16 Feb
2006
16 Feb
'06
7:10 a.m.
Hello all,
i'm want to use the asoundrc for my music production. I've read the docs under
http://alsa.opensrc.org/. While testing the funktionality, i came into the
first problem. when trying the examples. if i try
aplay test.wav
everything went fine. but after
aplay -D hw:0,0 test.wav
or
aplay -D hw:1,0 test.wav // this is the onboard soundcard
i get
Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono
aplay: set_params:901: Channels count non available
and no sound will be played
googling doesn't really enlightened me. any hints ?
sizu c~
BTW: is there a way to search the LAU archive. Haven't found that.
16 Feb
16 Feb
9:54 a.m.
conrad berh?rster wrote:
i'm want to use the asoundrc for my music
production.
Why? What do you want to do that does not work with the default
configuration?
if i try
aplay test.wav
everything went fine. but after
aplay -D hw:0,0 test.wav
i get
Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono
aplay: set_params:901: Channels count non available
"hw" goes directly to the hardware, without automatic sample format
conversion. Your hardware cannot play mono data.
HTH
Clemens
12:26 p.m.
Hallo Clemens,
Am Donnerstag 16 Februar 2006 15:53 schrieb Clemens Ladisch:
Well, how can i obtain a special channel. i thought hw:0,0 means first card,
first channel. is this a general misunderstanding.
thanks c~
conrad berh?rster wrote:
well, i have 2 soundcards and want to capture from both at the same
time with
jack. i don't need sample sync (or such bad things ;=) ) only need the
signal, each channel individual but into one app, i have written.
i'm want to use the asoundrc for my music
production.
Why? What do you want to do that does not work with the default
configuration? if i try
aplay test.wav
everything went fine. but after
aplay -D hw:0,0 test.wav
i get
Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz,
Mono aplay: set_params:901: Channels count non available
"hw" goes directly to the hardware, without automatic sample format
conversion. Your hardware cannot play mono data.
HTH
Clemens
4:15 p.m.
New subject: [linux-audio-user] Re: asoundrc and aplay
2006/2/16, conrad berhörster <conrad.berhoerster(a)gmx.de>de>:
well, i have 2 soundcards and want to capture from
both at the same time
with
jack. i don't need sample sync (or such bad things ;=) ) only need the
signal, each channel individual but into one app, i have written.
Two explain Pauls Jedi-magic:
If you use jack, you want sample-sync. If you don't want sample-sync
and just want to record two independent streams into your software
(assuming you don't want to use it for music[*]) you just have to open
both sounddevices the alsa way and process according to the
alsa-interface.
[*] If you want to do music with it, you want sample-sync...
Enough things have been said about way two soundcards without syncing
can't be used as one multichannel device, so I won't repeat it here.
I just want to add another trick I learned from my hardware-dealer who
syncs ISDN-cards: Maybe its not enough to just couple their clocks,
they also change the driver to mask the IRQ's from the slave cards and
just react on the master-IRQ and then the driver automaticly
proccesses the data on all cards.
Speaking in the audio-speach of el-cheapo-multichannel: It's not
enough to solder the crystals together, there may be a need to modifiy
the driver. Of course this would make a new soundcard out of it which
exposes its multi-channels to jack directly through the
alsa-interface, no need for .asoundrc-fiddling...
I haven't tested that but I think if it works for ISDN-cards it should
work with soundcards too.
Arnold
--
visit http://dillenburg.dyndns.org/~arnold/
---
Wenn man mit Raubkopien Bands wie Brosis oder Britney Spears wirklich
verhindern könnte, würde ich mir noch heute einen Stapel Brenner und
einen Sack Rohlinge kaufen.
5:42 p.m.
