seems
to be quite high already).
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management.
This part I didn't understand, sorry. What do you mean by external connection
management? Could you give examples of other JACK apps that do it?
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you have a clear opinion
about this and I'd like to consistently conform to it.
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On Wed, Aug 29, 2007 at 12:52:21PM +0200, Jan Kasprzak wrote:
I have read the FAQ, where it is described how to join several sound cards
into one output. What I would like to do is the opposite:
I have a 4-channel sound card, and I would like to use it as two
independent sinks, with
On Wed, Aug 29, 2007 at 03:08:43PM +0300, Tanu Kaskinen wrote:
You can load
module-alsa-sink with only two channels, and with
channel_map you could probably even control which channels
go to which physical output. Then you would only need to
load another instance of module-alsa-sink
On Wed, Oct 10, 2007 at 10:29:55PM -0700, Jim Carter wrote:
pulseaudio: pulsecore/mutex-posix.c:98: pa_mutex_unlock: Assertion
`pthread_mutex_unlock(m-mutex) == 0' failed.
[snip]
Does anyone have any idea what's going on here?
My crystal ball says that you have a Debian version of
libtool,
On Sun, Dec 16, 2007 at 06:15:59PM -0500, Ritesh Kumar wrote:
BTW, what does pacmd's set-default-sink/source do? What is the meaning of
the default sink? I figured out that alsamixer (through the pulse alsa
plugin) always changes the mixer controls for the current default sink as
per the
(or some ultra fancy sound card) - a DSP card isn't
required, equalizing doesn't eat CPU power awfully much.
Hope this helps.
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On Wed, Jan 09, 2008 at 09:15:33AM +0100, Adam Sulmicki wrote:
Folks,
I have followed the suggestions given by Tanu Kaskinen, sadly I'm having
problems. With new settings the program crashes with memory protection
violation. Despite repeated attempts, I am unable to produce a meaningful bt
On Thu, Jan 10, 2008 at 06:32:14PM +0200, Tanu Kaskinen wrote:
Then, record the output:
pacat -r --device=ladspa_out.monitor noise.raw
Whoops! It seems that the you have to use the alsa sink's
monitor source. The command above records the unfiltered
noise.
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than me, I
just tried audacity's spectrogram for the first time.
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(replace 0 with something else
if the sound card doesn't have index 0) and if some slider
shows positive gain, lower it. The gain is shown in the
Item: line in upper left corner.
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network
traffic could be prevented with firewalling in the Linux
machine.
What really is needed is a proper virtual pulse sound card
for Windows.
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the instructions
at
http://www.pulseaudio.org/wiki/PerfectSetup#ALSAApplications.
Or maybe you've done that already.
Rest of the file seemed to be fine.
If problems arise, output of pulseaudio -vv will be useful.
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be hw:0 and Delta will be hw:1. This isn't
necessary a valid assumption.
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On Tue, Jan 29, 2008 at 10:01:48AM -0600, Dewey Smolka wrote:
On Jan 29, 2008 5:02 AM, Tanu Kaskinen [EMAIL PROTECTED] wrote:
I tried to run jack on top of pulse, and it seems that it
doesn't work at all because the pulse device doesn't support
mmap access that jack needs. So we can
-user way just doesn't cut it. This isn't a
fatal mistake (if it's mistake at all, I don't know what
kind of setup you actually have/need), and you don't
necessarily need to fix it. Some information on this matter:
http://www.pulseaudio.org/wiki/SystemWideInstance
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should contact Martin-Éric Racine. Irc is
preferred, he's always on Freenode and IRCnet with nick
Q-FUNK.
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be a
way to properly bridge sources to sinks, but because it
wasn't Lennart who I heard this from, this is just a rumor
(there seems to be a lot of those nowadays...).
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as needed you have full control over
what goes where, without any drop-outs.
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on actual time, but
just the observed difference in the clocks.
So resampling is always done only once (afaik).
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On Mon, Feb 18, 2008 at 10:37:09PM +0200, Tanu Kaskinen wrote:
If I understood you correctly, then the answer is that there
is no final hardware matching stage. In your example the
sound card reports that it uses sampling rate of 48000 Hz,
and pulseaudio trusts that it's correct information
On Mon, Feb 18, 2008 at 01:19:43PM -0800, Erich Boleyn wrote:
Tanu Kaskinen [EMAIL PROTECTED] wrote:
On Mon, Feb 18, 2008 at 11:56:05AM -0800, Erich Boleyn wrote:
When Pulseaudio converts from one format to another at the sound output
(say for an ostensibly 16-bit/48-khz DAC
be annoying.
If all the receiving machines use simple alsa sinks with
default configuration, (and the same sample rate and sample
size), I would guess that their internal latencies would be
pretty much the same.
any experience with this?
No, I haven't used RTP myself.
