but still one thing is not certain. what does the following (in
snd_hdsp_initialize_firmware) set?
#ifdef SNDRV_BIG_ENDIAN
hdsp_write(hdsp, HDSP_jtagReg, HDSP_BIGENDIAN_MODE);
#endif
does it switch the access to big-endian? if yes, then we need
cpu_to_xxx things
Paul Davis wrote:
i have been using alsamixer to control the playback volumes, and it
seems to work fine. amixer reports the values correctly as well, so
try
amixer cget numid=11 # channel 1 playback volume
numid=11,iface=MIXER,name='Chn',index=1
; type=INTEGER,access=rw---,values=1
Thomas and Ico:
can you please try to boot your systems with the pci=biosirq command
line flag?
from my discussions with alan cox on the linux kernel mailing list, it
appears that the PCI configuration information for the card isn't
complete, and only the BIOS knows how to figure out the
i'm new to this list. i'm writing because i have a hammerfall dsp
multiface soundcard with a pcmcia interface and would like to run it
under linux. is it already possible to test the new drivers? i think i
will need some help to get it to run for i have not so much experience
in solving difficult
I'm just starting to write my first alsa-lib midi applications. I'm
wondering what the acronym ppq stands for and what it means (especially
in relation to tempo). Unfortunately, I didn't find any docu on the
topic of tempo.
int ppq; /* time resolution, ticks/quarter */
[ ... ]
the
ppq stands for parts per quarter. as the header file suggests
without much explanation, its length of a quarter note expressed in
clock ticks. if the meter is 4/4 and the tempo is 120bpm, then a
quarter note lasts 0.5 seconds. if the clock used for timing ticks 100
times per second, there are 50
I am trying to read and write raw audio from my soundcard which is an es1371.
the code i use is as follows :
this the ALSA development mailing list. If you have questions, please
at least use the ALSA API, and not the outdated-but-supported OSS API.
ya i know the code is not optimized but then
Hi,
I was very excited and pleased to see hdsp driver had reached cvs (thank you
Paul !) , so I gave it a try. Here are the results.
I manage to insmod everything needed except the snd-hdsp module. I get the
following in syslog:
May 16 12:11:18 satellite kernel: RME Hammerfall-DSP: no cards
I changed my lilo.conf accordingly.
The original entry had a nobiospnp option (nobiospnp was there since
initially that prevented me from booting due to fact that long time ago
Mdk froze at boot time when it came to probing pcmcia service, however
I've changed it and it made no difference).
I
latency
calculations.
Developers and users interested in Jack should sign up to
[4]jackit-devel, our mailing list.
Contributors
* Paul Davis -- principal author
* Kai Vehmanen -- design, debugging, general code-fu
* Steve Harris -- dsp mastery
* Andy Wingo
It seems to me that the pcm_peak code in aplay doesn't work or maybe i
dont understand it's actual function. It shows the same thing over and
over no matter what the volume of the audio going through it (captured
from mic).
then its probably not working on your system for some other reason,
My library version is 0.5.10b and the driver version is 0.5.12a.
0.5 is no longer really supported. You should be using 0.9.X. The web
site says on the front page:
*** N.B. The 0.5.x series is considered deprecated and is no longer
supported by ALSA developers ***
The author of the
you do it using ~/.asoundrc to define a PCM device that has the
channel characteristics you want.
next, you'll be asking how. that's the part i can't help you with. the
archives have some examples, and there are several people on the list who
can tell you.
aha, so the idea is that an
personally, i don't like either of them very much, which is partly why
i wrote JACK.
ok. you are now using: ALSA kernel - ALSA lib - JACK.
thats the current config, except that we now have
solaris audio - JACK
as well. and we use very, very little of alsa-lib with JACK unless the
user
There is a prominent link on
http://www.alsa-project.org/documentation.php3#0.9doc
Howto use the ALSA API - Paul Davis has also written a brief explanation.
http://www.op.net/~pbd/alsa-audio.html#captureex
Right here, on this mailing list, and in private mail to Philipp, and
in mail to LAD, I
the names above are not defined on all cards (except for default).
i think we need a kind of config database for each card.
this is useful not only for pcm but also for parsing the mixer
structure.
i think this is the wrong solution. i believe the correct solution is
to provide a **simple** way
Starting sound driver: snd-hdsp PCI: Found IRQ 10 for device 00:0d.0
PCI: Sharing IRQ 10 with 00:07.2
PCI: Sharing IRQ 10 with 00:07.3
ALSA ../../alsa-kernel/pci/rme9652/hdsp.c:452: wait for FIFO status = 0
failed
after 100 iterations
looks like a multiface
done
When this happens, the card will
Paul Davis wrote:
Isn't there any option to use a 'virtual' ALSA device? Like snd-pcm-oss, but
then the other way around. snd-oss-pcm module? Then you could use ALSA with
a working OSS driver, and no working ALSA driver...
