[pulseaudio-discuss] Alsamixertest

2010-09-27 Thread David Henningsson
So over the previous weeks I've been working on a small script which 
tests whether the ALSA mixer lives up to PA's expectations. If you are 
familiar with dbmeasure or dbverify by Lennart Poettering, this 
application's purpose is very similar, but this one is hopefully easier 
to set up, more user friendly, and also tests that the names of the 
volume controls are correct.
My hope is that this will aid as a debugging tool for all these 
everything below 20% of my speaker is muted, and then 21% blows my 
speakers bugs.


To use the tool, you'll need some kind of loopback. You can e g use a 
loopback cable and connect that between line in and line out, or test 
your laptop's internal speakers with your laptop's internal mic (just 
stop humming when you do so :-) ). Just set up the recording levels 
appropriately.


Alsamixertest is available for Ubuntu Lucid and Ubuntu Maverick from 
these PPAs:

Lucid: https://launchpad.net/~diwic/+archive/ppa
Maverick: https://launchpad.net/~diwic/+archive/maverick

For other distributions, download the tarball:
https://launchpad.net/~diwic/+archive/ppa/+files/alsamixertest_47.14.tar.gz
Unpack and read the readme file for compilation and install instructions.

When it is installed, run alsamixertest -r for a small tutorial and 
alsamixertest -h for command line options help.


Looking forward to your comments about this new little tool! I think it 
should be considered beta quality at this point.


--
David Henningsson, Canonical Ltd.
http://launchpad.net/~diwic
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Re: [pulseaudio-discuss] Alsamixertest

2010-09-27 Thread Colin Guthrie
'Twas brillig, and David Henningsson at 27/09/10 09:53 did gyre and gimble:
 So over the previous weeks I've been working on a small script which
 tests whether the ALSA mixer lives up to PA's expectations. If you are
 familiar with dbmeasure or dbverify by Lennart Poettering, this
 application's purpose is very similar, but this one is hopefully easier
 to set up, more user friendly, and also tests that the names of the
 volume controls are correct.
 My hope is that this will aid as a debugging tool for all these
 everything below 20% of my speaker is muted, and then 21% blows my
 speakers bugs.
 
 To use the tool, you'll need some kind of loopback. You can e g use a
 loopback cable and connect that between line in and line out, or test
 your laptop's internal speakers with your laptop's internal mic (just
 stop humming when you do so :-) ). Just set up the recording levels
 appropriately.
 
 Alsamixertest is available for Ubuntu Lucid and Ubuntu Maverick from
 these PPAs:
 Lucid: https://launchpad.net/~diwic/+archive/ppa
 Maverick: https://launchpad.net/~diwic/+archive/maverick
 
 For other distributions, download the tarball:
 https://launchpad.net/~diwic/+archive/ppa/+files/alsamixertest_47.14.tar.gz
 Unpack and read the readme file for compilation and install instructions.
 
 When it is installed, run alsamixertest -r for a small tutorial and
 alsamixertest -h for command line options help.
 
 Looking forward to your comments about this new little tool! I think it
 should be considered beta quality at this point.

Cool. Packaged up for Mandriva too now. What license do you use.
Couldn't seem to find any reference to it...

Not tried it yet but will try and take it for a spin soon.

Cheers

Col


-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mageia Contributor [http://www.mageia.org/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

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Re: [pulseaudio-discuss] Alsamixertest

2010-09-27 Thread Colin Guthrie
'Twas brillig, and Colin Guthrie at 27/09/10 10:34 did gyre and gimble:
 'Twas brillig, and David Henningsson at 27/09/10 09:53 did gyre and gimble:
 So over the previous weeks I've been working on a small script which
 tests whether the ALSA mixer lives up to PA's expectations. If you are
 familiar with dbmeasure or dbverify by Lennart Poettering, this
 application's purpose is very similar, but this one is hopefully easier
 to set up, more user friendly, and also tests that the names of the
 volume controls are correct.
 My hope is that this will aid as a debugging tool for all these
 everything below 20% of my speaker is muted, and then 21% blows my
 speakers bugs.