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the resampling after all, or what
part have I understood wrong?
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machines. There would be one
module-combine instance per stream (I guess that in your
case this means one per sending machine).
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) and configure the programs to
use the spdif device directly.
If I used anything besides nothing or channels=2, I don't get any audio.
Here I'm a bit confused - what do you give that parameter
to? I thought you were talking about module-detect, but it
doesn't accept that parameter.
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/var/log/syslog has more error messages. If they
aren't informative enough, you can try increased verbosity
by starting with the parameter -vv.
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real-time
scheduling in system-wide daemon. Maybe configuring the
rtprio resource limit for user pulse in
/etc/security/limits.conf could work.
The rest of the mail was good advice.
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line in daemon.conf and set the value to yes.
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the easiest I could
come up with.
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tries to connect to. Otherwise, if you decide to
manage the routing information in a separate program, I
don't think you can use the client API in any way, so you'll
have to use some other ipc solution.
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it would hang?
Just a wild guess... Maybe resuming from suspended state
doesn't work for a reason or another? Try commenting out
module-suspend-on-idle in /etc/pulse/default.pa.
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set arbitrary properties to
streams, by using environment variables, or
programmatically. As mentioned in
https://tango.0pointer.de/pipermail/pulseaudio-discuss/2008-May/001777.html
you should probably use the property media.role.
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null_sink1.monitor | encoder1
parec -d null_sink2.monitor | encoder2
etc...
If you record while nothing is played, zeros are generated
just like with the pipe sink too, but at least the timing is
sensible.
See parec --help for available options and the default
values for the stream format.
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surround
streams, if you have a surround sound card. Unfortunately
this affects the streams from the remap sinks too (I think
this will be fixed real soon now in svn, if Lennart approves
the new behaviour). Fortunately the feature can be disabled
in daemon.conf (disable-remixing).
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by pointing
them at front:1 and rear:1. Are there equivalent values for side:1 and
center/lfe:1? I tried side:1 without avail so that can't be it.
I don't know about that.
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2009/3/31 Matt Price matt.pr...@utoronto.ca:
I need audio for both functions, but only have one audio card. Now,
this card has 8 audio channels, but what i really need is 2 stereo
outputs. Is it possible, with pulse, to choose which channels to which
to route streams? I don't see anything in
2009/5/5 Markus Feldmann feldmann_mar...@gmx.de:
Markus Feldmann schrieb:
doesn't say anything to me. I mean the values following after
...control= 11.621622,10,4.594594,2.702703,...
Which value is for which frequency ?
What are the minimums and maximums ?
The frequencies are documented
2009/5/5 Markus Feldmann feldmann_mar...@gmx.de:
But here comes some questions:
I am using pulseaudio 9.10 under debian.
I go through the guide,
http://www.pulseaudio.org/wiki/Modules#module-ladspa-sink
but it doesnt works for me.
1.) If i create ~/.pulse/default.pa then pulseaudio
doesn't
2009/5/5 Markus Feldmann feldmann_mar...@gmx.de:
I get a step further.
i corrected the name of the master to,
master=alsa_output.pci_1002_4383_sound_card_0_alsa_playback_0
is there a possibility to make the master name more dynamically ?
Maybe it will be changed in the future.
If I
want
to cooperate (well, it could use the kill command or something...).
PA cooperates by listening for device release requests using D-Bus, so
Jack has to use D-Bus too. And apparently jackdbus is the only
implementation that does that.
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?
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/local/src/pulseaudio/pulseaudio/configure: line 5866: `LT_PREREQ(2.2)'
My guess would be that you don't have new enough libtool. I think the
check tries to make sure you have at least libtool 2.2, but maybe the
check itself doesn't work with earlier versions?
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over D-Bus, like trasferring audio streams).
I'm being paid to implement the D-Bus module by the end of August.
More specific information isn't available - the work will begin
probably next monday.
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2009/6/28 Timothy J Massey tmas...@obscorp.com:
E: module-remap-sink.c: Failed to parse module arguments.
sink_properties was added to module-remap-sink after the release of
0.9.15, so it's available only in the git version.
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to fix typos and other obvious mistakes yourself.
[1] http://pulseaudio.org/wiki/DBusInterface
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, if I calculated this correctly. For short calls this shouldn't
be too much. Also, if the output device (from which you record using
the monitor source) and the mic device are on the same card, they may
share the clock, in which case no drift happens.
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solution for this.
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that your problems have very little to do with pulseaudio,
since (at least in the example) you record directly from the tuner,
instead of through pulseaudio. Only the delay issue may have something
to do with pulseaudio.
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is farting ;) Since the flow is always from source to
sink, you of course do parec | pacat twice, with different devices.
