Currently the default sink policy is simple: either the user has
configured it explicitly, in which case we always use that as the
default, or we pick the sink with the highest priority. The sink
priorities are currently static, so there's no need to worry about
updating the default sink when sink
ing about what to do
about that, so I'll submit that patch separately later, or a
different patch that achieves the same goal.
Tanu Kaskinen (2):
refactor default sink/source handling
core, device-port: check availability when choosing the default device
src/modules/dbus/iface-core.c
On Wed, 2017-02-01 at 16:39 -0500, sean darcy wrote:
> On 01/28/2017 07:22 AM, Tanu Kaskinen wrote:
> > I suspect
> > {fd19a5b3f9ab48aeae18d687a1e5c0cc}unix:/run/user/1001/pulse/native
> > comes from the X server. On your Fedora laptop, module-x11-publish puts
> >
On Wed, 2017-02-01 at 09:44 +0100, Ivan Kanis wrote:
> Hello,
>
> I am trying to get sound output from the speaker plugged on my computer
> and the Chromecast audio in the living room.
>
> I have launched the program pulseaudio-dlna and I can hear audio on the
> Chromecast when I select it as pla
On Tue, 2017-01-31 at 17:29 +0530, kranthi kumar wrote:
> Hi All,
>
> I am very new to pulseaudio environment. I have working ucm scripts for
> playback and capture in HiFi profile. The requirement is that I need to
> split the scripts into codec specific and machine specific. Codec and
> machine
Clang didn't like the variable length array:
pulsecore/iochannel.c:358:17: error: fields must have a constant size:
'variable length array in structure' extension will never be supported
uint8_t data[CMSG_SPACE(sizeof(int) * nfd)];
^
Commit 451d1d6762 introduced the variab
We disallow autospawning for root, but when using systemd socket
activation to start pulseaudio, that replaces the autospawning
mechanism, and there was no similar "root protection" in socket
activation. This patch disables the socket activation for root.
Thanks to Felipe Sateler for coming up wit
On Tue, 2017-01-31 at 11:18 -0300, Felipe Sateler wrote:
> On 31 January 2017 at 10:45, Tanu Kaskinen wrote:
> > On Mon, 2017-01-30 at 10:17 -0300, Felipe Sateler wrote:
> > > On 28 January 2017 at 11:24, Ahmed S. Darwish
> > > wrote:
> > > > On Sat, Ja
On Mon, 2017-01-30 at 14:54 +0530, Arun Raghavan wrote:
> On Thu, 8 Sep 2016, at 04:36 PM, Tanu Kaskinen wrote:
> > It was reported in bug 93006 that on hardware that has HDMI and analog
> > outputs on different cards, the HDMI sink is chosen by default even when
> > headphone
On Mon, 2017-01-30 at 14:37 +0530, Arun Raghavan wrote:
>
> On Thu, 8 Sep 2016, at 04:36 PM, Tanu Kaskinen wrote:
> > +void pa_core_update_default_sink(pa_core *core) {
> > +pa_sink *best = NULL;
> > +pa_sink *sink;
> > +uint32_t idx;
>
On Sun, 2017-01-29 at 10:36 +0900, Takashi Sakamoto wrote:
> On Jan 29 2017 01:08, Tanu Kaskinen wrote:
> > The pa_channel_map_init_extend() call later in the function crashes if
> > if ss->channels is greater than PA_CHANNELS_MAX.
> >
> > Reported here:
> > ht
On Sat, 2017-01-28 at 17:30 -0800, Klaus Badelt wrote:
> Great workaround for now.
>
> Do you see a chance of using up to 32 channels of any capture/playback are
> used, instead of ignoring it?
