Re: [pulseaudio-discuss] Progress of 'dynamic default sink' or 'device preference list'?

2010-12-01 Thread Colin Guthrie
'Twas brillig, and Ng Oon-Ee at 01/12/10 07:48 did gyre and gimble:
 Hi all,
 
 Now that 0.9.22 is finally out, I'm wondering what happened to what
 Colin mentioned some time back about 'device preferences'. As I
 understood at the time, assuming 3 possible sound devices (which may or
 may not be plugged in):-
 1. New streams would be automatically sent to the highest priority
 device on starting
 2. Streams which have been manually moved would stay where they've been
 moved to.
 3. When another device which has a higher priority is plugged in, all
 streams (except for those in 2.) are moved to it.
 
 This scheme was mentioned in preference to a generic move all sound to
 latest plugged-in-device scheme, I believe.
 
 Are there any plans in this direction?

Not done much on it TBH. Was wanting to discuss things with Lennart
further as he never quite acked the approach (not the finer details
anyway) and there are still some caveats to the approach (like dealing
with apps that produce multiple streams where you want to control them
independently - although I think that can and should be mitigated in an
app specific way).

I may just crack on an do it over the winter seeing as it's getting
really cold now and worry about the finer details later.

Col


-- 

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Re: [pulseaudio-discuss] Card sometimes doesn't create sinks/sources

2010-12-01 Thread Colin Guthrie
'Twas brillig, and Colin Guthrie at 07/11/10 20:51 did gyre and gimble:
 [Resend: Didn't get through moderation last time... 2nd time lucky]
 
 Hi,
 
 This is a strange one that I've seen happen very occasionally, but
 cannot nail down or reliably reproduce.
 
 I've got a set of USB speakers and when coming back from suspend with
 them plugged in (they were not plugged in prior to suspending but have
 been previously with the same instance of PA).
 
 The card is recognised and appears under Configuration tab or pacmd
 list-cards. It shows it's profiles correctly and selects the one I want,
 but it does not load the sinks and sources (in this case, just sinks).
 
 Unplugging and replugging doesn't help. Changing profile doesn't help.
 
 Everything looks correct, but the sinks just don't appear.
 
 Restart PA daemon and all is well - the sinks appear as expected.
 
 I've attached the log output, but the only strange thing in there to me
 is the events suppressed message. Sadly it's not clear exactly which
 event is suppressed when it prints that message which really doesn't
 help debugging... someone should really fix that with the log
 suppression technique currently in use!


Just as an update to this issue, it appears to related to DBus
connections being lost on a suspend/resume cycle.

org.pulseaudio.Server is not on the session bus. The reason the sinks
are not created is due to the reservation protocol to allow graceful
handover to other applications. Because this fails to init, we do not
create the sink.

Not sure if this is a DBus of PA problem yet tho'.

Thanks to David Henningsson for helping debug this when it cropped up
again this morning.

Col

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Re: [pulseaudio-discuss] [PATCH] alsa-sink: take base volume into account when applying hw volume

2010-12-01 Thread Jyri Sarha


On Tue, 30 Nov 2010, Colin Guthrie wrote:


Any comments on this one Jyri?

It seems add code to sink_write_volume_cb() to mirror the normal method
sink_set_volume_cb() so looks OK to me but not followed the path through
to see if this is supposed to be factored in already in some other way...

Your thoughts would be appreciated.




Yes,
The base volume stuff appeared some time between the first version of sync
volume and my upstream rebase. AFAIK, the fix is correct. However, there
is still the double use of string bug (originally coming from me, but it
looks quite different now). The bug is not serious, the print just is not 
very informative.


Am currently working on a problem caused by safety-margin not
taken into account when rewinding. When I get this fixed I'll add a fix 
for double use bug too.


Cheers,
Jyri



Col


'Twas brillig, and Juho Hämäläinen at 25/11/10 13:15 did gyre and gimble:

Currently if sink base volume differs from 0dB and sync-volume is used,
wrong volume values are written to hw. This patch fixes that.