New subject: [linux-audio-user] Re: asoundrc and aplay
On Thu, 2006-02-16 at 22:14 +0100, Arnold Krille wrote:
I just want to add another trick I learned from my
hardware-dealer who
syncs ISDN-cards: Maybe its not enough to just couple their clocks,
they also change the driver to mask the IRQ's from the slave cards and
just react on the master-IRQ and then the driver automaticly
proccesses the data on all cards.
thats a very cool idea. i wish i had the time to add that to the RME
hammerfall/hdsp/madi drivers because they can all have IRQ's disabled or
enabled. you could then load the driver with an option to disable IRQs
for all but the first card. nice.
--p
6:23 p.m.
New subject: [linux-audio-user] Re: asoundrc and aplay
On Thu, 16 Feb 2006 17:45:56 -0500
Paul Davis <paul(a)linuxaudiosystems.com> wrote:
On Thu, 2006-02-16 at 22:14 +0100, Arnold Krille
wrote:
One would need to ensure that the start/stop time of the cards is right
in sync in the first place no? After that, if properly sample synced
everything is fine. I don't know how the hw works, but i imagine it has
a "pointer" into a sample buffer where it puts each new sampled value
(and another buffer with another "pointer" where it reads from, etc pp).
These "pointers" need to have an offset as small as possible between two
cards, ideally it would be 0 (So that in a sense, if both cards still
generated interrupts, they would happen at the exact same moment).
I always wondered about this part of ALSA. How do you start/stop
soundcards at the exact same moment? Does stuff like the multi or
whatwasitsname ALSA pcm plugin care about this?
Flo
--
Palimm Palimm!
http://tapas.affenbande.org
I just want to add another trick I learned from
my hardware-dealer who
syncs ISDN-cards: Maybe its not enough to just couple their clocks,
they also change the driver to mask the IRQ's from the slave cards and
just react on the master-IRQ and then the driver automaticly
proccesses the data on all cards.
thats a very cool idea. i wish i had the time to add that to the RME
hammerfall/hdsp/madi drivers because they can all have IRQ's disabled or
enabled. you could then load the driver with an option to disable IRQs
for all but the first card. nice. 
17 Feb
17 Feb
4:22 a.m.
New subject: [linux-audio-user] Re: asoundrc and aplay
Hi
Can't we link the two Hammer falls with a link cable (I seem to have one
here)
I think it uses the adat 2 input.
Or am I on the wrong track here (no pun intended!)
cheers
Bob
Paul Davis wrote:
On Thu, 2006-02-16 at 22:14 +0100, Arnold Krille wrote:
I just want to add another trick I learned from my
hardware-dealer who
syncs ISDN-cards: Maybe its not enough to just couple their clocks,
they also change the driver to mask the IRQ's from the slave cards and
just react on the master-IRQ and then the driver automaticly
proccesses the data on all cards.
thats a very cool idea. i wish i had the time to add that to the RME
hammerfall/hdsp/madi drivers because they can all have IRQ's disabled or
enabled. you could then load the driver with an option to disable IRQs
for all but the first card. nice.
--p
16 Feb
16 Feb
7:54 p.m.
New subject: [linux-audio-user] Re: asoundrc and aplay
Hello Arnold,
Am Donnerstag 16 Februar 2006 22:14 schrieb Arnold Krille:
2006/2/16, conrad berhörster
<conrad.berhoerster(a)gmx.de>de>:
Thanks very much, Paul was a little bit to
"short" ;=)
well, i have 2 soundcards and want to capture
from both at the same time
with
jack. i don't need sample sync (or such bad things ;=) ) only need the
signal, each channel individual but into one app, i have written.
Two explain Pauls Jedi-magic: If you use jack, you want sample-sync. If you
don't want sample-sync
and just want to record two independent streams into your software
(assuming you don't want to use it for music[*]) you just have to open
both sounddevices the alsa way and process according to the
alsa-interface.
No, this is not quite right for me. sample-sync is not the only
reason, why
someone uses jack. i use it i my app, because
- easy programming interface , better than alsa.
