a new source that have "echo
cancelled" in their names. Set them as default and move all
applications to them.
--
Alexander E. Patrakov
d 4.0 Output +
> Analog Stereo Input (priority 1265, available: unknown)
> output:iec958-stereo: Digital Stereo (IEC958) Output (priority 38268,
> available: unknown)
> output:iec958-stereo+input:analog-stereo: Digital Stereo (IEC958) Output +
> Analog Stereo Input (priority 5565, available: unknown)
> output:hdmi-stereo: Digital Stereo (HDMI) Output (priority 38668, available:
> unknown)
> output:hdmi-stereo+input:analog-stereo: Digital Stereo (HDMI) Output + Analog
> Stereo Input (priority 5965, available: unknown)
> output:hdmi-surround: Digital Surround 5.1 (HDMI) Output (priority 33568,
> available: unknown)
> output:hdmi-surround+input:analog-stereo: Digital Surround 5.1 (HDMI) Output
> + Analog Stereo Input (priority 865, available: unknown)
> off: Off (priority 0, available: unknown)
> active profile:
> sinks:
> alsa_output.pci-_00_1b.0.hdmi-stereo/#0: Built-in Audio Digital Stereo
> (HDMI)
> sources:
> alsa_output.pci-_00_1b.0.hdmi-stereo.monitor/#1: Monitor of Built-in
> Audio Digital Stereo (HDMI)
> ports:
> analog-input-mic: Microphone (priority 8700, latency offset 0 usec,
> available: no)
> properties:
> device.icon_name = "audio-input-microphone"
> analog-output-headphones: Headphones (priority 9900, latency offset 0 usec,
> available: no)
> properties:
> device.icon_name = "audio-headphones"
> analog-output: Analog Output (priority 9900, latency offset 0 usec,
> available: unknown)
> properties:
>
> iec958-stereo-output: Digital Output (S/PDIF) (priority 0, latency offset 0
> usec, available: unknown)
> properties:
>
> hdmi-output-0: HDMI / DisplayPort (priority 5900, latency offset 0 usec,
> available: yes)
> properties:
> device.icon_name = "video-display"
> device.product.name = "SAMSUNG"
Don't bother. This is an issue common to 99% of USB webcams. Namely,
those not listed explicitly in /usr/share/alsa/cards/USB-Audio.conf.
Both the analog and the fake S/PDIF microphones ultimately point to
exactly the same ALSA hardware device.
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Alexander E. Patrakov
at it fixes detection of
recording devices. It was caused by incompatibility of the old code
with GStreamer 1.16.x. Could you please make sure that you have at
least that version?
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CV: http://u.pc.cd/wT8otalK
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t, use speex-float-5. There is no point in going
higher than that.
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quot;default.pa is too static and cannot configure hot-plugged things"
problem.
An obvious proposed addition to the API would be an ability to list
persistent loopbacks, including those that refer to a source or a sink
that doesn't exist at the moment.
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пн, 20 мая 2019 г. в 15:35, Georg Chini :
>
> On 20.05.19 10:47, Alexander E. Patrakov wrote:
> > пн, 20 мая 2019 г. в 13:17, Georg Chini :
> >> Hi Alexander, Tanu,
> >>
> >> would it make sense to give filters some old data for preconditioning
monstrate the problem. Please note that it is not
suitable for inclusion of PulseAudio.
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вс, 5 мая 2019 г. в 22:58, Georg Chini :
>
> On 05.05.19 18:41, Alexander E. Patrakov wrote:
> > вс, 5 мая 2019 г. в 01:41, Georg Chini :
> >> On 04.05.19 20:54, Alexander E. Patrakov wrote:
> >>> сб, 4 мая 2019 г. в 20:25, Georg Chini :
> >>>>
вс, 5 мая 2019 г. в 01:41, Georg Chini :
>
> On 04.05.19 20:54, Alexander E. Patrakov wrote:
> > сб, 4 мая 2019 г. в 20:25, Georg Chini :
> >> On 04.05.19 16:42, Alexander E. Patrakov wrote:
> >>> сб, 4 мая 2019 г. в 16:17, Georg Chini :
> >>>> H
сб, 4 мая 2019 г. в 20:25, Georg Chini :
>
> On 04.05.19 16:42, Alexander E. Patrakov wrote:
> > сб, 4 мая 2019 г. в 16:17, Georg Chini :
> >> Here is the new version of the header file, based on your feedback.
