Hi,
If I have a USB soundcard with say 8 channels, how do I route channels 7+8 to a
stereo capture device?
Regards
/R
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Setting
options snd-usb-audio index=5
in alsa-base.conf seems to do the trick, thanks!
Regards
/R
-Original Message-
From: Ralf Mardorf
Sent: Sunday, 22 November 2020 16:10
To: alsa-user@lists.sourceforge.net
Subject: Re: [Alsa-user] USB ALSA card number
Hi,
I'm using the below
I have a system where I need USB attached audio devices to start numbering from
ALSA card5 and upwards (i.e. card0 to card4 are reserved). Is this possible?
Regards
/Robert
The information in this email (including any attachments) may contain
confidential and/or proprietary material. Any
Hi Clemens,
> The dmix plugin plays shared-memory tricks with the ring buffer and therefore
> requires to run on top of a hw device.
> It might be possible to run dmix on a virtual loopback sound card and route
> that one to the LADSPA plugin, but I guess using PulseAudio would be a better
>
I’d like to know if there is any way to get the following sound chain through
ALSA:
Mediaplayer -> dmix -> LADSPA -> plughw:0,0
For my application it is crucial that the LADSPA plugin be applied AFTER dmix.
Regards
/Robert
The information in this email (including any attachments) may contain
On a similar note, the sample rate seems to be specified when instantiating the
PCM plugin, but I’m not aware if frames per buffer is ?
From: Robert Bielik
Sent: Monday, 18 May 2020 10:39
To: alsa-user@lists.sourceforge.net
Subject: ALSA PCM plugin lifetime
Hi all,
I have a system setup where
Hi all,
I have a system setup where I need a post-processing ALSA PCM plugin. But how
is the lifetime managed by the ALSA server? Will the plugin only be
instantiated whilst there is an audio stream running?
Is it possible to setup an ALSA configuration that keeps the PCM plugin
instantiated,
> So I guess its a ALSA version issue ☹
ALSA version where hints work is 1.1.3 and where they don't version is 1.0.29.
/R
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Whereas on the target platform:
Advanced Linux Sound Architecture Driver Version k4.1.25.
So I guess its a ALSA version issue ☹
Regards
/R
>
> Regards
> /Robert
>
> > -Original Message-----
> > From: Robert Bielik
> > Sent: den 6 september 2018 14:46
> &g
orge.net
> Subject: Re: [Alsa-user] List devices
>
> Robert Bielik wrote:
> > I cannot get it listed with ALSA API function snd_device_name_hint.
>
> This is undocumented; see the "hint" entries in
Hi,
I have a dmix plug that's connected to hw dev 1 (/etc/asound.conf):
pcm.i2s_play {
type dmix
ipc_key 1024
ipc_key_add_uid 0
slave {
pcm {
type hw
card 1
}
Hmm... I was a bit too fast there...
> aplay -L
> null
> Discard all samples (playback) or generate zero samples (capture)
> pulse
> PulseAudio Sound Server
Can you try playing through pulseaudio with:
aplay -D pulse test.wav ?
Regards
/R
Yó napot kivánok!
Take a look at https://alsa.opensrc.org/Dmix , dmix is the ALSA plugin you
should use for this.
Regards
/Robert
> -Original Message-
> From: Csányi Pál [mailto:csanyi...@gmail.com]
> Sent: den 10 februari 2018 12:02
> To: Alsa User
> I'm trying to reorder my soundcards on a RPi so that the I2S based cards
> always is index zero. I looked at the docs
> (https://alsa.opensrc.org/MultipleCards), which just says, "easy peasy, just
> use options snd slots=this, that". Problem is that nowhere is it documented
> WHAT "this, that"
Hi,
I'm trying to reorder my soundcards on a RPi so that the I2S based cards always
is index zero. I looked at the docs (https://alsa.opensrc.org/MultipleCards),
which just says, "easy peasy, just use options snd slots=this, that". Problem
is that nowhere is it documented WHAT "this, that" is!
Would the dshare plugin do this for you ?
http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html
Regards
/Robert
> -Original Message-
> From: Samuel Nicholas [mailto:nicholas.sam...@gmail.com]
> Sent: den 1 februari 2018 22:09
> To: alsa-user@lists.sourceforge.net
> Subject:
I want to setup a system where JACK manages a low latency path for audio
in/out, and where ALSA apps use the ALSA JACK PCM plugin
(http://jackaudio.org/faq/routing_alsa.html)
Is it possible on the ALSA side to setup dmix to use the "jack" type plugin as
backend ? Or will the jack plugin do the
> > is there some ALSA plugin that can coalesce buffering ? Meaning that
> > the plugin can take f.i. larger period_size than what the dmix device
> > is working with ?
>
> What problem would that solve?
Not sure. It would allow clients connecting to that device to have a more
relaxed callback
Hi Clemens,
> >> Is there any other plugin doing the same thing as dmix... but working ?
>
> Yes, dmix with a larger buffer size (i.e., more periods).