 To use the tool, you'll need some kind of loopback. You can e g use a
 loopback cable and connect that between line in and line out, or test
 your laptop's internal speakers with your laptop's internal mic (just
 stop humming when you do so :-) ). Just set up the recording levels
 appropriately.

 Alsamixertest is available for Ubuntu Lucid and Ubuntu Maverick from
 these PPAs:
 Lucid: https://launchpad.net/~diwic/+archive/ppa
 Maverick: https://launchpad.net/~diwic/+archive/maverick

 For other distributions, download the tarball:
 https://launchpad.net/~diwic/+archive/ppa/+files/alsamixertest_47.14.tar.gz
 Unpack and read the readme file for compilation and install instructions.

 When it is installed, run alsamixertest -r for a small tutorial and
 alsamixertest -h for command line options help.

 Looking forward to your comments about this new little tool! I think it
 should be considered beta quality at this point.
 
 Cool. Packaged up for Mandriva too now. What license do you use.
 Couldn't seem to find any reference to it...
 
 Not tried it yet but will try and take it for a spin soon.

OK, for reference here is my output:

I don't actually have a specific headphones profile (no separate mixer
for headphones) but the results are more or less the same with and
without the -p argument (the only thing different appears to be the
range of the PCM tests). Ultimately the results seem to indicate that
Master is OK, but out by a bit, but PCM is all kinds of broken.

FWIW, the testing for -18.0, expected -21.04 measure -20.03dB is all a
little confusing. When testing for -18.0, why is 021.04 expected?
Perhaps some more explanation would be nice as to why the expected
values differ from the test value.

Anyway the output:

[co...@jimmy pulse]$ alsamixertest -p analog-output-headphones
INFO:root:Running initial test signal.
INFO:root:Testing that Master actually mutes the signal
INFO:root:Running test for mixer Master, level -3.00dB
INFO:root:Running test for mixer Master, level -4.50dB
INFO:root:Running test for mixer Master, level -6.00dB
INFO:root:Running test for mixer Master, level -9.00dB
INFO:root:Running test for mixer Master, level -10.50dB
INFO:root:Running test for mixer Master, level -12.00dB
INFO:root:Running test for mixer Master, level -15.00dB
INFO:root:Running test for mixer Master, level -16.50dB
INFO:root:Running test for mixer Master, level -18.00dB
ERROR:root:Error 6: Mixer Master has invalid dB data for dB=-18.0,
expected -21.04dB but measured -20.03dB.
INFO:root:Running test for mixer Master, level -21.00dB
ERROR:root:Error 6: Mixer Master has invalid dB data for dB=-21.0,
expected -24.04dB but measured -22.72dB.
INFO:root:Running test for mixer Master, level -22.50dB
ERROR:root:Error 6: Mixer Master has invalid dB data for dB=-22.5,
expected -25.54dB but measured -24.09dB.
INFO:root:Running test for mixer Master, level -24.00dB
ERROR:root:Error 6: Mixer Master has invalid dB data for dB=-24.0,
expected -27.04dB but measured -25.40dB.
INFO:root:Running test for mixer Master, level -27.00dB
ERROR:root:Error 6: Mixer Master has invalid dB data for dB=-27.0,
expected -30.04dB but measured -28.16dB.
INFO:root:Running test for mixer Master, level -28.50dB
ERROR:root:Error 6: Mixer Master has invalid dB data for dB=-28.5,
expected -31.54dB but measured -29.30dB.
INFO:root:Running test for mixer Master, level -30.00dB
ERROR:root:Error 6: Mixer Master has invalid dB data for dB=-30.0,
expected -33.04dB but measured -30.70dB.
INFO:root:Running test for mixer Master, level -33.00dB
ERROR:root:Error 6: Mixer Master has invalid dB data for dB=-33.0,
expected -36.04dB but measured -33.43dB.
INFO:root:Running test for mixer Master, level -34.50dB
ERROR:root:Error 6: Mixer Master has invalid dB data for dB=-34.5,
expected -37.54dB but measured -34.67dB.
INFO:root:Running test for mixer PCM, level -2.00dB
ERROR:root:Error 6: Mixer PCM has invalid dB data for dB=-2.0, expected
-5.04dB but measured -3.05dB.
INFO:root:Running test for mixer PCM, level -4.00dB
ERROR:root:Error 6: Mixer PCM has invalid dB data for dB=-4.0, expected
-7.04dB but measured -3.12dB.
INFO:root:Running test for mixer PCM, level -6.00dB
ERROR:root:Error 6: Mixer PCM has invalid dB data for dB=-6.0, expected
-9.04dB but measured -3.11dB.
INFO:root:Running test for 

