David Henningsson | 29 Jan 16:26 2015

[PATCH] alsa-mixer: Make speaker unavailable when Line Out is plugged in

ALSA mutes speaker when Line Out is plugged in by default, so
we should follow that convention.

Signed-off-by: David Henningsson <david.henningsson <at> canonical.com>
 src/modules/alsa/mixer/paths/analog-output-speaker-always.conf | 8 ++++++++
 src/modules/alsa/mixer/paths/analog-output-speaker.conf        | 8 ++++++++
 2 files changed, 16 insertions(+)

diff --git a/src/modules/alsa/mixer/paths/analog-output-speaker-always.conf b/src/modules/alsa/mixer/paths/analog-output-speaker-always.conf
index a8e5d05..b6d9f94 100644
--- a/src/modules/alsa/mixer/paths/analog-output-speaker-always.conf
+++ b/src/modules/alsa/mixer/paths/analog-output-speaker-always.conf
 <at>  <at>  -33,6 +33,14  <at>  <at>  state.unplugged = unknown
 state.plugged = no
 state.unplugged = unknown

+[Jack Line Out]
+state.plugged = no
+state.unplugged = unknown
+[Jack Line Out Front]
+state.plugged = no
+state.unplugged = unknown
 [Element Hardware Master]
 switch = mute
 volume = merge
diff --git a/src/modules/alsa/mixer/paths/analog-output-speaker.conf b/src/modules/alsa/mixer/paths/analog-output-speaker.conf
index 2c51f6c..39193dd 100644
(Continue reading)

Tanu Kaskinen | 29 Jan 12:53 2015

[PATCH] alsa-mixer: Add lineout to surround21 mappings

This makes the analog-surround-21 mapping consistent with other
surround mappings. I'm not sure if this makes any practical
difference, though.
 src/modules/alsa/mixer/profile-sets/default.conf                   | 2 +-
 src/modules/alsa/mixer/profile-sets/force-speaker-and-int-mic.conf | 2 +-
 src/modules/alsa/mixer/profile-sets/force-speaker.conf             | 2 +-
 3 files changed, 3 insertions(+), 3 deletions(-)

diff --git a/src/modules/alsa/mixer/profile-sets/default.conf b/src/modules/alsa/mixer/profile-sets/default.conf
index bfb54c9..2d6720e 100644
--- a/src/modules/alsa/mixer/profile-sets/default.conf
+++ b/src/modules/alsa/mixer/profile-sets/default.conf
 <at>  <at>  -118,7 +118,7  <at>  <at>  priority = 10
 [Mapping analog-surround-21]
 device-strings = surround21:%f
 channel-map = front-left,front-right,lfe
-paths-output = analog-output analog-output-speaker
+paths-output = analog-output analog-output-lineout analog-output-speaker
 priority = 8
 direction = output

diff --git a/src/modules/alsa/mixer/profile-sets/force-speaker-and-int-mic.conf b/src/modules/alsa/mixer/profile-sets/force-speaker-and-int-mic.conf
index 9c57a20..41924f4 100644
--- a/src/modules/alsa/mixer/profile-sets/force-speaker-and-int-mic.conf
+++ b/src/modules/alsa/mixer/profile-sets/force-speaker-and-int-mic.conf
 <at>  <at>  -37,7 +37,7  <at>  <at>  priority = 10
 [Mapping analog-surround-21]
 device-strings = surround21:%f
 channel-map = front-left,front-right,lfe
(Continue reading)

David Henningsson | 28 Jan 23:14 2015

[RFC PATCH 0/6] LFE filter


Hui and I have been working on some LFE filter patches lately, and this
is our first draft for review/feedback.

First, I have greedily stolen the math from CRAS, because CRAS is BSD and
as I understand we don't have a problem with merging more liberal licenses.

The LFE filter is implemented in the resampler, which means it is done for
every sink-input rather than every sink - this might mean some additional
CPU processing if many different streams play back at the same time, but
putting it on the sink side would disable the possibility to mix a 2.0 stream
with a 2.1 stream.

The rewind part is very drafty and untested, and I'm not sure I choose the
best design here. But at least this is something that could act as base for

Also, while testing we noticed that flat volumes acts a bit weird, making it
impossible to have a high subwoofer volume level but at the same time low
front speaker volume, but I'm not sure if that's a bug or just the (ahem)
normal weirdness of flat volumes.

