I'm testing a USB DAC that seems to work fine except it does not
produce sound. The only clue I can find is "config 0 descriptor??" in
dmesg below. This DAC needs to be unmuted before it will work but
alsamixer says "This sound device does not have any controls."
# dmesg
...
usb 1-2: new high-sp
Alfonso Arbona Gimeno gmail.com> writes:
>
> No, and that's whats bothering me. All the important packages (kernel,
> drivers, etc) are installed using apt-get and the like (except that
> horrible nvidia driver xD)
>
I also see this. It happened after the Debian 7.1 updates came out
yesterda
> Should 32/384 and 32/358.2 work?
>
> - Grant
Does anybody have an idea on this?
- Grant
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Should 32/384 and 32/358.2 work?
- Grant
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Is there a way to get the latest version of the AV200 virtuoso driver
for the Asus cards? Is this the best place to ask?
- Grant
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David G. Th
>>>>> Can the conversion to S24_3LE be lossy?
>>>>>
>>>>> - Grant
>>>>>
>>>>
>>>> The conversion is to 24 bits. So, if the conversion is from lower
>>>> numbers
>>>> of bits,it's lossl
>> Can the conversion to S24_3LE be lossy?
>>
>> - Grant
>>
>
> The conversion is to 24 bits. So, if the conversion is from lower numbers
> of bits,it's lossless, if from higher - it's lossy.
>
> Regards,
> Sergei.
I think mpd outputs in a 32-b
ault device and the following config:
# cat /etc/asound.conf
defaults.pcm.rate_converter "samplerate_best"
pcm.!default {
type plug
slave.pcm hw
}
This way, if the sample rate, format, or number of channels is not
supported, they are converted, otherwise they are not. If the
lughw or the default device
>> causes conversions to take place. How can I send unaltered digital
>> audio to the DAC?
>
> Well, it's not possible to send those samples to the DAC in totally
> "unaltered" form, as the device simply doesn't support that
m digital
> }
I think I'm starting to figure this out. Please correct me if I'm
wrong, but I think the default device converts both format and sample
rate, and plughw converts only format. This is what I
be bothered with the details of what
> sample formats the device supports and those devices will do automatic
> conversion if needed.
I'm trying to avoid having the computer make any changes to the
digital audio signal whatsoever. I don&
ce
>> "hw:0,0" does not support format 16: Invalid argument
>>
>> When trying to play a 24-bit file instead of 16-bit, "16:" is replaced
>> by "24:" in the error message. The DAC supports 16-bit and 24-bit
>> audio, and the manufacturer has a
16: Invalid argument
When trying to play a 24-bit file instead of 16-bit, "16:" is replaced
by "24:" in the error message. The DAC supports 16-bit and 24-bit
audio, and the manufacturer has assured me that it enumerates with
Win
ve.pcm digital
> }
I'm trying to write a config in asound.conf that is equivalent to
hw:0,0. Is what you wrote above equivalent to hw or plughw? If
plughw, how would you write one that is equivalent to hw?
- Grant
--
>>> Hey Grant!
>>> Yes there is a difference. plughw automatically does some conversions for
>>> you and some looking after you. I'm not sure what exactly this comprised.
>>> But I _THINK_! it was samplerate conversion?, channel-counting and opening
>&
> Hey Grant!
> Yes there is a difference. plughw automatically does some conversions for
> you and some looking after you. I'm not sure what exactly this comprised.
> But I _THINK_! it was samplerate conversion?, channel-counting and opening
> the device, so it won't co
Is there a difference between specifying an audio device with
"plughw:0,0" and "hw:0,0"?
- Grant
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> rate 88200
> }
> }
> }
>
> Is there any way to resample conditionally? For example, resample
> 16/44.1 to 24/88.2, but don't resample a 24/96 source.
>
> Is there any sort of a UI for setting the resampling frequency?
>
> Can pulseaudio or any others do eithe
For example, resample
16/44.1 to 24/88.2, but don't resample a 24/96 source.
Is there any sort of a UI for setting the resampling frequency?
