Re: Bugs on aspire one A150

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Hi,
On Mon, Mar 16, 2009 at 06:15:39PM +0100, Takashi Iwai wrote:> 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.)
Yes, Capture Volume does influence i-Mic level.The Digital Capture control, however, doesn't influence level.As doesn't the Mic Boost Capture control (probably about e-Mic only?).
> > > > 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.
Oh, you mean "inverted" as in "_hardware_ channel which provides oppositesample values as compared to the other channel"?
> 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.
That exactly matches my current stream of thought (while reading"one channel is inverted" above).
> So we need to pass some information to change this kind of thing...
That's something specific to ALC268 codec setup, right?("ALC268 digital mic == left plus right channel, but inverted setup"?)
> 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?
OK, I know what you're up to, I'll do this external proof ASAP,will take a couple more minutes.
Andreas_______________________________________________Alsa-devel mailing listAlsa-devel@xxxxxxxxxxxxxxxxxxxx://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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