[Alsa-user] Fwd: No Analog headphone with Realtek ALC898 in Laptop (MSI GS40)

2017-01-25 Thread Cristiano Rodrigues

Hi there!  Do you know if there is any news on this topic? I have the same
laptop and the same problem.  Please, find in the attached file a
alsa-info.sh output. It wold be very nice if we could find a solution for
this. Thank you very much for your time.  hi,  I'm having a problem
with the automatic codec configuration of my card.  Everything looks good
until I plug the headphones .  Speakers keep sounding and the Sound of the
Headphones is Unbearable loud,   it is like the headphone is not detected
but theres sound and it is horrible.   I did a codec dump following some
guides and it looks like the only recognized use for the headphones Jack
node is SPDIF. Meaning there isn't any headphone device working.  But , if
i change to "Digital Output"  the sound gets ok in the headphones (i would
say really good sound indeed).  Laptop has some kind of AMP (Sabre HIFI)
In other OS with proprietary drivers (Win...) it ask me everytime (With a
pop Up) what device i'm plugin.. Hifi Headphones/ speakers/ or SPDIF so
it's clear that the jack shares the conector of 3 types of outputs.  First
time ever i see somthing like this. It is like the Digital SPDIF and the
headphone jack are hardwired internally and the priority is SPDIF. Worst
engineering decision  ever  i can provide whatever log or dump it needs to
help to "fix" this thing.   i just want the headphone to stop the speaker
and change the device to HEADPHONE when i plug it in the jack. I don't care
about the AMP/DAC or anything else.  As far as I know this is a common
problem with recent MSI laptops .   thank you in advance.


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alsa-info.txt.XlleCuH39s
Description: Binary data
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Re: [Alsa-user] CD format: Sample format non available ???

2017-01-25 Thread chris hermansen
Sean and list,

On Jan 25, 2017 09:07, "sean darcy"  wrote:

On 01/23/2017 07:44 PM, chris hermansen wrote:
> Sean and list
>
>
>
> On Jan 23, 2017 16:06, "sean darcy"  > wrote:
>
> alsa-lib-1.1.1
>
> aplay -l
>  List of PLAYBACK Hardware Devices 
> card 0: DragonFly [AudioQuest DragonFly], device 0: USB Audio [USB
> Audio]
>Subdevices: 1/1
>Subdevice #0: subdevice #0
> .
>
> [video@sixcore]$ aplay -D hw:0,0 -v   ~/2-15_Route_66.wav
> Playing WAVE '/home/video/2-15_Route_66.wav' : Signed 16 bit Little
> Endian, Rate 44100 Hz, Stereo
> aplay: set_params:1297: Sample format non available
> Available formats:
> - S24_3LE
>
> Any help appreciated.
>
>
> Your DragonFly looks like mine, only accepts 24 bit word length.
>
> Use plughw rather than he.
>
>

Thanks for the quick repsonse.


Most welcome!


Now I can't get alsa to use the sync resampler.

aplay -v -D plughw:1,0 2-15_Route_66.wav
Playing WAVE '2-15_Route_66.wav' : Signed 16 bit Little Endian, Rate
44100 Hz, Stereo
Plug PCM: Linear conversion PCM (S24_3LE)



Sean, what is it you are trying to do?  Are you just trying to convert the
16 bit to 24 bit?  Or are you trying to upsample as well?

If just 16 -> 24, can you explain why you want something other than linear?
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Re: [Alsa-user] CD format: Sample format non available ???

2017-01-25 Thread sean darcy
On 01/23/2017 07:44 PM, chris hermansen wrote:
> Sean and list
>
>
>
> On Jan 23, 2017 16:06, "sean darcy"  > wrote:
>
> alsa-lib-1.1.1
>
> aplay -l
>  List of PLAYBACK Hardware Devices 
> card 0: DragonFly [AudioQuest DragonFly], device 0: USB Audio [USB
> Audio]
>Subdevices: 1/1
>Subdevice #0: subdevice #0
> .
>
> [video@sixcore]$ aplay -D hw:0,0 -v   ~/2-15_Route_66.wav
> Playing WAVE '/home/video/2-15_Route_66.wav' : Signed 16 bit Little
> Endian, Rate 44100 Hz, Stereo
> aplay: set_params:1297: Sample format non available
> Available formats:
> - S24_3LE
>
> Any help appreciated.
>
>
> Your DragonFly looks like mine, only accepts 24 bit word length.
>
> Use plughw rather than he.
>
>

Thanks for the quick repsonse.

Now I can't get alsa to use the sync resampler.

aplay -v -D plughw:1,0 2-15_Route_66.wav
Playing WAVE '2-15_Route_66.wav' : Signed 16 bit Little Endian, Rate 
44100 Hz, Stereo
Plug PCM: Linear conversion PCM (S24_3LE)


even though:

cat asound.conf
#
# Place your global alsa-lib configuration here...
#
defaults.pcm.rate_converter "samplerate_best"

and

ls /usr/lib64/alsa-lib/
libasound_module_conf_pulse.so
libasound_module_ctl_oss.so
libasound_module_ctl_pulse.so
libasound_module_pcm_oss.so
libasound_module_pcm_pulse.so
libasound_module_rate_samplerate_best.so
libasound_module_rate_samplerate_linear.so
libasound_module_rate_samplerate_medium.so
libasound_module_rate_samplerate_order.so
libasound_module_rate_samplerate.so
smixer

What am I missing this time ?

sean


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