On Mon, 31 May 2004, Roc Wu wrote:
> Hello alls:
>
> Sorry for post the mail again. I posted it several
> days ago, but no response. Anybody can give me some
> hints?
>
> I cross compiled the alsa-lib-1.04 to arm platform.
> The 2.6.6 kernel including alsa driver is ok on our
> ARM board, so I w
Hello alls:
Sorry for post the mail again. I posted it several
days ago, but no response. Anybody can give me some
hints?
I cross compiled the alsa-lib-1.04 to arm platform.
The 2.6.6 kernel including alsa driver is ok on our
ARM board, so I want to test the driver. how to do it?
I think maybe t
Hello alls:
I cross compiled the alsa-lib-1.04 to arm platform.
The 2.6.6 kernel including alsa driver is ok on our
ARM board, so I want to test the driver. how to do it?
I think maybe the first step is to crosscompile the
alsa utils to play a wave audio file.
CC=arm-linux-gcc ./configure --host
On Sat, 6 Mar 2004, Jaroslav Kysela wrote:
>> On the Echoaudio cards the sample rate is a global setting, so I need a way
>> to prevent an app to change the rate if someone else already set it (the
>> card han several devices/subdevices). I was thinking to put a simple
>> "if (IsAudioOpen()) rate
Clemens Ladisch wrote:
Caleb Crome wrote:
Clemens Ladisch wrote:
However, Realplayer uses the OSS emulation through /dev/dsp0. To make
that work with your "file" device, add the following to the config
file:
pcm.dsp0 = file
Awesome! It works like a charm. Than you so much! The o
Caleb Crome wrote:
> Clemens Ladisch wrote:
> >However, Realplayer uses the OSS emulation through /dev/dsp0. To make
> >that work with your "file" device, add the following to the config
> >file:
> >
> > pcm.dsp0 = file
>
> Awesome! It works like a charm. Than you so much! The one thing that
>
Clemens Ladisch wrote:
Caleb Crome wrote:
pcm.file {
type file
slave {
pcm "hw:0,0"
}
file "/tmp/file.raw"
}
Alright, I did what you suggested, and real player comes up and connects
to the dummy sound card (I had to do a modprobe snd-pcm-oss and
snd-mixer-oss). However, I
Clemens Ladisch wrote:
ns wrote:
I have WDM driver for our lab's own propertiary sound
card(ADAT+SPDIF+DB...) and wanna write ALSA drv.
I have linux w/kernel 2.4.20 (original).
I know that I must patch kernel for ALSA and write ALSA driver.
HOw to do it fastest?
look into alsa-kernel/Documentati
ns wrote:
> I have WDM driver for our lab's own propertiary sound
> card(ADAT+SPDIF+DB...) and wanna write ALSA drv.
> I have linux w/kernel 2.4.20 (original).
>
> I know that I must patch kernel for ALSA and write ALSA driver.
> HOw to do it fastest?
look into alsa-kernel/Documentation/DocBook/wr
Caleb Crome wrote:
> >>>pcm.file {
> >>> type file
> >>> slave {
> >>> pcm "hw:0,0"
> >>> }
> >>> file "/tmp/file.raw"
> >>>}
>
> Alright, I did what you suggested, and real player comes up and connects
> to the dummy sound card (I had to do a modprobe snd-pcm-oss and
> snd-mixer-oss)
Hello alsa-devel,
HI all!
I have WDM driver for our lab's own propertiary sound
card(ADAT+SPDIF+DB...) and wanna write ALSA drv.
I have linux w/kernel 2.4.20 (original). I have patch it to RTLinux
which got work OK recently.
I know that I must patch kernel for ALSA and write ALSA driver. HOw to
Caleb Crome wrote:
Alright, I did what you suggested, and real player comes up and
connects to the dummy sound card (I had to do a modprobe snd-pcm-oss
and snd-mixer-oss). However, I don't get a file called
/tmp/file.raw. Is there any way to check how the data is routed?
