More information about the 'missing sound' on the 'other' (not left
front/right front) channels:
I attempted to make the audio input channels go to various output
channels, and even that doesn't work correctly. They can only be
routed to the first two (LF/RF) channels (no others). As an assist t
Mark Knecht wrote:
> Hi. Does anyone else experience very slow responses from Alsa-Devel? I sent
> this email this morning at 9:16AM this morning and it just arrived back here
> and at my home account at 1:30PM. Most other reflectors I'm on normally have
> round trip times of just a minute or two.
At Wed, 10 Sep 2003 23:26:37 -0700 (PDT),
Tom Watson wrote:
>
> [1 ]
> More information about the 'missing sound' on the 'other' (not left
> front/right front) channels:
the attachech patch will fix this problem.
Takashi
revo-fix2.dif
Description: Binary data
At Thu, 11 Sep 2003 08:40:30 +0200,
Denis Sbragion wrote:
>
> Hello Takashi,
>
> At 19.59 10/09/2003 +0200, you wrote:
> ...
> > > 2) I use 'alsamixer'. Works fine. One problem: When I set the
> > > conversion rate to the SPDIF clock (it has another name IEC...) IT
> > > LOCKS AND STAYS THERE.
Hello Takashi,
At 19.59 10/09/2003 +0200, you wrote:
...
> 2) I use 'alsamixer'. Works fine. One problem: When I set the
> conversion rate to the SPDIF clock (it has another name IEC...) IT
> LOCKS AND STAYS THERE. No amount of diddling in 'alsamixer' yields
> anything. To get around this I r
Arve Knudsen wrote:
> What's the purpose of set_start_threshold?
The stream will automatically start when the buffer has been filled up
to the amount specified by start_threshold.
HTH
Clemens
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Brix wrote:
> i need information about obtaining the correct rawmidi interface
> name (such as "hw:0,0,1"), corresponding to the MIDI synth device,
> from the system;
What do you mean with "the MIDI synth device"? There could be more
than one, or none. Usually, you let the user choose the device.
Paul,
I'm pretty sure that I was not the intended audience for this level of
technical data, but it does raise the question (for me) as to whether this
problem might be created by using the 'wrong' (I.e. - undocumented
requirements) kernel?
The machine exhibiting the problem is running a pre
> > At lunch today I rewired my MIDI input from the keyboard
> controller to use
> > the MidiSport input instead of the HDSP 9652 input. In this
> configuration,
> > with only about 15 minutes of testing, I had no problems. I will do more
> > testing this evening, but possibly this data says it's
Paul Davis wrote:
> >When an interrupt for MIDI input occurs, the hdsp driver disables all
> >further MIDI interrupts until the current input data has been read.
>
> the hdsp driver is one of the first alsa drivers to use tasklets to
> handle MIDI without impacting audio latency (that is, we can co
Paul Davis wrote:
> >When an interrupt for MIDI input occurs, the hdsp driver disables all
> >further MIDI interrupts until the current input data has been read.
... and midi_input_trigger flushes the input FIFO if the interrupt bit
isn't set.
Unfortunately, this does not seem to be the bug we're
On Thu, 11 Sep 2003 14:12:08 +0200 (METDST)
Clemens Ladisch <[EMAIL PROTECTED]> wrote:
> What do you mean with "the MIDI synth device"? There could be more
> than one, or none. Usually, you let the user choose the device.
>
> As default name, you could use "default", which is "hw:0,0" by
> defaul
>
> AFAIK tasklets have higher priority than any user thread.
>
> From the error description I diagnose that either MIDI interrupts or
> the MIDI tasklet are suppressed while jack is running. It shouldn't be
> possible to prevent a tasklet from running unless the CPU usage is
> 100%, so I think it
On Thu, 11 Sep 2003 14:17:59 +0200 (METDST), Clemens Ladisch
<[EMAIL PROTECTED]> wrote:
Arve Knudsen wrote:
What's the purpose of set_start_threshold?
The stream will automatically start when the buffer has been filled up
to the amount specified by start_threshold.
Ok, obviously there's something
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?
What this value means varies depending on the sound card hardware being
used.
I think that this
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.
I've joined the following files :
- kernel strace through ksymoops
- /proc/asound/card0/ac97
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