[alsa-devel] [RFC PATCH] Inverted internal mic

Takashi Iwai tiwai at suse.de
Tue Feb 28 11:38:09 CET 2012


At Tue, 28 Feb 2012 10:54:00 +0100,
David Henningsson wrote:
> 
> On 02/28/2012 10:24 AM, Takashi Iwai wrote:
> > At Tue, 28 Feb 2012 09:57:56 +0100,
> > David Henningsson wrote:
> >>
> >> Hi,
> >>
> >> One of the more common problems on laptop machines, is that the internal
> >> mic provides a stereo signal but with one channel phase inverted, or
> >> differential output.
> >>
> >> This means problems for applications recording two channels but later
> >> merging them into one, leaving them with zero or near-zero output.
> >>
> >> There are various ways we can work around this in both the kernel,
> >> alsa-lib and PulseAudio layers. It's a matter of picking the right one.
> >> I'm leaning towards trying to fix it in the kernel's codec drivers, because
> >> 1) we already have quirking infrastructure there
> >> 2) we already have some working quirks already in that layer
> >> 3) it would benefit other sound applications that use ALSA directly.
> >>
> >> The downside to that is really that we're silencing out one channel for
> >> everyone, leading to no application being able to use both channels,
> >> even if they would implement some kind of
> >> "auto-detect-and-reverse-one-channel" functionality.
> >>
> >> For the most part, this has been some Acer Aspire machines running
> >> Realtek ALC268 / ALC269 / ALC271X / ALC272X, and for two of these we
> >> have proc coefs we can set, but for the other two these proc coefs, if
> >> they exist, are unknown to us.
> 
> A follow-up question here would be if it is easy for you (or Kailang?) 
> to figure this out, or if we should resort to the suggest workaround for 
> Realtek ALC268 and ALC272X as well?

Actually, Kailang (Cc'ed now) and I talked about it when I visited
Taipei in the last year, and he mentioned that there is no way to
detect the FM chip from the codec.  It seems that SSID listing is the
only way.

> >> Recently I came across a Conexant as well, and I decided to write a
> >> patch for it, that would take the approach that the internal mic is
> >> forced mono on the kcontrol, and make sure the right channel is always
> >> muted. The patch is verified by the reporter to fix this problem. It
> >> could use some perfection though - it would make sense to to the same to
> >> the internal mic boost as well, and the strcmp('Internal Mic') call
> >> could maybe be turned into something more elegant. But before going
> >> ahead with that, I'd like to hear your opinion on the matter, if you
> >> agree that this is a good approach to the problem?
> >
> > The remaining problem in the patch is that the signal is recorded only
> > on left when you record in stereo.  I guess the reporter tested only a
> > program like Skype, which uses only a mono stream.
> >
> > If this behavior, only one channel in a stereo stream, is acceptable,
> > it's a reasonable fix, I think.
> 
> That is indeed a disadvantage, but I don't know if we have something 
> better to offer? E g, we could try an alsa-lib ttable solution, but that 
> would then have a problem with knowing whether the Internal Mic is 
> currently selected or not.

Note that in alsa-lib, the HD-audio "default" is already set up to
copy left-channel for mono streams.  You can see a line setting
"route_policy" to "copy" in HDA-Intel.conf.

Thus, when ALSA apps run without PA, it'd work in both stereo and
mono.

> If the program:
> 
> - tries to record in mono:
>    * As the industry standard for mapping mono is to map it to the left 
> channel, I assume this is what ALSA/HDA does in this case as well, so 
> unchanged and working behaviour
> 
> - record in stereo, then take left channel only:
>    * unchanged and working behaviour
> 
> - record in stereo, then take right channel only:
>    * regression as the inverted channel will now be zero, but this would 
> be highly unconventional and unusual, I assume?
> 
> - record in stereo, then merge both channels:
>    * this is the normal and common case for VoIP applications going 
> through PulseAudio (Skype etc). Improvement, this will fix the bug; 
> although the result will be -6 dB compared to the original signal.
> 
> - record in stereo, keep signal in stereo for playback:
>    * this somewhat depends on how the playback is set up, but I would 
> say that it would be an improvement in most cases if the speakers are 
> relatively close to one another...I guess?

Well, I think this could be also worked around in PA.  Can PA switch
the downmixing to the mono to a simple left-channel copy, only for
certain devices, or at least via a config?

If this is feasible, the rest is how to identify such a device.
And this can be done equally well in user-space, too, as there is no
hidden information about it.

Of course, this is not the best of the best.  Ideally, the driver
provides multiple PCM streams and PA switches between them and
switches the downmixing simultaneously per input-source change.
But this needs a lot of works, and maybe requires a fundamental
infrastructure change, too.


thanks,

Takashi


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