David Henningsson (4):
  lfe-filter: Import code from the Chrome OS audio server
  lfe-filter: Enable LFE filter in the resampler
  lfe-filter: Add rewind support
  resampler: Make some basic functions for rewinding

Hui Wang (2):
(Continue reading)

David Henningsson | 28 Jan 21:10 2015

[PATCH] module-device-restore: Fix memory leak in read_sink_format_reply

This function could sometimes leak a perportentry.

Signed-off-by: David Henningsson <david.henningsson <at> canonical.com>
 src/modules/module-device-restore.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/src/modules/module-device-restore.c b/src/modules/module-device-restore.c
index 09ba60a..70a6517 100644
--- a/src/modules/module-device-restore.c
+++ b/src/modules/module-device-restore.c
 <at>  <at>  -1021,6 +1021,7  <at>  <at>  static void read_sink_format_reply(struct userdata *u, pa_tagstruct *reply, pa_s
         PA_IDXSET_FOREACH(f, e->formats, idx) {
             pa_tagstruct_put_format_info(reply, f);
+        perportentry_free(e);


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Tanu Kaskinen | 28 Jan 15:36 2015

Patch review status wiki page updated

Patch review status updated:


* 2015-01-28
   * 22 patches are pending review.
   * The oldest pending patch is 192 days old.

* 2015-01-19
   * 26 patches are pending review.
   * The oldest pending patch is 183 days old.

* 2015-01-13
   * 21 patches are pending review.
   * The oldest pending patch is 177 days old.

* 2015-01-08
   * 26 patches are pending review.
   * The oldest pending patch is 172 days old.

* 2014-12-29
   * 22 patches are pending review.
   * The oldest pending patch is 162 days old.

* 2014-12-22
   * 23 patches are pending review.
   * The oldest pending patch is 155 days old.

* 2014-12-15
(Continue reading)

sean darcy | 27 Jan 19:41 2015

paplay : connection refused

On Fedora 20, I'm trying to play a bunch of opus files with opusdec,
which outputs to /dev/dsp. I tried padsp, but that doesn't work:

padsp opusdec test.opus
Decoding to 48000 Hz (2 channels)
Cannot open /dev/dsp: Connection refused

So to troubleshoot I tried:

paplay test.wav
Connection failure: Connection refused
pa_context_connect() failed: Connection refused

  pulseaudio is running:

  pacmd ls
  Memory blocks currently allocated: 1, size: 63.9 KiB.
  Memory blocks allocated during the whole lifetime: 40, size: 2.1 MiB.
  Memory blocks imported from other processes: 0, size: 0 B.
  Memory blocks exported to other processes: 0, size: 0 B.
  Total sample cache size: 0 B.
  Default sample spec: s16le 2ch 44100Hz
  Default channel map: front-left,front-right
  Default sink name: alsa_output.pci-0000_00_14.2.analog-stereo
  Default source name: alsa_input.pci-0000_00_14.2.analog-stereo
  Memory blocks of type POOL: 1 allocated/1 accumulated.
  Memory blocks of type POOL_EXTERNAL: 0 allocated/0 accumulated.
  Memory blocks of type APPENDED: 0 allocated/0 accumulated.
  Memory blocks of type USER: 0 allocated/0 accumulated.
  Memory blocks of type FIXED: 0 allocated/39 accumulated.
(Continue reading)

Mario Sanchez Prada | 27 Jan 13:27 2015

No sound on Bluetooth headset with HSP/HFP profiles and RTL8723BE chipset


I'm seeing and strange issue happening with PulseAudio and the
integrated Bluetooth card of my barebone computer and, after a couple of
days debugging this, I still could not figure out what the problem is,
so I thought I would ask for some help here, in case someone was able to
provide some help.