Yes, if you write a patch for that :)
--
Tanu
https://www.patreon.com/tanuk
On Mon, 2017-01-30 at 17:35 +0100, Georg Chini wrote:
> Hello,
>
> in module-bluez5-device.c and module-bluez4-device.c, latencies for
> bluetooth are defined as follows:
>
> #define FIXED_LATENCY_PLAYBACK_A2DP (25 * PA_USEC_PER_MSEC)
> #define FIXED_LATENCY_PLAYBACK_SCO (125 * PA_USEC_PER_MSEC)
On Mon, 2017-01-30 at 10:17 -0300, Felipe Sateler wrote:
> On 28 January 2017 at 11:24, Ahmed S. Darwish wrote:
> > On Sat, Jan 28, 2017 at 04:00:31PM +0200, Ahmed S. Darwish wrote:
> > > Unless we want a restricting directive directly inside systemd,
> > > below trick seems to work here:
> > >
>
On Mon, 2017-01-30 at 17:28 +0100, Georg Chini wrote:
> On 27.01.2017 21:40, Georg Chini wrote:
> > On 03.01.2017 18:17, Tanu Kaskinen wrote:
> > > On Thu, 2016-12-15 at 12:34 +0530, Renjith Thomas wrote:
> > > > Issue: When HFP/HSP profile is used with certain
On Sun, 2017-01-29 at 23:23 +0530, Arun Raghavan wrote:
>
> On Sun, 29 Jan 2017, at 02:29 PM, Tanu Kaskinen wrote:
> > On Sun, 2017-01-29 at 10:29 +0530, Arun Raghavan wrote:
> > > This adds an "avoid-resampling" option to daemon.conf that makes the
> > >
On Fri, 2017-01-27 at 19:34 +0100, Steffen Pfendtner wrote:
> To catch up your points:
>
> I will synchronise the pactl to pacmd. pulseaudio-dlna is using pactl
> too. I just missed that point.
>
> I share your view regarding the location of the new functions. A new
> file set src/pulsecore/combi
On Sun, 2017-01-29 at 10:29 +0530, Arun Raghavan wrote:
> This adds an "avoid-resampling" option to daemon.conf that makes the
> daemon try to use the stream sample rate if possible (the device needs
> to support it, which currently only ALSA does), and there should not be
> any other stream connec
On Sat, 2017-01-28 at 12:47 -0800, Klaus Badelt wrote:
> Thank you! What's the behavior now with your fix? Can I find the code
> somewhere?
I sent the patch to the mailing list, the subject is
[PATCH] alsa-util: don't crash on devices with more than 32 channels
The expected behaviour with the pat
On Sat, 2017-01-28 at 13:26 +0530, Arun Raghavan wrote:
> This adds an "avoid-resampling" option to daemon.conf that makes the
> daemon try to use the stream sample rate if possible (the device needs
> to support it, which currently only ALSA does), and there should not be
> any other stream connec
On Fri, 2017-01-27 at 22:08 +0900, Takashi Sakamoto wrote:
> Hi Tanu and Klaus,
>
> On Jan 27 2016 20:26, Tanu Kaskinen wrote:
> > On Tue, 2017-01-24 at 11:35 -0800, Klaus Badelt wrote:
> > > The 32 channel limit excludes use of audio interfaces like for example
> &g
The pa_channel_map_init_extend() call later in the function crashes if
if ss->channels is greater than PA_CHANNELS_MAX.
Reported here:
https://lists.freedesktop.org/archives/pulseaudio-discuss/2017-January/027404.html
---
src/modules/alsa/alsa-util.c | 7 +++
1 file changed, 7 insertions(+)
On Thu, 2017-01-26 at 15:42 -0500, sean darcy wrote:
> On 01/26/2017 12:39 AM, Tanu Kaskinen wrote:
> > On Sat, 2017-01-21 at 16:18 -0500, sean darcy wrote:
> > > On Fedora 25, running PA 10.0.
> > >
> > > [video@sixcore ~]$ ps aux | grep pulse
> &g
Hi all,
In the "PA 10 : paplay can't connect !" thread I noticed worrying
netstat output:
[video@sixcore ~]$ netstat -l -x -p | grep pulse
(Not all processes could be identified, non-owned process info
will not be shown, you would have to be root to see it all.)
