At Mon, 16 Mar 2009 18:00:15 +0100,
Andreas Mohr wrote:
>
> Hi,
>
> On Mon, Mar 16, 2009 at 05:19:38PM +0100, Takashi Iwai wrote:
> > At Mon, 16 Mar 2009 17:06:35 +0100,
> > 私 wrote:
> > >
> > > > > What are "sliders"?
> > > >
> > > > Umm, volume level controls.
> > >
> > > Yes but there are many of such :)
> > >
> > > More exactly, from the driver perspective, there are no volume
> > > controls but only there are control elements with integer values.
> > > Do you mean "Capture Volume" control or which one?
>
> Hmm, ok, this needs to be more precise:
> In gamix (codec "HDA Intel : Realtek ALC268"), the Capture Volume control.
Yeah, that's more understandable :)
BTW, does "Capture Volume" influence on the recording level even for
the built-in mic, right? I'm asking this because the digital mic on
STAC/IDT codecs isn't controlled via "Capture Volume" control that is
bound to an ADC widget. (That's why "Digital Capture Volume" control
exists. It's a value used by alsa-lib softvol plugin for "default"
PCM.)
> > > And, is the behavior consistent regardless of the value high, i.e.
> > > the key is only whether the values for both channels are identical?
> >
> > BTW, what if you record with the following definition?
> > Put the below to ~/.asoundrc
> >
> > pcm.imix {
> > type plug
> > slave.pcm "hw"
> > ttable.0.0 0.5
> > ttable.0.1 -0.5
> > }
> >
> > and record like
> >
> > % aplay -Dimix -c1 foo.wav
>
> Does NOT exhibit the "equal sliders == no sound" bug (apart from this sliders
> are acting normally, i.e. slider low == no sound), despite being a
> "plug" type definition (this is what you wanted to discern, right? ;).
Interesting. This implies that one channel is inverted indeed.
As default the alsa-lib plugin downmixes a stereo stream to a mono
stream simply by left/2 + right/2. The above changes the routing
policy as left/2 - right/2.
So we need to pass some information to change this kind of thing...
But a question still remains; why conversion with sox worked.
Maybe it didn't mix? Or, the code alsa-lib could be buggy...
A simple test would be to just sum all 16bit samples in a stereo
stream file externally. That is, first record a RAW file via
% arecord -Dhw -traw -fdat foo.dat
Then create a mono stream just do 16bit left/2 + right/2 calculation
by any way (a good homework for kids :). Is it also problematic?
thanks,
Takashi
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