Re: [Alsa-user] CMI8788.

[Clemens Ladisch]

Mark A Jenks wrote:
> I have a Bluegears b-Enspirer 7.1 Sound Card CMI8788 that I am looking
> for some help/advice for.
>
> I would like to seperate out the stereo outs to multiple outputs, so I
> can use it as a backend for my house with zones.

The hardware cannot control the four analog outputs independently, but
it's possible to use plugins for ALSA-aware applications.
Put the following into your ~/.asoundrc or /etc/asound.conf:

pcm_slave eightchannels {
pcm "hw:0,0"# or "hw:1,0" for the second card
channels 8
}
pcm.stereo1 {
type plug
slave.pcm {
type dshare
slave eightchannels
}
bindings [ 0 1 ]
}
pcm.stereo2 {
type plug
slave.pcm { type dshare slave eightchannels }
bindings [ 2 3 ]
}
pcm.stereo3 {
type plug
slave.pcm { type dshare slave eightchannels }
bindings [ 4 5 ]
}
pcm.stereo4 {
type plug
slave.pcm { type dshare slave eightchannels }
bindings [ 6 7 ]
}

then use device names "stereo1" to "stereo4" instead of "default" or
"hw:x".

> I also would like to leave the Digital out as an 7.1 to work with my
> stereo for Watching movies.

The SPDIF output is independent; it mirrors the front analog output only
when the "spdif" device is not being used.


HTH
Clemens

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Re: [Alsa-user] CMI8788.

[Clemens Ladisch]

Mark A Jenks wrote:
> I tried those commands in my 1.0.16 alsa, and it doesn't like them.

Sorry, didn't test them.
Try this:

pcm_slave eightchannels {
pcm "hw:0,0"# or "hw:1,0" for the second card
channels 8
}
pcm.stereo1 {
type plug
slave.pcm {
type dshare
ipc_key 8788
slave eightchannels
bindings [ 0 1 ]
}
}
pcm.stereo2 {
type plug
slave.pcm {
type dshare
ipc_key 8788
slave eightchannels
bindings [ 2 3 ]
}
}
pcm.stereo3 {
type plug
slave.pcm {
type dshare
ipc_key 8788
slave eightchannels
bindings [ 4 5 ]
}
}
pcm.stereo4 {
type plug
slave.pcm {
type dshare
ipc_key 8788
slave eightchannels
bindings [ 6 7 ]
}
}


HTH
Clemens

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Re: [Alsa-user] ALSA vs M-Audio fast track ULTRA

[Clemens Ladisch]

V Gabriele De Palo wrote:
> # cat /proc/asound/cards
> ...
>  1 [Ultra  ]: USB-Audio - Fast Track Ultra
>   M-Audio Fast Track Ultra at usb-:00:1d.7-3, high 
> speed

It seems the device is more or less class compliant, so, in theory, it
could work.

> # speaker-test -c 2 -D hw:1
>
> ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition 
> 'defaults.namehint.extended'

There is some problem with your alsa-lib installation that is unrelated
to the sound hardware.

Try reinstalling alsa-lib and/or alsa-utils, if possible.
Remove /etc/asound.conf or ~/.asoundrc, if you have them.


HTH
Clemens

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Re: [Alsa-user] SPDIF woes

[Clemens Ladisch]

Jason Gauthier wrote:
> I can't find many cards except for the Creative Labs X-Fi that seems to
> fit both.. and from what I read SPDIF passthrough doesn't work.I'm
> not 100% sure what that means. (Pretty new to this level of audio)

SPDIF was designed to transport two channels of 16-bit uncompressed
audio at 48 kHz (this is 192,000 bytes per second).  If you want to
transport more channels (4.0/5.1/7.1), the data has to be compressed
with a codec like Dolby Digital (AC-3) or DTS, and a special control bit
must be set in the SPDIF stream to indicated that the data is
compressed.  (Since nobody wants to pay for an encoder license, this is
mostly interesting for DVDs, which the audio data already in compressed
form.)

A sound card that can do SPDIF passthrough (usually called DTS/AC3
passthrough) can two things:
1) it does not modify the audio data in any way (every bit is needed for
   decoding), and
2) it allows to set the "non-audio" control bit.

You don't need this feature if all you want is stereo.

> Does anyone have any recommended cards that fit both PCI Express, USB,
> or firewire and SPDIF out?

USB:  There are many devices with SPDIF output, but there isn't any one
that does support passthrough in Linux.

PCI-E:  The X-Fi has SPDIF, and it should not take more than another few
years before Creative adds passthrough support to the driver.  The Asus
Xonar DX and D2X cards do support passthrough, but you need a recent
enough kernel; see
.

Firewire:  don't know, there is probably nothing that would be cheaper
than the Xonar DX.


HTH
Clemens

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Re: [Alsa-user] Which process is using an ALSA driver.

[Clemens Ladisch]

Eliot Blennerhassett wrote:
> can anyone tell me how to find out which processes are using an ALSA
> driver and preventing it from being unloaded.
>
> I know that "lsof /dev/snd"  will tell me who has these files open, but
> often this shows nothing and still the driver can't be unloaded.
>
> Where else should I look?

The device files for the OSS emulation are in not in /dev/snd but in
/dev, try "lsof /dev/dsp*".

> Is there an equivalent to lsof for memory mapped regions?

Yes, it's called lsof. ;-)  Memory-mapping requires an open file handle.


HTH
Clemens

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Re: [Alsa-user] SPDIF not working with AD1989B (Asus P5Q-E motherboard)

[Clemens Ladisch]

Jason Tackaberry wrote:
> I am testing with mplayer, and ensuring the digital device (as shown by
> aplay -l) is being used.  In my case it's device 1, and I am specifying
> -ao alsa:device=hw=0.1 on the mplayer command line.

Try "spdif" instead of "hw:0,1".


HTH
Clemens

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Re: [Alsa-user] How to interrupt a blocking call to snd_pcm_writei

[Clemens Ladisch]

Florian Winter wrote:
> Suppose, an ALSA playback device is opened in blocking mode, and one
> thread calls snd_pcm_writei. If the snd_pcm_writei call blocks, because
> the internal buffer of the ALSA device is full, is there a way by which
> another thread can interrupt the call, so it returns immediately without
> writing any data to the device?

No.  As far as the user-space library is concerned, ALSA is not thread
safe.

You have to use non-blocking mode and poll() so that you can send a
message through a file descriptor (e.g. pipe or eventfd).


Regards,
Clemens

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Re: [Alsa-user] em28xx_alsa: disagrees about version of symbol xxx after upgrading CentOS 5.2 ALSA .14 for ALSA .17

[Clemens Ladisch]

Robert Vincent Krakora wrote:
> em28xx_alsa: disagrees about version of symbol snd_pcm_new

This means that the em28xx_alsa module that you're trying to load and
the currently loaded snd_pcm module have been compiled for different
kernels.

It is possible that the old ALSA modules are still loaded; try unloading
them, or rebooting.


HTH
Clemens

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Re: [Alsa-user] How to interrupt a blocking call to snd_pcm_writei

[Clemens Ladisch]

Florian Winter wrote:
> I can't derive from the documentation when exactly my poll() call will
> wake up.

It can wake up whenever the device generates an interrupt, i.e., at
period boundaries.

> Is there a way to tell ALSA something like: "I want to write at
> least N frames to the device without blocking. Please don't wake up my
> poll() call until I can do this" (assuming that N is a meaningful value,
> i.e. less than the total buffer size)?

You can use snd_pcm_sw_params_set_avail_min(), but it won't have any
effect if N is lower then the period size.

> Will the ALSA device always become writeable when N bytes can be
> written, or is the ALSA implementation (driver implementation) free to
> decide when and when not to block?

All drivers behave the same in this regard because waking up is handled
by the ALSA framework.

Please note that you can be waken up when an error occurs before N
frames have become available.


HTH
Clemens

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Re: [Alsa-user] ALSA timer interface and accuracy

[Clemens Ladisch]

Julien Claassen wrote:
> Does the ALSA timter interface take its timing info from the soundcard
> (sample clock) or from the system's timer (RTC)?

ALSA's timer interface can use
1) the system timer (which is not the RTC timer),
2) the RTC timer (if it is available),
3) a sound card timer, if the sound card has a programmable timer (very
   few actually have one),
4) a PCM slave timer, which generates timer ticks at period boundaries
   (PCM slave timers are available for every hardware sound device, but
   only while running, i.e., while sound is actually being played/recorded).

> If it is the system: Is there a way to get a clock source from the soundcard
> using the ALSA API?

The device name of the timer device "hw" has several parameters:
1) hw:CLASS=1,DEV=0
2) hw:CLASS=1,DEV=1
3) hw:CLASS=2,CARD=xxx,DEV=yyy
4) hw:CLASS=3,CARD=xxx,DEV=yyy,SUBDEV=zzz


HTH
Clemens

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Re: [Alsa-user] ALSA timer interface and accuracy

[Clemens Ladisch]

Julien Claassen wrote:
> If I'd like to use the "PCM slave timer", how do I access that? Or were the
> instructions you gave me at the end of your mail already for that prupose?

Yes.  For the values of the CARE/DEV/SUBDEV parameters, see
/proc/asound/timers.


HTH
Clemens

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Re: [Alsa-user] choice of a new soundcard

[Clemens Ladisch]

Vedran Miletić wrote:
> Drivers that support hardware mixing:
> * snd_emu10k1 - ...
> * snd_cs46xx - ...
> * snd_au88x0 - ...

* snd_ymfpci - for YMF7xx chips.  Like most of the others, has been
  discontinued and must be bought used.  Used on several cards, e.g.,
  Hoontech SoundTrack Digital XG (most of those cards have "XG" in
  their name).

If you want a supported wavetable synthesizer, get a SB Live! or Audigy.
If you want to do bit-exact recording from the SPDIF input, get a
YMF754B-based card.  Otherwise, the chip doesn't matter.

> Not sure about:
> * snd_ice1712

Does _not_ support hardware mixing.

> Hardware does support it, and it did work on Windows.

The Windows driver, like the drivers of most other cards, claims to
support multiple streams and does software mixing in the driver itself
instead of letting Windows do the mixing.  Since Windows assumes that no
driver would do such a silly thing, the driver shows up as "supports
hardware mixing".