unix 2 [ ACC ] STREAM
On Thu, 2017-01-26 at 10:57 -0600, Dean Brundage wrote:
> Hey Tanu. Thanks for the reply. Response inline. --Dean
>
> On Wed, Jan 25, 2017 at 11:19 PM, Tanu Kaskinen wrote:
>
> > On Thu, 2017-01-19 at 11:15 -0600, Dean Brundage wrote:
> > > Hello pulseaudio-discuss
On Wed, 2017-01-25 at 14:06 +0500, Dmitry Vinokurov wrote:
> Hi,
>
> I've faced strange PulseAudio monitor device (i.e. audio input device which
> plays sound sent to speaker) behaviour on Ubuntu 16.04. I've reduced code
> from my real project to simple example based on code from PulseAudio docs
>
On Tue, 2017-01-24 at 11:35 -0800, Klaus Badelt wrote:
> The 32 channel limit excludes use of audio interfaces like for example (my)
> RME HDSPe MADI (64x64), all other MADI interfaces (incl. RME MADIface), RME
> Fireface UFX+ (94x94), MOTU 1248 (32x34), MOTU 112D (112), Focusrite Red
> (64x64), Pr
On Mon, 2017-01-23 at 17:24 +0100, Pascal Heinrich wrote:
> Hi,
>
> in my setup I have multiple tunnel-sinks defined and they are combined
> via module-combine-sink.
> All is running great and syncronous.
>
> Now sometimes the master pulseaudio server with mpd install comes up
> before all slav
On Mon, 2017-01-23 at 13:59 +0500, Alexander E. Patrakov wrote:
> 2017-01-23 13:38 GMT+05:00 Tanu Kaskinen :
> > If we set magic_number to zero, the code will deadlock, because the
> > thread that is waiting for us to set magic_number to non-zero will
> > never progress.
>
On Mon, 2017-01-23 at 05:24 +, Mahendran, Dandapani (D.) wrote:
> Hi,
>
> [I am resending this mail with reduced content/log]
>
> Using "module-null-sink" we face the following issue of pulseaudio
> getting stuck in rewinding process.
> The trigger point is that the input audio stream is stop
On Sat, 2017-01-21 at 16:18 -0500, sean darcy wrote:
> On Fedora 25, running PA 10.0.
>
> [video@sixcore ~]$ ps aux | grep pulse
> video 2319 0.0 0.0 433508 12212 ?Ssl 15:51 0:00
> /usr/bin/pulseaudio --daemonize=no
> video 2358 0.0 0.0 130028 4892 ?S15:51 0
On Thu, 2017-01-19 at 11:15 -0600, Dean Brundage wrote:
> Hello pulseaudio-discuss,
>
> I am setting up a small linux computer (C.H.I.P.) to be a bluetooth
> receiver using pulseaudio. The device is not running X so I am working
> through pactl and configuration files. Everything is working exce
On Tue, 2017-01-17 at 19:53 -0800, Klaus Badelt wrote:
> ---
> src/pulse/sample.h | 2 +-
> 1 file changed, 1 insertion(+), 1 deletion(-)
>
> diff --git a/src/pulse/sample.h b/src/pulse/sample.h
> index 4299eec..613c3e8 100644
> --- a/src/pulse/sample.h
> +++ b/src/pulse/sample.h
> @@ -125,7 +125
On Tue, 2017-01-17 at 17:52 +0530, Arun Raghavan wrote:
>
> On Tue, 17 Jan 2017, at 12:12 PM, Tanu Kaskinen wrote:
> > Also remove the minimum supported PulseAudio version. The current
> > minimum libpulse version is 5.0, but if it's written on the page, the
> > in
On Tue, 2017-01-17 at 11:18 +0100, Michael Renner wrote:
> Moin,
>
> my new project seems to be easy, but the setup still does not work. On a
> Raspberry Pi (raspbian jessie with backports) runs pulseaudio 7.1.2 on a
> USB sound card. On a notebook I installed ubuntu xenial with Pulseaudio 8.
>
>
On Mon, 2017-01-16 at 16:37 -0600, A. Wilcox wrote:
> Hello,
>
> While packaging PulseAudio for our distribution, I noticed the test
> suite failing in two different ways; both of these failures are in
> thread-test.
>
> The first failure is related to pulseaudio's usage of rand().
> According to
If we set magic_number to zero, the code will deadlock, because the
thread that is waiting for us to set magic_number to non-zero will
never progress.
