[alsa-devel] wrong decibel data?

[Colin Guthrie]

'Twas brillig, and Raymond Yau at 14/06/10 13:36 did gyre and gimble:
> 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
> }

I've made this change on my system and while previously my UI had no
"Base Volume" displayed (because all my "h/w" (I include softvol in
that) controls had their dB value >0.

Now that this change is live, I have a base volume present in my GUI (at
around the 64% mark with the cubic scale we've already discussed). When
I set my volume ot the base volume, the h/w controls are all set to 0dB
which is exactly as expected.

I fail to see the point here? The base volume is clearly exposed to the
as the recommended point on the scale at which no clipping occurs.

I really don't get where your complaint is.

Col





-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

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