[pulseaudio-commits] [SCM] PulseAudio Sound Server branch, master, updated. v0.9.19-402-g323f0ad
Lennart Poettering
gitmailer-noreply at 0pointer.de
Sun Feb 21 11:11:10 PST 2010
This is an automated email from the git hooks/post-receive script. It was
generated because of a push to the "PulseAudio Sound Server" repository.
The master branch has been updated
from f6f957b53cc8ed945364ef5d51546f59a4c653be (commit)
- Log -----------------------------------------------------------------
323f0ad build-sys: fix check for pthread_setaffinity_np()
2b2149d build-sys: Add missing profile and alsa-mixer/paths to src/Makefile.am
609a005 pacmd: don't enter busy loop when reading from stdin very early
83eb8af daemon: first take name on the bus, then return in starter process
ea29b11 scache: when playing a sample from the cache make sure not queue them up when the sink is suspended
3f1c90b sample-cache: use the sample name as unmodified fallback in the properties when playing a sample from the cache
857ff2c stream-restore: be a little bit more verbose why we don't reastore a sink
7098d60 proplist: explicitly mention a role 'test'
534def2 vala: fix wrapping of port setting calls
27427c1 channelmap: Use "Subwoofer" as pretty name for LFE
c25665e vala: fix definition of the GLib mainloop adapter
57e0e2c vala: fix definition of INVALID_INDEX
c2079d7 shm: don't complain about missing SHM segments
f9f2d6f pacmd: store away fd type
d57ba82 pacat: always fully fulfill write requests
b9bcc7c fdsem: be more verbose when reading from eventfd fails.
ac76dbd dbus: first restart timer, then dispatch it
69ddfaf padsp: emulate /dev/audio, too
80093c9 build-sys: add gobject to build dependencies
abfd2aa alsa: use default output port names
dfa00bf alsa: cover "Internal Mic" elements
67e3925 alsa: cover 'Shared Mic/Line in', 'Analog Source'
14e4369 alsa: cover "Desktop Speaker" mixer elements
-----------------------------------------------------------------------
Summary of changes:
configure.ac | 3 +-
src/Makefile.am | 3 +
src/daemon/main.c | 18 +-
src/modules/alsa/alsa-mixer.c | 3 +-
src/modules/alsa/mixer/paths/analog-input-aux.conf | 4 +
src/modules/alsa/mixer/paths/analog-input-fm.conf | 4 +
...put-mic.conf => analog-input-internal-mic.conf} | 8 +-
.../alsa/mixer/paths/analog-input-linein.conf | 4 +
.../alsa/mixer/paths/analog-input-mic-line.conf | 4 +
src/modules/alsa/mixer/paths/analog-input-mic.conf | 4 +
.../alsa/mixer/paths/analog-input-tvtuner.conf | 4 +
.../alsa/mixer/paths/analog-input-video.conf | 4 +
src/modules/alsa/mixer/paths/analog-input.conf | 3 +
.../alsa/mixer/paths/analog-input.conf.common | 25 +++
...ker.conf => analog-output-desktop-speaker.conf} | 15 +-
.../mixer/paths/analog-output-headphones-2.conf | 5 +
.../alsa/mixer/paths/analog-output-headphones.conf | 5 +
.../mixer/paths/analog-output-lfe-on-mono.conf | 4 +
.../alsa/mixer/paths/analog-output-mono.conf | 4 +
.../alsa/mixer/paths/analog-output-speaker.conf | 5 +
src/modules/alsa/mixer/paths/analog-output.conf | 4 +
src/modules/alsa/mixer/profile-sets/default.conf | 14 +-
src/modules/module-stream-restore.c | 2 +-
src/pulse/channelmap.c | 2 +-
src/pulse/proplist.h | 2 +-
src/pulsecore/core-scache.c | 10 +-
src/pulsecore/dbus-util.c | 29 ++--
src/pulsecore/fdsem.c | 41 ++++-
src/pulsecore/play-memblockq.c | 7 +-
src/pulsecore/play-memblockq.h | 18 +-
src/pulsecore/play-memchunk.c | 3 +-
src/pulsecore/play-memchunk.h | 15 +-
src/pulsecore/shm.c | 2 +-
src/tests/once-test.c | 2 +-
src/utils/pacat.c | 47 +++--
src/utils/pacmd.c | 187 +++++++++++---------
src/utils/padsp.c | 55 ++----
vala/libpulse.vapi | 10 +-
38 files changed, 366 insertions(+), 213 deletions(-)
copy src/modules/alsa/mixer/paths/{analog-input-mic.conf => analog-input-internal-mic.conf} (92%)
