On Fri, 12 Sep 2003 16:51:26 +0200 (METDST), Clemens Ladisch
<[EMAIL PROTECTED]> wrote:
Arve Knudsen wrote:
I did a little test where I set start threshold to buffer size,
acquire memory using snd_pcm_mmap_begin (number of returned frames
equals buffer size), fill the buffer and do a snd_pcm_mmap
If one is outputing 5.1 audio, and one wishes the adjust the volume, one
currently has to adjust 4 sliders. (Front stereo or Master/PCM, Surround
stereo, Center, Lfe.).
I think that is would be really good if the Master volume control was
abstracted. So, for example, mixer controls changed to: -
Hi,
does anybody have more complete documentation of the ALSA control/mixer
interface. I am constantly fighting with the interface as many functions have
no Doxygen documentation.
All I can do is guessing from alsalib sources and alsamixer code. Any help is
greatly appreciated. I am especially
Takashi Iwai <[EMAIL PROTECTED]> writes:
> > regardince M-Audio buggy support, i got a strange bug report about
> > snd-ice1724 in alsa-drivers-0.9.6 on kernel 2.4.2x, see mdk bug #5084:
> > http://qa.mandrakesoft.com/show_bug.cgi?id=5084
>
> it's ice1712.
>
> and, i still suspect it's a mixer c
>When an interrupt for MIDI input occurs, the hdsp driver disables all
>further MIDI interrupts until the current input data has been read. I
>don't know why it does this, but you may try to remove/disable lines
>3181, 3182, 3188, and 3189 in hdsp.c.
the hdsp driver is one of the first alsa driver
At Fri, 12 Sep 2003 16:44:38 +0100,
James Courtier-Dutton wrote:
>
> Takashi Iwai wrote:
> > At Fri, 12 Sep 2003 13:46:39 +0100,
> > James Courtier-Dutton wrote:
> >>
> >>Zero as a minimum value is not very meaningfull in audio terms. With a
> >>mixer, the zero dB point is probably more use. The
Takashi Iwai wrote:
At Fri, 12 Sep 2003 13:46:39 +0100,
James Courtier-Dutton wrote:
Zero as a minimum value is not very meaningfull in audio terms. With a
mixer, the zero dB point is probably more use. The minimum value of a
slider should be the equivalent of mute, so I think that instead of a
Brix wrote:
> I'm working on an ear trainer using the rawmidi interface.
>
> Because i would let this application used by newbies, i need a
> method to obtain (from files in /proc/asound, i think) the name of
> a working MIDI Synth device.
It isn't possible to automatically determine which of the
Arve Knudsen wrote:
> I did a little test where I set start threshold to buffer size,
> acquire memory using snd_pcm_mmap_begin (number of returned frames
> equals buffer size), fill the buffer and do a snd_pcm_mmap_commit.
> After the commit I check the state of the card, which is
> SND_PCM_STATE_
At Fri, 12 Sep 2003 07:00:11 -0700 (PDT),
Tom Watson wrote:
>
> [1 ]
> Well, I finally got this beast working in time for my demo tomorrow.
> Turns out that the AK4355 codec has not one, not two, but THREE reset
> bits in its registers. They ALL need to be 1 if you want the beast to
> function.
At Fri, 12 Sep 2003 13:46:39 +0100,
James Courtier-Dutton wrote:
>
> Jaroslav Kysela wrote:
> > On Fri, 12 Sep 2003, James Courtier-Dutton wrote:
> >
> >
> >>There is a MAJOR problem with the current alsa mixer.
> >>How do I set a mixer volume to the 0 db point. I.E. no attenuation, and
> >>no g
Well, I finally got this beast working in time for my demo tomorrow.
Turns out that the AK4355 codec has not one, not two, but THREE reset
bits in its registers. They ALL need to be 1 if you want the beast to
function. In my persuit of all of this I added some code to the
'ice1724.c' to print ou
Jaroslav Kysela wrote:
On Fri, 12 Sep 2003, James Courtier-Dutton wrote:
There is a MAJOR problem with the current alsa mixer.
How do I set a mixer volume to the 0 db point. I.E. no attenuation, and
no gain.
Currently, we might have a value from 0 to 100%. Percent of what?
Percent of the volume
On Fri, 12 Sep 2003, Takashi Iwai wrote:
> At Fri, 12 Sep 2003 12:31:32 +0200 (CEST),
> Jaroslav wrote:
> >
> > On Fri, 12 Sep 2003, Takashi Iwai wrote:
> >
> > > > I have plans to extend the mixer interface to pass (and handle) the dB
> > > > resolution. But it will be done completely in the user
On Fri, 12 Sep 2003 10:52:22 +0200
Takashi Iwai <[EMAIL PROTECTED]> wrote:
> are you sure aplay segfaults with the same oops?
Sorry, I haven't checked, I'll tell you in a few hours.
> the oops in resample_expand() was known in the ealier 2.6.0-test
> kernels when it's build without frame pointer
At Fri, 12 Sep 2003 12:31:32 +0200 (CEST),
Jaroslav wrote:
>
> On Fri, 12 Sep 2003, Takashi Iwai wrote:
>
> > > I have plans to extend the mixer interface to pass (and handle) the dB
> > > resolution. But it will be done completely in the user space. The alsa-lib
> > > will analyze information fr
At Fri, 12 Sep 2003 09:23:13 +0200 (CEST),
Jaroslav wrote:
>
> On Fri, 12 Sep 2003, James Courtier-Dutton wrote:
>
> > There is a MAJOR problem with the current alsa mixer.
> > How do I set a mixer volume to the 0 db point. I.E. no attenuation, and
> > no gain.
> > Currently, we might have a valu
At Fri, 12 Sep 2003 02:58:15 +0200,
Olivier Blin wrote:
>
> Hi,
>
> With snd-intel8x0 driver from alsa 0.9.6 and a Sis7012 chipset, some
> sound applications cause kernel oops. For example, aplay segfaults
> with some files (KDE_Startup.wav) and SDL_OpenAudio() segfaults.
are you sure aplay segf
On Fri, 12 Sep 2003, James Courtier-Dutton wrote:
> There is a MAJOR problem with the current alsa mixer.
> How do I set a mixer volume to the 0 db point. I.E. no attenuation, and
> no gain.
> Currently, we might have a value from 0 to 100%. Percent of what?
Percent of the volume range.
> What t
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