[pulseaudio-discuss] Some info on the RME Hammerfall DSP

2010-09-27 Thread David Kastrup

I have had very limited success finding any information about what to do
about the RME Hammerfall DSP Multiface II card in connection with
Pulseaudio (running on Ubuntu Lucid 10.04).

Basically, the best advice I had been able to find has been putting the
lines

load-module module-alsa-sink sink_name=dsp_out device=hw:DSP format=s32le 
channels=18 
channel_map=left,right,aux0,aux1,aux2,aux3,aux4,aux5,aux6,aux7,aux8,aux9,aux10,aux11,aux12,aux13,aux14,aux15
 tsched=0
load-module module-alsa-source source_name=dsp_in device=hw:DSP format=s32le 
channels=18 
channel_map=left,right,aux0,aux1,aux2,aux3,aux4,aux5,aux6,aux7,aux8,aux9,aux10,aux11,aux12,aux13,aux14,aux15
 tsched=0

into the default configuration.  That works reasonably well, but one
gets glitches most particularly when playing Flash audio.

I just now figured out that removing the tsched=0 lines works fine as
long as you run

amixer -D hw:DSP cset numid=23 on

before starting pulseaudio.

amixer -D hw:DSP cget numid=23

now displays

numid=23,iface=CARD,name='Precise Pointer'
  ; type=BOOLEAN,access=rw--,values=1
  : values=on

where the default is 'off'.  Which does not appear to disturb Jack.  But
pulseaudio most certainly does not seem to like it, with tsched=1
suffering much worse than tsched=0.

Setting this to 'On' makes pulseaudio work worse with tsched=0, and
apparently have no problems with tsched=1.

All this is rather crazy, and I have no idea where to report it such
that a future version of Ubuntu will run out of the box.

Is there any way I can get Pulseaudio to set this particular control
when opening the hw:DSP device?

Thanks

-- 
David Kastrup

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Re: [pulseaudio-discuss] Some info on the RME Hammerfall DSP

2010-09-27 Thread David Kastrup
David Kastrup d...@gnu.org writes:

[...]

 load-module module-alsa-sink sink_name=dsp_out device=hw:DSP format=s32le 
 channels=18 
 channel_map=left,right,aux0,aux1,aux2,aux3,aux4,aux5,aux6,aux7,aux8,aux9,aux10,aux11,aux12,aux13,aux14,aux15
  tsched=0
 load-module module-alsa-source source_name=dsp_in device=hw:DSP format=s32le 
 channels=18 
 channel_map=left,right,aux0,aux1,aux2,aux3,aux4,aux5,aux6,aux7,aux8,aux9,aux10,aux11,aux12,aux13,aux14,aux15
  tsched=0

 into the default configuration.  That works reasonably well, but one
 gets glitches most particularly when playing Flash audio.

 I just now figured out that removing the tsched=0 lines works fine as
 long as you run

 amixer -D hw:DSP cset numid=23 on

 before starting pulseaudio.

 amixer -D hw:DSP cget numid=23

 now displays

 numid=23,iface=CARD,name='Precise Pointer'
   ; type=BOOLEAN,access=rw--,values=1
   : values=on

 where the default is 'off'.  Which does not appear to disturb Jack.  But
 pulseaudio most certainly does not seem to like it, with tsched=1
 suffering much worse than tsched=0.

