On 12-06-08 05:52, Rene Herman wrote:
Oh, by the way, upon rereading:
On 11-06-08 17:41, Dominique Michel wrote:
You must add some device definitions in /etc/modules.d/alsa (or
whatever file
your distribution is using):
## ALSA portion
alias snd-card-0 snd-...
options snd-... index=0
I don't think we are in disagreement in substance. I was trying to give a
larger framework for all of those not as familiar with the general workings
and decisions behind computer audio as currently implemented. Just works
is very important to most users. But isn't right for some of us.
Somewhere
On 12-06-08 08:28, Demian Martin wrote:
I don't think we are in disagreement in substance. I was trying to
give a larger framework for all of those not as familiar with the
general workings and decisions behind computer audio as currently
implemented. Just works is very important to most
Florian Winter wrote:
- What is the dmix plugin and what are the benefits of using it?
- Is it possible to disable the dmix plugin?
- What consequences does disabling the dmix plugin have? What essential
features of ALSA will be missing without it?
The dmix plugin allows multiple
Again thanks you for your answer. You have been of great help. Using
plughw instead of default seems to do the trick.
I have some more questions:
Is there a way to find out which soundcards support hardware mixing (and
have support for it implemented in their corresponding ALSA drivers)? Is
it
On 12-06-08 16:14, Florian Winter wrote:
I have some more questions:
Is there a way to find out which soundcards support hardware mixing (and
have support for it implemented in their corresponding ALSA drivers)?
Not all that easily from the code. The ALSA soundcard matrix at:
On 12-06-08 17:09, Rene Herman wrote:
On 12-06-08 16:14, Florian Winter wrote:
I have some more questions:
Is there a way to find out which soundcards support hardware mixing (and
have support for it implemented in their corresponding ALSA drivers)?
Not all that easily from the code.
dmix *is* an application, conceptually it is not different to esd, artsd
or pulseaudio opening the ALSA hardware directly. It just happens to be
an ALSA plugin and is a part of the signal chain that ALSA-lib sets up
by default.
You can either:
a) configure dmix using a custom .asoundrc to
Hola Jochen and all,
yes, of course it is a trade-off between xruns and delay, but i do
that adaptively as well - start with a quite low framing, measure the
drop-out rate and reopen the soundcard in case of too much drop-outs.
this only impacts the quality of the start-up phase and
On Monday 09 June 2008, Bill Unruh wrote:
That will at most give you digitization noise which at 16 bit is 96dB
below full signal. Ie, it is much less than the tape hiss from a tape
recorder for example.
Not that tape hiss should be a standard we compare everything to :)
A little peeve with
Chris,
A little peeve with some so called pro audio servers is their
inability to act as a 'digital wire', ie: what goes in comes out,
totally unchanged. As an example, there are times you may want the
same exact 16 bits you send out of app to arrive at the audio device
unmolested. Jack,
On Thursday 12 June 2008, Grant wrote:
I'm trying to get music from mpd to my USB DAC in 100% untouched
form. My mpd.conf is as follows:
audio_output {
type alsa
name USB Monica
device hw:0,0
format 44100:16:2
}
Don't know anything about mpd but you can set up an asoundrc plug in
alsa
On Thursday 12 June 2008, Florian Faber wrote:
What makes you think converting a 16 bit unsigned integer to a IEEE
32 bit float and back would change the value?
Should have used a 24 bit example. I'm of the opinion that with it the
process is not always a bit perfect translation. But I'm open
Hello,
I've been receiving some Australian help on getting flite working on the
H2210 ipaq. (thanks for that!)
But it's not 100% yet on the ipaq:
It appears that mono audio goes to the null device.
Stereo audio goes to the default device.
flite generates mono audio.
When I use the mono to
On Thu, 2008-06-12 at 14:10 -0400, Chris Smith wrote:
On Thursday 12 June 2008, Florian Faber wrote:
What makes you think converting a 16 bit unsigned integer to a IEEE
32 bit float and back would change the value?
