[alsa-devel] Maximum 50 dB gain in ALSA softvol plugin

[Ricard Wanderlof]

On Mon, 15 Jan 2018, Jaroslav Kysela wrote:

> > The usecase we have actually calls for 55 dB of gain maximum, but I was 
> > thinking that looking at the code the maths can actually handle 90 dB so 
> > it would be good 'reasonable limit' - it's a nice round number and fits in 
> > 16 bits (signed).
> > 
> > The usecase we have is using digital MEMS microphones ((2S connected, 
> > with no integral amplification) and when the sound source is far away, a 
> > fair amount of gain can be needed.
> 
> Wow, the signal quality from few bits per sample must be really good ;-)
> It would be better to use ADC with higher resolution (24-bit) and low
> noise floor (but I see the possible costs requirements). This is why we
> have such good audio in collaborative conferences :-)

Ok, I wasn't expecting this kind of discussion on this subject ... 

It's true that for best performance, analog microphones (even analog MEMS) 
trump digital MEMS microphones when it comes to SNR, but in many cases, 
the SNR of the room itself will be the limiting factor anyway.

The point is this: a typical digital MEMS microphone has 0 dBFS output at 
120 dB SPL. Looking at one typical recording usecase, the background noise 
level in a quiet room is 40 or 50 dB SPL, with speech at a distance of a 
meter or so being around 60 dB SPL. So in this case we're 60 dB below full 
scale for the speech signal, with the acoustic noise floor being 10 or 20 
dB below that. The noise floor of a MEMS microphone of this type is 
equivalent to about 30 dB SPL, so the acoustic noise is still higher than 
the electrical noise in the system.

Now, assuming we actually want to hear what is going on, so we bring up 
the gain by 60 dB in order to get the speech to full scale. That 
means that the noise floor comes up too, and there's not much we can do 
about that anyway as it's part of the acoustic scene we're capturing. The 
fact that we loose bit depth is not of much consequence, as the acoustic 
noise in the scene is above the resulting quantization noise anyway, in 
much the same way that adding dither to a signal masks the quantization 
noise. The acoustic SNR is 10-20 dB so we could actually represent the 
resulting signal with 8 bits and still be fine.

A side note here is that many PC's, especially laptops, do not provide 
much playback gain, so there is a point in bringing up signals to close to 
0 dBFS.

The bottom line is that, yes, we loose bit depth when we apply gain, but 
that itself doesn't impact the sound quality, as long as the noise level 
is above the quantization noise to start with, which it is in many 
microphone cases. And we really might need a large gain in certain 
situations.

On Mon, 15 Jan 2018, Takashi Iwai wrote:

> [ ... ] the amplification in softvol is really dumb, and such a high 
> gain like 90dB is doubtful whether it's really useful.  As Jaroslav 
> already suggested, we need a better setup to get more meaningful 
> results.

In what way is it dumb? Amplification is just a multiplication with a gain 
factor, and the softvol plugin seems to do that fine. I just tested 
patching it to 90 dB gain max, and applying a (24 bit) sine wave at -90 dB 
and amplifying it by 90 dB using softvol, and I couldn't see any odd 
artfacts.

Of course, for the usecase I described, an AGC of some form, or dynamic 
compressor would probably be better, but for a static gain, softvol could 
very well be employed.

Ok ... bottom line ... I'd like to increase the maximum potential gain for 
softvol above the current 50 dB. I suggested 90 dB because that's how much 
the algorithm can handle, and that's what I figured the MAX_DB_UPPER_LIMIT 
in pcm/pcm_softvol.c should reflect. Furthermore, I can't see any problem 
with increasing the limit; it does not degenerate the algorithm or cause 
potential problems for existing users. Admittedly, this is not kernel 
code, but if it were I would expect that the point of view would be that 
the code should not make assumptions on user's policies but only its own 
technical limitations. And if the input is 32 bit audio, we'd still end up 
with about 16 bits of resolution when applying 90 dB of gain, so 
mathematically it's not unreasonable.

The only problem I can foresee is that it means that all future changes to 
the algorithm might need to accommodate the specified maximum gain to 
avoid annoying users who are actually using it, which might be a problem 
somewhere down the line.

But for my usecase I'd be happy just pushing the limit to 60 or 70 dB if 
that's more acceptable.

/Ricard
-- 
Ricard Wolf Wanderlöf                           ricardw(at)axis.com
Axis Communications AB, Lund, Sweden            www.axis.com
Phone +46 46 272 2016                           Fax +46 46 13 61 30