 Setting this to 'On' makes pulseaudio work worse with tsched=0, and
 apparently have no problems with tsched=1.

Might likely be added as a PostScriptum to Trac ticket #764.

 Is there any way I can get Pulseaudio to set this particular control
 when opening the hw:DSP device?

That I still want to know.

-- 
David Kastrup

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Re: [pulseaudio-discuss] Some info on the RME Hammerfall DSP

2010-09-27 Thread Daniel Chen
[I don't know whether this was moderated or whether you're a
subscriber, so I added you.]

On Mon, Sep 27, 2010 at 7:52 AM, David Kastrup d...@gnu.org wrote:
    amixer -D hw:DSP cget numid=23

 now displays

 numid=23,iface=CARD,name='Precise Pointer'
  ; type=BOOLEAN,access=rw--,values=1
  : values=on

 where the default is 'off'.  Which does not appear to disturb Jack.  But
 pulseaudio most certainly does not seem to like it, with tsched=1
 suffering much worse than tsched=0.

 Setting this to 'On' makes pulseaudio work worse with tsched=0, and
 apparently have no problems with tsched=1.

There are two places where this control can be set (i.e., the
default changed), via alsactl's init db or in the driver itself.
Because Maverick is in deep freeze, and because Lucid is in an even
deeper freeze (having been released months ago), the best thing to do
at this point is to file a bug using ubuntu-bug alsa-base so that
the appropriate mixer information can be uploaded to Launchpad, and a
patch can be generated for one of the above contexts.  Personally, the
former makes more sense and is less likely to cause regressions in the
short term.  The main thing to test then becomes confirming that jackd
works correctly with that setting.  After sufficient testing, the
driver could be updated.


 All this is rather crazy, and I have no idea where to report it such
 that a future version of Ubuntu will run out of the box.

There isn't an easy way to set everything up nicely for this
particular card yet.  The changes will touch alsa-lib and alsa-utils
(or possibly alsa-driver, the latter depending on which route above is
taken).


 Is there any way I can get Pulseaudio to set this particular control
 when opening the hw:DSP device?

Sure, one can hack the mixer profile paths to set it, but that isn't
the recommended way.  It should be done via alsactl init.

Best,
-Dan
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Re: [pulseaudio-discuss] Some info on the RME Hammerfall DSP

2010-09-27 Thread David Kastrup
Daniel Chen seven.st...@gmail.com writes:

 [I don't know whether this was moderated or whether you're a
 subscriber, so I added you.]

Reading through gmane.

 On Mon, Sep 27, 2010 at 7:52 AM, David Kastrup d...@gnu.org wrote:
    amixer -D hw:DSP cget numid=23

 now displays

 numid=23,iface=CARD,name='Precise Pointer'
  ; type=BOOLEAN,access=rw--,values=1
  : values=on

 where the default is 'off'.  Which does not appear to disturb Jack.  But
 pulseaudio most certainly does not seem to like it, with tsched=1
 suffering much worse than tsched=0.

 Setting this to 'On' makes pulseaudio work worse with tsched=0, and
 apparently have no problems with tsched=1.

 There are two places where this control can be set (i.e., the
 default changed), via alsactl's init db or in the driver itself.
 Because Maverick is in deep freeze, and because Lucid is in an even
 deeper freeze (having been released months ago), the best thing to do
 at this point is to file a bug using ubuntu-bug alsa-base so that
 the appropriate mixer information can be uploaded to Launchpad, and a
 patch can be generated for one of the above contexts.  Personally, the
 former makes more sense and is less likely to cause regressions in the
 short term.  The main thing to test then becomes confirming that jackd
 works correctly with that setting.  After sufficient testing, the
 driver could be updated.