The problem was reported here:
https://lists.freedesktop.org/archives/pulseaudio-discuss/2017-January/027368.html
---
src/tests/thread-test.c | 5
Dear Sir/Madam,
I'm pleased to inform you that the version 10.0 of PulseAudio is now
officially ready for your consumption. Here's a list of some things you
can expect from the new version:
* Automatically switch Bluetooth profile when using VoIP applications
* New module for prioritizing passt
Also remove the minimum supported PulseAudio version. The current
minimum libpulse version is 5.0, but if it's written on the page, the
information will inevitably get out of date sooner or later.
---
doc/README.html.in | 6 +++---
1 file changed, 3 insertions(+), 3 deletions(-)
diff --git a/doc/
On Tue, 2015-07-28 at 09:24 +0200, Peter Mattern wrote:
> Stumbled upon two more things in the meantime:
> Link "PulseAudio" in section "Overview" points to the downloads, the one
> in "Requirements" to PulseAudio's homepage. This should be the other way
> around.
> Link "gtkmm" in section "Requi
On Thu, 2017-01-12 at 21:56 -0600, Hajime Fujita wrote:
> On Jan 12, 2017, at 7:46 PM, Tanu Kaskinen wrote:
> > And so ends this journey of several years... I pushed these patches to
> > the "next" branch now. Thank you all for your work and patience!
>
> Awesome!
On Thu, 2017-01-12 at 22:41 +0100, Rikard Söderström wrote:
> ---
> man/pulse-daemon.conf.5.xml.in | 2 +-
> 1 file changed, 1 insertion(+), 1 deletion(-)
>
> diff --git a/man/pulse-daemon.conf.5.xml.in b/man/pulse-daemon.conf.5.xml.in
> index ff5a2936..1c0de9e4 100644
> --- a/man/pulse-daemon.co
On Tue, 2017-01-10 at 22:23 -0600, Hajime Fujita wrote:
> Hi Tanu and Anton,
>
> > On Jan 10, 2017, at 5:57 PM, Tanu Kaskinen wrote:
> >
> > On Thu, 2017-01-05 at 22:34 +0100, Anton Lundin wrote:
> > > On 06 November, 2016 - Hajime Fujita wrote:
> > >
On Wed, 2017-01-11 at 15:19 +0500, Alexander E. Patrakov wrote:
> Yes, I see the contradiction. Thanks for noticing it. Indeed, I have
> misremembered the forum post that prompted me to investigate the bug.
>
> In that forum post, the microphone actually worked, but the speakers
> didn't, because
BugLink: https://bugs.archlinux.org/task/52484
---
src/modules/module-zeroconf-publish.c | 4 +++-
1 file changed, 3 insertions(+), 1 deletion(-)
diff --git a/src/modules/module-zeroconf-publish.c
b/src/modules/module-zeroconf-publish.c
index e9710292a..482881c5b 100644
--- a/src/modules/module-
On Wed, 2017-01-11 at 17:22 +0530, Arun Raghavan wrote:
> On Wed, 11 Jan 2017, at 12:04 PM, Tanu Kaskinen wrote:
> > The uninitialized error struct might be the cause for the crash
> > reported here: https://bugs.archlinux.org/task/52484
> >
> > ---
> > src/modu
---
Makefile.am | 1 +
1 file changed, 1 insertion(+)
diff --git a/Makefile.am b/Makefile.am
index 13bc469db..14054acc4 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -16,6 +16,7 @@
ACLOCAL_AMFLAGS = -I m4
EXTRA_DIST = \
+ AGPL \
bootstrap.sh \
git-version-gen \
L
On Wed, 2017-01-11 at 17:44 +0530, Arun Raghavan wrote:
> On Wed, 11 Jan 2017, at 03:45 AM, Tanu Kaskinen wrote:
> > Thanks for the patch!