Can pulseaudio or any others do either of these?
- Grant
--
Come b
Can I specify a dmix rate_converter for sox "very high quality"?
- Grant
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B to Linux?
>
> Thanks,
> Grant
How about EMU 1212 PCI 24-bit recording support? Does it work?
- Grant
--
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is the only developer event you need to a
I need to get an ADC for copying vinyl records to the computer. I've
read that the EMU 0404 USB is excellent for this, but I've read mixed
reports on Linux compatibility. Will the device currently record from
the analog inputs to 24/96 or 24/192 and output via USB to Linux?
Tha
em produce sound
except for S24_3LE and S16_LE. How can I output S24_3LE directly to
the device for testing?
- Grant
>> 2. Why is so much more CPU being used to play files with mpd than with
>> mplayer? If I remove the asound.conf entries, mpd no longer uses more
>> tha
default {
>>> type hw
>>> card 0
>>> }
>>>
>>> and tested with mplayer but that yielded no sound at all. I also tried
>>> this:
>>>
>>> defaults.pcm.rate_converter "samplerate"
>>>
>>> but that also
bit even though I've specified "format
S24_3LE"?
2. Why is so much more CPU being used to play files with mpd than with
mplayer? If I remove the asound.conf entries, mpd no longer uses more
than 2% CPU.
- Grant
ound at all. I also tried this:
>
> defaults.pcm.rate_converter "samplerate"
>
> but that also yielded no sound from mplayer. Can I get around this static?
>
> - Grant
The following patch completely fixes this problem.
https://bugtrack.alsa-project.org/als
e "front:CARD=Juli,DEV=0" obtained with "aplay -L"
I get:
# aplay -L
null
Discard all samples (playback) or generate zero samples (capture)
# aplay -l
List of PLAYBACK Hardware Devices
card 0: USBDAC [Proton USBDAC], device 0: USB Audio [USB Audio]
Subdevices: 0/1
d at all, and apps set to OSS
to play sound perfectly as always:
pcm.!default {
type plug
slave.pcm {
type dmix
ipc_key 1024
slave {
pcm "hw:0,0"
format S24_LE
rate 96000
}
}
}
- Grant
--
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h more sophisticated options.
>>
>> That sounds good, but what else can I try?
>
> what I'd do as a first step is trying to get your own
> distribution / installation / customizations out of the
> mix.
>
> So get some proven Live CD/DVD (NOT Gentoo-based!) and
>
ice - but that's
> all.
>
> Your problem can NOT be due to a poor resampling algorithm: it must
> be a plain BUG somewhere!
>
> For the moment I urge you to forget about resampling algorithms and
> seek for the real source of your problems instead.
>
> Trying t
this last file, you don't have to restart alsa, it is enough
> to just restart your programs.
>
> Ciao,
> Dominique
Thanks Dominique. Not restarting alsasound will be nice.
- Grant
--
Crystal Reports - New F
exactly
> my ultimate goal), I got "static". :-(
>
> (maybe there is a solution to that too, but I was short
> of time and just gave up trying).
>
> BTW: isn't it possible to tell dmix to run itself at some
> specific (e.g. 96K) sample rate?
I've tried that lik
ot; as per default?
I did that by removing /etc/asound.conf and doing 'aplay file.wav',
but it produces the same static problem. I really think dmix is not
using the specified samplerate or speexrate.
- Grant
> If that works, next step is trying to upsample but using the default
>
t seen any evidence that my
defaults.pcm.rate_converter is being obeyed. With it set to
"samplerate_best" I get the same 'lsof|grep aplay' output I posted
before. There is no mention of libsamplerate or speex and there is
static in the sound produced by aplay.
- Grant
t; Does this mean dmix is using speex? If so, what else could be causing
>> >> my static problem? I basically hear static whenever dmix is involved.
>> >> If I have mpd resample with libsamplerate, I get no static.
>> >>
>> >> - Grant
>> >
causing
>> my static problem? I basically hear static whenever dmix is involved.