Well, I have no idea how to do this right.  One obvious snag is that a
setting like Precise Pointer is likely there for a reason.  Why would
anyone ask for imprecise pointers (the default) without there being an
actual advantage?  I have not yet found a rationale why this setting is
there in the first place.  I can't just believe there is a shoot
yourself in the foot variable in a driver with a default of yes.

I also have no idea about Maverick at all, and how it would cater for
the Hammerfall with regard to Pulseaudio, defaults and so on.

Would it be considered useful if I made the jump to Maverick prerelease
right now?  Who will then be doing the testing for a putative backport?

-- 
David Kastrup

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Re: [pulseaudio-discuss] Some info on the RME Hammerfall DSP

2010-09-27 Thread Daniel Chen
On Mon, Sep 27, 2010 at 8:32 AM, David Kastrup d...@gnu.org wrote:
 I also have no idea about Maverick at all, and how it would cater for
 the Hammerfall with regard to Pulseaudio, defaults and so on.

The modifications that you made would still be necessary in Maverick.


 Would it be considered useful if I made the jump to Maverick prerelease
 right now?  Who will then be doing the testing for a putative backport?

My opinion is that it isn't absolutely necessary to move to Maverick
(I dogfood the LTS releases).  All the modifications that I suggested
are possible in Lucid, too.  It's probable that [Launchpad's]
~ubuntu-audio-dev will have a staging PulseAudio package (based on
Maverick's) for Lucid soon.

Best,
-Dan
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Re: [pulseaudio-discuss] Some info on the RME Hammerfall DSP

2010-09-27 Thread David Kastrup
Daniel Chen seven.st...@gmail.com writes:

 On Mon, Sep 27, 2010 at 8:32 AM, David Kastrup d...@gnu.org wrote:
 I also have no idea about Maverick at all, and how it would cater for
 the Hammerfall with regard to Pulseaudio, defaults and so on.

 The modifications that you made would still be necessary in Maverick.


 Would it be considered useful if I made the jump to Maverick prerelease
 right now?  Who will then be doing the testing for a putative backport?

 My opinion is that it isn't absolutely necessary to move to Maverick
 (I dogfood the LTS releases).  All the modifications that I suggested
 are possible in Lucid, too.  It's probable that [Launchpad's]
 ~ubuntu-audio-dev will have a staging PulseAudio package (based on
 Maverick's) for Lucid soon.

The recipe can slightly be improved using

amixer -D hw:DSP cset 'iface=CARD,name=Precise Pointer' On

since that would be more symbolical and would likely do nothing to cards
not in possession of such a control.  I did some more experiments, and
Flash Video continues to have skips.  However, the previous artifacts
sounded like wraparounds.  The current ones apparently correspond to
skipped frames.  I have no idea why Flash would skip frames when there
seems to be enough buffer of original content available, but maybe the
codecs don't keep up.

Right now I have a hard time convincing firefox/pulseaudio to try the
builtin sound card for comparison.

-- 
David Kastrup

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Re: [pulseaudio-discuss] Some info on the RME Hammerfall DSP

2010-09-27 Thread David Kastrup
Daniel Chen seven.st...@gmail.com writes:

 On Mon, Sep 27, 2010 at 8:32 AM, David Kastrup d...@gnu.org wrote:
 I also have no idea about Maverick at all, and how it would cater for
 the Hammerfall with regard to Pulseaudio, defaults and so on.

 The modifications that you made would still be necessary in Maverick.


 Would it be considered useful if I made the jump to Maverick prerelease
 right now?  Who will then be doing the testing for a putative backport?

 My opinion is that it isn't absolutely necessary to move to Maverick
 (I dogfood the LTS releases).  All the modifications that I suggested
 are possible in Lucid, too.  It's probable that [Launchpad's]
 ~ubuntu-audio-dev will have a staging PulseAudio package (based on
 Maverick's) for Lucid soon.