> >
> > On Thu, 2017-01-05 at 20:13 +0100, Steffen Pfendtner wrote:
> > > Subject: [PATCH] added two new commands to native API
The uninitialized error struct might be the cause for the crash
reported here: https://bugs.archlinux.org/task/52484
---
src/modules/module-zeroconf-publish.c | 2 ++
1 file changed, 2 insertions(+)
diff --git a/src/modules/module-zeroconf-publish.c
b/src/modules/module-zeroconf-publish.c
index
On Mon, 2017-01-09 at 11:57 -0300, Felipe Sateler wrote:
> On 8 January 2017 at 17:05, Tanu Kaskinen wrote:
> > Hi all,
> >
> > The first draft of the 10.0 release notes is now complete:
> > https://www.freedesktop.org/wiki/Software/PulseAudio/Notes/10.0/
> >
On Tue, 2017-01-10 at 04:35 +0500, Alexander E. Patrakov wrote:
> 2017-01-09 1:05 GMT+05:00 Tanu Kaskinen :
> > Hi all,
> >
> > The first draft of the 10.0 release notes is now complete:
> > https://www.freedesktop.org/wiki/Software/PulseAudio/Notes/10.0/
> >
>
On Thu, 2017-01-05 at 22:34 +0100, Anton Lundin wrote:
> On 06 November, 2016 - Hajime Fujita wrote:
>
> > This patch set adds a support for UDP version of RAOP (so called
> > raop2). Most of the RAOP devices (e.g. AppleTV, AirportExpress,
> > third party AV receivers) today use UDP version, so th
Thanks for the patch!
On Thu, 2017-01-05 at 20:13 +0100, Steffen Pfendtner wrote:
> Subject: [PATCH] added two new commands to native API to control the combine
> sink slaves after the combine sink has been created
There's a misunderstanding: you edited the command line interface, not
the native
On Wed, 2017-01-04 at 11:55 -0500, da...@mandelberg.org wrote:
> > So the suggestion here is to rename the
> > flag and invert its meaning, so that we can keep it off by default and
> > thus have PulseAudio fill all the output channels by default, like it
> > does now.
>
> Done.
Thanks a lot for
Hi all,
The first draft of the 10.0 release notes is now complete:
https://www.freedesktop.org/wiki/Software/PulseAudio/Notes/10.0/
If you think something should be added or removed or changed, please
reply to this mail or just edit the wiki.
--
Tanu
https://www.patreon.com/tanuk
_
On Thu, 2017-01-05 at 19:12 -0300, Gustavo Duarte wrote:
> Hi all,
>
> We are facing a problem with audio intput (mic), on a laptop with sound
> device ALC283.
>
> When the laptop boots connected to power line, all works fine.
>
> When the laptop boots without power line connection, sound input
On Wed, 2017-01-04 at 19:57 +0100, Linux User #330250 wrote:
> The one thing that remains unclear to me so far is how it autoloads in
> the first place, because I cannot find any raop modules in the config
> files in /etc/pulse. I assume that module-raop-discover does the magic,
> but it does it
On Wed, 2017-01-04 at 16:19 +, Marc Warne wrote:
> Hi,
>
> I have a PC that just has its own internal sound card, nothing special.
> However, this card (HDA Intel) has three output ports: analogue, HDMI
> and optical.
>
> I have been trying to find a way to get output on both the analogue a
On Sun, 2017-01-01 at 19:30 +0100, Linux User #330250 wrote:
> Hello!
>
>
> I have an Apple Airport Express (1st version) and wanted to finally use
> its RAOP function in Linux. But I have a couple of problems with it and
> don't know how to get past them... so I thought, I'll ask for help.
>
On Sun, 2017-01-01 at 09:20 -0500, Robert Drury wrote:
> pulseaudio 1.1 - Is there a way to set in a configuration file the
> volume that I want for an application when it starts up?
Unfortunately not.
--
Tanu
https://www.patreon.com/tanuk
___
pulsea
On Mon, 2016-05-23 at 16:57 +0200, Pierre Ossman wrote:
> I have a bunch of patches waiting in bugzilla to improve the new tunnel
> modules. We've been running with these patches for one release now (6
> months) and this makes the new tunnel modules work at least as well as
> the old one.
>
> I'm
On Tue, 2017-01-03 at 15:44 +0100, Georg Chini wrote:
> Hi Tanu,
>
> finally I am finding the time and motivation to continue on the loopback
> patches.
Great!
