Hi all !
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Is someone writing drivers for Echoaudio cards ? Perhaps
I'll buy one soon and I can try to write drivers if nobody
alse is working on it.
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On 20-Feb-2003 David Olofson wrote:
On Thursday 20 February 2003 10.17, Giuliano Pochini wrote:
Is someone writing drivers for Echoaudio cards ? Perhaps
I'll buy one soon and I can try to write drivers if nobody
alse is working on it.
I have an old Layla20 and intend to write a driver
Is there any chance to write a driver in C++ ? I'll get an Echoaudio
card soon, so I started to study the Writing an ALSA driver tutorial
by T.Iwai and the C++ sources of the generic driver by Echoaudio. That
drivers do not use exceptions (VVindos don't like it), but things as
simple as a new do
I wrote a very minimal driver that does nothing and I put it
in /pci directory. How do I compile it ?
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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 alsa-kernel
I can't compile spin_lock(lock):
echoaudio.c:237: dereferencing pointer to incomplete type
I try to compile my module with the same args, defines and includes of
azt3328.c. Spinlock operations expands to exactly the same stuff
according cpp, but azt3328 compiles fine. What am I missing ?
On 10-Mar-2003 Takashi Iwai wrote:
At 06 Mar 2003 22:46:45 +,
Giuliano Pochini wrote:
I can't compile spin_lock(lock):
echoaudio.c:237: dereferencing pointer to incomplete type
I try to compile my module with the same args, defines and includes of
azt3328.c. Spinlock operations
I have to allocate a page that the DSP will use to communicate with my
driver via bus mastering. Can I use snd_malloc_pci_pages() ?
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When an interrupt arrives and I call snd_pcm_period_elapsed(), what
function gets from alsa middle layer the new block to play and sets up
the hardware ?
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When an interrupt arrives and I call snd_pcm_period_elapsed(), what
function gets from alsa middle layer the new block to play and sets up
the hardware ?
Ok, nobody answered, perhaps my question was too stupid :))
I still haven't found in the docs how can I get from ALSA the
address of the
On 17-Mar-2003 Takashi Iwai wrote:
usually in the interrupt handler, you need to just send ack to the
middle layer by calling snd_pcm_period_elapsed(). then, the
middle-layer checks the current position by calling pointer callback,
and copy/send/set-silence in the necessary area.
I didn't
On mar, 2003-03-18 at 13:32, Takashi Iwai wrote:
also, don't forget to unlock the spinlock during calling
snd_pcm_period_elapsed() if a single lock is used for callbacks.
Yes, I'm following the tutorial and it's clear about that.
Now I have another weird problem. This is a peice of my hw_param
Look at the physical addresses. They're spaced by 4KB, but the periods
are 11KB long and 11026*344100 !! You can imagine how beatiful sound I
get... What am I missing ?
Ehm, I found the problem, ignore my previous msg.
Bye.
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This
I'm facing another *£%@ problem. When period size is a multiple or
submultiple of PAGE_SIZE, it works fine, but when it isn't, sound
clicks, pops, repeapeapeats, skps :(( And since the period never crosses
physically the page boundary because I split it when it happens, I can't
imagine what's
I'm facing another *£%@ problem. When period size is a multiple or
submultiple of PAGE_SIZE, it works fine, but when it isn't, sound
clicks, pops, repeapeapeats, skps :(( And since the period never crosses
physically the page boundary because I split it when it happens, I can't
imagine
They are different:
[aplay no options track.wav]
Mar 19 19:21:23 localhost kernel: pcm_hw_params (bufsize=88200 periods=3
persize=22052)
bufsize==88200 != 3*22052 == 66156
Note that there is no guarantee that the periods == integer value.
If your hardware doesn't allow to set
Ok, but how do I build the sg list when bufsize!=sz*periods ? Take
the example above: have I to build 4 periods and let the last one
smaller than 22052 ?
it depends on the hardware.
There are no hw constraints IFAIK, but the hw does not provide the
current dma address. It tells me how many
What are PUSH, RELEASE, SUSPEND, RESUME trigger commands supposed to do
?
