz test signal to the second computer was a very desperate, last attempt (and finally eye opener). The HF roll-off is a result of low-passing twice, first by DirectSound and then again due to the bug.

The 44.1->44.1 and 44.1->96 modes only undergo no or just one (proper) resampling by DirectSound. And the quality is fine!

This makes the topic title misleading. I propose changing it to "How I got 0wn3d by a b1tch ca113d ALC889".

This post has been edited by googlebot: Feb 11 2011, 03:32

[soulsearchingsun]

The HF roll-off is a result of low-passing twice, first by DirectSound and then again due to the bug.

Why should DirectSound do 20kHz lowpass-filtering if you're using 192kHz sample rate on the output?
Your signal will imho be lowpass-filtered twice, once in the output stage at around 40-50kHz (if your codec is good) and once in the input stage at 20kHz (recording at 44.1kHz). If this bug is really hardware (or driver) related the codec (dac side) is lowpass-filtering at 20kHz. I have no idea why this should be the case.

[googlebot]

Why should DirectSound do 20kHz lowpass-filtering if you're using 192kHz sample rate on the output?

Because the application outputs a 44.1 kHz stream. It's a side effect of upsampling.

Your signal will imho be lowpass-filtered twice, once in the output stage at around 40-50kHz (if your codec is good) and once in the input stage at 20kHz (recording at 44.1kHz).

The full chain is actually something like this:

Application (44.1 kHz) -> SRC (192 kHz) [LP@~22 kHz] -> DAC (oversampling to 192*n kHz) [1. digital LP@~96 kHz, 2. analog LP@~192*n/2 kHz] -> ADC [1. analog LP@~96*n/2 kHz, 2. digital LP@~48 kHz] -> Audacity (96 kHz)

This post has been edited by googlebot: Feb 11 2011, 13:29