[pulseaudio-discuss] On scaling the HRIR in module-virtual-surround-sink

[Alexander E. Patrakov]

2014-04-02 1:48 GMT+06:00 Niels Ole Salscheider
<niels_ole at salscheider-online.de>:

> Would you like to write such a patch or should I do it? Or do you have an idea
> for a better heuristic?

I do have an idea for a possibly-better heuristic (see below).
However, I have caught a cold, and there is also an extraordinary
number of DNS queries we (SafeDNS) get from Turkish users right now,
leading to server load issues. So, my health and my work are
priorities now :(

So, please submit a patch.

A possibly-better but untested heuristic is as follows, and I do
intend to implement it later, after the Turkish YouTube/DNS situation
settles.

1. Collect spectrum and inter-channel correlation data from some music
DVDs, take an average.

2. Synthesize a short signal that has roughly the same characteristics.

3. Filter it through the A-weighing filter (see the transfer function
in Wikipedia, it is a rational function => the filter is implementable
as IIR).

4. Store the result as a short sequence of samples in PA source code.

5. When loading a filter from a wav file, apply it to the test signal
from step (4).

6. Take the sum of squares of samples sent to the left headphone. This
(due to A-weighing) would represent the subjective loudness of the
filtered music.

7. Compare that to the sum of squares of samples sent through the
conventional downmixer. Compensate accordingly by scaling the loaded
filter.

The result should be that the audio filtered through your module and
through the conventional downmixer (in src/pulsecore/resampler.c), in
average for the music material, would be of the same subjective
loudness. I am not sure if that would be sufficient to eliminate the
clipping, but we can always multiply the result by the fudge factor
later if needed.

-- 
Alexander E. Patrakov