As a bit of extra context, I'm trying to use a Plantronics M50 headset
with the BT card coming with the RTL8723BE chipset of my Gygabyte Brix
2807 [1] and Fedora 21's package for PulseAudio 5.0, which backports the
patches needed to add back support for Bluez5 and headset profiles (see

The problem I'm seeing is that, after pairing and connecting my BT
headset to the computer I can neither get any sound out of the headphone
nor use the mic, even if I explicitly select them in the "sound" panel
of gnome-control-center, as long as the HSP/HFP profile is selected. If
I select the A2DP profile, though, I do get it to work as an output
device, but then I get silence again if I switch back to the HSP/HFP

Additionally, I've observed that PA will be basically sleeping while
playing a .wav file with paplay for as long as I keep the HSP/HFP
profile, resuming the play once I switch back to another output device
(e.g. HDMI). See attached the output of `thread apply all bt` after
manually breaking the sleeping execution in gdb, where it seems all the
threads are sleeping, hence the silence I guess.

Now, using an external BT usb dongle in my barebone instead of the
(Continue reading)

Alexander E. Patrakov | 25 Jan 19:52 2015

Some bogus ports


I have noticed some ports in my "pactl list cards" output, that I think 
should not be there.

1. iec958-stereo-input on my webcam
2. analog-output on my analog card (because this should be already 
covered by analog-output-lineout)

Attached are the "pactl list cards" output and pulseaudio verbose log.

Please note: I have some jack overrides installed in order to hide 
"front panel" jacks that my motherboard has, but the case doesn't have 
(so I can't connect anything to them). Besides, if I don't hide these 
jacks, they sometimes pick up spurious jack-detection events.

All of that is with today's git master, and linux-3.19.0-rc4


Alexander E. Patrakov
Card #0
	Name: alsa_card.pci-0000_00_03.0
	Driver: module-alsa-card.c
	Owner Module: 5
		alsa.card = "0"
		alsa.card_name = "HDA Intel HDMI"
		alsa.long_card_name = "HDA Intel HDMI at 0xf7e34000 irq 32"
(Continue reading)

Ville Skyttä | 25 Jan 12:42 2015

[PATCH] build-sys: Install bash completion to where bash-completion.pc says

Fall back to the previous /etc/bash_completion.d dir on failures
(either old bash completion or not installed).

Signed-off-by: Ville Skyttä <ville.skytta <at> iki.fi>
 Makefile.am  | 2 +-
 configure.ac | 6 ++++++
 2 files changed, 7 insertions(+), 1 deletion(-)

diff --git a/Makefile.am b/Makefile.am
index 75b784f..b39fc41 100644
--- a/Makefile.am
+++ b/Makefile.am
 <at>  <at>  -59,7 +59,7  <at>  <at>  endif
 cmakedir = $(libdir)/cmake/PulseAudio
 cmake_DATA = PulseAudioConfig.cmake PulseAudioConfigVersion.cmake

+bashcompletiondir= <at> bashcompletiondir <at> 
 dist_bashcompletion_DATA = shell-completion/bash/pulseaudio

diff --git a/configure.ac b/configure.ac
index 3c71472..a79dff8 100644
--- a/configure.ac
+++ b/configure.ac
 <at>  <at>  -1421,6 +1421,11  <at>  <at>  AC_ARG_WITH(


(Continue reading)

Dâniel Fraga | 23 Jan 21:02 2015

Bug: 2 seconds lag after application finished playing audio

	I'm using Pulseaudio 5.0 and all applications show a delay of
about 2 seconds after the audio finished playing.

	I searched for Google and it's exactly like what was reported

pa_simple_drain() takes over 2 seconds to complete!

	What you need to debug this?


	PS: I reported here, but nobody replied:



Linux 3.18.3: Diseased Newt
Bitcoin: 12H6661yoLDUZaYPdah6urZS5WiXwTAUgL

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Michael DePaulo | 22 Jan 03:42 2015

libsndfile CVE-2014-9496

Hi PulseAudio devs,

Can someone tell me whether PulseAudio can actually be affected by the
libsndfile vulnerability CVE-2014-9496?

"It looks like the affected code is in reading SD2 (Sound Designer II)
files and writing AIFF files".

I am thinking the answer is "no".

Currently I am maintaining both X2Go Client for Windows[1] and my
unofficial PulseAudio builds for Windows[2][3]. X2Go Client for
Windows bundles the PulseAudio builds. So I am trying to figure out
whether I urgently need to update them with the patched libsndfile


[1] http://wiki.x2go.org/doku.php/doc:release-notes-mswin
[2] https://build.opensuse.org/project/show/home:mikedep333:branches:home:mkbosmans:mingw32:pulseaudio
[3] http://code.x2go.org/releases/binary-win32/3rd-party/pulse/
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