Signed-off-by: Juho Hämäläinen ext-juho.hamalai...@nokia.com
---
  src/modules/alsa/alsa-sink.c |   10 --
  1 files changed, 8 insertions(+), 2 deletions(-)



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Re: [pulseaudio-discuss] [PATCH] alsa-sink: take base volume into account when applying hw volume

2010-12-01 Thread Colin Guthrie
'Twas brillig, and Jyri Sarha at 01/12/10 10:31 did gyre and gimble:
 
 On Tue, 30 Nov 2010, Colin Guthrie wrote:
 
 Any comments on this one Jyri?

 It seems add code to sink_write_volume_cb() to mirror the normal method
 sink_set_volume_cb() so looks OK to me but not followed the path through
 to see if this is supposed to be factored in already in some other way...

 Your thoughts would be appreciated.

 
 
 Yes,
 The base volume stuff appeared some time between the first version of sync
 volume and my upstream rebase. AFAIK, the fix is correct. However, there
 is still the double use of string bug (originally coming from me, but it
 looks quite different now). The bug is not serious, the print just is
 not very informative.
 
 Am currently working on a problem caused by safety-margin not
 taken into account when rewinding. When I get this fixed I'll add a fix
 for double use bug too.

Great, I'll push this fix for now then and look forward to your next
batch of fixes too :)

Cheers

Col



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Re: [pulseaudio-discuss] [ANNOUNCE] New Version Naming Scheme

2010-12-01 Thread Colin Guthrie
'Twas brillig, and Colin Guthrie at 28/11/10 15:31 did gyre and gimble:
 As you know I've been angling for some kind of solid way forward with
 version numbers for a while so this email is basically a description of
 what has been done about this and how we'll move forward.
 ...

OK, with no major problems with the two-part version scheme voiced, I've
pushed this to master now.

Nothing really major has changed so it shouldn't affect things
downstream in any way (that's the theory at least!).

Let me know if there are any strange build issues on master.

Col



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Re: [pulseaudio-discuss] Progress of 'dynamic default sink' or 'device preference list'?

2010-12-01 Thread Ng Oon-Ee
On Wed, 2010-12-01 at 09:20 +, Colin Guthrie wrote:
 'Twas brillig, and Ng Oon-Ee at 01/12/10 07:48 did gyre and gimble:
  Hi all,
  
  Now that 0.9.22 is finally out, I'm wondering what happened to what
  Colin mentioned some time back about 'device preferences'. As I
  understood at the time, assuming 3 possible sound devices (which may or
  may not be plugged in):-
  1. New streams would be automatically sent to the highest priority
  device on starting
  2. Streams which have been manually moved would stay where they've been
  moved to.
  3. When another device which has a higher priority is plugged in, all
  streams (except for those in 2.) are moved to it.
  
  This scheme was mentioned in preference to a generic move all sound to
  latest plugged-in-device scheme, I believe.
  
  Are there any plans in this direction?
 
 Not done much on it TBH. Was wanting to discuss things with Lennart
 further as he never quite acked the approach (not the finer details
 anyway) and there are still some caveats to the approach (like dealing
 with apps that produce multiple streams where you want to control them
 independently - although I think that can and should be mitigated in an
 app specific way).
 
 I may just crack on an do it over the winter seeing as it's getting
 really cold now and worry about the finer details later.

Thanks for the quick response Colin. Would user-input as to the design
of this behaviour be helpful, or just a hindrance?

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[pulseaudio-discuss] module-combine causes wine apps sound to crash [WAS: Latency control for wine ALSA emulation?]

2010-12-01 Thread Ng Oon-Ee
So after more testing, I figured out my automatic load-module
module-combine was causing the problem in this mail below.

The same does not occur when running on the sound card itself directly.
Is there any equivalent to tsched=0 for module-combine?

On Tue, 2010-11-30 at 09:40 +0800, Ng Oon-Ee wrote:
 Dammit, forgot to include links!
 