>> If I have mpd resample with libsamplerate, I get no static.
>>
>> - Grant
>>
>
> Yes, you are using speex.
I don't think my defaults.pcm.rate_converter is being obeyed. I
sw
urse YMMV.
>> >
>> >
>> > Ciao,
>> > Paolo.
>>
>> Is there any way to tell which resampler dmix is using? I have:
>>
>> defaults.pcm.rate_converter "speexrate_best"
>>
>> in /etc/asound.conf an
ell which resampler dmix is using? I have:
defaults.pcm.rate_converter "speexrate_best"
in /etc/asound.conf and I restarted alsa, but I still have the static problem.
- Grant
--
The NEW KODAK i700 Series Scann
only contains
that line and I don't have an .asoundrc. Even with:
defaults.pcm.rate_converter "samplerate_best"
I still get static and my CPU is idle. When I use samplerate_best in
mpd my CPU is maxed. I've also tried samplerate, samplerate_medium,
speex, and speex-float-
lso yielded no sound from mplayer. Can I get around this static?
- Grant
--
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? Can a DAC
>>work in Linux at 16-bit but not at 24-bit?
>>
>>- Grant
>
> Two stages-
> First test the card from aplay to make sure it does work at 24 bits, with 24
> bit files. If it does then you need the latest version of MPD to get 24 bit
> audio support (0.15.
ere should I look next? Can a DAC
>> >work in Linux at 16-bit but not at 24-bit?
>> >
>> >- Grant
>>
>> Two stages-
>> First test the card from aplay to make sure it does work at 24 bits, with 24
>> bit files. If it does then you need the latest versi
? Can a DAC
>>work in Linux at 16-bit but not at 24-bit?
>>
>>- Grant
>
> Two stages-
> First test the card from aplay to make sure it does work at 24 bits, with 24
> bit files. If it does then you need the latest version of MPD to get 24 bit
> audio support (0.15.
16-bit but not at 24-bit?
- Grant
--
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production scanning environment may not be a perfect world - but thanks to
Kodak, there's a perfect scanner
h support 24-bit playback and are
supported in Linux, but can I count on 24-bit *in* Linux?
- Grant
-
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Build the coolest Linux based applications
, and currently they're both
> dead. I also have none.
>
> I'm watching Blu-Ray. Might be investing a bit in that...
Of course DVD-A is dead, but DVD-Video containing audio won't catch up
with this:
http://dvd-a.info/Index.asp?stitle=neil+young
Blu-Ray surely will but the above
d is a DVD drive and:
>>
>> http://dvd-audio.sourceforge.net/
>
> Interesting. Thanks.
Check this out:
http://dvd-a.info/Index.asp?stitle=neil+young
2 pages of mainly Neil Young DVD-A releases all containing
, like me, like very direct recordings, you're going to
> love the sound-quality even of the CD (as most of his releases, a HDCD).
Can you rip and play back 20-bit HDCD audio?
>> BTW, the entire Doors catalog is available on DVD-A in 24/96.
>
> For DVD-A you need a DVD-A pla
ame one
or two? I'm a huge fan.
BTW, the entire Doors catalog is available on DVD-A in 24/96.
- Grant
-
This SF.Net email is sponsored by the Moblin Your Move Developer's challenge
Build the coolest Linux based applica
ting a USB one. I found the Creative sb0490 for about $50 which
is said to work in Linux here:
http://ubuntuforums.org/showthread.php?t=584190
It goes to 24/96 in stereo and that's as high as my DAC will go so I'm
thinking it's about right. What do you guys think?
- Grant
ontained by the "bluegears
b-Enspirer" which is about $100. I've read that CMI cards have good
ALSA support. Does this sound like a good one?
- Grant
-
This SF.Net email is sponsored by the Moblin Your Move D
I literally
can't find it for sale online. The site does say "Linux driver
available" though.
- Grant
-
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ted in the "minimal" benefit of 24-bit source
material (DVD-A, etc).