HTH
Clemens

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Re: [Alsa-user] choice of a new soundcard

[Clemens Ladisch]

Vedran Miletić wrote:
> http://www.via.com.tw/en/products/audio/controllers/envy24/
> Claims that it has "20-channel, 26-bit wide built in mixer". What does
> that mean?

The chip plays 10 channels in parallel (8 analog, 2 digital) and records
12 channels in parallel (8 analog, 2 digital, 2 monitoring).  The two
monitoring channels result from mixing together the other 20 channels.

This is not the usual hardware mixing scenario because we don't have
several independent stereo streams, and the mixing result does not go
to the speaker outputs.


Regards,
Clemens

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Re: [Alsa-user] Definition of a frame

[Clemens Ladisch]

Bankim Bhavsar wrote:
> For a "stereo" DAC channel with 16-bit samples, number of bytes per
> frame = 4 bytes, correct?

Yes.


HTH
Clemens

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Re: [Alsa-user] AC-3 Passthrough doesn't works

[Clemens Ladisch]

[EMAIL PROTECTED] wrote:
> But when I try to replay a DVD with AC-3 data xine only tells me that
> the audio device is not available.

What happens when you try a command like this:

  aplay -D spdif something.wav


Best regards,
Clemens

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Re: [Alsa-user] setting ALSA_CARD from /etc/bashrc

[Clemens Ladisch]

Yan Seiner wrote:
> I've added a line to /etc/bash.bashrc that sets ALSA_CARD depending on
> the value of $DISPLAY.
>
> Now this works fine if I open a terminal window and run my apps.  But
> apps started from the menu don't have ALSA_CARD set.

You could set ALSA_CARD when X is started (in .xinitrc or something like
that).


HTH
Clemens

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Re: [Alsa-user] ICH7/ALC662 card won't initialise

[Clemens Ladisch]

Lindsay Roberts wrote:
> I have a board

Which one?

> This is an Intel 945GC chipset (ICH7) with a Realtek ALC662 on it,
> both of which seem to be supported in alsa.
>
> Leaving the dist to its own devices sees it load snd_intel8x0, and
> error out as follows:
>
> codec_ready: codec is not ready[0x870]
>
> Assuming what I know about the division between snd_hda_intel and
> snd_intel8x0 is correct this doesn't suprise me a great deal because
> the ALC662 is actually a Realtek hda chip, not an ac97 chip.
>
> Seems like the pci id registered for the ICH7 in snd_hda_intel is
> 8086:27d8 and the one in snd_intel8x0 is 8086:27de . The device on my
> board is of course pci id 8086:27de . Hacking in that pci id to
> snd_hda_intel was completely unenlightening:
>
> hda-intel: ioremap error

The BIOS is responsible for switching the controller into either AC97
or HDA mode.  It appears this was not done correctly in your case.
As the chip is currently configured, you _do_ have an AC97 controller
(without an AC97 codec).

I have no explanation for this, except maybe a wrong BIOS (for a
different mainboard), or a hardware error.

Did it ever work?

> One final juicy tidbit is that it reports no subsystem vendor/device
> (zeroes all round).

IIRC these IDs are to be initialized by the BIOS, too.


Best regards,
Clemens

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Re: [Alsa-user] Audio on ECS A780GM-A motherboard

[Clemens Ladisch]

Laurent E wrote:
> - According to the command "cat /proc/asound/card0/codec#0", the chip
> is a "SigmaTel ID 7645". [...]
> 1/ Is it possible that the chip is incorrectly detected?

Your driver doesn't know about your chip.

Support for this chip was added in May:


Update your kernel, or try to apply the patch manually.


HTH
Clemens

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Re: [Alsa-user] Digital bit perfect ouptut with ALSA

[Clemens Ladisch]

Paulo Moura Guedes wrote:
> - does Linux/ALSA features dynamic sample rates?

The default device (named "default") uses automatic resampling, but the
"spdif" device does not.

> - is it possible to set the bit-depth? (in my case to 24 bit)

Yes, if the hardware supports it.

> - what other variables do i have to consider in order to get bit transparent
> ouput, i.e., no resampling at all?

Just using the "spdif" device should work.  (Nowadays, there is no
device that would want to forgo the ability to output AC-3/DTS data.)

> - is it possible to check if ouput is bit-perfect?

Record it, then compare the data to the original.  (However, there are
sound chips that always resample when recording from a digital input.)


HTH
Clemens

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Re: [Alsa-user] Digital bit perfect ouptut with ALSA

[Clemens Ladisch]

Paulo Moura Guedes wrote:
> On Thursday 11 December 2008 07:35:25 Clemens Ladisch wrote:
> > The default device (named "default") uses automatic resampling, but the
> > "spdif" device does not.
>
> So, I have to manually set the sample-rate, depending on the files I will 
> play?

No, ALSA automatically switches the device's sample rate to that of the
audio data (or resamples to the nearest supported rate).

> > > - is it possible to set the bit-depth? (in my case to 24 bit)
> >
> > Yes, if the hardware supports it.
>
> It's a Benchmark DAC1 which supports 24 bit. Where can I set it?

You don't need to do this manually; ALSA automatically uses 24 bits.


In your case, the USB device is handled like a sound card; any other
sound card in the system doesn't affect it.


Best regards,
Clemens

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Re: [Alsa-user] Digital bit perfect ouptut with ALSA

[Clemens Ladisch]

Bill Unruh wrote:
> On Mon, 15 Dec 2008, Paulo Moura Guedes wrote:
> > [...]
> > The ASRC, as the name implies, is not syncronized to the clock of
> > the incoming digital signal. Therefore, its performance is independant of 
> > the
>
> This makes no sense at all. If the incoming signal is a digital signal, it has
> no clock.

An S/PDIF signal does have a clock (it uses biphase mark coding for
exactly this reason), and there are many DACs that derive their master
clock directly from the input signal.

(It's different in the case of USB where the sample clock is independent
of the bus clock.)

> > quality of that clock. In other words, it doesn't matter if the signal came
> > from a cheap transport with cheap cables, or from a $10,000 signal chain. 
> > The
>
> And what the hell do cables have to do with anything? Cables do NOT affect the
> timing.

If the cable's impedance is high enough, the transitions between the
signal levels can be delayed.  This wouldn't matter much except that
due to the biphase mark coding, the jitter is data dependent.

(It's a different question whether the distortions caused by the jitter
are actually audible.)


Best regards,
Clemens

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Re: [Alsa-user] How to order multiple usb devices

[Clemens Ladisch]

Ari Moisio wrote:
> # Usb-headset
> options snd-usb-audio index=2  vid=0x046d,0x0a02
> # M-audio fasttrack
> options snd-usb-audio index=3  vid=0x0763,0x2012

This does not work because the second 'options' line overrides the
first, and the product ID must be put into the 'pid' parameter.

Use this:

  options snd-usb-audio index=2,3 vid=0x046d,0x0763 pid=0x0a02,0x2012

(Since the vendor IDs are unique, you don't actually need 'pid' in this
case.)


HTH
Clemens

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Re: [Alsa-user] textual tool for pause/resume

[Clemens Ladisch]

davide.bonfa...@bticino.it wrote:
> I'm searching for a textual tool in order to play, pause and resume the
> streaming on a certain device.

Try mplayer.


HTH
Clemens

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Re: [Alsa-user] emu 0404 usb problems

[Clemens Ladisch]

Aleksander Kamenik wrote:
> Since the upgrade from Fedora 9 to Fedora 10, the EMU 0404 USB card does 
> not work any more.

Only in PulseAudio?

> The alsa version changed form .17 to 19.

There have been no relevant changes in either the ALSA driver or
alsa-lib.

> The gist of the problem ass told by Lennart:
> 
> Ah, that's interesting. Apparently your sound card does not support
> integral number of periods.

It should; the USB driver accepts nearly anything.

Please run PA with the environment variable LIBASOUND_DEBUG set to 1.
Alsa-lib will then output some information about the error on stderr.

> Also, under which project should this be filed?

alsa-lib


Best regards,
Clemens

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Re: [Alsa-user] Device exists, PCM does not

[Clemens Ladisch]

Evan Leibovitch wrote:
> # arecord -D hw:1,0 >/dev/null
> Recording WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
> arecord: set_params:932: Broken configuration for this PCM: no configurations 
> available

Try "arecord -D default:1".


HTH
Clemens

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Re: [Alsa-user] Surround over spdif

[Clemens Ladisch]

Floris wrote:
> Is there a way to encode a file on the fly?

http://wiki.alsa-project.org/main/index.php/A52_plugin


HTH
Clemens

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Re: [Alsa-user] broken digital sound again

[Clemens Ladisch]

James wrote:
> $ speaker-test -D iec958 -c 6
> ...
> Channels count (6) not available for playbacks: Invalid argument

S/PDIF does not support uncompressed surround sound; you have to play
stereo data (use "-c 2") or AC-3/DTS-compressed data.


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Clemens

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Re: [Alsa-user] xonar essence stx

[Clemens Ladisch]

jed wrote:
> Is support for this coming soon?

Yes.

> How does using ALSA compare to using drivers developed by the vender 
> specifically for Windows?
> i.e.
> Will I not be getting most out of this card (hardware-wise) if I'm not 
> using the drivers Asus developed, or is the difference negligible?

The driver supports all features of the hardware; the output quality is
the same as with the Windows driver.

Please note that the Linux driver does not have the "driver features",
which are implemented in software (Dolby, SVN, Xear, etc.).


Best regards,
Clemens

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Re: [Alsa-user] xonar essence stx

[Clemens Ladisch]

jed wrote:
> > > Is support for this coming soon?
> > 
> > Yes.
> 
> Awesome, is it likely to be within the next 6-months?

Actually, it looks like this week ...

> > Please note that the Linux driver does not have the "driver features",
> > which are implemented in software (Dolby, SVN, Xear, etc.).
> 
> Can you please be a little more specific about these "driver features"
> implemented in software?
> Or provide a link to where it's outlined?