 [1] -
 http://www.mail-archive.com/pulseaudio-discuss@mail.0pointer.de/msg06596.html
 [2] - http://pulseaudio.org/wiki/LatencyControl
 
 On Tue, 2010-11-30 at 09:39 +0800, Ng Oon-Ee wrote:
  Just recently I noticed my wine games weren't outputting sound. After
  some trouble-shooting (and remembering old issues I'd seen on this
  list), I realized that starting them with pavucontrol on fixed the
  sound, it would work just fine.
  
  I refer to [1] and [2] where Colin says this would be due to the
  reduction in latency. One workaround to this is tsched=0, but I haven't
  needed that before for ages.
  
  I'm on pulseaudio 0.9.22, just upgraded, also just upgraded wine to
  1.3.8. However I can't seem to find a combination of downgraded packages
  (0.9.21 and 1.3.6 respectively, which used to work) which can make
  things work without pavucontrol again. Could this be affected by for
  example lib32-alsa-lib or alsa-lib or alsa-utils (lib32-alsa-utils as
  well)?
  
  I'd rather not hardcode tsched=0 because I'm on a laptop =). Please let
  me know what other things I can do to debug, would a log be helpful
  (since the issue is pretty certainly the latency control), or should I
  get a wine log instead. That would probably be a waste of time since
  winealsa isn't maintained currently I believe.
 
 


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Re: [pulseaudio-discuss] [PATCH] alsa-sink: take base volume into account when applying hw volume

2010-12-01 Thread David Henningsson

On 2010-12-01 11:31, Jyri Sarha wrote:

Am currently working on a problem caused by safety-margin not
taken into account when rewinding. When I get this fixed I'll add a fix
for double use bug too.


Hmm, for tsched or non-tsched? I think we merged a patch related to this 
from pl bossart a while ago.


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Re: [pulseaudio-discuss] module-combine causes wine apps sound to crash [WAS: Latency control for wine ALSA emulation?]

2010-12-01 Thread Colin Guthrie
'Twas brillig, and Ng Oon-Ee at 01/12/10 10:59 did gyre and gimble:
 So after more testing, I figured out my automatic load-module
 module-combine was causing the problem in this mail below.
 
 The same does not occur when running on the sound card itself directly.
 Is there any equivalent to tsched=0 for module-combine?

Not that I'm aware of... maybe requires someone to give module-combine
some love? I've had issues with it crashing things long ago and not
really seen much in the way of fixes for it in the past little while, so
they could easily still be lurking there.

Col

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Re: [pulseaudio-discuss] Progress of 'dynamic default sink' or 'device preference list'?

2010-12-01 Thread Colin Guthrie
'Twas brillig, and Ng Oon-Ee at 01/12/10 10:57 did gyre and gimble:
 On Wed, 2010-12-01 at 09:20 +, Colin Guthrie wrote:
 'Twas brillig, and Ng Oon-Ee at 01/12/10 07:48 did gyre and gimble:
 Hi all,

 Now that 0.9.22 is finally out, I'm wondering what happened to what
 Colin mentioned some time back about 'device preferences'. As I
 understood at the time, assuming 3 possible sound devices (which may or
 may not be plugged in):-
 1. New streams would be automatically sent to the highest priority
 device on starting
 2. Streams which have been manually moved would stay where they've been
 moved to.
 3. When another device which has a higher priority is plugged in, all
 streams (except for those in 2.) are moved to it.

 This scheme was mentioned in preference to a generic move all sound to
 latest plugged-in-device scheme, I believe.

 Are there any plans in this direction?

 Not done much on it TBH. Was wanting to discuss things with Lennart
 further as he never quite acked the approach (not the finer details
 anyway) and there are still some caveats to the approach (like dealing
 with apps that produce multiple streams where you want to control them
 independently - although I think that can and should be mitigated in an
 app specific way).

 I may just crack on an do it over the winter seeing as it's getting
 really cold now and worry about the finer details later.
 
 Thanks for the quick response Colin. Would user-input as to the design
 of this behaviour be helpful, or just a hindrance?