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setvolume control callback always get uc-value.integer.value[x]==0. Any
ideas ?
Bye.
-
[EMAIL PROTECTED] Giu]$ amixer
Simple mixer control 'PCM',0
Capabilities: pvolume
Playback channels: Front Left - Front Right
Limits: Playback
What are PUSH, RELEASE, SUSPEND, RESUME trigger commands supposed to do
?
* pcm stream pausing
imagine the pause-button is pushed and released.
these commands are issued only when SNDRV_PCM_INFO_PAUSE is set to
info field of snd_pcm_hardware_t.
- PAUSE_PUSH
pause the stream
-
On 24-Mar-2003 Takashi Iwai wrote:
Ok, PAUSE_PUSH--stop() and TRIGGER_STOP--stop()-and-reset dma
pointer. In my case RELEASE and START are the same thing.
note that resetting the dma pointer wouldn't be necessary in STOP
trigger (although it may be safer), since the dma pointer will be
On 23-Mar-2003 Giuliano Pochini wrote:
setvolume control callback always get uc-value.integer.value[x]==0. Any
ideas ?
Solved. I had to remove .access=SNDRV_CTL_ELEM_ACCESS_READWRITE and .index=0
from snd_kcontrol_new_t structure init. Strange.
Bye
snd_pcm_new() has two parameters for the number of substreams
for playback and capture. Yes, but what does it mean ? If I
have N playback substreams, can I service N applications that
use one stream ? Can channels be allocated on demand, to allow
the user to use 4 xmms or 1 xmms and a dvd player
snd_pcm_new() has two parameters for the number of substreams
for playback and capture.
its a language problem. its not the number of substreams as in the
quantity of substreams. rather, it means the ID of the substream.
Yes, but what does it mean ? If I
have N
On 26-Mar-2003 Takashi Iwai wrote:
the pcm substreams belong to each pcm stream. when you open a pcm
device, one pcm substream will be used at each time.
snd_pcm_new() takes the argument how many pcm substreams are created
for each pcm stream. you can give more than one here only if the
When an application opens a mono substream, the sound has to
be sent to one channel only or must the low level driver
transparently convert it to stereo ?
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On 28-Mar-2003 Paul Davis wrote:
one channel only.
[...]
...but, if you are doing mono playback, your sound driver has to setup
your soundcard to playback on both channels the same mono channel. The
Vortex
driver (still not in the main branch) for example splits the mono signal
on its internal
The documentation of my card says I should set pci latency to 0xC0. What
is that latency ? How do I change it ?
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Please give me some suggestions about how to build a control interface
for this thing. The card has Ni inputs and No ouputs. It can route any
input to any number of outputs and each connection have a different
gain. e.g:
---50%- Out0
/
In0 ---30%- Out3
\ /
\ -60%---
The driver is in alpha stage, there are a lot of problems, a lot of
things are not implemented yet and it may crash you printer too. It
works not too nicely on my Gina24. How want to waste some time ?
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Must the string returned by an enumerated _info control callback be
constant ?
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On 31-Mar-2003 Takashi Iwai wrote:
as we talked in the LAD meeting, there are some possibilities:
1. implement it on hwdep device
2. implement it as a new control element (e.g. CTL_ELEM_MATRIX)
3. introduce a flag to allow the indirect access with CTL_ELEM_INTEGER
4. use 1500+ mixer
On 31-Mar-2003 Jaroslav Kysela wrote:
the case 2 is, IMO, the most straightforward way. the info field will
have number of rows and columns, in addition to min, max and step
values. the change in the alsa core wouldn't be too much.
I think that we simply touch the barrier given by 'struct
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
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
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_pages(), dma is
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
buffer
On mer, 2003-06-04 at 17:17, Takashi Iwai wrote:
At Wed, 04 Jun 2003 17:57:32 +0200 (CEST),
wouldn't it better to put another Makefile in sub directories?
Sure. I followed the instrutions you wrote here a couple of months ago.