They're listed on this page:
http://www.asus.com/products.aspx?modelmenu=2&model=2711&l1=25&l2=150&l3=0&l4=0


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Clemens

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Re: [Alsa-user] Need help connecting spdif input

[Clemens Ladisch]

rek...@holisticode.se wrote:
> I have been searching the web for an example of how to get sourround
> sound from my toslink input [...] I realise that an a52 filter should
> be used,

The a52 plugin encodes PCM data into Dolby data, and as a plugin, it is
only used when an application uses the specified sound device.

What you want to do is to record from the digital input device, convert
the data from Dolby to PCM, then output this to the analog output
device.

It should be possible to do this by using arecord to capture raw data,
then piping it into ac3dec.

untested, assuming card 0:
arecord -D spdif:0 -f dat -t raw | ac3dec -D hw:0 -6


HTH
Clemens

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Re: [Alsa-user] xonar essence stx

[Clemens Ladisch]

jed wrote:
> Damn, that's a substantial amt of stuff that won't be available don't u 
> reckon?

Yes.


Best regards,
Clemens

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Re: [Alsa-user] Split stereo input?

[Clemens Ladisch]

deadrabbit wrote:
> I'm new to using ALSA, or at least to the internals of it. I'm trying
> to create a Icecast stream encoding computer for two audio streams.
> The computer as a single stereo line in, so I'm planning on feeding
> each mono stream through the single stereo input. I've looked at the
> asoundrc guide, but I'm not sure I understand it entirely: there's no
> documentation for some things, like "ttable". Will the config below
> work for splitting the stereo input in to two mono devices?

In theory, yes.

> Also, how do I know what the input device name is?

It's the string after the "pcm.".
In this case, "dshare" is the slave device used by the two other devices
which are named "leftx" and "rightx".


HTH
Clemens

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Re: [Alsa-user] anyone using the M-Audio Fast Track Ultra 8R (or any other 8in 8 out USB device)

[Clemens Ladisch]

sonof...@iinet.net.au wrote:
> I'm looking to move over to a device that I can use with both my linux desktop
> and a 2nd hand powerbook i bought recently. I have been more focussed on FFADO
> compatible as from what I understand I am more likely to get latency issues 
> on a
> USB device...
> 
> However, I have been using my M-Audio Delta 1010 for years now and have been 
> very
> happy with the audio quality and latency performance. This looks like a USB
> version but with preamps (which I don't need)

The USB latency depends on the USB protocol itself and on the driver;
these are the same for _all_ USB devices.

Furthermore, there is no supported 8+8 device at the moment.


Best regards,
Clemens

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Re: [Alsa-user] USB Sound Devices vs. On-Board Audio

[Clemens Ladisch]

Xelatec wrote:
> I have a case where an Centos 5.2 O/S on an Intel Atom motherboard
> properly recognizes the on-board Intel sound system. If I hot plug in 4
> USB sound devices they are configured and work.
> 
> But if I boot the system with the 4 USB sound devices attached, they are
> configured but the on-board Intel sound system is gone.

Please show the contents of /proc/asound/cards in this case.
Are there any error messages in the system log?


Best regards,
Clemens

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Re: [Alsa-user] Griffin Imic

[Clemens Ladisch]

gary wrote:
> gary wrote:
> > Now for the problem. I can't talk to the iMic. Using
> > krecord -d /dev/dsp4 .
> > 
> > I get the response
> > ioctl SNDCTL_DSP_SETMFT: Input/output error

There should be an error message in the system log (probably
/var/log/messages, or at the end of the output of dmesg).

> I loaded the same two devices on Suse 10. That got a bit further. The 
> error message is the usb sound module can't be loaded.

There should be an error message in the system log.


Best regards,
Clemens

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Re: [Alsa-user] Unable to configure ASUS A7V sound card with Alsa

[Clemens Ladisch]

sto...@safe-mail.net wrote:
> The card is enabled in the BIOS,

Is the entry "Enabled" or "Auto"?  The latter means the BIOS
automatically disables it when it finds another sound card, because,
apparently, it doesn't expect anybody would want to use it when there
is a _real_ sound card.

This is common with BIOSes for VIA chipsets.  The Linux kernel has code
to forcibly enable the AC'97 controller on VT8237-based mainboards.
In the case of older mainboards like yours, which sometimes didn't have
audio, this cannot be done automatically.

Also check that Sound Blaster emulation is disabled.

> but does not appear in the lspci output.

Please show the lspci output.

> I was also surprised to find that Asus supplies drivers that support
> RedHat 6.2.

This is no surprise if you consider that RH 6.2 comes with kernel 2.2.14,
which didn't yet have ALSA.

I strongly suggest you forget those drivers and use the snd-via82xx
driver of your distribution, which is likely to be newer by about eight
years or so.


HTH
Clemens

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Re: [Alsa-user] Need help connecting spdif input

[Clemens Ladisch]

Mårten Gustafsson wrote:
> mor...@linux-hyfd:~> arecord -D spdif:0 -f dat -t raw | ac3dec -D hw:0 -6
> Recording raw data 'stdin' : Signed 16 bit Little Endian, Rate 48000 Hz, 
> Stereo
> 2.0 Mode 48.0 KHz  32 kbps Complete Main Audio Service
> unsupported 1/1 channels 6

Apparently, the recorded Dolby data is encoded as 2.0, not 5.1, and
ac3dec does not supported playing 2.0 data on six channels.
Try it without the "-6".


Best regards,
Clemens

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Re: [Alsa-user] ALSA support for old ISA card?

[Clemens Ladisch]

Landis McGauhey wrote:
> ALSACONF recognizes the soundcard as "SBAWE Creative SB AWE64 PnP" and
> completes its work without errors.  ... Command ALSAMIXER results in
> the error message "function snd_ctl_open failed for default:  no such
> device".

In theory, all those old ISA cards are still supported.
However, some distributions don't enable the ISA drivers.

Does the card appear in /proc/asound/cards?
If not, does it after running "modprobe snd-sbawe"?

> If you need some output from DMESG, just let me know.

This would be a good idea.


Best regards,
Clemens

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Re: [Alsa-user] modprobe snd-virtuoso error

[Clemens Ladisch]

Ron Pitts wrote:
> ...
> FATAL: Error inserting snd_virtuoso 
> (/lib/modules/2.6.27-11-generic/kernel/sound/pci/oxygen/snd-virtuoso.ko): 
> Unknown symbol in module, or unknown parameter (see dmesg)

The most likely cause of this problem is that you are trying to load a
module of a new driver version while some other sound module of the old
driver version is still loaded.

It should work after you unload all snd_* modules, or reboot.


HTH
Clemens

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Re: [Alsa-user] (no subject)

[Clemens Ladisch]

dennism...@gmx.net wrote:
> Can please a admin have a look at my mail?

There is no moderator; all such mails go into a black hole.

> the message was over the max size

Compress it, ot put it somewhere on the web.


HTH
Clemens

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Re: [Alsa-user] modprobe snd-virtuoso error

[Clemens Ladisch]

Ron Pitts wrote:
> Its loading now without error  however doesnt report its found my ASUS
> HDAV1.3 ..

It doesn't report anything when there is no error.
Does it appear in /proc/asound/cards?

Anyway, I've fixed a bug after 1.0.19 was released; please try
http://www.alsa-project.org/snapshot/files/alsa-driver-1.0.19.30.gf0b3d.325.gc46d0.tar.bz2

> Clemens, your the man when it comes to this card, I understand its only 10%
> finished at the moment?

You're the first to ever try this code; I'm not sure if anything works.

I estimate that analog sound output and video data passthrough should
work, and that HDMI sound output possibly works.  Video effects are not
supported at the moment because I don't know how to enable them.


Best regards,
Clemens

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Re: [Alsa-user] Diamond Xtreme XS51 ALSA support? (was "ALSA Support for old ISA Soundcard")

[Clemens Ladisch]

Landis McGauhey wrote:
> I've decided to replace the old ISA card with a PCI card (several PCI slots
> are available).  Can anyone happen to tell me if you have any ALSA
> experience with the Diamond Xtreme XS51?

According to the driver's .inf file, that card uses a CMI8738/8768
chipset, which is supported by ALSA.


HTH
Clemens

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Re: [Alsa-user] modprobe snd-virtuoso error

[Clemens Ladisch]

Ron Pitts wrote:
> [  653.504508] snd_mpu401_uart: Unknown symbol snd_rawmidi_receive
> ...

The sound modules cannot be loaded because other sound modules they
depend on cannot be loaded.

Please show the first few of these messages (for module "snd").


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Clemens

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Re: [Alsa-user] modprobe snd-virtuoso error

[Clemens Ladisch]

Ron Pitts wrote:
> Analogue Output is working fine now however no luck with the HDMI audio.

But HDMI video does work?
Does your HDMI display show if any audio is sent, or in which format?
What sample format did you try to play?

> I've rebuilt the alsa lib with full debugging turned on:
> No sound for HDMI but receiving the following in dmesg:
> 
> [   33.668846] : 4e 43 4f 4b   NCOK
> [   36.523606] message from Xonar HDAV HDMI chip received:
> [   36.523632] : 53 54 41 52 54 4f 4b  STARTOK
> [   36.581530] message from Xonar HDAV HDMI chip received:
> [   36.581551] : 32 34 50 4f 46 46 4f 4b   24POFF
> 
> Is the plug iec958, the HDMI sound component?

No, that is the SPDIF output.  The HDMI sound must be driven by the same
DMA channel as the analog outputs ("Multichannel") because it is the
only one that supports more than two uncompressed PCM channels.

> I take it the above commands are from the method xonar_hdav_init in
> virtuoso.c,

No, the commands sent by the driver are not shown.  The logged messages
are responses from the HDMI chip.

You can show the sent commands by adding the following at the end of
the hdmi_write_command function:

printk(KERN_DEBUG "sent: fb ef %02x %02x", command, count);
for (i = 0; i < count; ++i)
printk(" %02x", params[i]);
printk(" %02x\n", checksum);

> its been awhile since I hacked some C++ code!!

Well, the driver doesn't use C++ ...


Best regards,
Clemens

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Re: [Alsa-user] Unable to configure ASUS A7V sound card with Alsa

[Clemens Ladisch]

(please don't top-post)

sto...@safe-mail.net wrote:
> Any suggestions with this output?