Well I've gathered quite a lot of feedback already, but please feel free
to critique more! I think the comments on the blog article[1] and a
thread on this list which I've long since lost of my gmane cache were
the best references... i'll try and dig out the archive link to that
thread when I have a sec.

Col


1 http://colin.guthr.ie/2010/02/this-is-the-route-to-hell/


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[pulseaudio-discuss] Mic input volume controls

2010-12-01 Thread David Henningsson

Hi folks,

The way we control mic input is quite broken. I've tested here with both 
IDT codecs and Realtek codecs (the two most common HDA Codecs AFAIK) and 
as far as I see they're both broken in the same ways.


Let's assume that we have an internal mic and an external mic jack, with 
autoswitching in the kernel. We have a Mic Boost affecting the 
external mic jack, a Front Mic Boost affecting internal mic, and a 
Capture control affecting both.
Mic Boost and Front Mic Boost go from 0 to 60 dB in 20 dB steps, and 
Capture goes from 0 to 22.5 dB in 1.5 dB steps.


1)
The first problem is PA is unable to combine both of them correctly: for 
the first 0 to 22.5 dB, PA moves the Capture control only. After that 
say at 25 dB, Mic Boost moves to +20 dB, keeping Capture at 22.5 dB, 
then using softvol to compensate, instead of lowering Capture. I've 
been suggested that changing the order of Mic Boost and Capture 
improves this situation, so we should probably do that. Any objections?


2)
The harder problem is that PulseAudio does not know whether to control 
Mic Boost or Front Mic Boost, since it doesn't know what path is 
active at the moment. While we really need some kind of module, and 
appropriate kernel support, the question is what we do before that 
becomes sufficiently available. The next best solution, would probably 
be to have different profiles and manual autoswitching. While I yet have 
to look into the details, I guess we'll have to add different profiles 
for Mic, Front Mic, Rear Mic and so on, instead of having them in 
the same file as we have today. Any thoughts?


3)
It'd be great if we could present something else than Microphone 1, 
Microphone 2 and so on when we have more than one mic input. Any idea 
of where this name actually comes from, and how we can make it better?


Thanks/credits to Vitaliy Kulikov for helping with the analysis.


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Re: [pulseaudio-discuss] Mic input volume controls

2010-12-01 Thread Colin Guthrie
'Twas brillig, and David Henningsson at 01/12/10 13:55 did gyre and gimble:
 Hi folks,
 
 The way we control mic input is quite broken. I've tested here with both
 IDT codecs and Realtek codecs (the two most common HDA Codecs AFAIK) and
 as far as I see they're both broken in the same ways.
 
 Let's assume that we have an internal mic and an external mic jack, with
 autoswitching in the kernel. We have a Mic Boost affecting the
 external mic jack, a Front Mic Boost affecting internal mic, and a
 Capture control affecting both.
 Mic Boost and Front Mic Boost go from 0 to 60 dB in 20 dB steps, and
 Capture goes from 0 to 22.5 dB in 1.5 dB steps.
 
 1)
 The first problem is PA is unable to combine both of them correctly: for
 the first 0 to 22.5 dB, PA moves the Capture control only. After that
 say at 25 dB, Mic Boost moves to +20 dB, keeping Capture at 22.5 dB,
 then using softvol to compensate, instead of lowering Capture. I've
 been suggested that changing the order of Mic Boost and Capture
 improves this situation, so we should probably do that. Any objections?

With the direction argument passed to the relevant alsa calls as we
currently have, swapping them around should indeed help here yes. No
objection in principle for me. I guess we'll just statically swap them
around for now? In theory, I guess the order of the controlled element
pipeline could be dynamically adjusted so that biggest steps happen
first... but this approach would possibly introduce unwanted side effets.

 2)
 The harder problem is that PulseAudio does not know whether to control
 Mic Boost or Front Mic Boost, since it doesn't know what path is
 active at the moment. While we really need some kind of module, and
 appropriate kernel support, the question is what we do before that
 becomes sufficiently available. The next best solution, would probably
 be to have different profiles and manual autoswitching. While I yet have
 to look into the details, I guess we'll have to add different profiles
 for Mic, Front Mic, Rear Mic and so on, instead of having them in
 the same file as we have today. Any thoughts?

