Re: [Alsa-devel] Opening dev/dsp takes very long
> It's related to the snd-ice1712 driver and hardware which has the cs8427 > transciever. I've added cs8427_timeout module option, so you can I have M-Audio Audiophile and suffer from this also, but I'm not sure if it has cs8427. 2.4.24 with ll+pe alsa 0.9.8 If I'm not totally wrong, the earlier 0.9 series didn't suffer from this, but it's long time ago so can't remember. Tommi --- This SF.Net email is sponsored by: IBM Linux Tutorials Free Linux tutorial presented by Daniel Robbins, President and CEO of GenToo technologies. Learn everything from fundamentals to system administration.http://ads.osdn.com/?ad_id=1470&alloc_id=3638&op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Re: Again with the Delta 1010LT...
> I'm struggling to sync two Delta 1010LT cards. I've tried both the word
> clock and SPDIF approaches, without any success. Both the SPDIF Out and
> Word Clock Out connectors show no activity when inspected by an
> oscilloscope.
Have you tried to connect card 1's spdif out to 2's input and set 2's
clock to spdif in.
I think only way to get anything properly from spdif in is to clock the
card from the spdif in also.
However, sometimes my Audiophile card receivec stuff from spdif even it's
not clocked from it, but then I get pops and cracks on leftf channel, and
sometimes I hear nothing. But both issues disappear when clocking from
spdif.
> - Is the word clock output something you must switch on in the ALSA
> driver or is it supposed to start automatically when you have a valid
> clock signal on the card?
Don't know.
> - Do I have to do something special to get SPDIF output (I'm new to
No you don't. My .asoundrc has
pcm.spdif_test
{
type plug
ttable.0.8 1
ttable.1.9 1
slave.pcm {
type hw
card 0
device 0
}
}
and I can play & capture using this 'spdif_test' device. This is for
Audiophile, but since they use same chip, this should work for you.
> this)? I've fiddled with iecset from alsa-utils, and otherwise I
> connected the SPDIF Out to various inputs using the patchbay of
> envy24control. Still, no signal appears on the connector.
>From one card? You can't connect card's outputs to any inputs with
anvy24control within the same card. JACK can do that, but it's a different
story.
You can connect inputs to outputs directly with the envy24control and it's
done in hardware.
Tommi
ps. One more thing, make sure you use the right connectors when connecting
physically the spdif inputs and outputs.
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Re: [Alsa-devel] Multiple Delta 1010LT woes: UPDATE
> Are you in possesion of multiple cards at the moment? If so, have you > tested any multitrack recording of analog inputs from both cards > simultaneously yet? I have 1 1010LT now, and am planning on a second > one within the month, to create a 16 track live recording studio. This I have successfully used Delta 1010LT and Delta Audiophile 2496 (2 ana + 2 digi channels) together to have 10 analog channels (1010LT has 8 analog, right?). I synced the cards through the S/PDIF since 2496 doesn't have word clock i/o. But I had to reset alsa drivers after every capture, since I experienced sometimes some driver lockups. I think the spdif code had some bug and it was pathced later, but I didn't have the recording setup anymore to test if that patch helped. For recording I used ecasound. Also tried to hack around with .asoundrc and JACK, but no luck. This was about 6 months ago, things might be different now. Tommi Uimonen --- SF.Net is sponsored by: Speed Start Your Linux Apps Now. Build and deploy apps & Web services for Linux with a free DVD software kit from IBM. Click Now! http://ads.osdn.com/?ad_id=1356&alloc_id=3438&op=click ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] ice1712,how to use all 10 channels
Hi. If I understood correctly ice1712 is capable of hardware mixing 10 channels. How do I access these different channels from eg. aplay Like sblive can do: aplay -D front aplay -D rear simultaneously, without blocking each other (or can it?) I would need access to Audiophile 2496's analog & digital channels separately (channels 0,1 for analog; 8,9 for digital) without blocking. And I think the driver can handle this, but how Joe User can accomplish this without C++ knowledge. Or without .asoundrc knowledge.. Thanks! Tommi Uimonen ps. I know this can be done with JACK, but I'm not looking for that. --- The SF.Net email is sponsored by EclipseCon 2004 Premiere Conference on Open Tools Development and Integration See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. http://www.eclipsecon.org/osdn ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Re: ice1712 driver broken in 2.6.1-mm kernels
> > The one problem is I still can't play different wave files
> > simultaneously through the two outputs.