- Grant
>> I'm going to donate a 24-bit capable sound card to the mpd project to
>> add 24-bit support. Can anyone recommend an inexpensive one that is
>> 24-bit capable and PCI, PCIe, or USB?
I'm going to donate a 24-bit capable sound card to the mpd project to
add 24-bit support. Can anyone recommend an inexpensive one that is
24-bit capable and PCI, PCIe, or USB?
- Grant
-
This SF.Net email is sponsored b
usb-audio is definitely my USB DAC's driver and I've set this up
under /root/.asoundrc and /home/user/.asoundrc and rebooted:
pcm.usb-audio {
type hw
card 0
}
I'm using alsa-lib-1.0.16. Please let me know if you can help
- Grant
-
>> Is there any way to test the digital contents of a FLAC file against
>> the digital stream sent to the sound card for a match?
>>
>> - Grant
>>
>> -
>> Check out the new SourceFor
Is there any way to test the digital contents of a FLAC file against
the digital stream sent to the sound card for a match?
- Grant
-
Check out the new SourceForge.net Marketplace.
It's the best place to buy or sell ser
ame the plug usb-audio_44 then your mpd.conf device has to match
> (not usb_audio_44, but usb-audio_44).
>
> First make sure you get sound when you play to the non-plug device
> (usb-audio).
I don't get sound from usb-audio, making sure the name matches. Could
the .asoundrc s
"usb_audio_44"
}
I also tried .asoundrc with usb_audio instead of usb-audio and mpd.conf with:
type "usb-audio_44"
Do you see the problem?
- Grant
-
Check out the new SourceForge.net Marketp
>> On IEEE 32 bit floats the mantissa is 23 bit, so there might be
>> situations where you loose the LSB.
>
> And that was the only point - a "pro audio chain" should be able to
> support "digital wire" ca
ve {
pcm my_device
rate 44100
}
}
Relevant mpd.conf section:
audio_output {
type "alsa"
name "USB Monica"
device my_device_44"
}
I've also tried:
t "44100:16:2"
}
I know this isn't the mpd list, but does that look like it should
accomplish this? Is there any way to verify that this is happening?
- Grant
-
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Where is my output sample rate defined? I'm trying to make sure mpd
isn't resampling my music before it's sent to the USB DAC.
- Grant
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>> Is running jack, pulseaudio, or esound enough to jeopordize the bits,
>> or must the audio player support the sound server in order for that to
>> happen?
>>
>> - Grant
>>
>>
>
> I think you're asking, if the sound server isn't used by
nning jack, pulseaudio, or esound enough to jeopordize the bits,
or must the audio player support the sound server in order for that to
happen?
- Grant
-
Check out the new SourceForge.n
ushing through our
> ears makes a sound louder than -96 db. :-)
>>
>> Other than what's been mentioned, ALSA will do a perfect job of
>> passing the audio straight through to the DAC?
>>
>
> Yes. Think of alsa as a lazy worker; it does the least amount of work it
lume control. I now use mpd
>> which does not use a software mixer, the volume control doesn't work,
>> and that is good. It sounds a lot clearer.
>>
>> Can you guys tell me how else I should inspect my system's
>> configuration for digital manipulation of my mu
doesn't work,
and that is good. It sounds a lot clearer.
Can you guys tell me how else I should inspect my system's
configuration for digital manipulation of my music? Would pulseaudio
introduce this type of manipulation?
- Grant
)
On Thu, 2004-05-13 at 17:29, Jason Grant wrote:
> I've been unable to record on mic2 with my Asus P4P800 Deluxe
> motherboard. I've tried all sorts of combinations with amixer, but I've
> basically been assuming that the following setting should switch on mic2
> rather
I've been unable to record on mic2 with my Asus P4P800 Deluxe
motherboard. I've tried all sorts of combinations with amixer, but I've
basically been assuming that the following setting should switch on mic2
rather than the default mic1:
amixer cset iface=MIXER,name='Mic Select' 1
I've found th
clude
kernel parameters properly (I mean CONFIG_DEVFS_FS
in info.c and probably others).
Thank you in advance,Grant L.
Atoyan
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