Do I have to do any
Caleb Crome wrote:
> Arek Korbik wrote:
> >pcm.file {
> >type file
> >slave {
> >pcm "hw:0,0"
> >}
> >file "/tmp/file.raw"
> >}
>
> what do I do with that bit of configuration data? I found the
> /usr/share/alsa/alsa.conf, and there is a similar pcm.file entry
> in that, bu
At Thu, 22 Apr 2004 14:24:30 -0700,
Caleb Crome wrote:
>
> Arek,
> That looks like what I want. However, I'm sorry to be such an
> ignoramus -- what do I do with that bit of configuration data? I found
> the /usr/share/alsa/alsa.conf, and there is a similar pcm.file entry in
> that, but i
Arek,
That looks like what I want. However, I'm sorry to be such an
ignoramus -- what do I do with that bit of configuration data? I found
the /usr/share/alsa/alsa.conf, and there is a similar pcm.file entry in
that, but it seems to take parameters. Since I have no sound card, how
do I co
On Thu, 2004-04-22 at 14:44, Caleb Crome wrote:
> I'd like to create a dummy audio device that can record the PCM data
> written to it into a file. So, the dummy device would appear as a sound
> card to the audio program (Real Player for example), and start dumping
How about 'file' type plugin
StreamRipper: Close, but no cigar. That only works for MP3 streams,
not for other streams, such as RealPlayer. I have looked at
streamripper before and it doesn't do what I want as far as I can tell.
In the case of RealPlayer, the audio goes from Real->PCM->MP3, which I
admit isn't great fo
>1) Open my Player, tune to my favorite station. Use my new device for
>audio output.
>2) do something like: dd if=/dev/audiocapturepcmout count=xxx | lame
>--output myfile.mp3
streamripper is a much better option for this. its specially designed
for this task. you are also talking about mp3->P
Hello,
I've looked through the archives and I couldn't find what I'm
looking for, so here goes:
I'd like to create a dummy audio device that can record the PCM data
written to it into a file. So, the dummy device would appear as a sound
card to the audio program (Real Player for example),
On 06-Mar-2004 Adam Tla/lka wrote:
>> > On the Echoaudio cards the sample rate is a global setting, so I need a way
>> > to prevent an app to change the rate if someone else already set it (the
>> > card han several devices/subdevices). I was thinking to put a simple
>> > "if (IsAudioOpen()) rate
On Sat, 6 Mar 2004, Adam Tla/lka wrote:
> On Sat, Mar 06, 2004 at 11:59:45AM +0100, Jaroslav Kysela wrote:
> > On Sat, 6 Mar 2004, Giuliano Pochini wrote:
> >
> > > On the Echoaudio cards the sample rate is a global setting, so I need a way
> > > to prevent an app to change the rate if someone el
On Sat, Mar 06, 2004 at 11:59:45AM +0100, Jaroslav Kysela wrote:
> On Sat, 6 Mar 2004, Giuliano Pochini wrote:
>
> > On the Echoaudio cards the sample rate is a global setting, so I need a way
> > to prevent an app to change the rate if someone else already set it (the
> > card han several devices
On Sat, 6 Mar 2004, Giuliano Pochini wrote:
> On the Echoaudio cards the sample rate is a global setting, so I need a way
> to prevent an app to change the rate if someone else already set it (the
> card han several devices/subdevices). I was thinking to put a simple
> "if (IsAudioOpen()) rate_min
On the Echoaudio cards the sample rate is a global setting, so I need a way
to prevent an app to change the rate if someone else already set it (the
card han several devices/subdevices). I was thinking to put a simple
"if (IsAudioOpen()) rate_min=rate_max=current_rate;" in pcm_open callback,
but
Bauke Jan Douma wrote:
> I have two soundcards in my system, a cmipci(card) and
> an intel8x0 (on board).
>
> I have the audio cable of my CD/ROM drive wired onto the
> cmipci. Attached to line-out of the cmipci is just my
> headphones.
