#941: No sound from Ice1712 chipsets - can help so it can maybee work
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Reporter: mith
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Owner: lennart
Type: defect
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Status: new
Milestone:
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Component: module-alsa-*
Keywords: Digigram VX442 Hoontech Soundtrack Audio DSP 24 IC Ensemble Generic
device M-Audio Midiman MAudio Delta 44 66 410 1010 1010LT 1010-LT
Audiophile 2496 DiO 2496 Seasound Solo Sonorus Medi/o Terratec
EWS88D EWS88MT EWX24/96 DMX 6Fire Event Electronics EZ8 ALSA Pulseaudio
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Hi, I have no sound from with my M-audio Delta 1010 LT soundcard from
fresh install in ubuntu (eighter one of these) 10.04, 10.10 and 11.04.
Not working from fresh install in ubuntustudio 10.04, 10.10 also.
I have reported this bug to Ubuntu team on theirs Launchpad site but with
no luck resolving this as an permanet fix.
As I have understand, You develop Pulseaudio and not Ubuntu as I first
believed (im new to Linux) and I think its more important to report the
bug here so it can be fixed. And if it gets fixed here, it will apply to
all Linux distros using Pulseaudio/Alsa.
Actually Ice 1712 chipsets are on a lot of soundcards so if this will get
fixed A LOT of soundcards and their users will get working sound.
Ive managed to follow some guides when ive searched myself over the net
how to get audio working with this card and it worked.
Now im reporting this so it can be implemented so it can be working again
with all soundcards using Ice 1712 chipset (now date I know about 17
soundcards that will work if this will be fixed - thats some!)
I myself are an soundtechnican and musican doing music on the computer so
I have knowledge in how to set the default mixer for best results and also
some tips about setting the right latency.
As I have discovered it can get to work but with audio glitches, have you
ever thought of increasing the buffer size that the soundcard uses? If its
to low, of course the sound will glitch.
I would recommend the following: If you only uses the computer not in
making music etc, you can set a high buffer for the soundcard.
Its only when you are making music you are in need of a low-latency
responsive system. otherwise you can set the buffer size to 512, 768 or
1024.
The slower the computer and older, the better to have an high latency
value. Because if the computer is to slow the audio will glitch. Therefore
its good? to set a high value like 512 to have older computers able to
playback sound.
As said, its only usefull to have a low latency when you are working with
music applications or videoediting etc. But the normal user dont do music
on the computer. I think its better to have sound working with low latency
than not working at all.
512 as a value should be fine, 768 as buffer size to help really older
computers (before 2002 etc) to work and 1024 for even older..
Just sharing my thoughts on this.
I can contribute to this in order to try making it work if you help me
what to do so you can use it later in future Pulse audio releases that
many Linux distrubitations use so many can get some sound..
So here I am willing to help make this work!
Thankful for co-operation!
With best regards
Mikael T.
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Ticket URL: <http://pulseaudio.org/ticket/941>
PulseAudio <http://pulseaudio.org/>
The PulseAudio Sound Server
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