> Sorry for the long delay, I hope you still remember some of it. I have a few
> questions regarding the thread separation. You write (
On Fri, 2016-12-30 at 12:05 +0900, Takashi Sakamoto wrote:
> In alsa-lib, snd_pcm_hw_params() internally calls snd_pcm_prepare(), thus
> user space applications have no need to call snd_pcm_prepare() after calls
> of snd_pcm_hw_params(). An explicit calls of snd_pcm_prepare() is expected
> in a cas
On Tue, 2017-01-03 at 10:57 -0600, Andrew Eikum wrote:
> On Tue, Jan 03, 2017 at 06:29:39PM +0200, Tanu Kaskinen wrote:
> > The first release candidate for PulseAudio 10.0 is now out. As always,
> > the purpose of the release candidate is to get some testing done before
> >
On Thu, 2016-12-15 at 12:34 +0530, Renjith Thomas wrote:
> Issue: When HFP/HSP profile is used with certain BT chipsets, the
> audio sounds heavily distorted, with very slow playback full of noise.
> During recording, the samples are dropped and it distorts the recorded
> audio samples.
>
> The ro
Hi all!
The first release candidate for PulseAudio 10.0 is now out. As always,
the purpose of the release candidate is to get some testing done before
the final version is released. So, please test this on your own
machine, and if you're a distro maintainer and you have some suitable
channel for p
On Mon, 2017-01-02 at 10:19 +0100, Pierre Ossman wrote:
> On 31/12/16 17:26, Ahmed S. Darwish wrote:
> > On Fri, Dec 30, 2016 at 05:52:36PM +0200, Tanu Kaskinen wrote:
> > > diff --git a/src/tests/memblockq-test.c b/src/tests/memblockq-test.c
> > > index fc83d99..2a9
On Sun, 2017-01-01 at 09:02 +0200, Ahmed S. Darwish wrote:
> On Sun, Jan 01, 2017 at 04:17:16AM +0200, Tanu Kaskinen wrote:
> > On Sat, 2016-12-31 at 18:26 +0200, Ahmed S. Darwish wrote:
> > > Hi!
> > >
> > > On Fri, Dec 30, 2016 at 05:52:36PM +0200, Tanu K
On Sat, 2016-12-31 at 18:26 +0200, Ahmed S. Darwish wrote:
> Hi!
>
> On Fri, Dec 30, 2016 at 05:52:36PM +0200, Tanu Kaskinen wrote:
> > The intuitive meaning of "missing" would be the difference between
> > tlength and the current queue length, and that&#
The function isn't used anywhere else than memblockq-test. Also, the
function is confusing, because it defines "missing" differently than
pa_memblockq_pop_missing(). pa_memblockq_missing() calculated the
missing amount like this:
missing = tlength - length,
where "length" is the current queue
The intuitive meaning of "missing" would be the difference between
tlength and the current queue length, and that's how memblockq-test
assumed pa_memblockq_pop_missing() to define the term "missing", but
that was an incorrect assumption, causing the last
pa_memblockq_pop_missing() return value asse
memblockq-test started failing after commit 5d3d4f597 (Revert
"memblockq: remove internal "missing" state variable"). In this case the
bug was in the test.
Tanu Kaskinen (2):
memblockq: remove pa_memblockq_missing()
memblockq-test: fix incorrect assumption of pa_m
On Wed, 2016-12-28 at 19:21 +0200, Ahmed S. Darwish wrote:
> On Wed, Dec 28, 2016 at 04:09:54PM +0200, Tanu Kaskinen wrote:
> > This reverts commit 74251f07864c63439ea847d7287024ac54578d64.
> >
> > The reverted commit was not intended to make any behavioral changes, but
>
---
AGPL| 661
LICENSE | 4 +
2 files changed, 665 insertions(+)
create mode 100644 AGPL
diff --git a/AGPL b/AGPL
new file mode 100644
index 000..dba13ed
--- /dev/null
+++ b/AGPL
@@ -0,0 +1,661 @@
+GNU
We're trying to relicense qpaeq to LGPL, but until that is successfully
completed, we should document the qpaeq license properly.
Also, the opening sentence of LICENSE was false.