>
> I also have a TV-card.
> An audio cable runs from the TV-
[Sorry about those earlier mails; something went wrong with
permissions on my mail-editor; the editor failed, but mail
was still sent out to some bogus (and some not so bogus)
addresses]
Hi,
I have two soundcards in my system, a cmipci(card) and
an intel8x0 (on board).
cmicpi is /dev/dsp0, inte
Hi,
I have two soundcards in my system, a cmipci(card) and
an intel8x0 (on board).
cmicpi is /dev/dsp0, intel8x0 is /dev/dsp1.
In .asoundrc I have:
pcm.cmipci { type hw ; card 0 ; device 0 }
ctl.cmipci { type hw ; card 0 ; device 0 }
pcm.intel8x0 { type hw ; card 1 ; device 0 }
ctl.intel8x0 {
Hi,
I have two soundcards in my system, a cmipci(card) and
an intel8x0 (on board).
cmicpi is /dev/dsp0, intel8x0 is /dev/dsp1.
In .asoundrc I have:
pcm.cmipci { type hw ; card 0 ; device 0 }
ctl.cmipci { type hw ; card 0 ; device 0 }
pcm.intel8x0 { type hw ; card 1 ; device 0 }
ctl.intel8x0 {
Jiang Jiang (HangZhou) wrote:
> I find that I can use alsamixer to adjust the volumn of line
> input. But how can I set the volumn of line input by program?
See the source of the amixer utility.
> And how to capture data from line input?
See the source of the arecord utility.
> What's more, whe
I'm newbe to alsa driver. I find that I can use alsamixer to adjust the
volumn of line input. But how can I set the volumn of line input by program?
And how to capture data from line input?
What's more, where can I find the explaniation of every item which
alsamixer lists ?
Thanks if anyone
At Mon, 24 Nov 2003 00:01:53 +0100,
Christian Esken wrote:
>
> On Tuesday 21 October 2003 11:16, Tommi Sakari Uimonen wrote:
> > > 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 .
> >
> > htt
On Tuesday 21 October 2003 11:16, Tommi Sakari Uimonen wrote:
> > 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/
Nice hint. But your comment i
> 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
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This SF.net email is sponsored by OS
dear all,
I want to write a mixer using alsa lib api. Now I am
reading the example program alsamixer.c and amixer.c,
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 .
best wishes
Marco Guo
_
On Sat, 26 Jul 2003, Prince John wrote:
> Hi,
> I have a few queries on buffer pointer updation during
> playback operation with DMA transfer. I have gone
> through the documents, but some are not very clear.
> I have 64KB DMA buffer and 16KB period size. I'm using
> runtime->dma_addr as the DMA s
Hi,
I have a few queries on buffer pointer updation during
playback operation with DMA transfer. I have gone
through the documents, but some are not very clear.
I have 64KB DMA buffer and 16KB period size. I'm using
runtime->dma_addr as the DMA source (physical
address). Is this right? This address
On Tue, 2003-06-03 at 16:30, Florian Schmidt wrote:
> hi.. maybe i can add something.. here's a snippet from my .asoundrc
>
> pcm.!default {
> type hw
> card 0
> }
>
> ctl.!default {
> type hw
> card 0
> }
>
>
> this creates a "default" pcm device which can poi
On 03 Jun 2003 12:06:18 -0700
Mark Knecht <[EMAIL PROTECTED]> wrote:
> Jaroslav,
>This did not work for me. I made the edits to the .asoundrc file,
> tried aplay with and without the -D, then tried restarting alsa and
> tried aplay again. It didn't work.
hi.. maybe i can add something.. here
On Tue, 2003-06-03 at 11:58, Jaroslav Kysela wrote:
> >
> > Will this make device 1 default for all applications that require sound but
> > do not ask for anything specific in terms of interfaces?