ALSA has a significantly different internal architecture than OSS
i currently writing a plugin for alsa9 and i'm thinking of various
paramters which could be set and if there are usefull for example does
someone know 'good' values for snd_pcm_sw_params_set_xfer_align(), in the
code samples it varies from 1 to 4, is it also usefull to set it to higher
values for
yes. see the internals of JACK's jack/drivers/alsa/alsa_driver.c
Honestly, I'm afraid to look at that file too carefully since it is GPL
and PortAudio is BSD. This puts me in kind of a weird situation.
nobody is going to come after you for an extremely similar
implementation. the GPL doesn't
client alsa_pcm: inprocess client, execution_order=0.
ALSA: poll time out polled for 44800.813378
driver wait function failed, exiting
telling signal thread that the engine is done
jack main caught signal 1
which means that the JACK alsa driver/client didn't return from poll
on the ALSA streams
* To [EMAIL PROTECTED] ([EMAIL PROTECTED]) wrote:
I am stumped. This short and simple program fails for me:
Wingo solved the mystery for me. Apparently you are required to do an
mmap_begin and mmap_commit before you can call snd_pcm_start.
or more generally: you have to put some data in
is it possible with alsa to use different settings for capture and
playback, either periods or buffer size or both ?
if yes, i might try and add that to jack, but since this is my first
undertaking with alsa programming, i thought i'd ask first :)
i got this idea from glame, which uses 2 and 4
I got around my previous problems (readn) in an output/capture
application with an envy24
(ice1712) by bypassing the plugin and going straight to hw:0,0. Had to
change my code a bit
but it's now **almost** working.
i humbly suggest that you:
(1) read the source code for JACK's ALSA
I have a small perl script which I am working on. The idea is to autodetect
a card/s and create a modules.conf based on that either at configure or as
a seperate process.
IIRC, this has already been written, but its not up-to-date. its in
the alsa-conf CVS module.
--p
I did, but the Jack introduction frankly scared me off a bit. I'm
sure that it's not its intention, but the talk about necessary mods
to the kernel and synchronisation issues between channels (IO)
seemed a huge amount of bother to go to at the time.
no kernel mods are necessary for JACK. kernel
(snd_force_firmware, Force a reload of the I/O box firmware);
+MODULE_PARM_SYNTAX(snd_force_firmware, SNDRV_ENABLED , SNDRV_BOOLEAN_FALSE_DESC);
MODULE_AUTHOR(Paul Davis [EMAIL PROTECTED]);
-MODULE_DESCRIPTION(RME Hammerfall DDSP);
+MODULE_DESCRIPTION(RME Hammerfall DSP);
MODULE_LICENSE(GPL);
MODULE_CLASSES
This link is labled Howto use the ALSA API, and it points to
http://www.op.net/~pbd/alsa-audio.html;. The example programs presented
there do not work with the current alsa library. This link should at
least be moved out of section ALSA 0.9.x Developer documentation. It's
quite confusing as it is
why doesn't this work:
snd_ctl_open (handle, hw:0,0, ...)
it means that you can't use the same standard name format for control
devices as PCM devices, which is a bit of problem for programs that
need to open both but only want to require the user to specify a
single name ...
--p
some functions take an int parameter called dir (e.g.:
snd_pcm_hw_params_set_rate()).
What is the meaning of that parameter?
its to let you know if the actual value set is above, below or equal
to the one you asked for.
--p
---
This sf.net
* remove some printk's
* fix rate rules so that OSS emulation works correctly
Index: hdsp.c
===
RCS file: /cvsroot/alsa/alsa-kernel/pci/rme9652/hdsp.c,v
retrieving revision 1.8
diff -u -u -r1.8 hdsp.c
--- hdsp.c 14 Jul 2002
i've been wondering what to do about setting sample rates in the
hammerfall and h-dsp drivers. fernando and gary from CCRMA have made
the excellent suggestion that rather than have the driver pretend that
all rates are available all the time, as it currently does, that we
have a control switch to
I am in a situation where I intend to call snd_pcm_oss_write
directly.
