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    SubjectRe: [alsa-devel] Bugs on aspire one A150
    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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