Hi everybody,
I have a question for you guys.
(I'm new to this list and hope this is the right place to ask
and not an FAQ... I've tried to search to see if this question
had been answered previously, but I have been not able to find
what I was looking for).
I'm thinking about using a
Hi everybody,
first of all, I'd like to thank you all so much for the prompt
and many replies I've got... Wow, they came in faster then I was
able to read! :-)
(there's still someone claiming that Linux has no support?! :-)
On Wednesday 21 November 2007 22:19, Sergei Steshenko wrote:
On Thursday 22 November 2007 11:39, Rene Herman wrote:
I suppose your external DAC has no actual WordClock (BNC connection) output?
indeed, it does not. As it does not have an SPDIF output, either...
but I'm a DIY guy (with an EE degree...) and can add whatever output
I need or even build a
On Wednesday 21 November 2007 23:49, Bill Unruh wrote:
The clock jitter tends to be in the ppm range. This means that the
frequency jitter is very low (if I believe the ppm then at the level of
-120dB)
which is completely inaudible. My cheap Transit card reliably gives me noise
On Thursday 22 November 2007 21:01, Vladimir Mosgalin wrote:
I meant M-Audio Revolution 7.1, and quite possibly M-Audio Revolution 5.1
allow you to use external clock source.
In theory, yes. In practice, I wasn't able to make my M-Audio Audiophile
USB get clock from external source. Well
On Tuesday 27 November 2007 10:08, mike wrote:
sound works on the most applications only the flash plug in for my
firefox, there is like mute. it's unhelpful to load the OSS-emulation.
(yesterday it was also a sdl based game without sound. but today it
works(i'm wondering)).
anyone a idea?
Hi everybody,
after your kind suggestions, I have done some search in the pro
and semi-pro sound card arena to find a suitable and affordable
card for my needs.
In the end, I have found a few possible candidates. According to
the manufacturer specs, all of them are capable of bitperfect
Hi Chris,
On Tuesday 27 November 2007 18:59, Chris Stranex wrote:
I can vouch for the Juli@ working on linux. I've used it before and
it's been okay EXCEPT the fact that any use of the midi port (even
connecting something to it with aconnect or qjackctl) caused my system
to freeze up. Other
On Friday 30 November 2007 13:15, Sergei Steshenko wrote:
I am sorry, but all the above said can be summarized in very few sentences:
1) any phase/frequency modulation produces a signal with infinite
spectrum;
[...]
well, sure... as far as one knows what all that means. I tried
to explain
ouch, some mails from the list get caught in the spam filter.
This will be a very late answer... :-(
On Thursday 22 November 2007 20:49, you wrote:
Just a random thought on jitter - if you like music recorded originally
before the digital era - don't bother.
I.e. analog tape recorder wow
On Friday 30 November 2007 18:18, Bill Unruh wrote:
Nevertheless, from a perceptual point of view, reasonable amounts
of analogue artifacts (modulation) seems to be much more benign
to our ears (brain, actually) than most currently common levels of
digital artifacts.
Your evidence for
On Saturday 1 December 2007 02:15, Rene Herman wrote:
Hey, don't apologise, I'm enjoying these messages. I do absolutely need to
comment on that last bit there though as all the hype I'm _ever_ hearing is
quite the other way around -- how old analogue is so obviously better then
anything
On Tuesday 18 December 2007 02:02, Tom Lanyon wrote:
cable. I can play audio files using DTS and AC3 passthrough direct to
the receiver without issue. However, playing non encoded audio (MP3s,
FLAC audio, etc) is distorted through all volume levels. It also
[...]
A quick follow up to my
On Wednesday 19 December 2007 03:00, Tom Lanyon wrote:
Interestingly, when I use mplayer to decode my DTS test file and play
via SPDIF, it comes out distorted. If I tell mplayer to use DTS
passthrough on the same file, it sounds perfect.
I tested playing the same files as decoded PCM
On Wednesday 19 December 2007 12:03, you wrote:
I tried using 'plughw' but mplayer tells me it can't find the plughw
lib:
[AO_ALSA] alsa-lib: pcm.c:2105:(snd_pcm_open_conf) Cannot open shared
library /usr/lib64/alsa-lib/libasound_module_pcm_plughw.so
mmmh... do you have a 64bit
On Tuesday 15 January 2008 01:40, Tom Enderlin wrote:
Tried moving all the mic and mic boost sliders in alsamixer to their max
settings and it still didn't record any audio, just background static.
I also use Debian. Some time ago, after some upgrade I have experienced
similar problems with
Hi everybody,
I'm in the process of putting together a new PC which will be
used both as a workstation and HTPC.
For the case, I was thinking about the Antec Sonata 550 Plus,
which is supposed to be good and quiet (nonetheless, any other
suggestion is welcome :-). As for the CPU, I was
On Wednesday 12 March 2008 19:32, Nigel Henry wrote:
On my Debian installs I've had to add my user name to the audio group to get
audio apps working as user, whereas on my Fedora installs there is no audio
group in /etc/group, and users being able to use audio apps normally only
accessable
On Friday 13 June 2008 05:41, Philippe MONROUX wrote:
I'm running debian etch with 2.6.18-6-686 kernel.
I have a asus p5kc mother board
lspci give:
00:1b.0 Audio device: Intel Corporation Unknown device 293e (rev 02)
...
So I tried to load snd-hda-intel without and with this
On Saturday 14 June 2008 15:40, Philippe MONROUX wrote:
Lenny don't solve the problem...
try kernel 2.6.25 from Sid. You can download the linux-image-,
linux-headers- and linux-kbuild- .deb files and install them
with dpkg -i ... on Lenny without dependency problems.