AFAIK, this is not a supported use of the alsa-lib API. There is no
guarantee that your application will continue to work in future
versions of alsa-lib.
What exactly happens is - the open and all the initial ioctl
Am reading audio using snd_pcm_readi() and using short buffer times
that result in Broken pipe errors being returned. Occasionally losing
audio is not a problem, but do I have to completely reinitialise whenever
this occurs?
it depends on the sw params you set up (if any). i think the default
is
Hello to all,
who has already done DTMF recognition with ALSA ( OSS )?
I searched for examples in google, but I have found that
only in TODO lists ;-(.
Is this so difficult or why doesn't exist anything about that?
its a relatively trivial FFT problem. use FFTW to analyse the signal,
then look
I have the working version of the template for the new improved docs.
Can people who have a spare moment have a look through it for obvious
mistakes. Any additions to the text will also be appreciated.
I've looked at it so many times now I need a fresh eye to spot my mistakes.
there also needs to be a place for information about card-specific
control switches too.
Are there many cards that have controls switches? If there are not then
I can see a reason to make that component optional to the template.
How should I describe them? I have never used a card that has
I try to do IO with the async transfer method, but I don't get it work.
As artwork I use the pcm example, but it doesn't work too (pcm -m async).
Apparently the callback gets not called.
Is there a bug concerning this?
i hope you have a good reason for using the async method, along with a
very
The way you'd do this is to link the capture and playback streams of
the device you use with snd_pcm_link(). They will start together and
we still haven't established the timing for this. its possible that
its not fast enough for the two streams to be started within
1sample, particularly if and
By the way, I tried to search for the same code in rme9652.c, and found
something similar, but it does not seem to activate both capture and
playback with a single write to the control register. I'm not sure how
the hardware works (nor the driver) though so I may be wrong.
there are no
I've looked through some driver code, and if I understand it correctly,
it seems like some drivers, for example the ens1370, check in the
trigger function the substream is linked with other substreams on the
yes, sorry, i forgot about this aspect of things. i was thinking of
snd_pcm_link()
I am working with Alsa 0.9.0beta12 version and I am also using posix
threads (libpthread) in my project. In the release version of the
program I link both libraries (libpthread, libasound) dynamically and
everything works well. In the debug version however, I have to do static
linking of the
i would personally work from a complete set of snd-* modules, and use
modinfo(1) + perl to get the information
Isn't that what happens in the INSTALL file anyway?
it doesn't look like it. there are several cards missing there, and it
looks hand-written rather then fetched from the source.
Do you mean that I should be parsing each driver file seperately?
not really, just do a single pass over each one of them at some point
in time. store the results, and use them. this should get you started
(its not perfect, but it doesn't do a bad job.
#!/usr/local/bin/perl
while() {
This is what I want to do: guitar into the line input on the live drive,
that going into ardour, through a few ladspa effects, and out to the
speakers. I can't seem to do this. The only way I seem to be able to
get the input into ardour is to have it going through the speakers as
well so that
Sorry I am new to ALSA so forgive me if this is obvious.
don't be sorry about that. instead, be sorry for this: whether you
know it or not, the email program you are using is sending out copies
of your mail in both plain text and HTML formats. increasingly on the
net, there are filters being put
Ok set the option ( you guessed it Outlook Express ) to text only. Sorry
don't have a go at me as it is the default setting I believe.
Option 1 or 2 would be what I am looking at. What I have is a program with
multiple threads needing to play PCM streams to the same device.
there are some
So does JACKs allow two separate applications to stream PCM audio at the
same time to the same device and if so will it then overlay them? If it will
why would two threads each registering with JACKs not work.
it might, but it would be very inefficient. every JACK connection
requires a new
i'm not exactly shure, what is double-buffers... the turtle beach
pinnacle/fiji has shared memory, that can be accessed
simultanously from the pc the sound card.