copy src/modules/alsa/mixer/paths/{analog-output-speaker.conf => analog-output-desktop-speaker.conf} (86%)
-----------------------------------------------------------------------
commit 14e4369bee6a808ca1f57945e7fe7165f7f18a9e
Author: Lennart Poettering <lennart at poettering.net>
Date: Mon Feb 15 22:23:39 2010 +0100
alsa: cover "Desktop Speaker" mixer elements
As exposed by really old Microsoft USB sound systems
diff --git a/src/modules/alsa/alsa-mixer.c b/src/modules/alsa/alsa-mixer.c
index 8bbd1f5..cb814af 100644
--- a/src/modules/alsa/alsa-mixer.c
+++ b/src/modules/alsa/alsa-mixer.c
@@ -1784,7 +1784,8 @@ static int path_verify(pa_alsa_path *p) {
{ "analog-output-lfe-on-mono", N_("Analog Output (LFE)") },
{ "analog-output-mono", N_("Analog Mono Output") },
{ "analog-output-headphones-2", N_("Analog Headphones 2") },
- { "analog-output-speaker", N_("Analog Speaker") }
+ { "analog-output-speaker", N_("Analog Speakers") }
+ { "analog-output-desktop-speaker", N_("Analog Speakers 2") }
};
pa_alsa_element *e;
diff --git a/src/modules/alsa/mixer/paths/analog-output-speaker.conf b/src/modules/alsa/mixer/paths/analog-output-desktop-speaker.conf
similarity index 88%
copy from src/modules/alsa/mixer/paths/analog-output-speaker.conf
copy to src/modules/alsa/mixer/paths/analog-output-desktop-speaker.conf
index aea7853..9f18b25 100644
--- a/src/modules/alsa/mixer/paths/analog-output-speaker.conf
+++ b/src/modules/alsa/mixer/paths/analog-output-desktop-speaker.conf
@@ -14,7 +14,7 @@
# along with PulseAudio; if not, write to the Free Software Foundation,
# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
-; Path for mixers that have a 'Speaker' control
+; Path for mixers that have a 'Desktop Speaker' control
;
; See analog-output.conf.common for an explanation on the directives
@@ -37,9 +37,9 @@ override-map.2 = all-left,all-right
switch = off
volume = off
-; This profile path is intended to control the speaker, not the
-; headphones. But it should not hurt if we leave the headphone jack
-; enabled nonetheless.
+; This profile path is intended to control the desktop speaker, not
+; the headphones. But it should not hurt if we leave the headphone
+; jack enabled nonetheless.
[Element Headphone]
switch = mute
volume = zero
@@ -49,6 +49,10 @@ switch = mute
volume = zero
[Element Speaker]
+switch = off
+volume = off
+
+[Element Desktop Speaker]
required = any
switch = mute
volume = merge
diff --git a/src/modules/alsa/mixer/paths/analog-output-headphones-2.conf b/src/modules/alsa/mixer/paths/analog-output-headphones-2.conf
index f2fd31c..580c798 100644
--- a/src/modules/alsa/mixer/paths/analog-output-headphones-2.conf
+++ b/src/modules/alsa/mixer/paths/analog-output-headphones-2.conf
@@ -55,6 +55,10 @@ override-map.2 = all-left,all-right
switch = off
volume = off
+[Element Desktop Speaker]
+switch = off
+volume = off
+
[Element Front]
switch = off
volume = off
diff --git a/src/modules/alsa/mixer/paths/analog-output-headphones.conf b/src/modules/alsa/mixer/paths/analog-output-headphones.conf
index 2131cfe..79ebc7f 100644
--- a/src/modules/alsa/mixer/paths/analog-output-headphones.conf
+++ b/src/modules/alsa/mixer/paths/analog-output-headphones.conf
@@ -55,6 +55,10 @@ volume = zero
switch = off
volume = off
+[Element Desktop Speaker]
+switch = off
+volume = off
+
[Element Front]
switch = off
volume = off
diff --git a/src/modules/alsa/mixer/paths/analog-output-lfe-on-mono.conf b/src/modules/alsa/mixer/paths/analog-output-lfe-on-mono.conf
index 0a43e27..67ee32f 100644
--- a/src/modules/alsa/mixer/paths/analog-output-lfe-on-mono.conf
+++ b/src/modules/alsa/mixer/paths/analog-output-lfe-on-mono.conf
@@ -58,6 +58,10 @@ volume = merge