- take some other apps and get/set data from/to them
i need that inputs not in a harddisk recording / pro audio way. all i need is
getting the signal into the app, use some effects and put them out again. no
sample sync, no need for "realtime". all inputs are independed.
Than i have read http://alsa.opensrc.org/TwoCardsAsOne.
is this page wrong!
i thought, one big advantage of jack is, to handle more than one card.
[*] If you want to do music with it, you want sample-sync...
Enough things have been said about way two soundcards without syncing
can't be used as one multichannel device, so I won't repeat it here.
yes, i
know. i have read them too. maybe you can explain, what you mean
exactly with the notion "multichannel device". Is it just a couple of IOs,
without sync, or are the requirements for pro audios (with sync).
if all this stuff isn't possible, do you have any idea to fix the problem.
thanks c~
20 Feb
20 Feb
7:48 a.m.
New subject: [linux-audio-user] Re: asoundrc and aplay
While most of your questions are already answered...
2006/2/17, conrad berhörster <conrad.berhoerster(a)gmx.de>de>:
Well "one multichannel-device" means one device as seen by the
software and exposed by the driver which can have any number of more
the two channels. Well, strictly it is more than one channel, so even
a stereo-device is already a multichannel device...
This is done in hardware by either on big'n'good AD/DA-unit which does
multiplexing of the IO's or by multiple AD/DA-units that are
sample-synced by a common clock. [1]
The problem as already explained a lot of times is to _really_ sync
several devices, which hw-manufactures solve in hardware on the
chip/board but is very hard / impossible to do in software.
Enough things
have been said about way two soundcards without syncing
can't be used as one multichannel device, so I won't repeat it here.
yes,
i know. i have read them too. maybe you can explain, what you mean
exactly with the notion "multichannel device". Is it just a couple of IOs,
without sync, or are the requirements for pro audios (with sync). if all this stuff isn't possible, do you have any
idea to fix the problem.
- learn the alsa-interface and use some jitter-buffers
- try to run several jackd (one for each hardware-device you want to
use) and connect your app with all of them (again you have to use
jitter-buffers)
- get a device with the desired number of inputs and outputs (easiest
but expensive solution)
Enough said, I should work now,
Arnold
[1] While I believe that most soundcards will use one fast ADC (for
input) and multiplex the incomming signals, I am working here with
some devices that reach a samplerat of 4GHz by multiplexing the one
input-signal to four 1Ghz ADC's. Quite the opposite but really works
:-)
--
visit http://dillenburg.dyndns.org/~arnold/
---
Wenn man mit Raubkopien Bands wie Brosis oder Britney Spears wirklich
verhindern könnte, würde ich mir noch heute einen Stapel Brenner und
einen Sack Rohlinge kaufen.
17 Feb
17 Feb
6:46 p.m.
New subject: [linux-audio-user] Re: asoundrc and aplay
On Thu, 2006-02-16 at 22:14 +0100, Arnold Krille wrote:
I just want to add another trick I learned from my
hardware-dealer who
syncs ISDN-cards: Maybe its not enough to just couple their clocks,
they also change the driver to mask the IRQ's from the slave cards and
just react on the master-IRQ and then the driver automaticly
proccesses the data on all cards.
Speaking in the audio-speach of el-cheapo-multichannel: It's not
enough to solder the crystals together, there may be a need to modifiy
the driver. Of course this would make a new soundcard out of it which
exposes its multi-channels to jack directly through the
alsa-interface, no need for .asoundrc-fiddling...
I haven't tested that but I think if it works for ISDN-cards it should
work with soundcards too.
Sounds good, if your time has no value.
Why not just buy a multichannel card, they are cheap these days?!?
Lee
20 Feb
20 Feb
7:40 a.m.
New subject: [linux-audio-user] Re: asoundrc and aplay
2006/2/18, Lee Revell <rlrevell(a)joe-job.com>om>:
On Thu, 2006-02-16 at 22:14 +0100, Arnold Krille
wrote:
<snipping my own blabla>
Sounds good, if your time has no value.
