Hi,
Realtek ALC650 is supported - but this is only analog part of sound
system, you need second part working - and this is soundstorm. But
Soundstorm is not supported very well. (Maybe i810 module is working
with it - I don't know, if yes, then only basic functions are
supported - no dolby
I'm running kernel 2.4.22 and alsa-0.9.7c on a Dell laptop, the driver used is
intel8x0.
At the time of suspend, if the snd-pcm-oss module is in use, then resume will cause X
server to crash and I get an oops in syslog. Suspend/resume has no problem when
snd-pcm-oss is not in use. I have tried
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
Hi all,
What would be an appropriate ALSA way of implementing a two-way
digital crossover PCM output? What I would like is for an ALSA PCM
device to provide two SPDIF outputs with high- and low-band audio for
the left and right channels, respectively. I'd like for this to be
transparent to the
I'm running kernel 2.4.22 and alsa-0.9.7c on a Dell laptop. I have problem on
suspend/resume when alsa modules loaded. I have done a few testings, below is my
findings:
If the snd-pcm-oss module is in use(by arts sound server) at the time of suspend,
kernel
oops upon resuming and X server
p z wrote:
Good luck.
When I told them my nForce board supported hardware mixing, based
upon
nVidia's documentation, I was told, flat out, I was wrong...
I hate to sound like a conspiracy theorist, but I think it was easy
for
them to patch up the i810 module to support the nForce, and
James Courtier-Dutton wrote:
I have 2 sound cards.
To output to front speakers of card0 I can do: -
front
or
front:0
For Card1
front:1
But for iec958 on Card1 if fails.
What name should I use for Card1 for IEC958 spdif digital out?
conf.c: parse_args() Parameter AES3 must be an
Anyone know the ETA for this driver?
--
Patrick Shirkey - Boost Hardware Ltd.
Http://www.boosthardware.com
Http://www.djcj.org - The Linux Audio Users guide
Being on stage with the band in front of crowds shouting, Get off! No!
We want normal music!, I
I have add code below to example program latency.c:
It just wrote first 10 samples to the file rawout in plain text
format.
I have simple program which helps me to analyse such files.
With that program I've found that some samples are lost (~ one per 4000)
(My settings: latency_min =
At Mon, 20 Oct 2003 15:48:55 +0800,
Gou Zhuang wrote:
I'm running kernel 2.4.22 and alsa-0.9.7c on a Dell laptop. I have problem on
suspend/resume when alsa modules loaded. I have done a few testings, below is my
findings:
If the snd-pcm-oss module is in use(by arts sound server) at the
snd_pcm_playback_drain() holds card-power_lock until the substream
has been drained completely. This prevents any other stream on that
card from preparing during this time (which is very annoying if one is
playing several small files which fit entirely into one buffer on a
multi-stream card like
Hello,
Question 1:
How do create a pointer and allocate memory to variables of type
snd_seq_event to generate events by software? The function
snd_seq_create_event doesn't exist anymore...
Question 2:
snd_seq_event * event;
snd_seq_event_input( handle, event );
This allocates memory, but how
Hi,
I am trying to build an application using ALSA, and which requires low
playback delays.
My first idea was to have a small hardware buffer to limit the playout
delays. But then if for some reason the buffer gets full, the card stops,
and the next writei will block for about one second.
My
James Courtier-Dutton [EMAIL PROTECTED] writes:
The nforce motherboards actually have 2 audio PCI devices. One is
the codec controller, which uses the alsa snd-intel8x0. It works ok
in linux apparently, but I am not sure about whether it has
multiopen/hardware mixing features or not as I
All,
We've noticed a problem with the newest version of Alsa while doing some
stress testing of our devices. Basically, the scenerio is we're playing
through a large library of .ogg files over the weekend. When we checked
on the progress upon getting in on Monday, we discovered that the PCM
The left front channel is always mute when the M-Audio Revolution
(ice1724) driver is started/restarted with all mixer settings saved
on exit. The other channels are fine.
Moving the DAC volume slider in alsamixer unmutes this channel.
---
hi,
last weekend i grabbed my old aureal vortex 2 card and replaced my
current (lowcost) card with it. and after installing the brand new
alsa-0.9.7c i got a fully functional soundcard with full duplex and
hardware mixing. whoooho! i was in heaven... for about 30 seconds. then
my via kt133
On Mon, 2003-10-20 at 01:47, Nick Arnold wrote:
Yes I think so, but I think it'd be easier to look at the recordings at a
sample-by-sample level. The process described in this follow-up post is
pretty much what we've been doing:
From: [EMAIL PROTECTED]
To: Alsa-Devel [EMAIL
On Fri, Sep 26, 2003 at 11:41:38AM +0200, Takashi Iwai wrote:
At Thu, 25 Sep 2003 22:46:37 +0200,
Werner Schweer wrote:
Meanwhile i checked the us122 alsa patch from martin-langer
(http://www.langerland.de/us122/driver.html) with no effect.
Hi,
there's a new version of the us122
Hi Guys,
I have been using the alsa api for a while working on some multimedia
stuff.
I believe I have found a bug with the intel8x0 driver when running the
nforce chip-set.
The problem appears to be an uninitialized variable type problem. The
first time you call snd_pcm_mmap_begin() the
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