> >
> > If I start "aplay -D maudio_spdif 12.wav &" and then try to do "aplay -D
> > plug:maudio_analog 11.wav" at the same time, the second one just blocks
> > until the first one is done.
> >
> > This is unfortunate, since I'd really like to be able to run two-room
> > sound from my computer into different amplifiers and speakers. I'm sure
> > there's a way to do it, since the hardware is capable of it...
Now after thinking this, I think ecasound is the right tool for this.
Create a device to .asoundrc: (with correct card number of course)
pcm.2496_4ch
{
type plug
ttable.0.0 1
ttable.1.1 1
ttable.2.8 1
ttable.3.9 1
slave.pcm {
type hw
card 0
device 0
}
}
Then use ecasound with this device, and take input from two sources and
send them to the appropriate channels, first input to channels 1&2, second
to channels 3&4.
I don't know the right way to do this, but I know that it can be done.
Since ecasound can split 4 channel wav to separate channels, I don't see
why it wouldn't join two stereo samples to one 4 channel sample.
Maybe you'll have to ask this from ecasound-list
http://www.eca.cx/mlists.html
Tommi
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Re: [Alsa-devel] Re: ice1712 driver broken in 2.6.1-mm kernels
> > It should be -D maudio_spdif, so lose the plug: and it will work. I think > > the analog should also work without plug. > > Thanks! That did it, mostly... I now can play sound through the SPDIF! > > The analog doesn't work without "plug", I get an error "aplay: > set_params:827: Sample format non available". I think it's because the > 2496 driver only does 24bit, so 16bit pcm requires translation. Yes, of course. > But that's not a problem - plug works fine. > > The one problem is I still can't play different wave files > simultaneously through the two outputs. > > If I start "aplay -D maudio_spdif 12.wav &" and then try to do "aplay -D > plug:maudio_analog 11.wav" at the same time, the second one just blocks > until the first one is done. > > This is unfortunate, since I'd really like to be able to run two-room > sound from my computer into different amplifiers and speakers. I'm sure > there's a way to do it, since the hardware is capable of it... Well, some kind of dmix approach might do it. app1 >-- --> analog \/ dmix /\ app2 >-- --> spdif I don't know how to do it. I think this kind of plug device would be popular among 5.1 owners, where you could define what input goes to what output. Something like: -D dmix_rear -D dmix_front etc.. But how to separate them from the mix? Anyway, dmix allows multiple instances to use the soundcard, so it's a start. Maybe the plugins can be linked so that output of one goes to input of the other. I guess this can be done, but I'm not any .asoundrc magician. Or ttable & bindings... http://www.alsa-project.org/alsa-doc/doc-php/asoundrc.php3 Tommi --- The SF.Net email is sponsored by EclipseCon 2004 Premiere Conference on Open Tools Development and Integration See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. http://www.eclipsecon.org/osdn ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Re: ice1712 driver broken in 2.6.1-mm kernels
> Thanks for the tip. I tried your suggestion, but it doesn't work for
> me. My .asoundrc now has this in it:
>
> pcm.maudio_analog {
> type hw
> card 1
> device 0
> }
>
> pcm.maudio_spdif {
> type plug
> ttable.0.8 1
> ttable.1.9 1
> slave.pcm {
> type hw
> card 1
> device 0
> }
> }
>
> I can play sound fine through the analog channel:
> [EMAIL PROTECTED] thoffman]$ aplay -D plug:maudio_analog 12.wav
> Playing WAVE '12.wav' : Signed 16 bit Little Endian, Rate 44100 Hz,
> Stereo
>
> But when I try to use the spdif channel, I get an error:
>
> [EMAIL PROTECTED] thoffman]$ aplay -D plug:maudio_spdif 12.wav
> Playing WAVE '12.wav' : Signed 16 bit Little Endian, Rate 44100 Hz,
> Stereo
> aplay: pcm_write:1083: write error: Invalid argument
It should be -D maudio_spdif, so lose the plug: and it will work. I think
the analog should also work without plug.