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ke, 2009-07-01 kello 01:30 +0300, Tanu Kaskinen kirjoitti:
ti, 2009-06-30 kello 22:23 +0200, Lennart Poettering kirjoitti:
snip
Also, we have X11 root window properties for finding the server, we
should have that for the D-Bus server too, probably.
Hmm, I am not really convinced
that Skype doesn't go through Pulseaudio. I can imagine that there
could be some advantage of a setup like that, but no idea how
significant the advantage would be. But if this is a networking issue,
hw mixing won't help a bit.
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, right click the stream and you get the move menu.
Also, make sure the correct device is set as the default. In the Output
devices tab, right click the internal sound card and select the Default
checkbox.
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. If someone knows how to tell
aclocal to use both directories, I'd be interested. aclocal accepts the
-I parameter, but I don't know where that should be specified in the
build/configure scripts.
Hopefully this helps at least a little.
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to, 2009-07-16 kello 22:28 +0200, Malte Gell kirjoitti:
Tanu Kaskinen ta...@iki.fi wrote
I don't know. Pulseaudio doesn't use hw mixing even if it's available.
But hw mixing should allow you to run Skype and Pulseaudio side by side
so that Skype doesn't go through Pulseaudio. I can
to, 2009-07-16 kello 23:48 +0200, Malte Gell kirjoitti:
Tanu Kaskinen ta...@iki.fi wrote
The correct fix would of course be to fix the webcam drivers
- if only some sample rates work, opening the device with anything else
should fail.
So this is a issue for the snd-usb-audio author(s
as much as you can.
This way the requests for new data come to the clients in a smoother
manner.
HTH.
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point, other than
sample rate conversions (which are also done somewhat transparently by
the existing framework).
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from the bridge
to the kernel module and from there to the application.
If you haven't done so yet, read the libpulse documentation at
http://0pointer.de/lennart/projects/pulseaudio/doxygen/ (from the front
page go to the Asynchronous API section).
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at all. But I'm not an expert here.
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the hardware directly after all! Instead,
it seems to be hardcoded to use dmix. (Hey Adobe, WTF is so hard in
using the default device??) I added this to ~/.asoundrc:
pcm.!dmix {
type pulse
}
And it worked!
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to already be highly
ranked by Google...) to Adobe and the distro maintainers who added this
nonsense to their packages.
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is to have the code ready for review by
the end of this month.
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the
unrecognized name, but since the plugin works with stereo signals,
changing the name alone won't help. Patches welcome!
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either Skype or Flash myself, but I believe the situation is
such that Flash 10 should work fine (I think Youtube might work with
free alternatives too: gnash and/or swfdec), whereas Skype doesn't work
well with PulseAudio.
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with rewinds, seems very non-trivial to me. It's on my todo
list, but it looks like I will have very little time in near future to
work on this. If you took this project forward, I would be very
grateful!
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, but I don't feel like going
in the details of those if this works for you. If you're interested, I
just point you to this:
https://tango.0pointer.de/pipermail/pulseaudio-discuss/2009-August/004730.html
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function calls pertain to buffering, I also tried with tsched=0. It
worked. So, maybe this is a bug in the plug alsa plugin, ie. it
doesn't implement the *_near() functions properly.
Lennart, any insights?
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pe, 2009-10-02 kello 18:30 +0300, Tanu Kaskinen kirjoitti:
Sadly the audio has a very poor quality:
1) the audio is pulsing, it seems that it pump up and down the master
volume
2) the rate is wrong, because it seems an old 33 rpm listening at 45.
I don't have any idea about the volume
a
wiki page that you can hopefully use as a canned response in the future:
http://pulseaudio.org/wiki/DefaultDevice
Please improve it if needed.
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to do mixing or resampling or
sample format conversion, I'm surprised if the bytes are touched.
So, you had float32le tones. Is your sink also configured with that
format (see pacmd list-sinks)? Do the sample rates match between the
files and the sink?
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ma, 2009-11-30 kello 19:51 +, Neil Wilson kirjoitti:
2009/11/30 Tanu Kaskinen ta...@iki.fi:
Are you sure your files had the same sample format as what your sound
card uses? If pulseaudio doesn't need to do mixing or resampling or
sample format conversion, I'm surprised if the bytes
Since the stream identifiers (channels) are monotonically growing integer, it
isn't a good idea to use them as index to a dynamic array, because the array
will grow all the time. This is not a problem with client connections that
don't create many streams, but, for example, long-running clients
Hello,
Here's five patches fixing various crash-inducing bugs in
protocol-dbus.c and module-stream-restore.c.
I hope these patches come across well-formatted - I have never sent
inline patches by mail before today.
