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

[jaro1]

This area has also my interest, however the problem are non-existent sources,tests or other material about it. I'm still unclear in two things i've written earlier in HA about (this or other threads, don't know yet).

1. MF pipeline uses under its audio DSPs a resampler DMO, which can be used by developing audio apps and its accuracy (filter length) can be adjusted from 1 to 60 (default is 30). Don't know if this can be also adjusted with "normal" playback through e.g. WMP12, as this is to me the only known MF application.

2. I suppose this component isn't used for resampling when using other APIs, e.g. dsound, waveout, xaudio2 (Tuniac uses this "dsound" replacement for hardware acceleration under xp - this seems not to be the case with WV/7 however, but i'm not sure). But if this is true, where does the resampling take place in this case? In the APIs themselves or in mixing engine Audiodg.dll? And can the accuracy also be adjusted (under V/7)? Under XP this was possible, but there was another architecture and the resampling was probably situated in kmixer.sys

However i don't use JRMC, i've used it once to find out the resampler DMO precision, because it allowed me to include it into the audio path, not as DSP (all were disabled), but in file type handling it was possible to "pin" it to specified format, e.g. wav. Of course, disk writer was used for output.
As audio driver was hard-settled to 16/48, resampler DMO automatically processed audio to these values (so bit dither/bitreduction was applied also). As usual, swept24 (24/96) from here: http://src.infinitewave.ca/TestSignals.zip was used for resampling.
The result is here: http://img571.imageshack.us/i/1648b.gif
Maybe this is with the default setting (30), don't know as adjusting wasn't possible in JRMC.

This post has been edited by jaro1: Apr 17 2011, 11:25