> >> The main changes are:
> >>
> >
reason for the
filter to be able to influence it.
Therefore, I believe these two fields can be replaced by one,
max_rewind, which is the size of history, in samples, that the filter
is willing to keep. Zero means a non-rewindable filter.
--
Alexander E. Patrakov
_
пт, 3 мая 2019 г. в 23:04, Georg Chini :
>
> On 03.05.19 16:32, Alexander E. Patrakov wrote:
> > пт, 3 мая 2019 г. в 11:57, Georg Chini :
> >> The channel layout of input/output
> >> and the playback device is known to module-plugin-sink, so if
> >> a filter
be interesting to see what happens if unplugging the
headphones causes the number of channels (of the channel layout in
general) to change. Note that I don't have such setup, not even sure
how it is handled currently.
--
Alexander E. Patrakov
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пт, 3 мая 2019 г. в 09:31, Georg Chini :
>
> On 03.05.19 02:22, Alexander E. Patrakov wrote:
> > чт, 2 мая 2019 г. в 23:45, Georg Chini :
> >> Hi,
> >>
> >> I created a new module-plugin-sink and a small amplifier demo plugin
> >> based on the attach
ot have, because we want to keep it simple and stable) any place to
express such policy.
In other words, I am still very skeptical about the proposal, simply
because we don't have enough filters to test the validity of the
proposed interface.
--
Alexander E. Patrakov
_
вт, 30 апр. 2019 г. в 23:46, Georg Chini :
>
> On 30.04.19 19:23, Alexander E. Patrakov wrote:
> >> /* If set, this function is called in thread context when an update
> >> of the
> >>* filter parameters is requested, may be NUL
r to the old structure so that it can be freed in the
> main thread
> * using parameter_free(). If the old structure can be re-used, the
> function
> * may return NULL. */
> void *(*update_filter_parameters)(void *parameters, void
&
сб, 20 апр. 2019 г. в 15:20, Georg Chini :
>
> On 20.04.19 12:14, Alexander E. Patrakov wrote:
> > [forgot to CC the list - sorry for any duplicates]
> >
> > сб, 20 апр. 2019 г. в 14:39, Georg Chini :
> >
> >> I think using malloc() when a parameter changes
wo filter versions and
interpolate between them, in order to avoid clicks, because such
gradual changes are an expected thing in pro audio workflows.
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e.
>
> If you load it through a file like pulseaudio does, how do
> you pass the values to the filter? Or should the plugin itself
> load the file?
The plugin itself should load the file. The fact that the file does
not come with the plugin itself is a major bug that unfortunately
exists fo
сб, 20 апр. 2019 г. в 00:15, Georg Chini :
>
> On 19.04.19 18:23, Alexander E. Patrakov wrote:
> > пт, 19 апр. 2019 г. в 14:13, Tanu Kaskinen :
> >> If the plugin gets the number of bands during the
> >> initialization, it can create the appropriate number o
пн, 8 апр. 2019 г. в 15:05, Alexander E. Patrakov :
>
> пн, 8 апр. 2019 г. в 13:08, Alexander E. Patrakov :
>
> > I have looked again at the paper, and have a DSP question.
> >
> > The thesis starts with designing a shelf filter. That is, a filter
> > that contai
HDMI receivers, Hi-Fi
amplifiers) just do not have equalizers with a variable number of
bands, for usability reasons. I don't see a reason for PulseAudio to
be different.
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пн, 8 апр. 2019 г. в 13:08, Alexander E. Patrakov :
> I have looked again at the paper, and have a DSP question.
>
> The thesis starts with designing a shelf filter. That is, a filter
> that contains a flat area at low frequencies, a flat area at high
> frequencies (but with a
tain. Additionally I agree with Andrea that
> handling LADSPA plugins is somewhat cumbersome. From a
> user point of view, a module is much easier to handle.
I have looked again at the paper, and have a DSP question.
The thesis starts with designing a shelf filter. That is, a filter
that conta
ontrol is not going to work, as the user will not be
able to interact with it, if the screen reader speaks to the HDMI
monitor which pretends to accept sound but does not actually have
speakers.
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Russell Treleaven :
>
> Is there an open bug for this?
No.