Tanu Kaskinen (2):
LICENSE: add a clarification
LICENSE: add a note about qpaeq being licensed under AGPL
---
LICENSE | 4 ++--
1 file changed, 2 insertions(+), 2 deletions(-)
diff --git a/LICENSE b/LICENSE
index 6932317..7f98019 100644
--- a/LICENSE
+++ b/LICENSE
@@ -1,5 +1,5 @@
-All PulseAudio source files are licensed under the GNU Lesser General Public
-License. (see file LGPL for details)
+All P
On Wed, 2016-12-28 at 19:20 +0200, Ahmed S. Darwish wrote:
> Welcome back ;-)
>
> On Wed, Dec 28, 2016 at 04:09:55PM +0200, Tanu Kaskinen wrote:
> >
> > Previously pacat wrote at most pa_stream_writable_size() bytes at a
> > time, now with this patch it can write mor
On Tue, 2016-12-27 at 16:33 +, Andrea A wrote:
> I'm trying to write a pulseaudio module for an equalizer, I'm using as
> examples others modules on pulseaudio source code and this is my source code:
> https://github.com/andrea993/audioeqpro/blob/master/pulsemodule/module-test.c
>
> Now I'm t
Previously pacat wrote at most pa_stream_writable_size() bytes at a
time, now with this patch it can write more than that if there's more
input data available. Writing in bigger chunks is potentially a bit more
efficient.
---
src/utils/pacat.c | 4 ++--
1 file changed, 2 insertions(+), 2 deletions
stream to stop
if the the server doesn't have a fix applied.
[1]
https://lists.freedesktop.org/archives/pulseaudio-discuss/2016-December/027273.html
Tanu Kaskinen (2):
Revert "memblockq: remove internal "missing" state variable"
pacat: write as much as possible in one go
This reverts commit 74251f07864c63439ea847d7287024ac54578d64.
The reverted commit was not intended to make any behavioral changes, but
it broke at least the case where a client writes more data than the
server has requested.
BugLink: https://bugs.freedesktop.org/show_bug.cgi?id=99211
---
src/pul
On Tue, 2016-12-27 at 11:27 +0100, Pierre Ossman wrote:
> Ping!
>
> Could we get these in for 10.0? :)
I'm sure they won't be in the first release candidate, which I intend
to publish as soon as we have a fix for the streaming regression. As
for subsequent release candidates, maybe we can add the
On Tue, 2016-12-27 at 11:25 +0100, Pierre Ossman wrote:
> On 26/12/16 06:31, Ahmed S. Darwish wrote:
> >
> > But bq->requested has different semantics upon write index change
> > before and after the same commit:
> >
>
> More cleanup and comments of that code is probably needed to make it
> man
On Thu, 2016-12-22 at 18:14 +, peterte...@yahoo.com wrote:
> Hi,
>
> thanks for your response. That really helps.
>
> Now I call with "control=Master" and pulseaudio changes the alsa Master
> volume.
> PCM Playback Volume is not touched by pulseaudio.
>
> > Pulseaudio doesn't currently have
On Wed, 2016-12-21 at 14:54 +, peterte...@yahoo.com wrote:
> Hi,
>
> I'd like to provide an ALSA driver which is designed to support
> pulseaudio by default (if that is possible).
> The driver works fine with aplay and alsamixer. It provides 6
> channels and volume controls.
> - One 'Master Pl
On Tue, 2016-12-20 at 09:07 +, Ahmed S. Darwish wrote:
> Current pacat code reads whatever available from STDIN and writes
> it directly to the playback stream. A minimal buffer is created
> for each read operation; no further reads are then allowed unless
> earlier read buffer has been fully c
On Thu, 2016-12-15 at 22:02 -0500, David Mandelberg wrote:
> Thanks for the pointers! I like your proposal at
> https://bugs.freedesktop.org/show_bug.cgi?id=62588#c6.
>
> Do you know what is needed in order to make progress on some sort of
> solution? Is more consensus needed? Or work on code? Som
On Sat, 2016-12-17 at 01:43 +0200, Ahmed S. Darwish wrote:
> # Note: Documentation of `pa_stream_begin_write(stream, &buf, &nbytes)'
> # suggests to pass nbytes = -1. Doing so though lends to a huge returned
> # nbytes values. Calling a pa_stream_write() with such values chokes the
> # stream: maki
The file was missing from release tarballs.