>
> The applications should use 'default' device name in that case. You can
> override it, of cour
On Tue, 2003-06-03 at 11:36, Mark Knecht wrote:
> >
> > Add 'device 1' to {} section.
> >
> > Jaroslav
>
> Thanks Jaroslav!
>
> Will this make device 1 default for all applications that require sound but
> do not ask for anything specific in terms of i
On Tue, 3 Jun 2003, Mark Knecht wrote:
> > > .asoundrc from Alsa site:
> > >
> > > pcm.via82xx {
> > >type hw
> > >card 0
> > > }
> > >
> > > ctl.via82xx {
> > >type hw
> > >card 0
> > > }
> > >
> > > Maybe I need to i
> > .asoundrc from Alsa site:
> >
> > pcm.via82xx {
> >type hw
> >card 0
> > }
> >
> > ctl.via82xx {
> >type hw
> >card 0
> > }
> >
> > Maybe I need to include the ',1' interface somehow? How would I do that?
>
> Add 'd
> .asoundrc from Alsa site:
>
> pcm.via82xx {
>type hw
>card 0
> }
>
> ctl.via82xx {
>type hw
>card 0
> }
>
> Maybe I need to include the ',1' interface somehow? How would I do that?
>
> Thanks,
> Mark
Would this be co
On Tue, 3 Jun 2003, Mark Knecht wrote:
> Hi,
>Yesterday evening I found that I could use aplay -D hw:0,1 and get sound
> out of Alsa. This was cool, but I don't normally do anything with aplay.
> Other apps that use sound are still locking up hard.
>
>I thought the .asoundrc file was supp
Hi,
Yesterday evening I found that I could use aplay -D hw:0,1 and get sound
out of Alsa. This was cool, but I don't normally do anything with aplay.
Other apps that use sound are still locking up hard.
I thought the .asoundrc file was supposed to make this happen
automatically, but apparent
On Sat, 5 Apr 2003, Christian Esken wrote:
> Hi,
>
> my mixer (kmix) supports OSS and ALSA at the same time. This is good for a
> mixed-mode environment (e.g. if there is a TV card w/o ALSA support).
>
> Unfortunately this means that kmix finds some mixers twice.
> - One time the real thing (AL
>Hi,
>
>my mixer (kmix) supports OSS and ALSA at the same time. This is good for a
>mixed-mode environment (e.g. if there is a TV card w/o ALSA support).
>
>Unfortunately this means that kmix finds some mixers twice.
>- One time the real thing (ALSA)
>- One time the OSS-emulation
>
>This will conf
Hi,
my mixer (kmix) supports OSS and ALSA at the same time. This is good for a
mixed-mode environment (e.g. if there is a TV card w/o ALSA support).
Unfortunately this means that kmix finds some mixers twice.
- One time the real thing (ALSA)
- One time the OSS-emulation
This will confuse users.
On 02-Apr-2003 Takashi Iwai wrote:
>> Yes, otherwise snd_pcm_lib_malloc_pages() fails and hw_params
>> callback exits with an error.
>
> ok, then something overwrites the entry.
> how about to check runtime->dma_area at each callback?
Bug found. Alsaplayer calls hw_params two times with different
At Wed, 02 Apr 2003 17:18:13 +0200 (CEST),
Giuliano Pochini wrote:
>
>
> On 02-Apr-2003 Takashi Iwai wrote:
> >> > anyway, runtime->dma_area and runtime->dma_bytes MUST be filled
> >> > manually IFF you don't use snd_pcm_lib_malloc_pages().
> >> > (dma_area won't be needed if the mmap is not supp
On 02-Apr-2003 Takashi Iwai wrote:
>> > anyway, runtime->dma_area and runtime->dma_bytes MUST be filled
>> > manually IFF you don't use snd_pcm_lib_malloc_pages().
>> > (dma_area won't be needed if the mmap is not supported and copy
>> > callback is defined, though.)