>
> On Wed, Dec 26, 2018, 6:41 PM Alexander E. Patrakov
> wrote:
>>
>> David Davidović :
>> >
>> > Hi,
>> >
>> > I'm running a system-wide PulseAudio instance o
what is written right now
(not what is playing right now).
What would help is a "pactl list sinks" on both ends while music is
playing and pavucontrol is not running.
> Has anyone else had this issue and resolved it?
Yes, by closing pavucontrol.
--
Alexander E. Patrakov
__
ing
like --disable-running-from-build-tree or --enable-reproducible-build)
that explicitly disables running from build tree and thus enables a
reproducible build?
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On 11/16/18 11:13 AM, Tanu Kaskinen wrote:
On Thu, 2018-11-15 at 23:50 +0500, Alexander E. Patrakov wrote:
Alexander E. Patrakov :
Tanu Kaskinen :
On Sat, 2018-10-13 at 22:29 +0500, Alexander E. Patrakov wrote:
Note that I am unhappy with it, because it fixes only a particular
common problem
Alexander E. Patrakov :
>
> Tanu Kaskinen :
> >
> > On Sat, 2018-10-13 at 22:29 +0500, Alexander E. Patrakov wrote:
> > > On 10/7/18 6:27 PM, Alexander E. Patrakov wrote:
> > > > вс, 7 окт. 2018 г. в 16:42, Karl Ove Hufthammer > > > <mailto
Tanu Kaskinen :
>
> On Sat, 2018-10-13 at 22:29 +0500, Alexander E. Patrakov wrote:
> > On 10/7/18 6:27 PM, Alexander E. Patrakov wrote:
> > > вс, 7 окт. 2018 г. в 16:42, Karl Ove Hufthammer > > <mailto:k...@huftis.org>>:
> > >
> >
Mikael Nousiainen :
>
> On Thu, Nov 8, 2018, at 20:49,
> pulseaudio-discuss-requ...@lists.freedesktop.org wrote:
> > Message: 4
> > Date: Thu, 8 Nov 2018 21:31:52 +0500
> > From: "Alexander E. Patrakov"
> > To: General PulseAudio Discussion
>
pture" to
compensate. So the fix would be to make sure that monitor sources
capture not what was just written to them, but what was actually played
and can no longer be overwritten. I.e., change the latency from negative
to small positive.
--
Alexander e
e need to figure out whether this is due to a monitor or due to a
null sink. Could you please let the browser play to your speakers, and
WSJT-X record from the monitor of the sound card that the browser is
playing to? Does it work?
Is the webrtc feed public? I.e., can I run WSJT-X locally in order
up to speed with recent KDE and
GNOME), and it uses exactly the proposed architecture.
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Tanu Kaskinen :
>
Thanks for contributing to this thread :)
> On Mon, 2018-11-05 at 00:14 +0500, Alexander E. Patrakov wrote:
> > Andrea A :
> > > I'm writing a new equalizer module for PA,
> > > https://github.com/andrea993/audioeqpro/blob/master/pulsemodule/
ch, I will look. It will be slower than usual because
I will have to use Google Translate.
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publish your thesis online somewhere as a pdf and insert a
link to it in the source code as a comment.
P.S. in some countries, including Russia, such publication is required anyway.
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pulsea
d to care about rewinds and may_move_to and all other
pulseaudio-specific boilerplate. Sounds like a win-win situation.
Could you please investigate this approach?
Thanks again for attempting to replace the current equalizer with
something better.
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might
be bugs and corner cases that I haven't thought about. Testing and
criticizing is welcome, inclusion in the official tree is probably not a
good idea.
The patch is an alternative approach to what was solved by
https://lists.freedesktop.org/archives/pulseaudio-discuss/2018-October/030
On 10/7/18 6:27 PM, Alexander E. Patrakov wrote:
вс, 7 окт. 2018 г. в 16:42, Karl Ove Hufthammer <mailto:k...@huftis.org>>:
Alexander E. Patrakov skreiv 07.10.2018 09:44:
>
> I don’t understand why this is happening. Shouldn’t
> ‘remixing-use
вс, 7 окт. 2018 г. в 16:42, Karl Ove Hufthammer :
> Alexander E. Patrakov skreiv 07.10.2018 09:44:
> >
> > I don’t understand why this is happening. Shouldn’t
> > ‘remixing-use-all-sink-channels = no’ just affect *upmixing* of
> > sound,
> >
ed should work fine.