Hmm. Yes. I've tried running aplay with "chrt --rr 99" and it got way more
stable, even at as low a settings as 64 frames (period_size) + 2
> The reason is that for my project I need to have as low a latency as possible
> in
> the dmix chain. Is there any other plugin doing the same thing as dmix... but
> working ?
More specifically, I'd need a mixing plugin that does not do sample rate
conversion, i.e. each client connecting to
Hi, I'm using the audioinjector octocard on a R Pi 3, and I have a problem
where the system default dmix (dmix:0,0) plays just fine (via aplay), but my
own defined dmix device occasionally stops streaming with a xrun condition:
Status(R/W):
state : RUNNING
trigger_time: 13953.124684
);
playback.putBuffer(out_buffer);
}
}
Regards
/R
> -Original Message-
> From: Robert Bielik
> Sent: den 15 januari 2018 17:59
> To: alsa-user@lists.sourceforge.net
> Subject: RE: Strange i/o problem
>
> Yet more info, the output of snd_pcm_hw_params_dump and
>
age-
> From: Robert Bielik
> Sent: den 15 januari 2018 17:47
> To: Robert Bielik <robert.bie...@dirac.com>; alsa-user@lists.sourceforge.net
> Subject: RE: Strange i/o problem
>
> Ah, forgot to mention a couple of things, this is on a Raspberry Pi 3 with
> Raspbian Stretc
, 666, 722, 2.2253
min, mean, max, stddev: 656, 666, 680, 1.57805
min, mean, max, stddev: 643, 666, 683, 1.54424
(which to me looks more than OK)
> -Original Message-
> From: Robert Bielik [mailto:robert.bie...@dirac.com]
> Sent: den 15 januari 2018 17:41
> To: alsa-user@lists.sou
I have a strange problem: I'm trying to pipe audio input -> output using a I2S
device @48000 Hz and 32 frames buffer size and 2 periods, to get as low a
latency as possible.
It works nicely if I either:
1. Use capture + playback and record capture to a wav file (sounds fine).
2. Use playback
Ok, hehe... found the problem, I was running gdbserver as root so it was the
wrong .asoundrc I changed...
> -Original Message-
> From: Robert Bielik [mailto:robert.bie...@dirac.com]
> Sent: den 8 januari 2018 10:30
> To: alsa-user@lists.sourceforge.net
> Subject: [Alsa-
I've come across an odd behavior: If I add a dummy pcm in .asoundrc :
pcm.dummy {
type plug
slave.pcm "plughw:0,0"
}
I can see it listed with aplay -L.
However, my own code, which uses the same exact mechanism as aplay does
(snd_device_name_hint) does NOT list the dummy device.
age-
> From: Clemens Ladisch via Alsa-user [mailto:alsa-user@lists.sourceforge.net]
> Sent: den 7 januari 2018 12:25
> To: alsa-user@lists.sourceforge.net
> Subject: Re: [Alsa-user] Problems opening devices
>
> Robert Bielik wrote:
> > After this I try snd_pcm_open on t
Mind you, this works nicely:
> aplay -D default:CARD=MOXF6MOXF8 test.wav
So I guess I must be doing something wrong ☹
/R
> -Original Message-
> From: Robert Bielik [mailto:robert.bie...@dirac.com]
> Sent: den 7 januari 2018 10:16
> To: alsa-user@lists.sourceforge.net
&
Hi all,
I am implementing an ALSA backend for an in-house cross-platform C++ audio
framework, but I have serious problems doing ALSA. My hardware is a Raspberry
Pi 3 running Raspbian Stretch, having an I2S soundcard as the main card. I
enumerate the PCM devices with the snd_device_name_hint
I've used the ALSA LADSPA PCM plugin, and it works nicely. However, I'd like to
use LV2 plugins aswell. Is there such a project active somewhere ?
Rgrds
/R
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Dear Clemens,
Thank you so much for the alsaloop tip, I just ran it with:
> chrt -rr 70 alsaloop -f S32_LE -C plughw:0 -P plug:ladspa -l 48
And it works perfectly, exactly what I needed!
Regards
/Robert
> -Original Message-
> From: Robert Bielik [mailto:robert.bie...@
Hello Clemens,
> Otherwise, you have to do the capture and playback in software. See the
> alsaloop tool. What latency you can reach depends on how much other
> applications and drivers interfere with the scheduling; on the Pi, typical
> culprits are WiFi, ethernet, or USB.
> Also see
Hi all,
I want to route input to output with minimal possible latency, this will run on
a Raspberry Pi, and the latency should be < 1 ms.
I was thinking... if the ALSA capture and playback device is mmapped to the
same buffer area, this should be dealt with automatically. Is this possible ?
Hi all,
Started experimenting using LADSPA plugins with the ALSA LADSPA PCM plugin, and
it works nicely. However, it seems only the PCM interface is exposed from the
plugin.
Question is if there is a way to expose the LADSPA plugin parameters directly
to alsamixer ?
Regards
/Rob
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