override-map.1 = all
override-map.2 = all-left,all-right
+[Element Desktop Speaker]
+switch = off
+volume = off
+
[Element Front]
switch = off
volume = off
diff --git a/src/modules/alsa/mixer/paths/analog-output-mono.conf b/src/modules/alsa/mixer/paths/analog-output-mono.conf
index 542edc4..13a2d6a 100644
--- a/src/modules/alsa/mixer/paths/analog-output-mono.conf
+++ b/src/modules/alsa/mixer/paths/analog-output-mono.conf
@@ -55,6 +55,10 @@ volume = merge
override-map.1 = all
override-map.2 = all-left,all-right
+[Element Desktop Speaker]
+switch = off
+volume = off
+
[Element Front]
switch = off
volume = off
diff --git a/src/modules/alsa/mixer/paths/analog-output-speaker.conf b/src/modules/alsa/mixer/paths/analog-output-speaker.conf
index aea7853..b133a2e 100644
--- a/src/modules/alsa/mixer/paths/analog-output-speaker.conf
+++ b/src/modules/alsa/mixer/paths/analog-output-speaker.conf
@@ -55,6 +55,10 @@ volume = merge
override-map.1 = all
override-map.2 = all-left,all-right
+[Element Desktop Speaker]
+switch = off
+volume = off
+
[Element Front]
switch = mute
volume = merge
diff --git a/src/modules/alsa/mixer/paths/analog-output.conf b/src/modules/alsa/mixer/paths/analog-output.conf
index d7c1223..50fc88e 100644
--- a/src/modules/alsa/mixer/paths/analog-output.conf
+++ b/src/modules/alsa/mixer/paths/analog-output.conf
@@ -53,6 +53,10 @@ volume = zero
switch = mute
volume = off
+[Element Desktop Speaker]
+switch = mute
+volume = off
+
[Element Front]
switch = mute
volume = merge
diff --git a/src/modules/alsa/mixer/profile-sets/default.conf b/src/modules/alsa/mixer/profile-sets/default.conf
index 046938f..5d352f1 100644
--- a/src/modules/alsa/mixer/profile-sets/default.conf
+++ b/src/modules/alsa/mixer/profile-sets/default.conf
@@ -62,42 +62,42 @@ auto-profiles = yes
[Mapping analog-mono]
device-strings = hw:%f
channel-map = mono
-paths-output = analog-output analog-output-speaker analog-output-headphones analog-output-headphones-2 analog-output-mono analog-output-lfe-on-mono
+paths-output = analog-output analog-output-speaker analog-output-desktop-speaker analog-output-headphones analog-output-headphones-2 analog-output-mono analog-output-lfe-on-mono
paths-input = analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line
priority = 1
[Mapping analog-stereo]
device-strings = front:%f hw:%f
channel-map = left,right
-paths-output = analog-output analog-output-speaker analog-output-headphones analog-output-headphones-2 analog-output-mono analog-output-lfe-on-mono
+paths-output = analog-output analog-output-speaker analog-output-desktop-speaker analog-output-headphones analog-output-headphones-2 analog-output-mono analog-output-lfe-on-mono
paths-input = analog-input analog-input-mic analog-input-linein analog-input-aux analog-input-video analog-input-tvtuner analog-input-fm analog-input-mic-line
priority = 10
[Mapping analog-surround-40]
device-strings = surround40:%f
channel-map = front-left,front-right,rear-left,rear-right
-paths-output = analog-output analog-output-speaker analog-output-lfe-on-mono
+paths-output = analog-output analog-output-speaker analog-output-desktop-speaker analog-output-lfe-on-mono
priority = 7
direction = output
[Mapping analog-surround-41]
device-strings = surround41:%f
channel-map = front-left,front-right,rear-left,rear-right,lfe
-paths-output = analog-output analog-output-speaker analog-output-lfe-on-mono
+paths-output = analog-output analog-output-speaker analog-output-desktop-speaker analog-output-lfe-on-mono
priority = 8
direction = output
[Mapping analog-surround-50]
device-strings = surround50:%f
channel-map = front-left,front-right,rear-left,rear-right,front-center
-paths-output = analog-output analog-output-speaker analog-output-lfe-on-mono