But in reality this should be fixed in PulseAudio code. Maybe I will work
on it next week.
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bwoofer. And unfortunately, my (not too expensive) home
> theatre system doesn’t have any options for increasing this (for HDMI
> audio).
>
Are you sure that the subwoofer gets anything at all in this mode?
Anyway the suggestion is to try the
y library which all above support? I have not find
> anything. Nor ffmpeg nor gstreamer.
>
> Which means that pulseaudio cannot delegate codec selection, codec
> initialization and codec encoding to some external library (yet).
>
> Anyway, now in bluez source code is some kind of codec encoding support
> for android systems. Therefore should not be a wise idea to move all
> these codec A2DP mess into bluez source code? Because now part of A2DP
> is in bluez and another part in pulseaudio. And on android is everything
> in bluez now.
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2018-05-25 22:15 GMT+08:00 Arun Raghavan <a...@arunraghavan.net>:
> On Tue, 22 May 2018, at 4:32 PM, Alexander E. Patrakov wrote:
>> There is some Microsoft documentation that contradicts this code.
>>
>> https://msdn.microsoft.com/en-us/library/windows/des
> pulseaudio-discuss mailing list
> pulseaudio-discuss@lists.freedesktop.org
> https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
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; alternate_rate % 11025 == 0 && spec->rate %
> 11025 == 0)
> +if (alternate_rate % 11025 == 0 && spec->rate % 11025 == 0)
> alternate_rate_is_usable = true;
> -if (alternate_rate && alternate_rate % 4000 == 0 && sp
Please run: alsamixer -c C920, then press F4 and post a screenshot.
2018-03-08 12:39 GMT+08:00 Brandon Carpenter <elementary.eni...@gmail.com>:
> On Wed, Mar 7, 2018 at 8:02 PM Alexander E. Patrakov <patra...@gmail.com>
> wrote:
>> Please run "alsa-info.sh". It w
HD Pro Webcam C920. The camera on the C920 works flawlessly.
> However, the builtin microphone only worked occasionally, and recently
> stopped working altogether.
Please run "alsa-info.sh". It will create a URL with more details. Post it here.
--
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___
2017-12-29 20:37 GMT+08:00 Tanu Kaskinen <ta...@iki.fi>:
> On Fri, 2017-12-29 at 11:46 +0800, Alexander E. Patrakov wrote:
>> 2017-12-28 18:09 GMT+08:00 Tanu Kaskinen <ta...@iki.fi>:
>> > The Intel HDMI LPE driver works in a peculiar way when the HDMI cable is
&g
e, based on the number of samples written and the wall-clock
time. If the card stalls or eats samples too quickly, set a flag that
it misbehaves, accept the xrun, and then set it to off when
convenient.
Sorry, I have no time to help with the code :(
--
Alexander E. Patrakov
_
e to load it into a distribution-provided
pulseaudio.
In other words, modules are used by PulseAudio not as an "extension
point for third parties like you", but as a convenience for
distributors so that they can split the package and do not require
libraries th
lid. As I said,
naive attempts of attaching independently-tuned filters to each
channel are Just Wrong from the DSP viewpoint, because of phase.
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of mpv.
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mit/?id=04737989ec537e01d63c155807a5d19b0ec5b294
I think that, if you don't undo this commit, at least a better message
what exactly works is needed. I.e. how to test, given that the HDMI
sink does not have the formats set.
--
Alexander E. Patrakov
_
when I attempt to run
> pulseaudio with sudo I can't hear anything. Is there any detailed tutorial
> on how to accomplish this?
Why do you run qemu with sudo? This should not be needed. If this is
due to the graphics card pass-through, try to chown the vfio nodes to
your user instead.
2017-08-12 0:27 GMT+05:00 Markus Ebner <markus-eb...@web.de>:
> On 08/11/2017 06:51 PM, Alexander E. Patrakov wrote:
>>
>> 2017-08-11 21:48 GMT+05:00 Alexander E. Patrakov <patra...@gmail.com>:
>>>
>>> 2017-08-10 1:27 GMT+05:00 Markus Ebner
cations to
> common"
> +print " that have not been commited to the actuall common module."
> +print
> +print " First push your changes to common/ before pushing the
> changes"
> +print " to the module using it."
2017-02-14 23:13 GMT+05:00 Fatima Castiglione Maldonado 发
<castiglionemaldon...@gmail.com>:
> it did not work
> maybe module is not installed?