The AC'97 controller does not show up.  Please make sure that SB
emulation is disabled.


Best regards,
Clemens

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Re: [Alsa-user] Analog line in output via IEC958?

[Clemens Ladisch]

Antony Gelberg wrote:
> I have an onboard NVidia sound card using snd_hda_intel.  Full info about my
> setup is at
> http://www.alsa-project.org/db/?f=f407c7aceb5abe36b68731ff50fd1d21f272fb98
> 
> I can output everything to a HT amp via the IEC958 optical out by means of the
> .asoundrc that you see in the above link.  I would like to plug something into
> the analogue line in and have it be output via the IEC958 as well.  However,
> it is only output via the analogue line out, not the optical.  How can I get
> this to work?

The ALC883 cannot do this alone.  You have to run a program to record
from the line in and to play the data back to the SPDIF output.

"arecord -f dat | aplay -D spdif" might work, but then you cannot use
these devices with other programs.


HTH
Clemens

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Re: [Alsa-user] How to set USB-Headset volume with udev and amixer

[Clemens Ladisch]

Jens Rutschmann wrote:
> I have a C-Media USB Headphone Set which is working well. One issue though is 
> that when plugging it in the mixer levels are initialized to bad values. The 
> "Speaker" level is set to 100% (*very* loud, like in "ouch !!!") while the 
> "Mic 
> Capture" level is set to 0%.
> 
> Therefore I set the mixer levels by hand (using alsamixer) until now. So I 
> wrote 
> a script which is invoked by udev. The script tries to set the levels using 
> amixer.
> [...]
> So I wrote the following script and udev rule:
> 
> ATTR{idProduct}=="000c", ATTR{idVendor}=="0d8c", ACTION=="add", 
> RUN+="/usr/local/bin/headset_volume.sh"
> ...
> The script is executed since I get the log file, but amixer obviously isn't 
> executed correctly,

I don't know why your script doesn't work, but the following udev rule
works for me, and it does not need to be adjusted for the specific sound
device but just requires that you've previously saved the the correct
settings with "alsactl store":

 SUBSYSTEM=="sound", ACTION=="add", KERNEL=="controlC[0-9]*", 
RUN+="/usr/sbin/alsactl restore $number"


HTH
Clemens

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Re: [Alsa-user] Problem with SoundBlaster Live

[Clemens Ladisch]

Matias D'Ambrosio wrote:
>  I have been using this system for months, no problem (and many distros 
> before), but a few days ago sound stopped working, probably after an update.
>  I have an SB Live, I'm running Debian testing AMD64 kernel  2.6.26-1-amd64 
> #1 
> SMP.
>  lspci shows the card:
> 
> 01:07.0 Multimedia audio controller: Creative Labs SB Live! EMU10k1 (rev 0a)
> 
>  cat /proc/asound/modules gives no output
> ...
>  dmesg output:
> http://pastebin.ca/1365751

> [9.394958] gameport: EMU10K1 is pci:01:07.1/gameport0, io 0xb800, 
> speed 1201kHz

The card is there ...

> [   11.280004] AC'97 0 does not respond - RESET
> [   11.280010] AC'97 0 access is not valid [0x0], removing mixer.
> [   11.280842] ACPI: PCI interrupt for device :01:07.0 disabled
> [   11.280857] EMU10K1_Audigy: probe of :01:07.0 failed with error -5

... but the AC'97 codec on your card doesn't seem to work.

This is probably a driver bug (try a newer kernel, or some older/newer
live CD to be sure), but a hardware defect is also possible.


Best regards,
Clemens

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Re: [Alsa-user] Problems with ALC650

[Clemens Ladisch]

Jonathan Black wrote:
> I've been struggling as of lately to get ANY sound through the lime 
> green output on the back of my motherboard.
> 
> lsmod | grep snd shows:
> 
> snd_mpu401  6252  0
> snd_cs4232 11380  0
> snd_via82xx20920  2

Please show the output of /proc/asound/cards.  Is the AC'97 controller
the default (first) sound card?  If yes, please show the output of
"amixer contents".


Best regards,
Clemens

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Re: [Alsa-user] building soc module drivers

[Clemens Ladisch]

Geiger Ho wrote:
>   I would like to build the WM8510 SOC sound driver for my ARM platform. 
> After downloading alsa-driver-1.0.19, "./configure 
> --with-cross=arm-linux- --with-kernel=/home/source/linux-2.6.22.19 
> --with-build=/home/source/linux-2.6.22.19 --with-cards=all", and then 
> make. I get no wm8510.c compiled.

Is the kernel in /home/source/linux-2.6.22.19 configured for your target
machine?


Best regards,
Clemens

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Re: [Alsa-user] Can't open Asus Xonar D1 PCM output 6-channel in non-interleaved mode

[Clemens Ladisch]

Oliver Stephenson wrote:
>Now though, I'm trying to open the PCM output in 6-channel mode 
> non-interleaved, and it all falls over here:
> 
>  if ((err = snd_pcm_hw_params_set_access(playback_handle, hw_params, 
> SND_PCM_ACCESS_RW_NONINTERLEAVED)) < 0)
>  {
>   fprintf(stderr, "cannot set access type (%s)\n", snd_strerror (err));
>   return false;
>  }
> 
>What gives? Is this a feature of the hardware (C-Media chipset) or is 
> there a workaround?

This chip, like pratically every other sound chip, supports only
interleaved samples.

Try adding "plug:x" in front of the device name.


HTH
Clemens

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Re: [Alsa-user] Trying to use two sound cards at the same time

[Clemens Ladisch]

Poulet Fou wrote:
> I have two sound cards, one attached to pc speakers and one attached to a
> sound system in another room.
> Following instructions on the alsa project wiki, I am trying to use two
> sound cards (V8237 and CA0106).
> I configure my ~/.asoundc but I still can't manage to send a sound to both
> cards at the same time. Below is my ~/.asoundrc content.
> When I tested directly the ca0106 card (hw:0,0) with a play, it didn't work
> so I thought it might be some sort of rate or conversion problem. I added
> the pcm.via and pcm.audigy in my asoundrc file and then, when I test each
> card individually ("via" and "audigy") with aplay it works well.
> When testing the "multi" pcm with aplay (  aplay -vD multi question.wav ), I
> get the following error
> aplay: set_params:959: Nombre de canaux non disponible (Channels count non
> available)

Your "multi" device is a four-channel device.  Try "plug:multi".


HTH
Clemens

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Re: [Alsa-user] pcm_min.c trouble with modification

[Clemens Ladisch]

MK wrote:
> I want to write a simple ALSA application. I have downloaded pcm_min.c
> from ALSA documentation and try to do some modifications in code like
> change channels to stereo or sampling rate to 44100Hz. But every time
> program fails. Can anybody explain me why this is happening?

Please show your changes.


Best regards,
Clemens

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Re: [Alsa-user] AMT8

[Clemens Ladisch]

immanuel litzroth wrote:
> >Apr 13 12:08:39 voodochile kernel: [  169.622067] usb 2-10: new full speed 
> >USB device using ohci_hcd and address 5
> >Apr 13 12:08:39 voodochile kernel: [  169.844566] usb 2-10: configuration #1 
> >chosen from 1 choice
> >Apr 13 12:08:39 voodochile kernel: [  169.847594] snd-usb-audio: probe of 
> >2-10:1.0 failed with error -5
> >Apr 13 12:08:39 voodochile kernel: [  169.847616] snd-usb-audio: probe of 
> >2-10:1.1 failed with error -5
> >Apr 13 12:08:44 voodochile kernel: [  174.656857] usb 2-9: new full speed 
> >USB device using ohci_hcd and address 6
> >Apr 13 12:08:44 voodochile kernel: [  174.879643] usb 2-9: configuration #1 
> >chosen from 1 choice
> >Apr 13 12:08:44 voodochile kernel: [  174.882987] snd-usb-audio: probe of 
> >2-9:1.0 failed with error -5
> >Apr 13 12:08:44 voodochile kernel: [  174.883013] snd-usb-audio: probe of 
> >2-9:1.1 failed with error -5
> What's up with that?

I'd guess the driver doesn't claim all the interfaces that the device
has.  This shouldn't matter.

> Then if I do and aplaymidi -l it lists the following ports:
>  40:0AMT8 AMT8 MIDI 1
>  40:1AMT8 AMT8 MIDI 2
>  40:2AMT8 AMT8 MIDI 3
>  40:3AMT8 AMT8 MIDI 4
>  40:4AMT8 AMT8 MIDI 5
>  40:5AMT8 AMT8 MIDI 6
>  40:6AMT8 AMT8 MIDI 7
>  40:7AMT8 AMT8 MIDI 8
>  40:8AMT8 AMT8 Broadcast
>  44:0AMT8 AMT8 MIDI 1
>  44:1AMT8 AMT8 MIDI 2
>  44:2AMT8 AMT8 MIDI 3
>  44:3AMT8 AMT8 MIDI 4
>  44:4AMT8 AMT8 MIDI 5
>  44:5AMT8 AMT8 MIDI 6
>  44:6AMT8 AMT8 MIDI 7
>  44:7AMT8 AMT8 MIDI 8
>  44:8AMT8 AMT8 Broadcast
> Is it possible to have other names generated for these ports through
> some magic?

You could add some entries to the snd_usbmidi_port_info array in the
driver source ...

> Preferably each port would be named after the synth connected to it.

Changing port names at runtime isn't supported by the driver.

> Then when I play the devices I get messages in /var/log/messages saying
> Apr 11 23:03:56 voodochile kernel: [14700.085974] rtc: lost 28 interrupts
> Apr 11 23:04:26 voodochile kernel: [14730.054964] rtc: lost 28 interrupts
> Apr 11 23:04:56 voodochile kernel: [14760.023963] rtc: lost 28 interrupts
> What's up with that?

Every thirty seconds, something on your computer eats some CPU and
disables all interrupts for about 27 milliseconds.  If you have a laptop,
this it probably some fan control or power management stuff done by the
BIOS.