I guess splitting them out makes sense so as to keep the pipeline for
each clean. Would you still envisage keeping each mic as a separate
port? (I guess the only other option would be to check if both mics
could be opened at the same time in the profile probing to allow for two
alsa-sources to be loaded at the same time. But that's likely going to
lead to a lot of other problems, so perhaps best to leave that until UCM
can just tell us without probing).

 
 3)
 It'd be great if we could present something else than Microphone 1,
 Microphone 2 and so on when we have more than one mic input. Any idea
 of where this name actually comes from, and how we can make it better?

I think ultimately from the table in ./modules/alsa/alsa-mixer.c

There is a difference between internal and external microphones, so I
guess making the distinction between front and rear and such like
wouldn't hurt anyway... that said, WTF is the diffrence between a front
and rear mic physically on a laptop? Do they *really* make laptops with
two mics on them for this purpose? (from the names I'd have to expect a
yes answer there, but seeing as my laptop has no built in mic, I'd say
having two is just showing off :D) But more seriously, why are there
front and rear mics? Is this for stereo recording? If so should these
inputs be handled as a single, multichannel source rather than separate
mono or stereo ones? I mean we don't expose the Rear playback so why
do we do it on mics? As you can tell, my foo for recording is rather weak :D

Col

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Re: [pulseaudio-discuss] [PATCH] alsa-sink: take base volume into account when applying hw volume

2010-12-01 Thread Jyri Sarha



On Wed, 1 Dec 2010, David Henningsson wrote:


On 2010-12-01 11:31, Jyri Sarha wrote:

Am currently working on a problem caused by safety-margin not
taken into account when rewinding. When I get this fixed I'll add a fix
for double use bug too.


Hmm, for tsched or non-tsched? I think we merged a patch related to this from 
pl bossart a while ago.




Sorry, should have been more specific. I was talking about
sync-volume-safety-margin, not rewind_safeguard related to tsched mode
rewinding. However, the two things are closely related in this problem
case. When DMA buffer is rewound the sync-volume events should rewound at
the same time. This works already now.  However the volume event rewinding
is not modifying the new expiry time with sync-volume-safety-margin
depending on whether we are changing the volume up or down.

Cheers,
Jyri
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Re: [pulseaudio-discuss] Mic input volume controls

2010-12-01 Thread Mark Brown
On Wed, Dec 01, 2010 at 03:09:10PM +, Colin Guthrie wrote:

 wouldn't hurt anyway... that said, WTF is the diffrence between a front
 and rear mic physically on a laptop? Do they *really* make laptops with
 two mics on them for this purpose? (from the names I'd have to expect a
 yes answer there, but seeing as my laptop has no built in mic, I'd say
 having two is just showing off :D) But more seriously, why are there
 front and rear mics? Is this for stereo recording? If so should these
 inputs be handled as a single, multichannel source rather than separate
 mono or stereo ones? I mean we don't expose the Rear playback so why
 do we do it on mics? As you can tell, my foo for recording is rather weak :D

You might have more than two microphones built in to use them for beam
forming applications in which case you're looking at a many channel
array and do want everything in sync.  On the other hand you may be
doing something a lot less sophisticated and just have several sets of
microphones for different physical use cases (eg, one set for use in
tablet mode with the screen fully folded over and another for use in
laptop mode) in which case you'd probably have only one in play at
once.  Or even some combination of the above :)
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Re: [pulseaudio-discuss] Progress of 'dynamic default sink' or 'device preference list'?

2010-12-01 Thread Colin Guthrie
'Twas brillig, and Ng Oon-Ee at 01/12/10 15:53 did gyre and gimble:
 1 http://colin.guthr.ie/2010/02/this-is-the-route-to-hell/

 Thanks Colin, I'll definitely try and contribute ideas, at the least.
 May I assume this ML is better for it than your blog? =)

General praise can go on my blog, criticism can come here :D

But more seriously, yeah, this thread is as good as any.

Col

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