[alsa-devel] Bugs on aspire one A150

[Takashi Iwai]

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