>>
>> I use snd_pcm_lib_malloc
At Wed, 02 Apr 2003 12:55:04 +0200 (CEST),
Giuliano Pochini wrote:
>
>
> On 02-Apr-2003 Takashi Iwai wrote:
> > sorry, which fields do you mean exactly?
> > runtime->private_data and runtime->hw ?
>
> Yes.
>
> > anyway, runtime->dma_area and runtime->dma_bytes MUST be filled
> > manually IFF yo
On 02-Apr-2003 Takashi Iwai wrote:
> sorry, which fields do you mean exactly?
> runtime->private_data and runtime->hw ?
Yes.
> anyway, runtime->dma_area and runtime->dma_bytes MUST be filled
> manually IFF you don't use snd_pcm_lib_malloc_pages().
> (dma_area won't be needed if the mmap is not s
At Wed, 02 Apr 2003 12:33:18 +0200 (CEST),
Giuliano Pochini wrote:
>
>
> On 02-Apr-2003 Takashi Iwai wrote:
> >> I don't set .copy callback and ->dma_area should be 0 because the card
> >> uses a sg-list. #:(
> >
> > no, dma_area should not be zero. you shouldn't change the value, if
> > you use
On 02-Apr-2003 Takashi Iwai wrote:
>> I don't set .copy callback and ->dma_area should be 0 because the card
>> uses a sg-list. #:(
>
> no, dma_area should not be zero. you shouldn't change the value, if
> you use snd_pcm_lib_malloc_pages(). in the case of sg-buffer,
> dma_area will hold the vir
At 02 Apr 2003 00:17:49 +,
Giuliano Pochini wrote:
>
>
> > My driver do not work anymore with xmms (alsa-xmms plugin) and
> > alsaplayer. [...]
>
> I tried to force the alsaplayer plugin to set S16_LE format (data is BE
> on powerpc) and it do not spit that errors anymore, but it do not play
> My driver do not work anymore with xmms (alsa-xmms plugin) and
> alsaplayer. [...]
I tried to force the alsaplayer plugin to set S16_LE format (data is BE
on powerpc) and it do not spit that errors anymore, but it do not play
anything and in the logs alsa prints:
ALSA ../alsa-kernel/core/pcm_l
My driver do not work anymore with xmms (alsa-xmms plugin) and
alsaplayer. I have no idea when that happened because I always use
alsa-utils to make tests and I made a lot of changes since the last time
I tested xmms and ap. Alsaplayer writes this:
[EMAIL PROTECTED] Giu]$ alsaplayer
ALSA lib pcm
Jaroslav Kysela wrote:
On Fri, 21 Mar 2003, Pieter Palmers wrote:
Hi all,
I face the following problem: The card I'm writing a driver for (SAM9707
based) doesn't support any form of DMA transfer. It requires you to
transfer the PCM data through a 'rep outsw' like mechanism. It raises an
inter
At Fri, 21 Mar 2003 13:31:02 +0100 (CET),
Jaroslav wrote:
>
> On Fri, 21 Mar 2003, Pieter Palmers wrote:
>
> > Hi all,
> >
> > I face the following problem: The card I'm writing a driver for (SAM9707
> > based) doesn't support any form of DMA transfer. It requires you to
> > transfer the PCM dat
On Fri, 21 Mar 2003, Pieter Palmers wrote:
> Hi all,
>
> I face the following problem: The card I'm writing a driver for (SAM9707
> based) doesn't support any form of DMA transfer. It requires you to
> transfer the PCM data through a 'rep outsw' like mechanism. It raises an
> interrupt when it ne
Hi all,
I face the following problem: The card I'm writing a driver for (SAM9707
based) doesn't support any form of DMA transfer. It requires you to
transfer the PCM data through a 'rep outsw' like mechanism. It raises an
interrupt when it needs new data. The transfer flow is as follows:
1) al
On Thu, 6 Mar 2003, Ivica Bukvic wrote:
> Hi all,
>
> I have this small question:
>
> Is it possible (I presume it is) to alter mixer settings in Alsa by
> invoking some kind of a system call using shell (i.e. how the RME hdsp
> can have its stuff altered)?