Some cards/chipsets (RME PST, ESS solo1) allow one to transfer
data for playback (record is similar) using a memory-mapped
Paul wrote:
cards that do not use DMA should generally be considered inferior
because of the extra CPU cycles they force on the host system.
not generally. i.e. the pinnacle/fiji way is to map a piece of memory into
the pc's memory space.
thus an application using alsa-mmap can write (or
Thanks for your reply and the education. My main reason behind asking
this question is that I am using the RME PST card presently to do
CPU controlled transfer of data to the memory mapped buffers in those
cards. I am interested in using the CPU to do the transfer since
I have heard that DMA or
Exactly. I would prefer the second as if I needed to add a third, 4th its
takes no effort and I don't reinvent the wheel as they say.
Does alsa support 2 or is this just JACK's.
No, ALSA does not support it *in general*. Some interfaces will work
this way, some will not. In theory, the ALSA
On Fri, 2 Aug 2002, Pete Black wrote:
They indicated 1-2 weeks until driver release, but it is now going on 3
weeks i think, so I don't hold out too much hope for a prompt release.
They have released it here:
http://www.echoaudio.com/Download/Developer/EchoGenericDriverA0.zip
wow.
now
I am going to set up a sourceforge project, and admin an email list and
website for the development of this driver. As 'project admin', i will
be happy to make contact with Echo corporation and request a hardware
loan on behalf of the project.
i personally think doing this on sf.net is
Hi, this may well be nothing to do with alsa but I thought you may know
what is happening.
If I make a system call from one thread whilst recording or playing in
another thread using the alsa lib, then when I try to snd_pcm_close I get:
ALSA lib pcm_hw.c:145:(snd_pcm_hw_hw_free)
known bugs i know of (and am working on) are:
hdsp
- cold+warm boot required to get output working
- 96kHz and 88.2kHz not accessible via PCM interface
rme9652
- 96kHz and 88.2kHz not accessible via PCM interface
wavefront
- cs4232 PCM output does not work at all (produces white
What is the purpose of the ctl.rme9652 declaration? My plugin devices
seem to work without it.
It is a configuration for the onboard mixer. Very little is known about
why we use it. Some programs (JACK) won't work without it.
not really.
the control device for a card is the way that
There are 3 standard PCM ioctls, and a driver could potentially support others
via its own ioctl function registered in the snd_pcm_ops_t structure. Could
anyone give me a hint regarding how an application could then call such
standard and other ioctls?
i'm not going to do that, but i will point
Recently I started to collect some info on how to make a ALSA driver for
the Digigram VXpocket V2. Digigram decided to provide information
this PCMCIA card, on condition that a moderator is the single point of
contact with Digigram.
glad they came to their senses. the existence of the RME
hmm, it's not so easy, since this will break also the assumption of
constant period frames by applications. if we introduce the
time-based period size, it won't work as compatible as older one,
e.g. jack wouldn't run properly if the period size changes
dynamically.
this isn't true, or if it is
after recompiling a kernel on my laptop, my RME HDSP configuration is
broken. Does anybody have a working setup with the RME PCMCIA card and
would be willing to share some information about the process of
initialization off list?
i don't have the working setup (with pcmcia), but i would
Now the snd-hammerfall-mem install o.k. but snd-hdsp segfaults (and I
have to reboot in order to get rid of the module).
well, pcmci-cs may be at fault, but this is still a bug in the
driver. it shouldn't segfault. i guess i've assumed that certain
things are always true and that this is broken
For some reason, the hotplug script can't successfully load the snd-hdsp
driver but a simple
modprobe snd-hdsp does work. This is true at boot time (coldplug) and
upon insertion (hotplug). Can anyone offer any insight into this?
i know nothing about the hotplug script of pcmcia-cs, but i bet
Patrick -
I wanted to add this information to the h-dsp page, but when i try to
get there from the card matrix, it seemed more work to use that form
since i'm already on this list.
Can you please add this note to the Known Issues for the H-DSP:
- the PCMCIA interface requires a kernel that
About OSS configuration, it was explained separately in another part
after the sentence above.
but we don't provide any such script ... without it and without the
modules.conf entries, any OSS program will fail, right?