URL:https://bugs.launchpad.net/bugs/648992

-- 
David Kastrup

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[pulseaudio-discuss] udev detection for 1 card with multiple devices

2010-09-27 Thread abraham duenas
Hi all,

I have this sound card with several hw devices attached, this is, it's only
one card but I can configure/route all its numerous pcm/hw devices to have
different sink/sources. For saying something I have hw:0,0 for
playback/capture, on hw:0,1 only capture, hw:0,2 again for playback/capture,
on hw:0,3 hdmi and so on. I've configured this card in the past using
module-alsa-sink/source for each hw device and it works pretty fine.
Recently I tried to use the udev module instead of module-alsa-sink/source
but only the first hw device is recognized (hw:0,0) correctly for
playback/capture... I'm wondering if this is the intended way (only first
available device will be used) or it should also be capable of detecting the
other devices or I need to do something else (like writing udev-rule?) but
i'm no expert on udev stuff...
FWIW I can see that all hw devices are correctly listed in /dev/snd
directory... (I have /dev/snd/controlC0 and all of the /dev/snd/pcmC0D*
entries)

Any ideas or help will be appreciated!

Thanks,
Abraham
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Re: [pulseaudio-discuss] udev detection for 1 card with multiple devices

2010-09-27 Thread Colin Guthrie
'Twas brillig, and abraham duenas at 27/09/10 18:12 did gyre and gimble:
 Hi all,
 
 I have this sound card with several hw devices attached, this is, it's
 only one card but I can configure/route all its numerous pcm/hw devices
 to have different sink/sources. For saying something I have hw:0,0 for
 playback/capture, on hw:0,1 only capture, hw:0,2 again for
 playback/capture, on hw:0,3 hdmi and so on. I've configured this card in
 the past using module-alsa-sink/source for each hw device and it works
 pretty fine. 
 Recently I tried to use the udev module instead of
 module-alsa-sink/source but only the first hw device is recognized
 (hw:0,0) correctly for playback/capture... I'm wondering if this is the
 intended way (only first available device will be used) or it should
 also be capable of detecting the other devices or I need to do something
 else (like writing udev-rule?) but i'm no expert on udev stuff...
 FWIW I can see that all hw devices are correctly listed in /dev/snd
 directory... (I have /dev/snd/controlC0 and all of the /dev/snd/pcmC0D*
 entries) 
 
 Any ideas or help will be appreciated!

Yes this is possible and as you suspected it will involve writing a udev
rule, but also writing a specific mixer profile too.

If you look in your /usr/share/pulseaudio/alsa-mixer/ folder you'll see
two folders. One is called paths and contains fairly generic config
files to probe the various mixer elements in ALSA. What is likely more
interesting is the profile-sets folder. It can contain custom
configurations for more featureful and specialist cards such as yours.

The task you have to complete is two fold: First of all you need to
write your profile set itself. Take a look in the folder for examples.
The files are quite liberally commented and I think this represents the
only documentation (other than the source that parses them!).

Once you have written your profile set for your hardware you have to
tell PA to actually use it. This is where the udev rule comes in.

Look at the file /lib/udev/rules.d/90-pulseaudio.rules. It contains a
few lines to register the relevant profile set against your specific h/w
via PCI or USB ids.

Simply copy one of the three of four rules in there already and modify
to suit.

Once this is all working to your satisfaction, it would be great if you
could submit your profile set and udev rule to us so we can include it
in the standard install!

If you have any questions just ask here and myself and the other folks
here will do our best to help (tho' I personally do not have all that
much experience with the mixer profiles specifically, I can generally
poke about and find out more :))

Cheers

Col


-- 

Colin Guthrie
gmane(at)colin.guthr.ie
http://colin.guthr.ie/

Day Job:
  Tribalogic Limited [http://www.tribalogic.net/]
Open Source:
  Mageia Contributor [http://www.mageia.org/]
  PulseAudio Hacker [http://www.pulseaudio.org/]
  Trac Hacker [http://trac.edgewall.org/]

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