Use amixer.
> If so, what is the ran
Hi all,
I have this small question:
Is it possible (I presume it is) to alter mixer settings in Alsa by
invoking some kind of a system call using shell (i.e. how the RME hdsp
can have its stuff altered)?
If so, what is the range of values that describe the loudest and softest
levels?
Finally, a
On Mon, Mar 03, 2003 at 09:21:24AM +0100, Giuliano Pochini wrote:
>
>
> What is alsa-kernel ? :))
>
alsa soundcard drivers, which are also part of the linux-2.5.x kernel tree
bye,
martin
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On 02-Mar-2003 Jaroslav Kysela wrote:
> On 2 Mar 2003, Giuliano Pochini wrote:
>
>> I wrote a very minimal driver that does nothing and I put it
>> in /pci directory. How do I compile it ?
>
> alsa-driver or alsa-kernel ? I recommend using alsa-driver directory for
> tests.
Driver. What is als
On 2 Mar 2003, Giuliano Pochini wrote:
> I wrote a very minimal driver that does nothing and I put it
> in /pci directory. How do I compile it ?
alsa-driver or alsa-kernel ? I recommend using alsa-driver directory for
tests.
1) update Makefile (see ALSA extra code section)
2) add module depende
I wrote a very minimal driver that does nothing and I put it
in /pci directory. How do I compile it ?
Bye.
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Welcome to geek heaven.
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On Thu, 9 Jan 2003, Gerard Janssen wrote:
> Hi Jaroslav,
>
> Many thanks for your reply. This realy solved my problem.
>
> I have been able to implement a new PCM-stream and send it to empty FXBUS
> registers. However, I could not find .asoundcr (I am using SUSE 8.1 with
> ALSA_rc6). Where is it
Hi Jaroslav,
Many thanks for your reply. This realy solved my problem.
I have been able to implement a new PCM-stream and send it to empty FXBUS
registers. However, I could not find .asoundcr (I am using SUSE 8.1 with
ALSA_rc6). Where is it? In stead, I adapted alsa.conf, emu10k1.conf and
added a
On Wed, 8 Jan 2003, Gerard Janssen wrote:
> Hi,
>
> By making slight changes in emufx.c, I could use all four spdif stereo
> channels of the sblive! that are present on the audio-ext connector. These
> spdif0..3 outputs can be accessed via: EXTOUT_TOSLINK_L,R ;
> EXTOUT_HEADPHONE_L,R ; EXTOUT_REA
Hi,
By making slight changes in emufx.c, I could use all four spdif stereo
channels of the sblive! that are present on the audio-ext connector. These
spdif0..3 outputs can be accessed via: EXTOUT_TOSLINK_L,R ;
EXTOUT_HEADPHONE_L,R ; EXTOUT_REAR_L,R and EXTOUT_CENTER,LFE. By re-routing
"spdif", "fr
I have noticed that some (most) low level drivers support multiple open,
that is a sound card which is already opened, can be opened again.
I noticed that also my rme96 driver supported this sort of, although it
probably does not feel very good about it. If the sound card is in ADAT
mode, it i
Hi,
At Mon, 23 Dec 2002 14:19:38 +0100,
Gerald Grabner wrote:
>
> Hi,
>
> I'm trying to set PPQ for the alsa sequencer queue but it doesn't
> work. Here is the piece of code:
>
>snd_seq_queue_tempo_set_tempo (queue_tempo, tempo) ;
>snd_seq_queue_tempo_set_ppq (queue_tempo, ppq ) ;
>
Hi,
I'm trying to set PPQ for the alsa sequencer queue but it doesn't
work. Here is the piece of code:
snd_seq_queue_tempo_set_tempo (queue_tempo, tempo) ;
snd_seq_queue_tempo_set_ppq (queue_tempo, ppq ) ;
snd_seq_set_queue_tempo (seq_handle, queue_id, queue_tempo) ;
It works for tempo,
Hi,
I want to play different wave file on two channel at the same time, that's
one wave file for left channel, and the other for right channel, can I
achieve this with the alsa lib on linux? Need I open the sound device twice?