When I do cvscompile --with-cards=powermac,echoaudio[...] it stops
In v0.9.4 Rules.make changed. I used to put the files of my driver
in the alsa-driver-xxx/pci directory, but they don't compile anymore:
bash-2.05$ make
Coping file alsa-kernel/core/pci/echoaudio.c
cp: cannot stat `/home/pochini/soft/alsa-driver-0.9.4/alsa-kernel/pci/echoaudio.c': No
such file
On 04-Jun-2003 Jaroslav Kysela wrote:
I tried to replace Rules.make with an older version and it works
fine. How do I compile the driver for 0.9.4+ ? I also would like
to group all files in a subdir.
Do you have a file alsa-driver/pci/echoaudio.patch?
Yes... I removed it and now it
How can I tell make dep to ignore some of the files listed in
snd-mychip-objs ?
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On 09-Jun-2003 Jaroslav Kysela wrote:
How can I tell make dep to ignore some of the files listed in
snd-mychip-objs ?
It is not possible. Could you send me your code? I will try to look what
is wrong.
There's nothing wrong. The driver is based on the C++ code written by the
manufacturer.
On 11-Jun-2003 Sundaranathan S wrote:
Hi All,
Can anyone tell me how to test alsa-lib APIs. If there is any
documentation available please let me know.
Yes, the official documentation is here:
http://www.alsa-project.org/documentation.php3
Bye.
My card which can enter a low power state. It turns
off some components but it leaves the PCI bus interface
active. To bring the card back to life it has to be
reinitialized from scratch. What D state is it ? PCI
power management papers say that already in D1 state
i/o space is disabled. I'm a
On 12-Jun-2003 Jaroslav Kysela wrote:
On 12 Jun 2003, Gorm David Lai wrote:
I tried your example. Aplay is rather stupid, so it is seems it needs a
Not too stupid, but you should use right device (plughw vs. hw) and
probably right alsa-lib configuration (~/.asoundrc).
The format written in
Some of the cards by Echoaudio have a midi port, but it's not MPC401
compatible. The card has two tiny buffers to send and receive bytes. Where
can I get infos, sample code, etc. to add midi support to my driver ?
The Layla24 also exists in PCMCIA edition. Just like above, I need docs...
Tnx.
On 18-Jun-2003 Warren Turkal wrote:
Any suggestions on how?
Ask him the address of his boss.
Bye.
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On 26-Jun-2003 Fred Gleason wrote:
I've got things working with my ASI6114. Playback works well.
Alsamixer shows me a bewildering array of controls, few of which seem to
do anything. How do I control stream levels?
I don't know that chip, but the names of the controls should be
A fellow testing my driver has a strange problem. lspci
shows the card is attached to IRQ10, but the driver reads
IRQ5 instead. He has another soundchip (CMI-something)
which uses IRQ5. My driver follows the tutorial by tiwai
closely and it's nearly identical to a lot of other
drivers. Any ideas,
I'm looking for a small program that reads and writes a few bytes through
the midi port.
--
Bye.
Giuliano.
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It's against alsa-lib 0.9.5. It also applies to the CVS version because
the files are identical.
--- ./src/control/control.c_old Sun Jul 13 14:01:10 2003
+++ ./src/control/control.c Sun Jul 13 14:09:31 2003
@@ -1699,10 +1699,10 @@
assert(obj);
if (obj-access
On 14-Jul-2003 James Courtier-Dutton wrote:
Worst case accuracy is about one period AFAIK. It depends on the sound chip
because the low level driver reads the DMA pointer (or something) from some
hardware register or it can know the DMA pointer when it receives an interrupt
at the end of a
On 05-Aug-2003 Antti Boman wrote:
This seems like the problem I've always had with my single processor
with JACK and SCHED_FIFO, with 2.4.20 kernel. I haven't tried the
realtime setting for a while, though.
Maybe there's some other thing causing these? Should we gather a bit
more in-depth
Please do not include the utils/mod-deps x86 binary executable in the stable
alsa-driver tarballs. cvscompile silently fails on non-x86 platforms.
--
Bye.
Giuliano.