Should have used a 24 bit example. I'm of the opinion that with it the
On Thursday 12 June 2008 20:10:04 Chris Smith wrote:
On Thursday 12 June 2008, Florian Faber wrote:
What makes you think converting a 16 bit unsigned integer to a IEEE
32 bit float and back would change the value?
Should have used a 24 bit example. I'm of the opinion that with it
the
On Thursday 12 June 2008, Florian Faber wrote:
On IEEE 32 bit floats the mantissa is 23 bit, so there might be
situations where you loose the LSB.
And that was the only point - a pro audio chain should be able to
support digital wire capability.
And as long as it doesn't support the sample
Chris,
On IEEE 32 bit floats the mantissa is 23 bit, so there might be
situations where you loose the LSB.
And that was the only point - a pro audio chain should be able to
support digital wire capability.
This has nothing to do with the original poster's issues, so I changed
the subject.
hi all
i am not quite sure, if this is the right place to ask, but from what i
heard, alsa is capable of doing resampling, when needed (e.g when
playing a 44.1 file on a card, that natively only runs at 48k).
however, i seem not to be able to use those capabilites, when running
jackd. no matter
On Thursday 12 June 2008, Florian Faber wrote:
So, please tell me - how should a 'pro audio chain' look like?
I'm not saying one should never work with floats - I'm sure there's a
very good reason for it, but the chain should still be able to
support digital wire capability if desired. You may
actually, my problem is not jackd related at all. i cannot use a
samplerate other than 48k with any application i tried, not only with
jackd. isn't alsa supposed to provide resampling, if necessary?
i forgot to mention the hardware:
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97
On Thu, Jun 12, 2008 at 4:30 PM, Roman Haefeli [EMAIL PROTECTED] wrote:
actually, my problem is not jackd related at all. i cannot use a
samplerate other than 48k with any application i tried, not only with
jackd. isn't alsa supposed to provide resampling, if necessary?
i forgot to mention
On Thu, 2008-06-12 at 17:24 -0700, Mark Knecht wrote:
On Thu, Jun 12, 2008 at 4:30 PM, Roman Haefeli [EMAIL PROTECTED] wrote:
actually, my problem is not jackd related at all. i cannot use a
samplerate other than 48k with any application i tried, not only with
jackd. isn't alsa supposed to
On Thu, Jun 12, 2008 at 6:05 PM, Roman Haefeli [EMAIL PROTECTED] wrote:
SNIP
are you saying, that it is simply not possible to run jackd over an alsa
plugin, that does resampling or mixing? if so, does that mean, that
Yes, I am saying that. Jack communicates directly with the hardware.
Jack
hi mark
thanks a lot for the detailed eplanation.
On Thu, 2008-06-12 at 18:45 -0700, Mark Knecht wrote:
Technically, I think you're looking for a Jack aware resampling
plugin. you would send your 44.1K sound file to that device and then
let it resample it to the Jack sample rate - 48K in
On Thu, Jun 12, 2008 at 7:00 PM, Roman Haefeli [EMAIL PROTECTED] wrote:
hi mark
thanks a lot for the detailed eplanation.
On Thu, 2008-06-12 at 18:45 -0700, Mark Knecht wrote:
Technically, I think you're looking for a Jack aware resampling
plugin. you would send your 44.1K sound file to
Hi all,
I'm running debian etch with 2.6.18-6-686 kernel.
I have a asus p5kc mother board
lspci give:
00:1b.0 Audio device: Intel Corporation Unknown device 293e (rev 02)
Subsystem: ASUSTeK Computer Inc. Unknown device 829f
Flags: bus master, fast devsel, latency 0, IRQ 15
I'm trying to get music from mpd to my USB DAC in 100% untouched
form. My mpd.conf is as follows:
audio_output {
type alsa
name USB Monica
device hw:0,0
format 44100:16:2
}
Don't know anything about mpd but you can set up an asoundrc plug in
alsa that will fix the rate from alsa to the
On Friday 13 June 2008, Grant wrote:
pcm.my_device
Sorry, thought it would be understood that 'my_device' is the alsa alias
(in my case the the name of the kernel module without the
leading snd-) for the hardware device in question (see your
modules.conf file or whatever is proper for your
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