Tommi
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Re: [Alsa-devel] Re: ice1712 driver broken in 2.6.1-mm kernels
> Just one problem remains to be solved for this card: The Audiophile
> 24/96 is a 4-in, 4-out card (stereo analog input and output, plus
> independent SPDIF input and output). But only one set of inputs and
> outputs is showing up for use:
>
> [EMAIL PROTECTED] thoffman]$ cat /proc/asound/pcm
> 01-00: ICE1712 multi : ICE1712 multi : playback 1 : capture 1
>
> Shouldn't there be separate devices for the analog & SPDIF inputs and
> outputs? If not, how do I send different signals to the analog &
> digital outputs?
You have to create .asoundrc to your home directory.
for spdif, put:
pcm.2496_spdif
{
type plug
ttable.0.8 1
ttable.1.9 1
slave.pcm {
type hw
card 1
device 0
}
}
(If I remember correctly, your card was no. 1, if not, change 'card 1')
This device can be used like this:
aplay -D 2496_spdif foo.wav
arecord -D 2496_spdif -f dat bar.wav
For combined analog & digital performance (4 channels):
pcm.2496_4ch
{
type plug
ttable.0.0 1
ttable.1.1 1
ttable.2.8 1
ttable.3.9 1
slave.pcm {
type hw
card 1
device 0
}
}
then record 4 channels:
arecord -D 2496_4ch -c 4 -f cd foobar.wav
I'm not sure how you can easily send different signals to analog & digital
outputs. I guess if you have 4 channel wav, the first two channels will be
played to analog and 3&4 to digital output.
Tommi Uimonen
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[Alsa-devel] Simple mixer API progress?
Since 1.0 is out now, what is planned with the simple mixer API? Is there any drafts anywhere and release schedule etc..? Tommi Uimonen --- The SF.Net email is sponsored by EclipseCon 2004 Premiere Conference on Open Tools Development and Integration See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. http://www.eclipsecon.org/osdn ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] how to write a mixer using alsa lib api ?
> and I want to know where I find some documnent to know > how to use these alsa lib api,namely,I want a > specification of these api,thanks a lots . http://www.alsa-project.org/alsa-doc/alsa-lib/ Tommi --- This SF.net email is sponsored by OSDN developer relations Here's your chance to show off your extensive product knowledge We want to know what you know. Tell us and you have a chance to win $100 http://www.zoomerang.com/survey.zgi?HRPT1X3RYQNC5V4MLNSV3E54 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] simple mixer API?
Hi. If I remember correct, a while ago there were some discussion about a simple mixer API for ALSA. What is the current status of this? With simple mixer API I mean some basic mixer commands that all audio cards would perform, like volume up/down, mute, maybe others? Maybe these simple commands could be 'mapped' to driver specific commands in some config file. And it would be up to the user/administrator to configure how these simple commands will be handled. Eg. 'vol up' = 'amixer [command to rise volume]' Tommi Uimonen --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] ALSA Version 0.9.7
> 2) Even nicer: How about a bunch of notes that highlight the changes > since the last. These could be posted on the alsa-project web site and > be helpful to all. Hopefully we won't need the entire CVS diff, but a > summary would be nice. I agree. Many times I have been longing for a changelog. It would be best if it was on alsa web page, so it wouldn't be neccessary to download the whole package just to see the changelog (but of course it should be also in the package) Yesterday I downloaded 0.9.7 just to see if there was any changelog, but didn't find it. Of course I might install it someday but since I don't know the benefits of the new version I'll postpone the installation. Tommi Uimonen --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] EWX 24/96 and SPDIF: solved? (with patch)
> With this patch my EWX works flawless with ALSA 0.9.7, anyway I can't > check if it is going to cause problems with other ICE1712 based > soundcards. I hope this change is the right one, and not something that > makes just my soundcard work. With 0.9.6 I experienced ice1712 driver hangups when I used two M-Audio cards connected together through SPDIF and the other card synced it's clock from SPDIF (Audiophile 2496 and 1010LT). Sometimes in envy24control the vu-meters freezed, and I needed to do /etc/init.d/alsa restart to get the cards to respond again. Sometimes the problem went away without restart when I tried to change the clock rate from envy24control, but the card didn't always respond, sometimes the clock changed fine and then the vu meters worked again. This, I think, caused also ecasound problems; it couldn't stop the recording and after killing the process I found out that it had not recorded anything even thougn the engine started fine. I quess the drivers were halted just after ecasound engine started, otherwise it might have reported some errors. (Btw Kai, if you read this, maybe ecasound could have some control ability to be sure that audio really get's written to harddisk. In my case it just wrote the wav headers.) I'm not sure if I used the cards right; modules.conf had settings for both cards, but when alsa started it reported the cards like this: Starting ALSA sound driver (version 0.9.6): ice1712 (ice1712) so I'm not sure if the other card was properly set up. (I assume that the parenthesis means that driver has some problems. I get (via82xx) if the motherboard AC97 is disabled from bios, and without parenthesis when it is enabled) But still I was able to record from both cards simultaneously when they were not freezed, and the sample clocks were synced fine. Tommi Uimonen ps. Kernel was 2.4.20 with LL patches. --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] arecord to support >2GB files?