Best regards,
Clemens

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Re: [Alsa-user] DELL SP2309W Monitor with Integrated microphone

[Clemens Ladisch]

Phil Gorbett wrote:
> I am having difficulty getting the microphone(s) going with this
> monitor, and get the get the following results when I disable pulseaudio
> and run arecord:
> 
> "arecord -v -f cd -D plughw:0 file.wav
> Recording WAVE 'file.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
> ALSA lib pcm_params.c:2135:(snd1_pcm_hw_refine_slave) Slave PCM not usable
> arecord: set_params:939: Broken configuration for this PCM: no configurations 
> available"

Please show the output of "lsusb -v" for this device.


Best regards,
Clemens

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Re: [Alsa-user] query alsa for supported sample rates and formats?

[Clemens Ladisch]

Matt Garman wrote:
> Is there a way to query alsa to see what sample rates and formats
> the sound hardware natively supports?

Try the attached program.


HTH
Clemens
/*
 * hw_params.c - print hardware capabilities
 *
 * compile with: gcc -o hw_params hw_params.c -lasound
 */

#include 
#include 

#define ARRAY_SIZE(a) (sizeof(a) / sizeof *(a))

static const snd_pcm_access_t accesses[] = {
SND_PCM_ACCESS_MMAP_INTERLEAVED,
SND_PCM_ACCESS_MMAP_NONINTERLEAVED,
SND_PCM_ACCESS_MMAP_COMPLEX,
SND_PCM_ACCESS_RW_INTERLEAVED,
SND_PCM_ACCESS_RW_NONINTERLEAVED,
};

static const snd_pcm_format_t formats[] = {
SND_PCM_FORMAT_S8,
SND_PCM_FORMAT_U8,
SND_PCM_FORMAT_S16_LE,
SND_PCM_FORMAT_S16_BE,
SND_PCM_FORMAT_U16_LE,
SND_PCM_FORMAT_U16_BE,
SND_PCM_FORMAT_S24_LE,
SND_PCM_FORMAT_S24_BE,
SND_PCM_FORMAT_U24_LE,
SND_PCM_FORMAT_U24_BE,
SND_PCM_FORMAT_S32_LE,
SND_PCM_FORMAT_S32_BE,
SND_PCM_FORMAT_U32_LE,
SND_PCM_FORMAT_U32_BE,
SND_PCM_FORMAT_FLOAT_LE,
SND_PCM_FORMAT_FLOAT_BE,
SND_PCM_FORMAT_FLOAT64_LE,
SND_PCM_FORMAT_FLOAT64_BE,
SND_PCM_FORMAT_IEC958_SUBFRAME_LE,
SND_PCM_FORMAT_IEC958_SUBFRAME_BE,
SND_PCM_FORMAT_MU_LAW,
SND_PCM_FORMAT_A_LAW,
SND_PCM_FORMAT_IMA_ADPCM,
SND_PCM_FORMAT_MPEG,
SND_PCM_FORMAT_GSM,
SND_PCM_FORMAT_SPECIAL,
SND_PCM_FORMAT_S24_3LE,
SND_PCM_FORMAT_S24_3BE,
SND_PCM_FORMAT_U24_3LE,
SND_PCM_FORMAT_U24_3BE,
SND_PCM_FORMAT_S20_3LE,
SND_PCM_FORMAT_S20_3BE,
SND_PCM_FORMAT_U20_3LE,
SND_PCM_FORMAT_U20_3BE,
SND_PCM_FORMAT_S18_3LE,
SND_PCM_FORMAT_S18_3BE,
SND_PCM_FORMAT_U18_3LE,
SND_PCM_FORMAT_U18_3BE,
};

static const unsigned int rates[] = {
5512,
8000,
11025,
16000,
22050,
32000,
44100,
48000,
64000,
88200,
96000,
176400,
192000,
};

int main(int argc, char *argv[])
{
const char *device_name = "hw";
snd_pcm_t *pcm;
snd_pcm_hw_params_t *hw_params;
unsigned int i;
unsigned int min, max;
int any_rate;
int err;

if (argc > 1)
device_name = argv[1];

err = snd_pcm_open(&pcm, device_name, SND_PCM_STREAM_PLAYBACK, 
SND_PCM_NONBLOCK);
if (err < 0) {
fprintf(stderr, "cannot open device '%s': %s\n", device_name, 
snd_strerror(err));
return 1;
}

snd_pcm_hw_params_alloca(&hw_params);
err = snd_pcm_hw_params_any(pcm, hw_params);
if (err < 0) {
fprintf(stderr, "cannot get hardware parameters: %s\n", 
snd_strerror(err));
snd_pcm_close(pcm);
return 1;
}

printf("Device: %s (type: %s)\n", device_name, 
snd_pcm_type_name(snd_pcm_type(pcm)));

printf("Access types:");
for (i = 0; i < ARRAY_SIZE(accesses); ++i) {
if (!snd_pcm_hw_params_test_access(pcm, hw_params, accesses[i]))
printf(" %s", snd_pcm_access_name(accesses[i]));
}
putchar('\n');

printf("Formats:");
for (i = 0; i < ARRAY_SIZE(formats); ++i) {
if (!snd_pcm_hw_params_test_format(pcm, hw_params, formats[i]))
printf(" %s", snd_pcm_format_name(formats[i]));
}
putchar('\n');

err = snd_pcm_hw_params_get_channels_min(hw_params, &min);
if (err < 0) {
fprintf(stderr, "cannot get minimum channels count: %s\n", 
snd_strerror(err));
snd_pcm_close(pcm);
return 1;
}
err = snd_pcm_hw_params_get_channels_max(hw_params, &max);
if (err < 0) {
fprintf(stderr, "cannot get maximum channels count: %s\n", 
snd_strerror(err));
snd_pcm_close(pcm);
return 1;
}
printf("Channels:");
for (i = min; i <= max; ++i) {
if (!snd_pcm_hw_params_test_channels(pcm, hw_params, i))
printf(" %u", i);
}
putchar('\n');

err = snd_pcm_hw_params_get_rate_min(hw_params, &min, NULL);
if (err < 0) {
fprintf(stderr, "cannot get minimum rate: %s\n", 
snd_strerror(err));
snd_pcm_close(pcm);
return 1;
}
err = snd_pcm_hw_params_get_rate_max(hw_params, &max, NULL);
if (err < 0) {
fprintf(stderr, "cannot get maximum rate: %s\n", 
snd_strerror(err));
snd_pcm_close(pcm);
return 1;
}
printf("Sample rates:");
if (min == max)
printf(" %u", min);
else if (!snd_pcm_hw_params_test_rate(pcm, hw_params, min + 1, 0))
printf(" %

Re: [Alsa-user] alsamixer does not save settings

[Clemens Ladisch]

Geoffrey Leach wrote:
> I'm havind a problem with alsamixer. It appears not to save the
> changes. The man page does not mention this, so perhaps its doing what
> it is designed to do.

It is only designed to change the sound card's mixer settings.

> In that case, can anyone point me to a tool that makes permanent
> (mute, in this case) settings?

"alsactl store" and "alsactl restore" do what you want.  In theory,
these calls should be made automatically by your distribution's shutdown/
startup scripts.


HTH
Clemens

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Re: [Alsa-user] DELL SP2309W Monitor with Integrated microphone

[Clemens Ladisch]

Paul Hartman wrote:
> On Fri, Apr 17, 2009 at 2:27 AM, Clemens Ladisch
>  wrote:
> > Phil Gorbett wrote:
> > > I am having difficulty getting the microphone(s) going with this
> > > monitor, and get the get the following results when I disable pulseaudio
> > > and run arecord:
> > >
> > > "arecord -v -f cd -D plughw:0 file.wav
> > > Recording WAVE 'file.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, 
> > > Stereo
> > > ALSA lib pcm_params.c:2135:(snd1_pcm_hw_refine_slave) Slave PCM not usable
> > > arecord: set_params:939: Broken configuration for this PCM: no 
> > > configurations available"
> >
> > Please show the output of "lsusb -v" for this device.
> 
> I'm not the OP but I have the same monitor and same problem. Here's
> the lsusb output for the microphone:
> 
> http://pastebin.com/f4c340014

These descriptors claim it is a simple standard-compliant device.

When you run that arecord command, is there any error message in the
system log (/var/log/messages, or the output of dmesg)?


Best regards,
Clemens

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Re: [Alsa-user] pcm_min.c trouble with modification

[Clemens Ladisch]

Michał Kowalczyk wrote:
>   SND_PCM_FORMAT_U8, //changing this type for any other gives an error in 
> the output: segmentation fault.

U8 has one byte per sample; if you try to use a bigger sample format,
you have to enlarge the buffer accordingly.

>   1, //argument responsible for channels; if I change it to 2(stereo) 
> program gives an segmentation fault error.

Using two channels would double the buffer size, too.

>   48000, //sampling rate; works good but if I modify this to lower value 
> the time of generating noise will be longer, don't know why.

If the sample rate is lower, you need more timer to play the same number
of samples.

> frames = snd_pcm_writei(handle, buffer, sizeof(buffer));

The last parameter is measued in frames, not in bytes.  When you use
another sample format than U8/S8 or more then one channel, you have to
divide the buffer size by the number of bytes per frame.


HTH
Clemens

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Re: [Alsa-user] emu10k firmware

[Clemens Ladisch]

Ari Moisio wrote:
>   Thanks:-) I  downloaded the relevant firmware package  from alsa-project 
> site, configured, compled and installed. Everthing is working again:)
> 
>   I only wonder if the driver itself loaded the firmware earlier because 
> there were no such issues with 2.4 kernels?

In older driver versions, the firmware was included directly in the
driver, but this was changed to allow the driver to be used with Debian.
It is still possible to compile the firmware into the kernel, but it
seems your distribution did not choose to do this.


HTH
Clemens

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Re: [Alsa-user] Musical Fidelity V-DAC, usb sound

[Clemens Ladisch]

Niels wrote:
> - This is with Kubuntu 9.04. For some reason it ships with ALSA 1.0.18rc3.
> Is it worth upgrading,

Probably not, AFAIK there haven't have any relevant changes.

> and how would I do that, other that wait for an automatic update?

Download, compile and install the 1.0.20 alsa-driver package.

> - speaker-test says it outputs to "Front left", but it actually outputs to
> both left and right.

