On Thu, Sep 21, 2006 at 02:09:30PM +0200, Tomas Carnecky wrote:
> If we're at it, can I snd_pcm_mmap_begin()/commit() more than 4096
> bytes? It seems as at it's possible since wine's alsa driver does that
> on the whole buffer, which can be more than 4096 bytes.
Certainly it can. My soundcard
On Wed, Jul 26, 2006 at 07:41:54AM -0700, Bill Unruh wrote:
> What makes you think that there is "aliasing"?
Clearly visible in the spectograms referred to in the original
message.
--
FA
Lascia la spina, cogli la rosa.
-
On Wed, Jul 26, 2006 at 02:04:24PM +0100, James Courtier-Dutton wrote:
> Alfons Adriaensen wrote:
> >
> >To summarize, aliasing is the result of the _sampled_ nature of a
> >digital signal, not of its numerical (digital) form.
> >
> >
>
> That i
On Wed, Jul 26, 2006 at 12:02:31PM +0200, Thomas Hood wrote:
> On 7/26/06, Alfons Adriaensen <[EMAIL PROTECTED]> wrote:
> > It could, even if the digital form of the signal is OK. Your card
> > may apply digital gain at the max setting (i.e. to allow you to
> > boost
On Wed, Jul 26, 2006 at 08:35:34AM +0200, Thomas Hood wrote:
> > Did you check the levels of your soundcard's mixer ? The spectrum
> > shows a signal that's rather badly distorted...
> >
>
> The levels in alsamixer are all set to 100 -- can this cause clipping,
It could, even if the digital form
On Tue, Jul 25, 2006 at 02:54:33PM +0100, Thomas Hood wrote:
> When msplaysweep.wav is played back under linux it sounds wrong -- as if it's
> aliasing (?) -- but perfect when played back under windows on the same
> machine. It sounds bad regardless of whether I use alsa (alsaplayer
> -oalsa / apl
On Tue, Apr 18, 2006 at 01:57:57PM +0200, Tobias Urban wrote:
> I have already tried the snd_pcm_drain() function but this did not work with
> my capture stream. Now I am trying a direct MMAP access - call hesync,
> receive pointer, begin, read sample data, commit. However, the official ALSA
> exa
On Thu, Feb 09, 2006 at 09:10:10AM -0500, Paul Fox wrote:
> > No idea, but I'm quite sure that most of the 'feautures' of
> > this device are not provided by the HW, but by the Windows
> > driver...
>
> do you have specific experience that suggests this to you? are you
> saying these devices
On Wed, Feb 08, 2006 at 11:06:55AM -0500, Paul Fox wrote:
> you're right, i forgot about that. i'm fairly prehistoric in my
> needs: i'd like to hear my music, in stereo :-), and i'd like
> the microphone input to work. (i'll hopefully be using that to
> drive lircd -- not for real audio input
On Wed, Feb 08, 2006 at 10:29:01AM -0500, Paul Fox wrote:
> can anyone recommend a readily available USB audio device that's
> known to work well with alsa? i've tried searching forums and
> such, but it's hard to get much information. in my experience,
> that either means that they all work well
On Mon, Jan 30, 2006 at 08:30:42PM -0500, Lee Revell wrote:
> Well you could try just upgrading the kernel and leaving userspace
> alone, that would be trivial to revert, and would rule out a problem at
> the kernel USB layer...
That could well be the next step.
--
FA
--
On Tue, Jan 31, 2006 at 09:17:45AM +0100, Clemens Ladisch wrote:
> You could try 1.0.11rc3, but your problem doesn't seem to be USB
> related.
>
> (You could try your internal sound card, just to be sure.)
No problem at all with that one (intel8x0).
> > > > - Exactly every 40 seconds something
Hello,
This weekend I installed SL10.0 on my Thinkpad. Everything runs fine
except for USB audio (EDIROL UA-2). I had it working perfectly with
-p 256 -n 3 on SL9.2, but now even -p 1024 gives problems, in two
ways. All this with jack 0.99.61 (same one as I used on 9.2 - it's
newer than the versio
On Wed, Feb 04, 2004 at 08:21:44AM -0800, Bill Unruh wrote:
> That "right thing" increases the noise level of the card and the
> distorition of the card. Just as resampling in time does it, so does
> resampling in amplitude. Now I do not know how the volumes on teh cards
> do things, but if they d
On Thu, Jan 08, 2004 at 03:39:07PM +0100, Jaroslav Kysela wrote:
> Basically, yes. We are able to mmap the DMA ring buffer to more
> applications at one time and mangle the stream parameters to satisfy
> user's requests. The nice thing is that the mangling is done on the user
> level, so the kerne
On Thu, Jan 08, 2004 at 03:04:03PM +0100, Jaroslav Kysela wrote:
> Oops. My fault. For capture is dsnoop plugin, of course (same syntax) the
> dhare plugin is for playback only.
>
> We have the asym plugin in CVS now, so you can combine dshare and dsnoop
> plugins for the full-duplex operation
On Thu, Jan 08, 2004 at 02:34:50PM +0100, Jaroslav Kysela wrote:
> You need to use the dshare plugin:
>
> pcm.in23 {
> type dshare
> ipc_key 321456 # any unique value
> ipc_key_add_uid true
> slave {
> pcm "hw:0,0"
> periods 0
Hello list,
I heard rumours of a new pcm API in the latest releases of ALSA.
Searched on the ALSA site for 'new api', nothing found.
Is there any document that describes this new API and the rationale
for it ?
TNX
--
Fons
---
This SF.ne
On Mon, Oct 20, 2003 at 09:23:26PM +0200, Frank Barknecht wrote:
> > Is there actually any documentation available about all this, apart from
> > the very cryptic 'asoundrc.txt' which is hidden deeply in the ALSA docs ???
>
> Yes, http://www.alsa-project.org/alsa-doc/alsa-lib/
> especially http:/
On Mon, Oct 20, 2003 at 04:44:29PM +0200, Frank Barknecht wrote:
> Jaroslav, you made my day! Thanks a lot. OSSdmix is working now on the Audiophile
> with the supplied asoundrc, yippee.
>
> ... lots of interesting stuff about dmix etc.
Things I see for the first time, and do not understand:
i
20 matches
Mail list logo