Thx!
Walter
_
On Sat, Dec 21, 2002 at 08:02:19PM +0800, ??? ??? wrote:
> Hi,
> I want to play different wave file on two channel at the same time, that's
> one wave file for left channel, and the other for right channel, can I
> achieve this with the alsa lib on linux? Need I open the sound device twice?
> T
Hi,all.
I'm a newbie of ALSA. I'm learning how to program with alsa-lib, I found that mmap is
a better way to used then write/read(oh, is that right?).
But I found that there are few cocumentation reference to it. So, I want to ask how to
use mmap.
I think a good and simple example is good for
Hello,
I'm developping an application where I sometimes have to change the hwparams
(or swparams) of an opened pcm device (playback or capture). At first, I
thought the simpler way to do that was to close the stream and reopen it. But
with alsa-0.9.0beta10a, closing and re-opening the device caus
Hi,
In my attempt to address the four SPDIF output ports of the SBLive!, I
made the following changes in the the emufx.c code. I did not to use the
playback registers or SND_EMU10K1_PLAYBACK_CHANNELS (as far as possible)
since I would like to omit volume and tone control.
if (emu->fx8010.extout_
Hi,
Takashi wrote
>Gerard Janssen wrote:
>>
>> I tried to find the spdif output port addresses by changing the register
>> addresses of EXTOUT_TOSLINK_L/R in emu10k1.h to all 16 possible output
>> address pairs between (00 - 1f). To do this, I removed the extout_mask
>> check in emufx.c in the
Hi,
Some time ago, I posted a similar message, but I never got a reply.
The SBLive! card can support four SPDIF output channels, indicated as
SPDIF#0 - 3, which are located on the 12 pins or the 40 pins external
connector (depending on the type of card).
When I play a file to "spdif" or "iec958:
On Mon, 12 Aug 2002, Anders Torger wrote:
> I'd like to be able to get the minimum period size for a pcm device, so
> then I use snd_pcm_hw_params_get_period_size_min(), however, that (of
> course) needs a pcm as the first argument.
>
> In my application it would be good if it was possible to
I'd like to be able to get the minimum period size for a pcm device, so
then I use snd_pcm_hw_params_get_period_size_min(), however, that (of
course) needs a pcm as the first argument.
In my application it would be good if it was possible to get this
information, even if the device in questio
Hi,
I am trying to send multiple digital audio streams to different SPDIF
outputs of my soundcard (SBLive! Player 1024, CT4830, Rev. 6). This
soundcard has a 40 pins external audio connector with four SPDIF outputs
indicated as SPDIF#0 - 3.
When I use the "front" channel in ALSA, this signal is
Hi,
I would like to send multiple digital audio streams to different SPDIF
outputs of my soundcard (SBLive! Player 1024, CT4830, Rev. 6). This
soundcard has a 40 pins external audio connector with four SPDIF outputs
indicated as SPDIF#0 - 3.
When I use the "front" channel in ALSA, this signal i
Hi,
I would like to send multiple digital audio streams to different SPDIF
outputs of my soundcard (SBLive! Player 1024, CT4830, Rev. 6). This
soundcard has a 40 pins external audio connector with four SPDIF outputs
indicated as SPDIF#0 - 3.
When I use the "front" channel in ALSA, this signal i
>> If someone has hardware with 1400+ controls, than memory allocation issue
>> is out of question. I wrote that we can optimize current code - add some
>> hash tables for faster lookups for example.
>
>well, about 200kB is not so big nowadays :)
>
>i thought the hardware doesn't need allocate 1
At Tue, 7 May 2002 11:55:51 +0200 (CEST),
Jaroslav wrote:
>
> On Tue, 7 May 2002, Takashi Iwai wrote:
>
> > > Card specific code may solve all that easily.
