[alsa-devel] wrong decibel data?

[Raymond Yau]

2010/6/14 James Courtier-Dutton <james.dutton at gmail.com>

> On 14 June 2010 11:22, Colin Guthrie <gmane at colin.guthr.ie> wrote:
> > 'Twas brillig, and James Courtier-Dutton at 14/06/10 09:56 did gyre and
> > gimble:
> >> If you use "alsamixer", dB values are shown so it is easy to find the
> >> 0dB "sweet spot".
> >> I think it is pulse audio that hides this information when it combines
> >> two alsa mixer controls into one pulseaudio control.
> >
> > But it doesn't hide it. It's shown very clearly in the volume control
> > GUIs as the Base Volume.
> >
> > Do you really think that most users look at the sliders to find the 0dB
> > point? Does gnome-alsa-mixer (the old one) expose this information? No.
> > Does kmix? No. So the vast, vast majority of users do not know where the
> > 0dB point is unless they use alsamixer.... and even if the user is
> > advanced enough to use alsamixer, then I'd still say a proportion of
> > users are just looking at how far up the slider is rather than looking
> > specifically for 0dB.
> >
> > So I'd argue the exact opposite of your claim. That with the base volume
> > clearly presented in the GUI, the h/w 0dB spot is much, much more
> > obvious to the vast majority of users.
> >
> > I really think this is a vast improvement over a complex balancing act
> > of getting two different sliders setup to get distortion free audio!
> >
> > Col
>
> One has very different problems with capture than one does with playback.
> With capture it is important to identify which are analog controls
> (applied to the analog part of the circuit) and which are digital
> controls (applied to the digital part of the circuit)
> So, for capture one might wish to adjust the analog control so that
> the signal going into the ADC is a suitable level, but once the signal
> is digital, one should really not adjust it further, and just record
> what you have.
> If one was to combine these two capture controls in one PA control, it
> would just be wrong.
>
>
The AC97 recording from line-in problem seem not related to capture gain
since you can set capture volume to 0dB

The HDA 's "PCM" softvol plugin is different from AC97 "PCM" Playback volume

But you can change the softvol plugin to add gain to emulate the clipping in
software side if PA developers did not have ac97 sound card ( clipping occur
in hardware side )

  /usr/share/alsa/cards/HDA-Intel.conf

HDA-Intel.pcm.front.0 {
    @args [ CARD ]
    @args.CARD {
        type string
    }
    type softvol
    slave.pcm {
        type hw
        card $CARD
    }
    control {
        name "PCM Playback Volume"
        card $CARD
    }
+      min_dB -46.5
+     max_dB 12.0
+      resolution 32
}