---
This sf.net email
This is a message from Dave Hinds concerning autoconfiguring the
PCMCIA. I looked for pci_register_driver and grep only found it in
the binaries of snd-hdsp. Has the driver been written with the
modern kernel API, Dave is referring to?
yes, all of ALSA does this. we use pci_module_init() and
Is there any know problems with 0.9.0rc2/Linux
2.4.19/RME HammerFall. I had a working application
using 0.9.0beta6/Linux/2.2.19/RME HammerFall, since I
am changing to a Intel P4 arch, I can't get the beta6
to work. It fails on an ioctl with something about a
PVERSION... I would like to use a
ALSA lib pcm_hw.c:579: (snd_pcm_hw_open)
SNDRV_PCM_IOCTL_PVERSION failed: Inappropriate ioctl
for device
failed to open audio device default
(Inappropriate ioctl for device)
snd_pcm_hw_open is looks like this:
snd_pcm_hw_open(pcmp, name, card, device,
subdevice, stream,
On Fri, 6 Sep 2002, Guilhem Tardy wrote:
Hi all,
While trying to understand snd_pcm_lib_write1(), I found that
runtime-xfer_align is set to period_size in pcm_native.c, but wonder if it
is
just a default?
Paul Davis' example (at http://www.op.net/~pbd/alsa-audio.html#playex) doesn
't
I understand that prepare() is used to restart playout or capture from a known
state in the driver, and wonder if it is OK to have it block until buffers
being currently played out have completed?
are you asking if a low level driver should do this? if so, then the
answer is no. the semantics of
Is it possible to adjust a stop threshold or something to get my
desired behaviour?
i think you should have set the start mode to START_DATA and the start
threshold appropriately. then the device would have started as soon as
the buffer was full.
--p
IMHO the current behaviour is the proper behaviour as implemented by other
file descriptors, and as mandated by POSIX.
http://www.opengroup.org/onlinepubs/007904975/functions/write.html
says regarding pipes, FIFOs and sockets:
| The write() function shall fail if:
the discussion here is about
Please understand that it's very hard to satisfy everybody and I'm not
sure it's a worthy goal.
Who do we know that would be unsatisfied by the proposed change in the
behaviour of poll(2) on an ALSA device?
--p
---
Sponsored by: AMD - Your
you probably are calling alsa from a shared object -- try
dlopen (libasound.so, RTLD_LAZY | RTLD_GLOBAL);
in your module's init function.
The need for RTLD_GLOBAL is somewhat ugly, usually one wants to load
plugins with RTLD_LOCAL.
the suggestion is not that *your* plugin should use
My knowledge of signals is that the period is the time it takes a signal to
complete 1 cycle. So what is the snd_pcm_hw_params_set_period used for?
Is the period a property of the signal, or an arbitrary property set by
the programmer for performance, fidelity, etc.
no, a period in ALSA is,
Paul Davis [EMAIL PROTECTED] writes:
you probably are calling alsa from a shared object -- try
dlopen (libasound.so, RTLD_LAZY | RTLD_GLOBAL);
in your module's init function.
The need for RTLD_GLOBAL is somewhat ugly, usually one wants to load
plugins with RTLD_LOCAL
Paul Davis [EMAIL PROTECTED] writes:
the problem is that libasound itself calls dlopen(), and the initial
(non-RTLD_GLOBAL) linkage hasn't put libasound's symbol into the
global namespace. hence, libasound's own dlopen'ed code can't access
libasound itself.
I know, that's why I suggested
1) build library with --with-compat-rc3 and place it to some
other directory
3) build library without --with-compat-rc3, place it to
/usr/lib as usuall
4) build alsa-utils and newer applications
5) build older applications with compatible library compiled
with --with-compat-rc3
Old and
this is the /usr/share/aclocal/alsa.m4 job to provide right linking
flags depending of the library
ah. typical autoconf confusion here.
autoconf looks in only ONE directory by default for *.m4 files. if it
was installed from a package, it probably looks in
/usr/share/aclocal. if it was
anyway, i agree with paul that pkg-config should be used instead,
which offers a lot more flexibility.
That's as maybe but it's non-standard still isn't it? Another dependency
in a complicated world. I think it's too soon.
its part of all distributions at this point. its about as standard
i don't want to blow my moderator points this time, so get busy. there
is lots of confusion, mis-information and general cluelessness in the
responses so far. get to it, provide me with some stuff to moderate
upward ...
--p
---
This sf.net
ALSA:
cat /proc/asound/devices shows only one raw midi interface. This is on
a two port device (but Muse lets me select either port). On the older
alsa two devices show up on the listing.
For multiport interfaces, the additional ports are available as subdevices
of the rawmidi
Hi, I'm trying to configure asound.conf to support two cards as one
(but with 3, 8 channel cards!) as suggested in the .asoundrc doc. When I
try to aplay the new device however I get:
ALSA lib pcm_multi.c:928:(_snd_pcm_multi_open) Invalid or missing
schannel for channel 0
aplay: main:462:
1. Is there a good guide to porting applications written for 0.5.x to
there are no good guide to anything related to ALSA at this time.