It is too old to accept arguments that I specified. Try without them:
pacmd load-module module-echo-cancel
--
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ks
behave differently than normal ones in aspects related to buffering.
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}
>
> +use_getsockopt = false;
> +if (pa_modargs_get_value_boolean(ma, "use_getsockopt", _getsockopt)
> < 0) {
> +pa_log("Invalid boolean value for use_getsockopt parameter");
> +goto fail;
> +}
> +
> m->userdata = u = pa_xnew0(struct userdata, 1);
> u->module = m;
> u->core = m->core;
> +u->use_getsockopt = use_getsockopt;
> u->loaded_device_paths = pa_hashmap_new(pa_idxset_string_hash_func,
> pa_idxset_string_compare_func);
>
> if (!(u->discovery = pa_bluetooth_discovery_get(u->core,
> headset_backend)))
> --
> 2.10.1
>
> ___
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I meant that these are latencies that are related to different parts
of the path.
Please send the patch, but I don't have the expertise to review it.
2017-01-30 22:46 GMT+05:00 Georg Chini <ge...@chini.tk>:
> On 30.01.2017 18:37, Alexander E. Patrakov wrote:
>>
>> I am n
to correct it?
>
> Regards
> Georg
>
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__
x1 + 1;
>
> pa_log_info("iteration %i (%i)", k, magic_number);
>
> --
> 2.11.0
>
> ___
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u Kaskinen <ta...@iki.fi>:
> On Tue, 2017-01-10 at 04:35 +0500, Alexander E. Patrakov wrote:
>> 2017-01-09 1:05 GMT+05:00 Tanu Kaskinen <ta...@iki.fi>:
>> > Hi all,
>> >
>> > The first draft of the 10.0 release notes is now complete:
>> &
ALSA device other than "hw:".
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e source audio.
>
> BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=62588
> BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=94563
>
> Suggested-by: Alexander E. Patrakov <patra...@gmail.com>
> ---
> src/pulsecore/resampler.c | 82
>
iscuss@lists.freedesktop.org
> https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
>
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://forum.xda-developers.com/showthread.php?p=29986442
(download attachment, because the app is no longer on Google Play)
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output), and its active port becomes unavailable?
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the v9.0 release.
There are no uninitialised value warnings with these patches. Thanks!
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12.06.2016 21:39, Alexander E. Patrakov пишет:
I have spent more time running pulseaudio (master + Tanu's patches)
under valgrind today.
This set of warnings has already been reported, and they appear every
time PulseAudio is started:
==15442== Conditional jump or move depends on uninitialised
16.06.2016 13:43, Ahmed S. Darwish пишет:
Hi :-)
On Sun, Jun 12, 2016 at 09:39:17PM +0500, Alexander E. Patrakov wrote:
I have spent more time running pulseaudio (master + Tanu's patches) under
valgrind today.
This set of warnings has already been reported, and they appear every time
13.06.2016 01:36, Tanu Kaskinen пишет:
On Sun, 2016-06-12 at 18:33 +0500, Alexander E. Patrakov wrote:
10.06.2016 22:55, Tanu Kaskinen пишет:
I want module-alsa-card to set the availability of unavailable
profiles before the initial card profile gets selected, so that the
selection logic can
13.06.2016 01:12, Tanu Kaskinen пишет:
I'd prefer to not add an assertion. Most of pa_hook_fire() calls in the
code base don't check the return value, and none of the calls use an
assertion.
OK.
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(mainloop.c:655)
==15374==by 0x510040B: pa_mainloop_dispatch (mainloop.c:898)
==15374==by 0x510080B: pa_mainloop_iterate (mainloop.c:929)
==15374==by 0x51008AF: pa_mainloop_run (mainloop.c:944)
==15374==by 0x406E3B: main (main.c:1141)
--
Alexander E. Patrakov
12.06.2016 19:16, Alexander E. Patrakov пишет:
10.06.2016 22:55, Tanu Kaskinen пишет:
The alsa card hasn't so far set any availability for profiles. That
caused an issue with some HDMI hardware: the sound card has two HDMI
outputs, but only the second of them is actually usable. The
unavailable
it a bit
later.