Doesn't it ever output to "front right"?  This looks like a mono mode.

What device name and number of channels did you tell speaker-test to use?

> - How do I disable dmix and other possible sources of signal degradation
> completely?

Dmix does not degrade the signal because its design requires it to run
with a sample rate and a sample format directly supported by the device.

You can disable dmix by telling your application(s) to use another
device name instead of "default".

> - How do I make sure ALSA doesn't resample or normalize the signal along the
> way?

Use a device name like "dmix" or "hw".
"default" and "plughw" include resampling, if needed.

In no case is there any normalization.

> - How do I test various sample rates and bit depths?

Play a file with the desired properties to the "hw" device.

> How do I get ALSA to tell me what the device supports?

Have a look into /proc/asound/card???/stream0.


HTH
Clemens

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Re: [Alsa-user] Capture w/ M-Audio Fast Track USB?

[Clemens Ladisch]

Michael B Allen wrote:
> Does capture work with the M-Audio Fast Track USB?

Maybe.  Does some capture device show up in the output of "arecord -l"?

> Can someone give me some advice or a pointer to some documentation
> that explains how the whole ALSA tool chain works wrt external USB
> sound recording?

You can tell any ALSA program to use a specific device by using a device
name like "plughw:x,y" if you know the card and device numbers.

If you're using PulseAudio, you have to tell it to use that device.


HTH
Clemens

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Re: [Alsa-user] alsa-utils-1.0.20.12.g730e compile problem

[Clemens Ladisch]

Raena wrote:
> I also installed the snapshot alsa-lib and that was fine but alsa-utils 
> would not configure. configure: error: panelw library not found I have 
> not been able to find out what package has this library. Here is my 
> configure output;
> ...
> checking for ncursesw5-config... yes 

This indicates that ncursesw is installed ...

> checking for new_panel in -lpanelw... no
> configure: error: panelw library not found

... but actually it isn't.

What is the output of the following commands?

  ncursesw5-config --libs
  ncursesw5-config --libdir
  locate libncurses.so
  locate libncursesw.so
  locate libpanel.so
  locate libpanelw.so


Best regards,
Clemens

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Re: [Alsa-user] alsa-utils-1.0.20.12.g730e compile problem

[Clemens Ladisch]

Raena Lea-Shannon wrote:
> Is there another ncurses-devel I have missed?

libncursesw5-dev?


HTH
Clemens

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Re: [Alsa-user] ymfpci drive capability - help needed with setup

[Clemens Ladisch]

robert crowther wrote:
> Can ymfpci access the WF-192XG wavetable? 

No.  While documentation for basic DS-XG wavetable functionality exists,
nobody has as yet added support for this to the driver.


Best regards,
Clemens

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Re: [Alsa-user] MOTU micro lite (USB, 5in, 5out)

[Clemens Ladisch]

Aidan Dixon wrote:
> I have a MOTU micro lite (USB, 5x MIDI IN and 5x MIDI OUT, Vendor ID
> 0x07fd, product ID 0x0001).  According to the kernel usb-audio device
> driver source code, this device should be supported as its USB vendor
> and product ID are listed in a table in linux/sound/usb/usbmidi.c.

It seems MOTU didn't quite grasp the concept of "product ID"; they use
the same ID 0x0001 for all their USB MIDI interfaces.  The driver checks
that the device is actually a FastLane before attaching; all other MOTU
models use a completely different protocol and are not supported.

> What are the issues beyond simply lack of information from MOTU? 

Lack of developers who would have time to reverse engineer it and to
write a driver.


Best regards,
Clemens

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Re: [Alsa-user] record vinyl at 24-bit 96khz

[Clemens Ladisch]

> scar wrote:
> > i was basically just looking for verification that the m-audio 2496 card
> > would suit my needs.

Yes, it certainly would.

gary wrote:
> I often wonder has VIA came out of nowhere with a DSP audio chip 
> (Envy24). Oversampled ADCs are not exactly rocket science, but to make 
> one with a product history of previous chips is suspicious.

VIA got the ICE1712 chip design when it took over IC Ensemble.  It then
develop a bunch of derived Envy24xxx chips from that.


Best regards,
Clemens

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Re: [Alsa-user] alsa-utils-1.0.20.12.g730e compile problem

[Clemens Ladisch]

Raena wrote:
> Clemens Ladisch wrote:
>> Raena wrote:
>>> checking for ncursesw5-config... yes
>> 
>> This indicates that ncursesw is installed ...
>> 
>>> checking for new_panel in -lpanelw... no
>>> configure: error: panelw library not found
>> 
>> ... but actually it isn't.
>> 
>> What is the output of the following commands?
>>   ...
>>   locate libpanel.so
>>   locate libpanelw.so
>
> ra...@kubuntu:~$ locate libpanel.so
> /usr/lib/libpanel.so
> /usr/lib/libpanel.so.5
> /usr/lib/libpanel.so.5.7

This is OK ...

> ra...@kubuntu:~$ locate libpanelw.so
> /usr/lib/libpanelw.so.5
> /usr/lib/libpanelw.so.5.7

... but this isn't: the libpanelw.so link is missing.

Please file a bug report with your distribution.


You could create the missing files yourself with these commands:

ln -s libpanelw.so.5 /usr/lib/libpanelw.so
ln -s libformw.so.5 /usr/lib/libformw.so
ln -s libmenuw.so.5 /usr/lib/libmenuw.so
ln -s libncursesw.so.5 /lib/libncursesw.so


(You don't actually need the new alsa-utils, except if you want to see
the Japanese translation of alsamixer. :-)


Best regards,
Clemens

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Re: [Alsa-user] dmix oddities

[Clemens Ladisch]

Grant wrote:
> > Are you sure the hardware actually supports S24_3LE?  Most 24 bit
> > soundcards require 24 bit audio to be packed into 32 bit words.  See
> > if you can output S24_3LE directly to the hw device.
> 
> I don't think a USB DAC could do 24-in-32 because of the USB 24-bit 
> limitation.
> 
> I've tried specifying S24_3LE, S24LE, S24_BE, S24_3BE, FLOAT_LE,
> FLOAT_BE, and S16_LE in asound.conf.  None of them produce sound
> except for S24_3LE and S16_LE.  How can I output S24_3LE directly to
> the device for testing?

24-bit USB devices use S24_3LE.
24-bit PCI cards usually use S32_LE (where only 24 bits are used).


HTH
Clemens

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Re: [Alsa-user] VIA8237 and no sound

[Clemens Ladisch]

Mike Knichel wrote:
> I have recently set up a computer for my daughter from old parts.  It
> uses the VIA 8237 chip.  I can get no sound from speakers.
> I even ran the alsa script and the results are here...

Your mixer settings seem to be correct.

However, some mainboards use another than the front output for their
front jack.  Try plugging the speakers into another output, and/or
changing the settings so that it play to more outputs (Exchange Front/
Surround, or Channel Mode 6ch, Surround Jack Mode Independent, Spread
Front to Surround and Center/LFE).


HTH
Clemens

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Re: [Alsa-user] VIA8237 and no sound

[Clemens Ladisch]

Mike Knichel wrote:
> I do not have front mounted speaker jacks.  Only the ones in the back.

I meant "front" as opposed to "surround" or "center/LFE".

If you have only one jack intended for speakers, then your description
looks as if that jack is connected to the "surround" output of the
codec.  Please try setting the "Surround Jack Mode" control to
"Independent" and enabling the "Exchange Front/Surround" control.


HTH
Clemens

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Re: [Alsa-user] Known C-Media 8738 issues?

[Clemens Ladisch]

Malte Gell wrote:
> when I use Skype 2.0.0.72 on Linux I get an error:
> 
> RtApiAlsa: callback thread error (RtApiAlsa: audio write error for device (C-
> Media CMI8738 (hw:CMI8738,0)): unknown error 405.) ... closing stream.

"Unknown error" isn't quite helpful.

> I did not have problems before when I use my good old Soundblaster Live! and 
> so I wonder, are there any known issues with the CMedia 8738 chip?

Nothing comes to mind.

> Do you think I could get rid of this Skype issue when I upgrade to a CMedia 
> 8768 chip?

That's exactly the the same hardware interface (and driver), so I'd
guess you'd get the same error.


Best regards,
Clemens


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Re: [Alsa-user] Known C-Media 8738 issues?

[Clemens Ladisch]

Malte Gell wrote:
> Clemens Ladisch  wrote
> > Malte Gell wrote:
> > > Do you think I could get rid of this Skype issue when I upgrade to a
> > > CMedia 8768 chip?
> > 
> > That's exactly the the same hardware interface (and driver), so I'd
> > guess you'd get the same error.
> 
> Ok, same driver, but different chip, isn't it?

The internals of the chip might be somewhat different, but the interface
to the driver is exactly the same (except for a bit to enable 8 channels),
and since the problem is apparently not with the hardware but with the
way how Skype accesses the device, the same thing is likely to happen
with the 8768.


Best regards,
Clemens

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Re: [Alsa-user] can't independently adjust Mic levels

[Clemens Ladisch]

scar wrote:
> so, i got my asus xonar dx ...
> the problem i am running into is i cannot independently adjust the left
> and right levels of the Mic capture.  well, i can, kind of, but it
> doesn't have any effect.

The microphone input is mono.  There are no left/right levels; the
driver pretending that they exist is a driver bug.  I'll fix that.


HTH
Clemens

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Re: [Alsa-user] can't independently adjust Mic levels

[Clemens Ladisch]

scar wrote:
> Clemens Ladisch @ 06/24/2009 12:53 AM:
> > The microphone input is mono.  There are no left/right levels; the
> > driver pretending that they exist is a driver bug.  I'll fix that.
> 
> but that port is also the line-in, so i need stereo

When used as line in, that port is stereo.  (It then also expects line
level and does not allow to change the volume.)


Best regards,
Clemens

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Re: [Alsa-user] can't independently adjust Mic levels

[Clemens Ladisch]

scar wrote:
> i have been experimenting, and what appears to be happening is this:
> while recording using the line, if the microphone levels are adjusted,
> the driver/card automatically jumps to recording from the mic.

Neither the driver nor the card do this; I'd guess this is a feature of
your mixer application.