+paths-output = analog-output analog-output-speaker analog-output-desktop-speaker analog-output-lfe-on-mono
priority = 7
direction = output
[Mapping analog-surround-51]
device-strings = surround51:%f
channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe
-paths-output = analog-output analog-output-speaker analog-output-lfe-on-mono
+paths-output = analog-output analog-output-speaker analog-output-desktop-speaker analog-output-lfe-on-mono
priority = 8
direction = output
@@ -105,7 +105,7 @@ direction = output
device-strings = surround71:%f
channel-map = front-left,front-right,rear-left,rear-right,front-center,lfe,side-left,side-right
description = Analog Surround 7.1
-paths-output = analog-output analog-output-speaker analog-output-lfe-on-mono
+paths-output = analog-output analog-output-speaker analog-output-desktop-speaker analog-output-lfe-on-mono
priority = 7
direction = output
commit 67e3925795ad939f2c9acb3aa93122a18672fade
Author: Lennart Poettering <lennart at poettering.net>
Date: Mon Feb 15 23:39:12 2010 +0100
alsa: cover 'Shared Mic/Line in', 'Analog Source'
https://bugzilla.redhat.com/show_bug.cgi?id=558638
diff --git a/src/modules/alsa/mixer/paths/analog-input.conf.common b/src/modules/alsa/mixer/paths/analog-input.conf.common
index fbdc8fb..66c3dda 100644
--- a/src/modules/alsa/mixer/paths/analog-input.conf.common
+++ b/src/modules/alsa/mixer/paths/analog-input.conf.common
@@ -277,6 +277,31 @@ name = input-microphone
[Option Digital Input Source:Digital Mic 2]
name = input-microphone
+;;; 'Analog Source'
+
+[Element Analog Source]
+enumeration = select
+
+[Option Analog Source:Mic]
+name = input-microphone
+
+[Option Analog Source:Line in]
+name = input-linein
+
+[Option Analog Source:Aux]
+name = input
+
+;;; 'Shared Mic/Line in'
+
+[Element Shared Mic/Line in]
+enumeration = select
+
+[Option Shared Mic/Line in:Mic in]
+name = input-microphone
+
+[Option Shared Mic/Line in:Line in]
+name = input-linein
+
;;; Various Boosts
[Element Capture Boost]
commit dfa00bf8f846d0e1ea0f47b09d114768a781abb7
Author: Lennart Poettering <lennart at poettering.net>
Date: Tue Feb 16 00:58:49 2010 +0100
alsa: cover "Internal Mic" elements
https://bugzilla.redhat.com/show_bug.cgi?id=562216
diff --git a/src/modules/alsa/mixer/paths/analog-input-aux.conf b/src/modules/alsa/mixer/paths/analog-input-aux.conf
index db78eb4..3a7cb7b 100644
--- a/src/modules/alsa/mixer/paths/analog-input-aux.conf
+++ b/src/modules/alsa/mixer/paths/analog-input-aux.conf
@@ -32,6 +32,10 @@ override-map.2 = all-left,all-right
switch = off
volume = off
+[Element Internal Mic]
+switch = off
+volume = off
+
[Element Line]
switch = off
volume = off
diff --git a/src/modules/alsa/mixer/paths/analog-input-fm.conf b/src/modules/alsa/mixer/paths/analog-input-fm.conf
index baf674a..7f150e3 100644
--- a/src/modules/alsa/mixer/paths/analog-input-fm.conf
+++ b/src/modules/alsa/mixer/paths/analog-input-fm.conf
@@ -32,6 +32,10 @@ override-map.2 = all-left,all-right
switch = off
volume = off
+[Element Internal Mic]
+switch = off
+volume = off
+
[Element Line]
switch = off
volume = off
diff --git a/src/modules/alsa/mixer/paths/analog-input-mic.conf b/src/modules/alsa/mixer/paths/analog-input-internal-mic.conf
similarity index 92%
copy from src/modules/alsa/mixer/paths/analog-input-mic.conf
copy to src/modules/alsa/mixer/paths/analog-input-internal-mic.conf
index 2a36f2f..70cd512 100644
--- a/src/modules/alsa/mixer/paths/analog-input-mic.conf
+++ b/src/modules/alsa/mixer/paths/analog-input-internal-mic.conf
@@ -14,12 +14,12 @@
# along with PulseAudio; if not, write to the Free Software Foundation,
# Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
-; For devices where a 'Mic' element exists
