James Shatto wrote:
On Tue, 5 Jan 2010 01:07:21 +0300
An St vit@gmail.com wrote:
Hello!
Please help. I can't get working microphone at HDMI output.
HDMI audio normally has some sort of limit in place. For my ATI
HD4550 video card, the audio has to be transmitted in an AC3
John Haxby wrote:
On the other hand, the hda hardware in the machines I've had is
seriously lacking in quality and I'd much rather pay more money and go
for something else, something that sounds nice.
Something which, alas, is rather impractical for 50% of all PCs sold
these days, which are
li...@lazygranch.com wrote:
I'm an ATI fan, but I have to agree Nvidia linux support is better. On Suse,
they has their own repository set up.
Heh. To be honest I always avoid Nvidia for Linux. Intel graphics
first, because they actively support open source Linux drivers, and
push forward
Daren Krive wrote:
I do not mean to be negative but I am seriously wondering. Do most
Linux users simply learn to live without sound? Or live with sound
that does not operate as expected? I have been searching for over a
year on how to get sound working on my LG QB01A9
Bill Unruh wrote:
24 bit quantization error ( being at 135dB below the full signal--
ie completely and utterly inaudible
Cumulative error in the (polyphase or not) digital filter arithmetic
is larger than sampling error. Still small, but how small depends on
how the filters are implemented.
Sergei Steshenko wrote:
On Sun, 21 Dec 2008 03:34:50 +
Jamie Lokier ja...@shareable.org wrote:
Even 2x upsampling introduces distortion.
No, it doesn't.
It can be implemeted as a pure linear transform with constant coefficients,
thus it would be a simple FIR LPF.
It depends
Bill Unruh wrote:
On Sun, 21 Dec 2008, Jamie Lokier wrote:
But then there's people who say that all analogue output above 20kHz
or so is pointless anyway. :-) (I have no opinion either way on this.
On the one hand, they could be right. The upper auditory limit is
well known
Dominique Michel wrote:
Upsampling does nothing for analog reproduction because you cannot get more
informations that what you get from the DAC output. Upsampling is only a
matter
of cost:
higher the ADC frequency, cheaper is the output filter for approximately the
same analog result at
Sergei Steshenko wrote:
The argument about 192 kHz is true, but why not run the DAC at the
'normal' 96 kHz then? I would prefer a 48 - 96 conversion over a 48 -
110 conversion!.
And how about interference: won't a 96 - 110 conversion give any
interference at 14 kHz? Right in the audible
[EMAIL PROTECTED] wrote:
I have some problems with the snd-hda-intel driver. The mixer controls
seem to be unpredictably appearing and disappearing from alsa-driver
version to version.
This does seem to be the bizarrest feature of snd-hda-intel that I've seen too.
Every version has a
Florian Faber wrote:
On Thursday 12 June 2008 20:10:04 Chris Smith wrote:
On Thursday 12 June 2008, Florian Faber wrote:
What makes you think converting a 16 bit unsigned integer to a IEEE
32 bit float and back would change the value?
Should have used a 24 bit example. I'm of the
Bill Unruh wrote:
I definitely agree that you want to avoid this . You might look at the
tests I did with the SOX rate conversion routines. Note that if you are
willing to take a long time, that rate conversion can be good. But it takes
a lot of processing and time-- doing a good job in real
Erik Slagter wrote:
So I guess it's up to you to ignore signals in threads that do not have
a signal handler installed by alsa. I guess you need to make a piece of
code that queries the current signal handler for SIGIO and if it's
non-null, leave it, otherwise have SIGIO ignored and do this
Gene Heskett wrote:
The fall time of that computer generated sawtooth is probably 100x the
fundamental frequency of the sawtooth. I suspect what your ears are hearing
is aliasing because some portions of that exceed the sampling frequency,
which would be non-harmonically related, and
Bill Unruh wrote:
Ok, say I'd be willing to pay 2k max for a complete 5.1 setup
including sound card, amp, and speakers. What would you suggest?
The 5.1 is most likely created by psychoacoustic effects, so it can
always only be simulated 5.1 - without a sub: no .1!
I have no idea what
Bill Unruh wrote:
Just to clarify, the crystal oscillators on the different cards to not run
at the same rate. Thus say one runs at 44100 and the other at 44050. That
means every second, the one card will bring in 50 fewer samples than the
other one. If you remove enough samples to flush the
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