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When I play something, depending on the buffer length and the period length,
sometimes ALSA complaints about a wrong hw_pointer value. pcm_hw_pointer()
returns the right value and the irq handler (not shown below) calls
snd_pcm_period_elapsed() only when the period is done. I can't understand
On 28-Aug-2003 Paul Davis wrote:
--*-- What is the best-supported *Pro Audio* device in terms of solid ALSA dri
ver support? Not interested in Sound Blasters and chips on the motherboard --
something along the lines of the Echo, M-Audio, or RME products. But the ALS
A drivers need to be stable
On 28-Aug-2003 Jaroslav Kysela wrote:
--*-- What is the best-supported *Pro Audio* device in terms of solid ALSA dri
ver support? Not interested in Sound Blasters and chips on the motherboard --
something along the lines of the Echo, M-Audio, or RME products. But the ALS
A drivers need to
On 01-Oct-2003 Nathaniel Gray wrote:
My sound card uses different devices for capture (device 0) and playback
(device 1). I have two issues with capture right now:
1. If I do something like:
arecord -f dat -D hw:0,1 -d 5 foo.wav
it seems to record for much longer than 5 seconds.
On Sat, 4 Oct 2003 16:58:12 -0500
Ryan Underwood [EMAIL PROTECTED] wrote:
i have an ES1983S Maestro-3i on a c600 dell laptop
i got this message in syslog :
ALSA sound/core/pcm_lib.c:214: Unexpected hw_pointer value (stream = 0,
delta: -944, max jitter = 1024): wrong interrupt
My hardware can start/stop multiple indipendent substreams at the same time. Does
the ALSA driver API support that feature ?
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On 15-Nov-2003 Jaroslav Kysela wrote:
On Sat, 15 Nov 2003, Giuliano Pochini wrote:
My hardware can start/stop multiple indipendent substreams at the same time.
Does the ALSA driver API support that feature ?
Yes, use snd_pcm_set_sync() function in open() callback and add necessary
code
If I try to register two different controls which have the same name, it does
not fails very nicely. It does not call the .dev_free callback and it doesn't
free a lot of stuff:
Nov 22 23:55:53 localhost kernel: ALSA ../alsa-kernel/core/memory.c:81: kmalloc(5)
from e501bc78 not freed
Nov 22
snd_ctl_elem_info_t.dimen contains the dimensions of the array of
controls. AFAIK there are no drivers which use that feature (but mine)
to look for examples. We have to agree about the order of them:
Array[d0][d1][d2][d3]
or
Array[d3][d2][d1][d0]
I used the first one. Is it ok ?
--
My driver supports several different cards of the same series. Most of the
code is common for all cards, so I put the code which is not common into
separate .c files. Well, now I need a way to tell make what is the card I
want the driver for.
--
Giuliano.
On Mon, 24 Nov 2003 21:25:00 +0100 (CET)
Jaroslav Kysela [EMAIL PROTECTED] wrote:
On Mon, 24 Nov 2003, Giuliano Pochini wrote:
snd_ctl_elem_info_t.dimen contains the dimensions of the array of
controls. AFAIK there are no drivers which use that feature (but mine)
to look for examples. We
The executable utils/mod-deps shouldn't be shipped with the alsa-driver
tarball. It's an x86 binary and it makes cvscompile to fail on other platforms.
--
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Does
Actually, I'm not completely opposed to the idea. But I am totally
clueless when it comes to writing device drivers, and would have no
idea where to start.
Start here: http://www.alsa-project.org/documentation.php3#Driver
You don't have to start writing the driver from scratch. You only
have
Echoaudio cards are grouped into two families: EchoGals (the old
discontinued serie) and the Echo24. The driver is almost identical for both
families, but some constants define at compile time. I currently select the
family by a hack in the makefile:
ifneq ($(CONFIG_SND_GINA24),)
EXTRA_CFLAGS +=
On Sat, 6 Dec 2003, Jaroslav Kysela wrote:
See alsa-kernel/pci/ens1370.c and alsa-kernel/pci/ens1371.c for an elegant solution.
Tnx.