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---
src/pulsecore/protocol-dbus.c |8
src/pulsecore/protocol-dbus.h |1 +
2 files changed, 9 insertions(+), 0 deletions(-)
diff --git a/src/pulsecore/protocol-dbus.c b/src/pulsecore/protocol-dbus.c
index e427bb1..95518a1 100644
--- a/src/pulsecore/protocol-dbus.c
+++
---
src/pulsecore/protocol-dbus.c |1 +
1 files changed, 1 insertions(+), 0 deletions(-)
diff --git a/src/pulsecore/protocol-dbus.c b/src/pulsecore/protocol-dbus.c
index 95518a1..582bc65 100644
--- a/src/pulsecore/protocol-dbus.c
+++ b/src/pulsecore/protocol-dbus.c
@@ -574,6 +574,7 @@ static
---
src/modules/module-stream-restore.c | 10 +-
1 files changed, 5 insertions(+), 5 deletions(-)
diff --git a/src/modules/module-stream-restore.c
b/src/modules/module-stream-restore.c
index 02c312e..ce92362 100644
--- a/src/modules/module-stream-restore.c
+++
---
src/modules/module-stream-restore.c |1 +
1 files changed, 1 insertions(+), 0 deletions(-)
diff --git a/src/modules/module-stream-restore.c
b/src/modules/module-stream-restore.c
index ce92362..a1273fe 100644
--- a/src/modules/module-stream-restore.c
+++
---
src/modules/module-stream-restore.c |9 +++--
1 files changed, 7 insertions(+), 2 deletions(-)
diff --git a/src/modules/module-stream-restore.c
b/src/modules/module-stream-restore.c
index a1273fe..becdb54 100644
--- a/src/modules/module-stream-restore.c
+++
is used
with three components (pa_asyncq, pa_shmasyncq and pa_rtpoll) of which
two (asyncq and shmasyncq) I know aren't multiple writer/reader safe. I
don't know the status of rtpoll.
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specify the sink or source too too:
pcm.speakers {
type pulse
device alsa_output.foobar42
}
HTH.
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proper latency accounting
and support for fixed blocksize filters. I believe those features are
going to give me headaches, because the pipeline will have to do
buffering and the sinks (and other components) have to take that into
account in their own operation...
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ti, 2009-12-22 kello 20:34 +, Colin Guthrie kirjoitti:
Now as PA will not add individual sinks for both the digital and analog,
you'll have to load a separate module-alsa-sink for the other one you
want - e.g. if auto-detection will load digital, you should add a
separate sink for analog.
/hvpanm.png
That should do the trick for the module-combine part.
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can add a separate sink for one
of the outputs manually in default.pa, but I don't think it will work
any better than creating the new profile.
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, and the
module would just use that property when setting the resampler of the
new sink input.
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tracks the option states along with port volume and mute.
Lennart, any comments? Would you accept a patch that made these changes?
I'm not saying that I would definitely write the patch, but it's not
impossible either.
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to be running pulseaudio 0.9.19. There were some bluetooth
fixes in 0.9.20; I recommend updating to 0.9.21. If the bugs are still
there, file a new ticket.
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,
it's a privacy issue on shared machines. You can try arguing against
that as long as nobody reports real cases of eavesdropping, but I think
this is a rather obvious problem...
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la, 2010-01-02 kello 02:56 +0100, Markus Rechberger kirjoitti:
On Sat, Jan 2, 2010 at 2:21 AM, Tanu Kaskinen ta...@iki.fi wrote:
pe, 2010-01-01 kello 16:58 -0500, Bill Cox kirjoitti:
Anyone out there every get hacked because you shared the Alsa back-end
with another user? Anyone?
I
it should work. I'm not very confident when speaking
about consolekit and boot/login processes, so I have to hope that the
system I described isn't too different from how things work in reality.
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la, 2010-01-02 kello 14:40 +, Colin Guthrie kirjoitti:
'Twas brillig, and Tanu Kaskinen at 01/01/10 03:52 did gyre and gimble:
Hi Colin,
I got a bit confused about the term port in your mail. Apparently you
used it to mean something else than pulseaudio's own device port
concept
to do that. Use switch = off
to set the element always off or switch = on to set it always on.
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. The current
symptoms don't sound like you've hit this issue yet, but I would think
that the system-wide pulseaudio instance will lose access to the sound
card as soon as someone logs in.
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ma, 2010-01-04 kello 23:10 +0100, Lennart Poettering kirjoitti:
On Fri, 01.01.10 05:52, Tanu Kaskinen (ta...@iki.fi) wrote:
- Track port volume and mute individually. That is, whenever the active
port is switched, the previous port's volume and mute state should be
saved to a database
ti, 2010-01-05 kello 09:44 +, Colin Guthrie kirjoitti:
'Twas brillig, and Tanu Kaskinen at 05/01/10 05:29 did gyre and gimble:
If I listen to music with headphones and find that the volume is not
perfect, I turn the music volume up or down. After this the volume is
perfect. Then I
going to tweak the
music player volume, which modifies the stream volume.
--
Tanu Kaskinen
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