0.9.0? I'm trying to fix mpg123 to work with ALSA 0.9.0 natively;
current stable version (0.59r) barfs when compiling all over
audio_alsa.c.
the API in 0.9.0
I would like to help with the Hammerfall DSP driver.
Is there any additional technical information? Could someone be so kind
and forward it to me? The driver is easily readable, but it wouldn't
hurt if I had the original documents still.
there are no documents. i have source code for a different
For synchronization purpose I need the current time of my sound card
(information on the sample that is currently played out might be
sufficient / ADAT-Synch timecode) I did anticipate that the functions
the code to read the ADAT sync timecode is written, but is not
exported to a usable part of
[Summary of this message:
I want to use alsa for timing of video frames, and want alsa to
trigger a wakeup every .01466 seconds. But the documentation is
sparse in this area]
I'm attempting to use alsa-lib as a timing tool for a specialized
movie playback engine. It _must_ tick/wakeup at
ALSA is *a* sound library. There are lots of things that it doesn't
I would really say: ALSA is *the* sound library (at least on Linux). Isn't it
in kernel
2.5+ ?
alsa-lib isn't in kernel 2.5, because its not part of the kernel.
alsa-lib doesn't contain any code to read or write audio files,
You see, if all apps are written to use the ALSA API, that's going to
be great for the purposes you have in mind, but totally awful for
those of us who want our audio apps to work in a sample synchronous
way and ignorant of the ultimate routing of their data. Many of us
don't think that an
2 clarifications:
It is not logical for every program to write support for esd, artsd, jack,
alsa, etc. Programs should write to ALSA and let ALSA do software mixing if
required. Windows provided this since DirectX (3?). Solaris provides this too
(esd apparently doesn't block on Solaris).
Its the same PCI ID, but insmod segfaults...
does the attched patch cure the segfault?
but it doesn't mean that your card will be supported by it :)
takashi - thanks for this patch. i didn't understand that
snd_hdsp_free() could be called when no hardware was detected. now i do :)
--p
its also worth noting that it too is not immune from missing xruns,
but there isn't anything we can do about kernel/driver code that
blocks interrupts for an entire buffer :(
I do not know how long an entire buffer is. I assume this will differ per
card, but how small could the worst-case
I have just got a couple of the new RME Hammerfall HDSP9652 cards, and
find that they are not supported by the snd-hdsp driver yet. The card is
basically the same hardware as the other hdsp cards but with the io
integrated on the card.
I would guess that it will need a different firmware and
actually, it can't. if the user space application is delayed for
precisely 1 buffer's worth of data, it will see the pointer at what
appears to the the right place and believe that no xrun has
occured. the only way around this is to provide either:
Well, but if you combine it with the current
I am currently taking the following approach: -
Always prepare 2 audio hardware periods of sample frames in advance
inside the user app.
1) snd_pcm_wait()
2) write()
3) prepare new sample frames, then go back to (1).
for lower latency, you'd do:
1) snd_pcm_wait()
2) prepare new sample frames
This is my personal preference. In this model the only service ALSA has to
supply are:
1. initial configuration/start/stop.
2. mmapable DMA buffer
3. fact and precise ioctl telling the current HW pointer in the buffer. If the
card is not queried each time, then the last period interrupt timestamp
Sorry, it's not as easy as you've described. It's not possible to invoke
any user code from the kernel code directly. There is a scheduler which is
informed that a task has been woken up. It depends on scheduler when the
task is really invoked. It's quite same as for the r/w model where the
Am I sending these queries about the operation of alsa through the API
to the right place ? I'm trying to use alsa for a real scientific
application and I'm starting to worry it simply isn't ready yet.
the API is ready. the documentation is not. some of us have been using
ALSA to develop
Ok, it's only simple example, that there are more solutions than Paul
suggests. I fully agree, that the callback model is suitable for the
perfect synchronization among more applications.
Let's be totally clear about this. its not just that the callback
model is suitable - the mserver model
Ok, thanks for the line out help, and I think I understand how the
matrix mixer works, and I can hear stuff through my monitors, but now I
can't figure out how to get alsa input or output functioning. I can
plug in my mic/pre and hear that through the monitors, and I can use the
oss compatability
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