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On Sun, 07 Dec 2003 14:19:28 -0500
Paul Davis [EMAIL PROTECTED] wrote:
i personally am heading towards believing that we made a design
decision that was wrong here. i now tend to think that the PCM
interface should not be involved with configuring the hardware at all,
and that this should be
On 08-Dec-2003 Arve Knudsen wrote:
Wether its done via the control or pcm interface, it'd be good to have a
loose coupling between configuration and streams, so one could could access
configuration space without locking a stream don't you think?
Yes, of course. Perhaps it can be done already
I have to save the settings of the mixer and I need a unique name for each
card. Does snd_ctl_card_info_get_id() return different ids if there are two
identical cards ?
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Attached is an update to the current status of the driver for Echoaudio
cards. That page is now generated dynamically and I don't know what
format do you prefer. I just grabbed the page and edited it.
--
Giuliano.
Title:
Advanced Linux Sound Architecture - Soundcard Matrix
On 19-Jan-2004 Michal Kostrzewa wrote:
Hello,
Please help me in that license related case:
I want to write a Lynx AES16 (http://www.lynxstudio.com/aes16.html) ALSA
driver (I'm from Warsaw University of Technology and we want to use this
card in our investigations). I wrote to Lynxstudio,
It fails at line #113 of /Makefile :
@for d in $(SUBDIRS); do if ! $(MAKE) -C $$d prepare; then exit 1; fi; done
I have no idea why $(SUBDIRS) is an empty string:
[EMAIL PROTECTED] alsa-driver-1.0.2]$ ./cvscompile --with-cards=powermac
--with-isapnp=no --with-sequencer=no
On Thu, 29 Jan 2004 14:38:14 +0100
Takashi Iwai [EMAIL PROTECTED] wrote:
note that --with-isapnp is ignored on 2.6 kernel.
the isapnp support depends only on the kernel itself.
Ok, I used that option only to avoid compiling useless stuff.
does the attached patch work?
Yes it works. I
--- pci/rme9652/hdsp.c_old Sun Feb 22 15:52:59 2004
+++ pci/rme9652/hdsp.c Sun Feb 22 15:57:43 2004
@@ -3810,8 +3810,8 @@
{
int mapped_channel;
-snd_assert(channel = 0 || channel hdsp-max_channels, return NULL);
-
+ snd_assert(channel = 0 channel
Is there a list anywhere listing the differences between OSS and ALSA
with regard to sound card hardware.
It would be nice to have a nice small list of all the sound hardware OSS
supports, but ALSA does not.
Then the intention would be to reduce that list to no entries by
On Wed, 25 Feb 2004, Gupta, Kshitij wrote:
hi,
I have a very trivial question about the dma transfers with respect
to ALSA framework.
Let me first explain a scenario
We have a Circularly linked Buffer pool
buf1buf2buf3 buf4 bufn
In very simple terms playing an
On 25-Feb-2004 Adam Tla/lka wrote:
On Tue, Feb 24, 2004 at 10:54:58PM +0100, Benno Senoner wrote:
for those that are too lazy to browse the forums:
http://www.4front-tech.com/forum/viewtopic.php?t=25
-
far more advanced ???
Ok I'd like see Ardour runnnig with multiple 24bit
On 04-Mar-2004 Takashi Iwai wrote:
It is wavetable synthesizer API not PCM API. This part of ALSA is still
in the development phase.
But when that gets fully developed, would I have to use the rawmidi API
to do this then?
I think that sequencer API is more appropriate, but as I
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
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())
On 09-Mar-2004 Ove Kaaven wrote:
Independently? No, our software mixing code resamples, adjusts volume,
and mixes the stream into the buffer in one step.
It should be more efficient than separate passes. It minimizes
memory i/o and some operation (multiply-add == volume-mix) can
be done
On 10-Mar-2004 Ove Kaaven wrote:
Well, the requirements that raised this thread should be fairly clear.
