[Alsa-user] No device for DAI tlv320aic23 (ARM,AT91,I2C)
I have problem with tlv32aic23 audio driver.
This codec works with mainboard with ARM AT91SAM9261 processor.
The codec is connected via I2C bus (control) and SSC-I2S (data).
I have got the following result:
++Starting kernel ...
++Uncompressing Linux done, booting the kernel
++Linux version 2.6.28-rc6 (r...@adamr) (gcc version 3.4.2)...
++CPU: ARM926EJ-S [41069265] revision 5 (ARMv5TEJ), cr=00053177
++Machine: Atmel AT91SAM9261-EK
++...
++...
++i2c /dev entries driver
++Advanced Linux Sound Architecture Driver Version 1.0.19.
++No device for DAI tlv320aic23
++...
++...
Some part of my code:
++//board-sam9261ek.c
++static struct i2c_board_info aspel_i2c_devs[] __initdata = {
++ {
++ I2C_BOARD_INFO(tlv320aic23, 0x1a),
++ }, {
++ I2C_BOARD_INFO(tmp101, 0x48),
++ },
++};
++static void __init ek_board_init(void)
++{
++...
++i2c_register_board_info(0, aspel_i2c_devs, ARRAY_SIZE(aspel_i2c_devs));
++...
++}
++//tlv320aic23.c
++...
++static int __init tlv320aic23_modinit(void)
++{
++ return snd_soc_register_dai(tlv320aic23_dai);
++}
++module_init(tlv320aic23_modinit);
++
++static void __exit tlv320aic23_exit(void)
++{
++ snd_soc_unregister_dai(tlv320aic23_dai);
++}
++module_exit(tlv320aic23_exit);
I did not modify ALSA code.
What should I do to run tlv32aic23 driver?
Any other initialization is needed?
Thanks for any help.
Adam R.
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Re: [Alsa-user] How to set USB-Headset volume with udev and amixer
Jens Rutschmann wrote on 24.03.2009 23:00:
Clemens Ladisch wrote on 20.03.2009 08:45:
I don't know why your script doesn't work, but the following udev rule
works for me, and it does not need to be adjusted for the specific sound
device but just requires that you've previously saved the the correct
settings with alsactl store:
SUBSYSTEM==sound, ACTION==add, KERNEL==controlC[0-9]*,
RUN+=/usr/sbin/alsactl restore $number
This one works perfectly.
I found out it did not work as perfectly as I thought in the first place:
Using the rule above made my built-in sound card (HDA Intel, with Analog
Devices
codec, in a Thinpad T61) create a cracking sound when (un)muting it.
So I refined the rule to only restore the mixer levels of the USB headset:
SUBSYSTEM==sound, ACTION==add, ATTRS{idProduct}==000c,
ATTRS{idVendor}==0d8c, KERNEL==controlC1, RUN+=/usr/sbin/alsactl restore 1
Perhaps this might help others.
Best regards,
Jens
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[Alsa-user] Trying to use two sound cards at the same time
Hello
I have two sound cards, one attached to pc speakers and one attached to a
sound system in another room.
Following instructions on the alsa project wiki, I am trying to use two
sound cards (V8237 and CA0106).
I configure my ~/.asoundc but I still can't manage to send a sound to both
cards at the same time. Below is my ~/.asoundrc content.
When I tested directly the ca0106 card (hw:0,0) with a play, it didn't work
so I thought it might be some sort of rate or conversion problem. I added
the pcm.via and pcm.audigy in my asoundrc file and then, when I test each
card individually (via and audigy) with aplay it works well.
When testing the multi pcm with aplay ( aplay -vD multi question.wav ), I
get the following error
aplay: set_params:959: Nombre de canaux non disponible (Channels count non
available)
Any help or suggestions are welcome
PF
~/.asoundrc content
pcm_slave.via_rate {
pcm hw:1,1
rate 48000
channels 2
}
pcm.via {
type plug
slave via_rate
}
pcm_slave.audigy_rate {
pcm hw:0,0
rate 48000
channels 2
}
pcm.audigy {
type plug
slave audigy_rate
}
pcm.multi {
type multi
slaves.a.pcm audigy
slaves.a.channels 2
slaves.b.pcm via
slaves.b.channels 2
bindings.0.slave a
bindings.0.channel 0
bindings.1.slave a
bindings.1.channel 1
bindings.2.slave b
bindings.2.channel 0
bindings.3.slave b
bindings.3.channel 1
}
ctl.multi {
type hw
card 0
}
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