---
src/Makefile.am | 1 +
1 file changed, 1 insertion(+)
diff --git a/src/Makefile.am b/src/Makefile.am
index 2d5bdd4..b1cd99b 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -91,6 +91,7 @@ MODULE_LIBADD = $(AM_LIBADD) libpulsecore-@PA_MAJORMINOR
Hi all,
A bug was recently filed about the top-level LICENSE file not
mentioning the AGPL licensing of qpaeq:
https://bugs.freedesktop.org/show_bug.cgi?id=92802
I was surprised to find out that qpaeq is licensed under AGPL. While we
could simply fix the top-level LICENSE file, I would prefer to c
On Tue, 2016-12-13 at 14:39 +0200, Ahmed S. Darwish wrote:
> On Mon, Dec 12, 2016 at 3:22 PM, Tanu Kaskinen wrote:
> >
> > What's the status? Did you give the alternative fix a try? Or should we
> > go with the ringbuffer-based fix?
> >
>
> pa_stream_begin
On Thu, 2016-12-15 at 11:11 +, Jay Cotton wrote:
> Hi guys:
>
>
> I was importing pulseaudio v9.0 and I noticed that the LICENSE file is
> changed, it refers to
>
> LICENSE.WEBKIT.
>
> That file is not in the source distro, but I found the patch that is supposed
> to put it in the distro.
On Tue, 2016-12-13 at 00:08 +0100, Erwan JACQ wrote:
> Hi all,
>
> To answer your requests, I could add a new boolean option in the config
> file.
> If this option is False (default), nothing changes compared to now.
> If this option is True, available hardware rates are used for the stream
> op
On Sun, 2016-12-11 at 22:06 +0100, Pali Rohár wrote:
> On Thursday 03 November 2016 14:53:42 Felipe Sateler wrote:
> > Excellent. Having the first RC within november/early december makes
> > me think having version 10 (or at least a late RC + cherry-picked
> > fixes) will be the best choice.
>
> I
On Mon, 2016-11-28 at 19:23 +0200, Ahmed S. Darwish wrote:
> On Mon, Nov 28, 2016 at 09:04:01AM +0100, David Henningsson wrote:
> > Wouldn't it be better if we had something like:
>
> By better, you mean simpler?
>
> > 1) Call pa_stream_begin_write to get a buffer.
> > 2) If we have half a fram
On Sat, 2016-12-10 at 16:37 +0200, Tanu Kaskinen wrote:
> On Thu, 2016-12-08 at 21:48 +0100, Steffen Pfendtner wrote:
> > I would like to extend the module-combine-sink to dynamically add and
> > remove the slaves. Especially with pulseaudio-dlna this would be a great
> > ben
On Sat, 2016-12-10 at 16:37 +0200, Tanu Kaskinen wrote:
> On Thu, 2016-12-08 at 21:48 +0100, Steffen Pfendtner wrote:
> > I would like to extend the module-combine-sink to dynamically add and
> > remove the slaves. Especially with pulseaudio-dlna this would be a great
> > ben
On Thu, 2016-12-08 at 21:48 +0100, Steffen Pfendtner wrote:
> I would like to extend the module-combine-sink to dynamically add and
> remove the slaves. Especially with pulseaudio-dlna this would be a great
> benefit.
This would be a very good feature to have.
> My first intention was to abuse th
On Thu, 2016-12-08 at 15:43 +0530, Nishit Sharma wrote:
> On Thu, Dec 8, 2016 at 3:37 PM, Tanu Kaskinen wrote:
>
> > On Thu, 2016-12-08 at 15:28 +0530, Nishit Sharma wrote:
> > > Hi All,
> > >
> > > I am facing noise issues when playing through Speaker/H
On Thu, 2016-12-08 at 11:32 +0530, Arun Raghavan wrote:
>
> On Fri, 2 Dec 2016, at 05:31 PM, Tanu Kaskinen wrote:
> > On Sun, 2016-11-27 at 22:27 +0100, Erwan JACQ wrote:
> > > You can use the default-sample-rate/alternate-sample-rate config values
> > > as mini
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