--
Alexander E. Patrakov
---
src/modules/alsa/module-alsa-card.c | 27 +++
1 file changed, 27 insertions(+)
diff --git a/src/modules/alsa/module-alsa-card.c
b/src/modules/alsa/module-alsa-card.c
index 1976230..323e08a 100644
--- a/src/modules/alsa
list_copy(data->proplist);
c->driver = pa_xstrdup(pa_path_get_filename(data->driver));
c->module = data->module;
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10.06.2016 22:55, Tanu Kaskinen пишет:
It's not being used any more.
Looks OK to me.
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Alexander E. Patrakov
---
src/pulsecore/card.c | 8
src/pulsecore/card.h | 5 -
2 files changed, 13 deletions(-)
diff --git a/src/pulsecore/card.c b/src/pulsecore/card.c
index a0c3d93
would like an extra confirmation whether in the
UCM case the sequence of mixer accesses is the same before and after the
patch.
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Alexander E. Patrakov
---
src/modules/alsa/module-alsa-card.c | 31 ++---
src/modules/bluetooth/module-bluez4-device.c | 31 +
src/modules/blu
added so that it is explicit that the hook is not expected to fail.
c->core = core;
c->name = pa_xstrdup(data->name);
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%s' not valid or not supported by device.",
default_profile);
+else {
+pa_log("Profile '%s' not valid or not supported by device.",
default_profile);
+pa_card_new_data_done();
+return -1;
+ }
22.05.2016 02:03, Alexander E. Patrakov wrote:
If a card has been hot-plugged after pulseaudio start, alsa-lib still has
old configuration in memory, which doesn't have PCM definitions for the
new card. Thus, this error appears, and the device doesn't work:
I: [pulseaudio] (alsa-lib)confmisc.c
/show_bug.cgi?id=54029
Signed-off-by: Alexander E. Patrakov <patra...@gmail.com>
---
src/modules/alsa/alsa-sink.c| 9 +
src/modules/alsa/alsa-source.c | 9 +
src/modules/alsa/module-alsa-card.c | 8
3 files changed, 26 insertions(+)
diff --git a/src/modules/als
.
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er frequency (120 Hz) is
wrong for laptop speakers.
- Finally, Windows 10 with a built-in USB audio driver does not synthesize
the LFE channel given a 5.1 card and a stereo audio stream by default.
Hides: https://bugs.freedesktop.org/show_bug.cgi?id=95021
Signed-off-by: Alexander E. Patrakov <patra
I move the stream to another sink, and
sometimes reappears when I move it back.
I will bisect later, as I know that in 8.0 and 93822f9 there was no such
bug.
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ic gain control and similar things) becomes unavailable, so it
may or may not be an option for your application.
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module-loopback
is, well, a module. You can remove this argument if you have no users
except root on your system.
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04.05.2016 11:00, Arun Raghavan wrote:
On 4 May 2016 at 11:23, David Henningsson <di...@ubuntu.com> wrote:
On 2016-05-04 06:49, Alexander E. Patrakov wrote:
Raymond produces too many irrelevant comments and seems not to understand
what he is talking about or what was already debugged.
Raymond produces too many irrelevant comments and seems not to
understand what he is talking about or what was already debugged. May I
request a ban in order to increase our signal-to-noise ratio?
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Alexander E. Patrakov
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19.04.2016 19:22, Luis Cavalheiro пишет:
Additional data: I use webrtc echo cancellation module. It may be relevant.
Yes, it is relevant.
Please pass this argument to module-echo-cancel:
aec_args="analog_gain_control=0\ digital_gain_control=0"
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Alexander E
source-outputs
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expressed in
https://lists.freedesktop.org/archives/pulseaudio-discuss/2015-February/023224.html
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nt and we'll get to them.
>
> If you think something is crucial for this release (critical bug fix,
> regression, etc.), please holler now.
We still don't have
https://freedesktop.org/software/pulseaudio/webrtc-audio-processing/webrtc-audio-processing-0.2.
en in linphone preferences.
So - no real (consistency-based) opinion here. If you can find anything
in the HIG that explains that the border should not exist in this case,
please add a link in the commit message, and/or quote directly, and push it.
--
Alexander E. Patrakov
---
src/pavuco
features externally to PulseAudio if it is easier. And an
external module SDK would also be a step forward.
I also think I should try porting the existing RAOP2 patch set to make
it an ALSA plugin instead, to set an example of the "externally where it
is easier" rule.
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Alexander E
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