> >
> > that's true.
> > but user may start alsamixer (or what else) and get huge amount of
> > bars (i thought), which should be avoided.
On Tue, 7 May 2002, Takashi Iwai wrote:
> > Card specific code may solve all that easily.
>
> that's true.
> but user may start alsamixer (or what else) and get huge amount of
> bars (i thought), which should be avoided.
We can avoid this in alsa-lib.
> > We need to separate in our minds the
At Mon, 06 May 2002 21:11:09 +0200,
Abramo wrote:
>
> Takashi Iwai wrote:
> >
> > so far, we have no way to distinguish the matrix elements from
> > others. imagine you implement all 1400+ elements as singletons.
> > what happens if you run alsamixer on that?
> > obviously showing all of them s
Takashi Iwai wrote:
>
> so far, we have no way to distinguish the matrix elements from
> others. imagine you implement all 1400+ elements as singletons.
> what happens if you run alsamixer on that?
> obviously showing all of them should be avoided.
Already now alsamixer does not show all the co
At Mon, 06 May 2002 14:19:14 +0200,
Abramo wrote:
>
> Takashi Iwai wrote:
> >
> > At Sun, 05 May 2002 21:32:12 -0400,
> > Paul Davis wrote:
> > >
> > > >On Sun, 5 May 2002, Paul Davis wrote:
> > > >
> > > >> >I suggest you to use the field index of struct sndrv_ctl_elem_id (one
> > > >> >byte pe
Takashi Iwai wrote:
>
> At Sun, 05 May 2002 21:32:12 -0400,
> Paul Davis wrote:
> >
> > >On Sun, 5 May 2002, Paul Davis wrote:
> > >
> > >> >I suggest you to use the field index of struct sndrv_ctl_elem_id (one
> > >> >byte per dimension).
> > >> >
> > >> >id.index = (source << 8) | destination;
At Sun, 05 May 2002 21:32:12 -0400,
Paul Davis wrote:
>
> >On Sun, 5 May 2002, Paul Davis wrote:
> >
> >> >I suggest you to use the field index of struct sndrv_ctl_elem_id (one
> >> >byte per dimension).
> >> >
> >> >id.index = (source << 8) | destination;
> >>
> >> but there are 1400+ possible
On Mon, 6 May 2002, Paul Davis wrote:
> >> there's no reason to do that. if you're going to require that the user
> >> merge two values into a single variable (as above), the user can put
> >> the merged result in control->value.integer.value[0]. i just wanted a
> >> way to avoid the user having
>> there's no reason to do that. if you're going to require that the user
>> merge two values into a single variable (as above), the user can put
>> the merged result in control->value.integer.value[0]. i just wanted a
>> way to avoid the user having to do this, because otherwise, it will be
>> im
On Sun, 5 May 2002, Paul Davis wrote:
> >On Sun, 5 May 2002, Paul Davis wrote:
> >
> >> >I suggest you to use the field index of struct sndrv_ctl_elem_id (one
> >> >byte per dimension).
> >> >
> >> >id.index = (source << 8) | destination;
> >>
> >> but there are 1400+ possible id's ...
> >
> >I
>On Sun, 5 May 2002, Paul Davis wrote:
>
>> >I suggest you to use the field index of struct sndrv_ctl_elem_id (one
>> >byte per dimension).
>> >
>> >id.index = (source << 8) | destination;
>>
>> but there are 1400+ possible id's ...
>
>I agree with Abramo. We can eventually optimize code to handl
On Sun, 5 May 2002, Paul Davis wrote:
> >I suggest you to use the field index of struct sndrv_ctl_elem_id (one
> >byte per dimension).
> >
> >id.index = (source << 8) | destination;
>
> but there are 1400+ possible id's ...
I agree with Abramo. We can eventually optimize code to handle such amo
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