[Alsa-user] VT1708 HD audio limitations?

2008-02-19 Thread Demian Martin
I have been trying to get this to play HD audio files at 24 bit depth and it
won’t at any sample rate including 44.1 KHz. The driver seems to be 16 bit
only but the chipset is capable of 24 bit audio. How do I configure
asound.conf or tweak the driver to resolve this? (It’s a similar but not
identical problem to the one I’m having with the ICE1724)

 1audio


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[Alsa-user] High sample rate audio problems with Juli@ ICE1724

2008-02-19 Thread Demian Martin
I’m trying to configure Alsa to send high definition audio files, 2 channel,
176.4 KHz/24 bit depth to the digital output of the card. I know the
hardware can do it correctly, I have confirmed that in Windows and confirmed
the stream with the external DAC. 

 

In Alsa no matter what I try the stream gets down converted to 88.2 KHz. Is
this an Alsa limitation or a driver issue? How do I work around this? I
don’t need dmix since it’s a single purpose music player and noting but the
selected stream should come out of the box. A direct to hardware route with
be the best option.

 

1audio


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Re: [Alsa-user] Need help with asound.conf

2008-02-19 Thread Adam Nielsen
>> For me, hw:0,1 is the SPDIF connector, so I just put this in
>> /etc/asound.conf:
>>
>>   defaults.pcm.card 0
>>   defaults.pcm.device 1
>>   
> What if I need a more generic version of asound.conf so it works without
> changes on machines where spdif is hw:0,2 ?

I'm certainly no expert, but to do this you may need to find the config
file in /usr/share/alsa/ that implements dmix (perhaps pcm/dmix.conf) or
anything that refers to defaults.pcm.card (or defaults.pcm.dmix.card)
and change that to pcm.spdif or whichever device you want to become the
default.

>> You can also set this if you want something other than 48kHz:
>>
>>   defaults.pcm.dmix.rate 48000
>>   
> I'm trying to make sense of the files in /usr/share/alsa and extrapolate
> the language so I can understand the flow and write an entire
> asound.conf script, but I was thinking there's a simpler way to do this,
> since the "spdif" pcm is already pointing to the right device.

Changing all references from the default PCM device to the SPDIF device
should work, but again, I'm not an expert at this :-)

Cheers,
Adam.


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Re: [Alsa-user] Need help with asound.conf

2008-02-19 Thread Radu Cristescu

Adam Nielsen wrote:

The problem is that I can't hear two concurrent audio streams at the
same time when using the default device. I tried setting things up using
dmix, but failed miserably. Any ideas what to do to make two
applications play simultaneously through spdif based on this
configuration? (It's mainly to get Flash to play through S/PDIF).



The default ALSA config dmixes everything at 48kHz so you shouldn't need
to set that explicitly.  All you should need to do is tell dmix to
output via SPDIF.

  
I wrote this asound.conf because for some reason ALSA used 44.1KHz and 
the sound card was locked to 48KHz, resulting in a very annoying 
artifact. The idea was to force it to covert everything to 48KHz when 
using spdif.

For me, hw:0,1 is the SPDIF connector, so I just put this in
/etc/asound.conf:

  defaults.pcm.card 0
  defaults.pcm.device 1
  
What if I need a more generic version of asound.conf so it works without 
changes on machines where spdif is hw:0,2 ?

You can also set this if you want something other than 48kHz:

  defaults.pcm.dmix.rate 48000
  
I'm trying to make sense of the files in /usr/share/alsa and extrapolate 
the language so I can understand the flow and write an entire 
asound.conf script, but I was thinking there's a simpler way to do this, 
since the "spdif" pcm is already pointing to the right device.

Make sure you use Flash >= 9 for ALSA support.
  

Of course :) I'm using Ubuntu 7.10 BTW.

Thanks,
   Radu
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[Alsa-user] Step-by-step configuring/testing a sound card

2008-02-19 Thread oxy
Hi all,
geting my sound card work has been confusing for me.
There are many forum threads, but none helps. More, maybe i've
messed up with the good config part following those instructions, who knows ...
Can somebody tell me a "linear" step-by-step method of
configuring/testing a sound card?

In my Linux box i do:
root%aplay -vv song.wma
   ALSA lib pcm_dmix.c:864:(snd_pcm_dmix_open) unable to open slave
   aplay: main:550: audio open error: Device or resource busy

My system: Debian 2.6.18.dfsg.1-18etch1

root%cat /proc/asound/card0/codec#* | grep Codec
   Codec: Generic 11c1 Si3054
   Codec: Analog Devices AD1986A

root% lspci | grep Audio
00:14.2 Audio device: ATI Technologies Inc SB450 HDA Audio (rev 01)

root%lshw
 ...
 description: Audio device
 product: SB450 HDA Audio
 vendor: ATI Technologies Inc
 physical id: 14.2
 bus info: [EMAIL PROTECTED]:14.2
 version: 01
 width: 64 bits
 clock: 33MHz
 capabilities: bus_master cap_list
 configuration: driver=HDA Intel latency=64
 resources: iomemory:d040-d0403fff irq:209


root% lsmod|grep ^snd
snd_hda_intel  17332  1
snd_hda_codec 137856  1 snd_hda_intel
snd_pcm_oss38368  1
snd_mixer_oss  15200  1 snd_pcm_oss
snd_pcm68676  3 snd_hda_intel,snd_hda_codec,snd_pcm_oss
snd_timer  20996  1 snd_pcm
snd47012  6
snd_hda_intel,snd_hda_codec,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer
snd_page_alloc 10184  2 snd_hda_intel,snd_pcm


Many questions like this can be found, but no solution (at least for
me). Thanks.

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