For example,
ALTERNATIVE 1
snd_pcm_set_volume(snd_pcm_t* pcm, int volume)
and
snd_pcm_set_pan(snd_pcm_t* pcm, int pan)
using whatever value range makes the most sense, and perhaps
On 10-Mar-2004 Ove Kaaven wrote:
Uhm... I think first of all we need a way to know how many
virtual channels are available (hw and sw)
For the EMU10K1, snd_pcm_info_get_subdevices_count() and
snd_pcm_info_get_subdevices_avail() works for me. But I suppose you're
right that this isn't a
On Mon, 22 Mar 2004, Pavana Sharma wrote:
Hello,
I am trying to export the controls to user space. I want to know the
complete list of controls
[...]
Where can I get the complete list of controls which an audio codec has
to support.
There is not such a list because they depend on the
My driver is almost complete and I'm working to make it acceptable for
inclusion in alsa-driver. It drives 8 different cards and it raises some
problems. The main file is echoaudio.c and is contains all the control
interfaces of all cards. It registers only the controls a card has at
runtime, so
When I try to load a module and it fails (snd-darla20 in this case) the
module is not unloaded. IIRC it didn't happen some months ago, but I have no
idea what is the cause because I changed a lot of stuff in the meantime
(kernel, compiler, modutils...). Is this the expected behaviour ?
[EMAIL
On 24-Mar-2004 Brian Furey wrote:
Hi all,
im using the ALSA driver 1.0.0 on two linux machines that connect a
voice-over-ip session. I need to know the minimum fragment size that
the alsa driver can take from the soundcard and place in the
sending/receiving buffer. Does it depend on the
On 25-Mar-2004 Brian Furey wrote:
Thanks Giuliano,
How can I get a dump of the current hardware configuration space?
Hmm, lspci ? :)
If you mean how to get the capabilities of the card, it's a bit
tricky. The caps are not a static thing. Each setting may rure
others. For example, a card has
I'm not sure if this is alsa's fault. It happens when I unload the module.
Yup, snd_gina24 is the driver I'm developing, but it also happens with
snd-powermac and when I try to load a module for a card I don't have.
Linux Jay 2.6.5-rc2-ben0 #1 SMP Sat Mar 27 09:10:55 CET 2004 ppc unknown
The hw supports all standard sample rates from 8KHz to 96KHz and continuous
mode, but the latter only in the range 25KHz-100KHz. What kind of constraint
I have to set ?
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On Sun, 28 Mar 2004 09:54:46 +0200 (CEST)
Jaroslav Kysela [EMAIL PROTECTED] wrote:
On Sat, 27 Mar 2004, Giuliano Pochini wrote:
I'm not sure if this is alsa's fault. It happens when I unload the module.
Yup, snd_gina24 is the driver I'm developing, but it also happens with
snd-powermac
I have some problems with 6 channel sis 7012 device - I'm using it as 3
stereo channels. I trigger 3 different instances of aplay at once and I get
underrun messages from aplay.
What does this mean ?
It happens because aplay can't send audio data as fast as the card needs it.
How to
On 11-May-2004 Niklas Werner wrote:
Hi *,
is there any progress on the status of full support for the newer devices
as snapper, etc (in the AlBooks,..). I'm running the 2.6.5 benh-kernel
from bitkeeper and basically the sound only works reliably when using the
oss emulation.
I have a
On Wed, 12 May 2004, James Courtier-Dutton wrote:
Just for general info.
pci10b5,1142 NOT-HANDLED lynxone
Add the LynxTWO, Lynx L22 and Lynx AES16. LynxStudio provides programming
info only under a NDA that doesn't allow the licencee to release anything
in source form.
--
Giuliano.
On Wed, 12 May 2004, James Courtier-Dutton wrote:
Are you saying that Opensound OSS linux drivers support the LynxTWO,
Lynx L22 and Lynx AES16 ? I don't think so.
From www.opensound.com homepage:
* OSS v3.99.1c [...] Beta drivers for LynxTWO professional soundcards
announced.
--
On 13-May-2004 Adam Tla/lka wrote:
What do you want to do with the mixer?
Hmm - typically people just want to control PCM volume.
When using OSS you can just do mixer ioctl's on the opened PCM fd.
Simple and clear.
No, it's not simple, unless the driver makes it simple
removing some
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