Current Lenny's Kernel
On Monday 16 June 2008 17:33, Paolo Saggese wrote:
Current Lenny's Kernel (2.6.24) provides ALSA 1.0.15, while
the 2.6.25 provides 2.6.25 (when Lenny will be released this
oops... of course I meant 2.6.25 provides ALSA 1.0.16.
Ciao,
Paolo.
--
Skype
On Saturday 02 August 2008 09:35, Mikhail Ramendik wrote:
I have debian etch, kernel 2,6,18, on a SiS964 motherboard. I want to use
spdif output.
It was working great for many months. However, recently (possibly after a
kernel security update) SPDIF output has stopped working.
mmh...
On Friday 22 August 2008 23:44, James Shatto wrote:
mplayer -dumpaudio -dumpfile sound_track.wav ./source.vob
Granted that my default distro supplied version of mplayer didn't work for
this.
I had to compile a custom version of mplayer from source. At least that's
how it
was for
On Wednesday 06 May 2009, Niels wrote:
- This is with Kubuntu 9.04. For some reason it ships with ALSA 1.0.18rc3.
Is it worth upgrading, and how would I do that, other that wait for an
automatic update?
don't know wether it may be worth it. You may check for yourself. Plug in
your vdac, open
On Tuesday 12 May 2009, Scott Barlow wrote:
I cannot get the HDMI audio to work with my ASUS M3N78-EM motherboard! I've
To get audio over HDMI may be not so straightforward at times, 'cause
it depends on a number of different things... not necessarily related
to the audio system (ALSA).
On Wednesday 13 May 2009, Takashi Iwai wrote:
BTW, did you install alsa-plugins-pph or so?
It's a resampler from speex, and it's usually much faster and good
enough.
upsampling to 24/192 (to an hi-end external DAC connected
via I2S from a Juli@) IME speexrate_best actually sounds
even
On Wednesday 13 May 2009, Scott Barlow wrote:
setup correctly. When using HDMI, it would not be displayed correctly, then
I guess you've already tried to fix that problem using nvidia-settings
and nvidia-xconfig before playing with the EDID, right? 8-)
when using VGA it would be. I used
On Wednesday 13 May 2009, Grant wrote:
defaults.pcm.rate_converter speexrate_best
in /etc/asound.conf and I restarted alsa, but I still have the static problem.
what do you have in your ~/.asoundrc ?
usually that's the best place to put your customizations.
(BTW: in Debian /etc/asound.conf
On Thursday 14 May 2009, Grant wrote:
I added this to /etc/asound.conf:
pcm.!default {
type plug
slave.pcm {
type dmix
ipc_key 1024
slave {
pcm hw:0
format S24_3LE
rate 96000
}
}
}
are you sure S24_3LE is the proper format for your card?
even if it is a 24-96 card, it may require
On Thursday 14 May 2009, Grant wrote:
Nevertheless, I guess these static problem is not related
to the resample algorithm you are using - unless the problem
is related to insufficient system resources. What CPU do you
have?
I'm using an AMD64 Athlon 3.1ghz CPU.
that one should have
On Thursday 14 May 2009, Grant wrote:
I've tried S24_3LE, S24LE, S24_BE, S24_3BE, FLOAT_LE, and FLOAT_BE.
None of them produce sound except for S24_3LE. S16_LE works, but
stills suffers from the static problem. Is there another format I
should try?
try the U (unsigned) ones... i.e. U16_LE,
On Friday 15 May 2009, Grant wrote:
So how come that you don't get 'em with mpd?!
In mpd I specify oss instead of alsa. If I specify alsa and involve
dmix, I get static in mpd too.
you say OSS... but was that the real oss (does it still exist for
modern kernels/hardware?!) or was it just
On Saturday 16 May 2009, Grant wrote:
If it's the latter (as usually is on modern systems), then its just
the input interface that changes, you're goin' trough ALSA anyway.
Also, I should mention something that contradicts this. When I define
a format for dmix in /etc/asound.conf that my
On Monday 18 May 2009, Johan Bakker wrote:
The problem is that i cannot output sound to HDMI and SPDIF simultaniously.
I tried .asoundrc with the copy and multi plugin but had no luck.
Is it possible to play sound on both devices simultaniously?
in principle it should be. Have you tried
On Tuesday 19 May 2009, Johan Bakker wrote:
But now everything is resampled. My receiver is getting my music at 48kHz now
(was 44.1kHz).
For my tv this is no problem. But i would like unchanged output on my
receiver (spdif). Is this possible? (Maybe i want too much)
I don't know (I'm NOT
Hi everybody,
in the last days I'm getting multiple (and I mean hundreds!) of
duplicates for some of the posts in this list.
Does it happen only to me (?!) or is there some problem with the list?
Ciao,
Paolo.
--
On Friday 22 May 2009, FabrÃcio Nihues wrote:
Front analog out works, could not play any dts/ac3 to analog outs
because was getting garbled sound,
I'd say of course... you'd need to decode that in order to play
it on anything but an A/V receiver supporting those formats (or
possibly one of
On Saturday 23 May 2009, FabrÃcio Nihues wrote:
The light of optical is on now... but don't send sound again... I'm
sending to iec958:CARD=default,DEV=0 or spdif:CARD=default,DEV=0, on
both I don't hear sound. And, if I change the receiver to my analog
try plughw:0,1 as the spdif out device.
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