> still remaining is the issue of not being able to adjust the left and
> right line volumes.  it would be nice to have that extra functionality.

The hardware does not have any mechanism to change the line input's
level.  You have to adjust the volume after recording.


Best regards,
Clemens

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Re: [Alsa-user] can't independently adjust Mic levels

[Clemens Ladisch]

scar wrote:
> Clemens Ladisch @ 06/26/2009 03:30 AM:
> > The hardware does not have any mechanism to change the line input's
> > level.  You have to adjust the volume after recording.
>
> shown in section 6.3.2 of the xonar dx manual is a picture of the levels
> for recording, indicating there is an adjustment for line-in.

The Windows driver does this in software.

> also throughout section 6, i am noticing a bunch of other options for
> sound/speaker settings that i am not sure how to take advantage of in
> linux or with ALSA.  can someone explain?

All those effect and encoders are implemented in software, too.


Best regards,
Clemens

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Re: [Alsa-user] Edirol UA-101: Does record work?

[Clemens Ladisch]

Phil wrote:
> I'm trying to find a way to get audio in over USB for a project, and I
> think that the UA-101 might be about the only Linux device that supports
> 6 or more channels of audio over USB.

Indeed.  There are other multichannel USB devices, but there are none
that have even rudimentary Linux support.

> As I understand from the kernel comments, there are glitches on playback
> but recording works, does this mean all 10 channels work on record?

>From what I've heard , recording works perfectly.

> If record does work, then that would probably be enough to buy the
> thing, and I'll sort out playback separately.

Somebody is currently trying to write a driver that would make playback
work, too.


Best regards,
Clemens

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Re: [Alsa-user] Mic not working with arecord on RedHat el5

[Clemens Ladisch]

Saleem Hasan wrote:
> I found that although aplay was working on my computer, arecord was not. I
> played back the wav files provided with the linux distribution for various
> system beeps and sounds. I am able to navigate through the alsamixer
> screen using tab, space, etc in order to select the mic, capture, mic
> boost, master and other settings but arecord still does not record. I have
> used the alsamixer -vv and the vu meter either stays at 0% or varies
> between 0-5% and the file when played back is only static. 

Try disabling "Headphone Jack Sense".


HTH
Clemens

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Re: [Alsa-user] KnoppMyth/LinHES Linux - Analog audio out to ear phones works but not to TV. Ubuntu 9-i386 LiveCD everything works fine!

[Clemens Ladisch]

Matt Snow wrote:
> I have a Asus Xonar DX sound card that was working perfectly in
> KnoppMyth/LinHES R5.5(Knoppix/debian based) after manually compiling
> alsa-driver 1.0.18b and 1.0.18 libs/utils/etc.
> After upgrading to KnoppMyth/LinHES R6(ArchLinux), the analog out
> plays audio perfectly through head phones or powered speakers, but the
> 3.5mm stereo-to-RCA cable going to RCA input on the TV doesn't produce
> any sound at all.

Isn't that the same jack on the card?  Or where have you connected the
headphones, the speakers and the TV?

What version of ALSA are you using?  (see /proc/asound/version)


Best regards,
Clemens

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Re: [Alsa-user] Fwd: KnoppMyth/LinHES Linux - Analog audio out to ear phones works but not to TV. Ubuntu 9-i386 LiveCD everything works fine!

[Clemens Ladisch]

Matt Snow wrote:
> On Thu, Jul 9, 2009 at 12:05 AM, Clemens Ladisch 
> wrote:
> > Matt Snow wrote:
> > > After upgrading to KnoppMyth/LinHES R6(ArchLinux), the analog out
> > > plays audio perfectly through head phones or powered speakers, but the
> > > 3.5mm stereo-to-RCA cable going to RCA input on the TV doesn't produce
> > > any sound at all.
> > 
> > Isn't that the same jack on the card?  Or where have you connected the
> > headphones, the speakers and the TV?
> 
> there are several analog jacks on the card for different channels. i'm
> using the front speaker jack, but have tried all jacks with the TV and
> ear phones.

So the headphones or speakers in the front jack work, and the to-RCA
cable in the *same* jack does not?  This is obviously a broken cable
(or TV).


Best regards,
Clemens

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Re: [Alsa-user] Fwd: KnoppMyth/LinHES Linux - Analog audio out to ear phones works but not to TV. Ubuntu 9-i386 LiveCD everything works fine!

[Clemens Ladisch]

Matt Snow wrote:
> On Fri, Jul 10, 2009 at 2:20 AM, Clemens Ladisch 
> wrote:
> > So the headphones or speakers in the front jack work, and the to-RCA
> > cable in the *same* jack does not?  This is obviously a broken cable
> > (or TV).
> 
> You might think that, but its fine. I take the exact same hdmi+dvi
> adaptor, and the rca-to-stero cable, plug in to either my macbook pro,
> or macbook with mini displayport adaptor to hdmi and I get audio and
> video.

What happens when you connect the MacBook to the TV with the HDMI+DVI
adapter and, at the same time, the Xonar DX to the TV with the RCA-to-
stereo cable?


Best regards,
Clemens

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Re: [Alsa-user] mmap_readi or mmap_begin

[Clemens Ladisch]

Marek Michalak wrote:
> Is there any difference if I write in my code:
> while (size > 0) {
> frames = size;
> err = snd_pcm_mmap_begin(handle, &my_areas, &offset, &frames);
> record_buffor(my_areas, offset, frames, &data->phase);
> snd_pcm_mmap_commit(handle, offset, frames);
> size -= frames;
> }
> OR
> while (rest > 0) {
> size_t c = (rest <= (off64_t)chunk_bytes) ? (size_t)rest :
> chunk_bytes;
> size_t f = c * 8 / bits_per_frame;
> snd_pcm_mmap_readi(audiobuf, f);
> write(fd, audiobuf, c);
> rest -= c;
> }

This depends on what the function record_buffor() does.

snd_pcm_readi() (or the equivalent snd_pcm_mmap_readi() from your second
example) copies data from the sound card's buffer into the user's buffer.
If record_buffor() copies the data elsewhere, too, then there would be
_no_ difference in what the code does.

Using a memory-mapped buffer makes sense only if you actually use the
data in the buffer instead of copying it, i.e., if you make your
computations directly on the data as it is in the buffer.

BTW: Using snd_pcm_mmap_readi() doesn't make much sense if don't also
use snd_pcm_mmap_begin() in your program, because snd_pcm_readi() does
the same but does not require a device that supports mmap.

> I want to have low latency capture and compute fft - is it good way to do
> this?

If your FFT algorithm
* can read the data directly from the sound card buffer,
* does not write to the sound card buffer, and
* can handle the period/buffer layout (which is hardware dependent; not
  all sound cards may support your desired period/buffers sizes),
then you can use snd_pcm_mmap_*.
Otherwise, you must use snd_pcm_readi().

Assuming that the FFT needs floating-point numbers, the int-to-float
conversion would be an algorithm that could be easily made to work with
mmap.  Or you could write the first step of the FFT so that it reads
integer data from the mmap buffer, converts it to floats, does the
computation, and writes the results to its own processing buffer.

BTW: When capturing, latency depends only on the period size, not on the
buffer size, so you should use as big a buffer as possible to avoid
overruns.

> Is true that snd_async_handler_t *ahandler is better for realtime system
> than nonblocking mode?

No.  Asynchronous handlers notify your program with a signal, but you
are not allowed to do anything useful inside the signal handler, so
you'd have to notify your own thread somehow.  Using blocking reads or
poll() is more efficient.


Best regards,
Clemens

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Re: [Alsa-user] Strange "knocking" while playing sound

[Clemens Ladisch]

Dennis Borgmann wrote:
> I am working with snd_pcm_mmap_begin and snd_pcm_mmap_commit - just the
> way it is done in the "pcm.c" under the "test"-directory of the
> alsa-lib. While working with snd_pcm_writei(), those problems did not
> occur, but I want to test this using the mmap technology.

This sounds like a problem with your mmap code.

As long as you are copying samples from some buffer into the sound
card's buffer, using mmap does not give you any benefit because this is
exactly the same as snd_pcm_writei() does.


Best regards,
Clemens

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Re: [Alsa-user] Terratec Phase28

[Clemens Ladisch]

Christoph Köditz wrote:
> is the driver in work or do you need some more informations about this 
> soundcard?

According to the driver's source, this card has been supported since 2005.


HTH
Clemens

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Re: [Alsa-user] Meaning of these options?

[Clemens Ladisch]

Geoffrey Leach wrote:
> --with-debug=full
> 
> Where does the output go?

Into the system log.

> --enable-dynamic-minors
> 
> What's a "dynamic minor" and what is the significance of enabling it?

>From linux/sound/core/Kconfig:
| config SND_DYNAMIC_MINORS
| bool "Dynamic device file minor numbers"
| help
|   If you say Y here, the minor numbers of ALSA device files in
|   /dev/snd/ are allocated dynamically.  This allows you to have
|   more than 8 sound cards, but requires a dynamic device file
|   system like udev.


HTH
Clemens

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Re: [Alsa-user] Fedora core 11. No sound

[Clemens Ladisch]

Dave Pawson wrote:
> 2009/7/24 Robert Hancock :
> > Looks like only the HDMI audio output is being detected. What kind of audio
> > outputs does the machine actually have?
> 
> It's an Asus p5Q Pro motherboard.
> $ cat /proc/asound/cards
>  0 [HDMI   ]: HDA-Intel - HDA ATI HDMI
>   HDA ATI HDMI at 0xfe8fc000 irq 17

This is your graphic card's HDMI output.

> I'm using the analogue outputs from the back to speakers.

Does the onboard HDA controller show up in the output of lspci?


Best regards,
Clemens

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Re: [Alsa-user] Fedora core 11. No sound

[Clemens Ladisch]

Dave Pawson wrote:
> 2009/7/24 Clemens Ladisch :
> > Does the onboard HDA controller show up in the output of lspci?
> 
> # lspci | grep Audio
> 00:1b.0 Audio device: Intel Corporation 82801JI (ICH10 Family) HD Audio 
> Controller
> 01:00.1 Audio device: ATI Technologies Inc R700 Audio Device [Radeon HD 4000 
> Series]

Both should have been picked up by the driver.