Most times you have either time and no money or money and no time. :-/
Why not just buy a multichannel card, they are cheap
these days?!?
Well, I would buy a bigger interface anytime (if I had the money and
the need (currently I am happy with my sblive and your
multichannel-patches:-) )).
But still it seems cheaper to buy three 25€ cards and do some
soldering than take on a credit to buy a 200€-pro-audio-card. I know
the quality is never the same, but...
Arnold
--
visit http://dillenburg.dyndns.org/~arnold/
---
Wenn man mit Raubkopien Bands wie Brosis oder Britney Spears wirklich
verhindern könnte, würde ich mir noch heute einen Stapel Brenner und
einen Sack Rohlinge kaufen.
16 Feb
16 Feb
7:57 p.m.
Am Donnerstag 16 Februar 2006 15:53 schrieb Clemens Ladisch:
conrad berh?rster wrote:
buuuuuttttt: the default of aplay
is hw:0,0 . on my first card, the first
subdevice is an mono channel. so, what puzzles me is the fact, that
aplay test.wav is ok but aplay -D hw:0,0 test.wav gives me this error.
Does this mean, that aplay with the -D option first reads all the data and
only send the raw data to the subdevice.
i'm want to use the asoundrc for my music
production.
Why? What do you want to do that does not work with the default
configuration?
if i try
aplay test.wav
everything went fine. but after
aplay -D hw:0,0 test.wav
i get
Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz,
Mono aplay: set_params:901: Channels count non available
"hw" goes directly to the hardware, without automatic sample format
conversion. Your hardware cannot play mono data.
HTH
what means HTH ;=)
Clemens
17 Feb
17 Feb
9:21 a.m.
conrad berh?rster wrote:
Hope this helps.
Clemens
Am Donnerstag 16 Februar 2006 15:53 schrieb Clemens
Ladisch:
The default device is not "hw:0,0" but "default" which gets eventually
mapped to "plug:hw:0,0".
"hw" goes directly to the hardware,
without automatic sample format
conversion. Your hardware cannot play mono data.
buuuuuttttt: the default of aplay is hw:0,0 . on my first card, the first subdevice is an mono
channel.
The subdevice number would be the third number (as in hw:0,0,0).
The first device on your first card has exactly one subdevice which has
exactly two channels.
A card has more than one device if it has several output/inputs that can
be used _independently_ (i.e., if the hardware can play from different
streams). For example, the analog and SPDIF outputs of Intel compatible
AC'97 controllers are available as hw:0,0 and hw:0,4.
A device has more than one subdevice if the hardware can mix several
streams for the same output.
HTH
what means HTH ;=)
6:47 p.m.
On Fri, 2006-02-17 at 01:54 +0100, conrad berhörster wrote:
buuuuuttttt: the default of aplay is hw:0,0 . on my
first card, the
first
subdevice is an mono channel. so, what puzzles me is the fact, that
aplay test.wav is ok but aplay -D hw:0,0 test.wav gives me this
error.
Wrong, the default of aplay is "default".
Lee
16 Feb
16 Feb
12:05 p.m.
New subject: [linux-audio-user] Re: asoundrc and aplay
Hi Conrad,
I've only seen .asoundrc used so far for the dmix plugin, and it always
relied on the 'default' output device.
Please also keep in mind asoundrc is considered 'consumer level' by many
who use Jack.
googling doesn't really enlightened me. any hints
?
Maybe he can enlighten you :) http://www.wakeuplaughing.com/
BTW: is there a way to search the LAU archive.
Haven't found that.
Try this: http://search.gmane.org/
Carlo
6998
days inactive
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days old
linux-audio-user@lists.linuxaudio.org
16 comments
9 participants
participants (9)
-
Arnold Krille -
Carlo Capocasa -
Clemens Ladisch -
conrad berhörster -
Florian Schmidt -
fsmith -
Lee Revell -
Paul Davis -
Reuben Martin