> >> I'm planning to use arecord for serious harddisk recording, but it can't > > that's a non-starter :) at the very least, you'd better make sure you > always use a period size larger than the maximum scheduling delay you > think you might encounter, since arecord writes to disk from the same > thread as one audio I/O is done from. Yes it doesn't sound sexy at all. But I tested arecord to perform nicely with 96khz 24channels 32bit data, on a ext2 filesystem (I found ext2 to be the best filesystem for this kind of writing). As I'm going to record max 12 channels, this seemed like a reasonable solution. After all, I didn't know ecasound can handle 96khz and I didn't have very promising results with ardour&jack, but the bottleneck was almost certainly my full ide-hd at the end of the disk (physically also at the end) and reiserFS, which causes overhaul to the process due journaling (I think, not 100% sure about this). > ecasound would be a much better idea, much, much better. I think I'll give ardour an another try and fall back to ecasound, which seems to be the right tool for now. Tommi Uimonen --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] arecord to support >2GB files?
> You can almost certainly just add > -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE > to the CFLAGS in the Makefile. (it would be good if this was default on > 32bit platforms). Thanks. This did the trick. Tommi --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] arecord to support >2GB files?
I guess the CVS doesn't update very rapidly. Since I last checked (5 mins ago) aplay.c was dated Aug 7) > Ecasound has support for LFS (compile with --with-largefile). Is there also --with-96khz option? :) If there is, or there will be soon (in two days), I'll change to that. 88200 would do fine also... And about GUI's for ecasound, are there such that won't affect the recording - I mean that the recording would be as "xrun free" as running ecasound in console? Tommi Uimonen --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] arecord to support >2GB files?
I'm planning to use arecord for serious harddisk recording, but it can't write files bigger than 2BG. After browsing through kernel mailing lists about LFS I concluded that the problem is arecord, not the underlying filesystem (and since I've had >8GB files on it :). I looked at the code, but didn't have any idea what to change. Tommi Uimonen --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] two soundcards as one, for recording?
Hi. I have M-Audio Delta 1010LT and Audiophile 2496 cards, running 2.6.0-test4. I tested the asoundrc magic described in http://article.gmane.org/gmane.linux.alsa.devel/1946/match=asoundrc+soundcards but without any success. Got the following error: arecord: set_params:787: Broken configuration for this PCM: no configurations available I can record from both (not at the same time of course) with 'arecord -D default:0 -c 8 [...]' (for 1010LT) and 'arecord -D default:1 -c 2 [...]' (for 2496), so the cards and their configuration work ok. I was hoping to get 10 channel recording done with these two cards. They sync together through SPDIF, so no drifting should occur. And if possible, get 12 channels by adding the free SPDIF input as one stereo source to the combination. (1010LT has 8 analog inputs, 2496 has two) The solution would be if arecord worked with JACK, but since I can't find any information on that, I'll have to live without. Any ideas? Tommi Uimonen --- This SF.net email is sponsored by: VM Ware With VMware you can run multiple operating systems on a single machine. WITHOUT REBOOTING! Mix Linux / Windows / Novell virtual machines at the same time. Free trial click here:http://www.vmware.com/wl/offer/358/0 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