What are the contents of /sys/module/snd_hda_intel/parameters/enable?


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Clemens

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Re: [Alsa-user] Fedora core 11. No sound

[Clemens Ladisch]

Dave Pawson wrote:
> 2009/7/24 Clemens Ladisch :
>> Dave Pawson wrote:
>>> # lspci | grep Audio
>>> 00:1b.0 Audio device: Intel Corporation 82801JI (ICH10 Family) HD Audio 
>>> Controller
>>> 01:00.1 Audio device: ATI Technologies Inc R700 Audio Device [Radeon HD 
>>> 4000 Series]
>>
>> Both should have been picked up by the driver.
>>
>> What are the contents of /sys/module/snd_hda_intel/parameters/enable?
> 
> Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y

Is there any error message in the system log when you do this:
rmmod snd-hda-intel
modprobe snd-hda-intel
dmesg | tail


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Clemens

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Re: [Alsa-user] Fedora core 11. No sound

[Clemens Ladisch]

Dave Pawson wrote:
> 2009/7/24 Clemens Ladisch :
> 
>> Is there any error message in the system log when you do this:
>>rmmod snd-hda-intel
> 
> $ rmmod snd-hda-intel
> ERROR: Module snd_hda_intel is in use

Please kill all applications that use it, then try again.
(I guess you could try to temporatily disable PulseAudio.)


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Clemens

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Re: [Alsa-user] IEC958 Playback Source = AC-Link or A/D Converter ?

[Clemens Ladisch]

Goga777 wrote:
> I have integrated VIA8237 sound card and I have the problem with Dolby 
> Digital audio through Spdif (mono and stereo is
> working through spdif) - I don't listen anything with DD with AC-Link and I 
> listen the sound like ЭырррЭ
> with A/D Converter/
> 
> My question - what should I choice in alsamixer - 'AC-Link' or 'A/D 
> Converter' for spdif output ?

Use "AC-Link"; "A/D Converter" is the analog input.

How exactly are you trying to play DD?


Best regards,
Clemens

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Re: [Alsa-user] Alsa module loading order query.

[Clemens Ladisch]

philicorda wrote:
> options snd-ice1712 index=3
> options snd-ice1712 index=4

The second options line overrides the first one; use instead:
  options snd-ice1712 index=3,4


HTH
Clemens

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Re: [Alsa-user] Interesting USB-Audio Device Problem

[Clemens Ladisch]

Dar Austin Zaccaro wrote:
> I'm trying to get my M-Audio FastTrack Ultra USB audio interface to work
> under GNU/Linux. JACK seems to recognize the card, however ALSA does not
> want to work with it. JACK prints the device as "hw:1 Fast Track Ultra"
> and `/proc/asound/cards` agrees with JACK:
> 
> 0 [Intel  ]: HDA-Intel - HDA Intel
>  HDA Intel at 0xf650 irq 22
> 1 [Ultra  ]: USB-Audio - Fast Track Ultra
>  M-Audio Fast Track Ultra at usb-:00:1a.7-4, high 
> speed
> 
> This is strange/interesting because `aplay -l` does NOT list the
> FastTrack Ultra:
> 
>  List of PLAYBACK Hardware Devices 
> card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog]
>   Subdevices: 1/1
>   Subdevice #0: subdevice #0

Please show the output of "lsusb -v" for this device.


Best regards,
Clemens

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Re: [Alsa-user] Interesting USB-Audio Device Problem

[Clemens Ladisch]

Dar Austin Zaccaro wrote:
> Here's the output of `lsusb -v`:

The MIDI interface is class compliant, and this is what's shown by Jack.
The audio interfaces are marked as vendor specific, so the driver needs
to be changed to support them.

Please try recompiling your kernel with the patch below.

Some M-Audio devices use big-endian samples, so beware of speaker-
killing noise.

It's possible that playback synchronization doesn't work completely.

Please check what happens when you try to play and record with different
sample rates.


HTH
Clemens

--- linux/sound/usb/usbaudio.c
+++ linux/sound/usb/usbaudio.c
@@ -2235,6 +2235,10 @@ static void init_substream(struct snd_us
case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */
subs->ops.retire_sync = retire_playback_sync_urb_hs_emu;
break;
+   case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra */
+   subs->ops.prepare_sync = prepare_playback_sync_urb;
+   subs->ops.retire_sync = retire_playback_sync_urb;
+   break;
}
}
snd_pcm_set_ops(as->pcm, stream,
@@ -2786,6 +2790,7 @@ static int parse_audio_endpoints(struct 
break;
case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */
case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */
+   case USB_ID(0x0763, 0x2080): /* M-Audio Fast Track Ultra */
/* doesn't set the sample rate attribute, but supports 
it */
fp->attributes |= EP_CS_ATTR_SAMPLE_RATE;
break;
--- linux/sound/usb/usbquirks.h
+++ linux/sound/usb/usbquirks.h
@@ -1864,6 +1864,33 @@
}
}
 },
+{
+   USB_DEVICE_VENDOR_SPEC(0x0763, 0x2080),
+   .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+   /* .vendor_name = "M-Audio", */
+   /* .product_name = "Fast Track Ultra", */
+   .ifnum = QUIRK_ANY_INTERFACE,
+   .type = QUIRK_COMPOSITE,
+   .data = & (const struct snd_usb_audio_quirk[]) {
+   {
+   .ifnum = 0,
+   .type = QUIRK_IGNORE_INTERFACE
+   },
+   {
+   .ifnum = 1,
+   .type = QUIRK_AUDIO_STANDARD_INTERFACE
+   },
+   {
+   .ifnum = 2,
+   .type = QUIRK_AUDIO_STANDARD_INTERFACE
+   },
+   /* interface 3 (MIDI) is standard compliant */
+   {
+   .ifnum = -1
+   }
+   }
+   }
+},
 
 /* Casio devices */
 {

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Re: [Alsa-user] 8x8 MIDI Interface

[Clemens Ladisch]

Arthur Peters wrote:
> What 8x8 MIDI interface should I use with ALSA (ideally USB)?

Good question.

> I already have a MOTU Express XT because the company I am working for
> (on this project) already has them for use with Macs. But they don't
> work with Linux. From the research I have done there is no fix to this
> at all.
> 
> Oddly however when you plug the MOTU unit into the USB port on linux
> it actually does connect and it appears as a 2x2 MIDI interface.

It's misdetected as a Fastlane.  This should not happen; please show
the contents of your /proc/bus/usb/devices.

> Are there any currently commercially available 8x8 interfaces that
> work with Linux?

The M-Audio Midisport 8x8, the Emagic Unitor8 and AMT8, and the Roland
UM-880 have been discontinued; you might find a used one.


HTH
Clemens

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Re: [Alsa-user] Multichannel hdmi video cards

[Clemens Ladisch]

Shane W wrote:
> I am looking for a standalone video card supporting
> multichannel HDMI at 96KHZ or 192KHZ output, similar to the
> Intel G45.
> ...
> Asus xonar hdav1.3, alsa support experimental but the
> hardware is certainly capable.

But the driver is not:
http://www.alsa-project.org/main/index.php/Matrix:Vendor-Asus

I'd strongly suggest a video card that supports HDMI audio output
by itself.


Best regards,
Clemens

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Re: [Alsa-user] ALSA, multiple recording apps, device busy, dsnoop

[Clemens Ladisch]

telmn...@757.org wrote:
> pcm.record_left {
>  ipc_key 234884
>  pcm "hw:0,0"
> 
> pcm.record_right {
>  ipc_key 2241234
>  pcm "hw:0,0"

dmix/dshare/dsnoop devices that want to share the same hardware device
must use the same ipc_key.


HTH
Clemens

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Re: [Alsa-user] Sound Card Configuration

[Clemens Ladisch]

John Beavers wrote:
> I have a soundblaster Live that I am trying to have the outputs used as
> independant virtual devices.
>
> aplay -l
> card 0: Live [Dell Sound Blaster Live!], device 0: emu10k1x [EMU10K1X Front]
> card 0: Live [Dell Sound Blaster Live!], device 1: emu10k1x [EMU10K1X Rear]
> card 0: Live [Dell Sound Blaster Live!], device 2: emu10k1x [EMU10K1X 
> Center/LFE]

Your card already has independent hardware devices.

> Basically, there are two X sessions running with two users logged into
> them (this is called multiseating).

To make a different device the default device for two different users,
put the following settings in their ~/.asoundrc files:

defaults.pcm.dmix.device 0

defaults.pcm.dmix.device 1

(This is user specific, not X session specific.)


HTH
Clemens

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Re: [Alsa-user] Sound Card Configuration

[Clemens Ladisch]

John Beavers wrote:
> I tried that, that particular suggestion didn't work,

It assumes that you use the default dmix configuration for your
sound card.

> but I changed it to 
> 
> pcm.!default rear:Live  and 
> 
> pcm.!default front:Live
> 
> which did work partially.  However, it broke sound in firefox.  It also
> messed up some flash programs.

The "front" and "rear" devices do not do sample rate conversion or
software mixing.

> Also, I was hoping to get the devices X session dependant.  I heard someone
> suggest chrooting each X session, copying the dev directory and linking
> /dev/dsp to /dev/dsp[x] (whatever number card you want for the given X
> session).

That suggestion is for programs using the OSS interface, but your
description above shows that Firefox and Flash use the ALSA interface,
so you should be able to get this to work with ALSA configuration files.

If you set an environment variable ALSA_DEVICE to 0 or 1 for each
session, the following definition in /etc/asound.conf should do what
you want:

pcm.!default {
type plug
slave.pcm {
@func concat
strings [
"dmix:CARD=Live,DEV="
{
@func getenv
vars [ ALSA_DEVICE ]
default 0
}
]
}
}


HTH
Clemens

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Re: [Alsa-user] audio loopback in linux

[Clemens Ladisch]

George Kuriakose wrote:
> I want to loop back audio coming in line-in to line-out with minimum
> latency. Is there any option in amixer or any other similar utility to
> achieve this?

Yes, but only if the hardware supports this.
If not, use some program to record and play the same data back.


Best regards,
Clemens

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