+; For devices where a 'Internal Mic' element exists
;
; See analog-output.conf.common for an explanation on the directives
[General]
-priority = 100
+priority = 90
name = analog-input-microphone
[Element Capture]
@@ -29,6 +29,10 @@ override-map.1 = all
override-map.2 = all-left,all-right
[Element Mic]
+switch = off
+volume = off
+
+[Element Internal Mic]
required = any
switch = mute
volume = merge
diff --git a/src/modules/alsa/mixer/paths/analog-input-linein.conf b/src/modules/alsa/mixer/paths/analog-input-linein.conf
index 4be5722..57568cc 100644
--- a/src/modules/alsa/mixer/paths/analog-input-linein.conf
+++ b/src/modules/alsa/mixer/paths/analog-input-linein.conf
@@ -31,6 +31,10 @@ override-map.2 = all-left,all-right
switch = off
volume = off
+[Element Internal Mic]
+switch = off
+volume = off
+
[Element Line]
required = any
switch = mute
diff --git a/src/modules/alsa/mixer/paths/analog-input-mic-line.conf b/src/modules/alsa/mixer/paths/analog-input-mic-line.conf
index f7f3085..fa680aa 100644
--- a/src/modules/alsa/mixer/paths/analog-input-mic-line.conf
+++ b/src/modules/alsa/mixer/paths/analog-input-mic-line.conf
@@ -32,6 +32,10 @@ override-map.2 = all-left,all-right
switch = off
volume = off
+[Element Internal Mic]
+switch = off
+volume = off
+
[Element Line]
switch = off
volume = off
diff --git a/src/modules/alsa/mixer/paths/analog-input-mic.conf b/src/modules/alsa/mixer/paths/analog-input-mic.conf
index 2a36f2f..9b8b75a 100644
--- a/src/modules/alsa/mixer/paths/analog-input-mic.conf
+++ b/src/modules/alsa/mixer/paths/analog-input-mic.conf
@@ -35,6 +35,10 @@ volume = merge
override-map.1 = all
override-map.2 = all-left,all-right
+[Element Internal Mic]
+switch = off
+volume = off
+
[Element Line]
switch = off
volume = off
diff --git a/src/modules/alsa/mixer/paths/analog-input-tvtuner.conf b/src/modules/alsa/mixer/paths/analog-input-tvtuner.conf
index 8531ec7..fae3ce8 100644
--- a/src/modules/alsa/mixer/paths/analog-input-tvtuner.conf
+++ b/src/modules/alsa/mixer/paths/analog-input-tvtuner.conf
@@ -32,6 +32,10 @@ override-map.2 = all-left,all-right
switch = off
volume = off
+[Element Internal Mic]
+switch = off
+volume = off
+
[Element Line]
switch = off
volume = off
diff --git a/src/modules/alsa/mixer/paths/analog-input-video.conf b/src/modules/alsa/mixer/paths/analog-input-video.conf
index 74c76f0..19f1809 100644
--- a/src/modules/alsa/mixer/paths/analog-input-video.conf
+++ b/src/modules/alsa/mixer/paths/analog-input-video.conf
@@ -31,6 +31,10 @@ override-map.2 = all-left,all-right
switch = off
volume = off
+[Element Internal Mic]
+switch = off
+volume = off
+
[Element Line]
switch = off
volume = off
diff --git a/src/modules/alsa/mixer/paths/analog-input.conf b/src/modules/alsa/mixer/paths/analog-input.conf
index 5055f90..3050738 100644
--- a/src/modules/alsa/mixer/paths/analog-input.conf
+++ b/src/modules/alsa/mixer/paths/analog-input.conf
@@ -32,6 +32,9 @@ override-map.2 = all-left,all-right
[Element Mic]
required-absent = any
+[Element Internal Mic]
+required-absent = any
+
[Element Line]
required-absent = any
commit abfd2aa0e712223a3573aa43468dacf3eee3c4e4
Author: Lennart Poettering <lennart at poettering.net>
Date: Tue Feb 16 00:59:24 2010 +0100
alsa: use default output port names
instead of coming up with pointless aliases, reuse the already established
names, for second headphones, and second speakers.
diff --git a/src/modules/alsa/alsa-mixer.c b/src/modules/alsa/alsa-mixer.c
index cb814af..93f2ed0 100644
--- a/src/modules/alsa/alsa-mixer.c
+++ b/src/modules/alsa/alsa-mixer.c
@@ -1783,9 +1783,7 @@ static int path_verify(pa_alsa_path *p) {
{ "analog-output-headphones", N_("Analog Headphones") },
{ "analog-output-lfe-on-mono", N_("Analog Output (LFE)") },
{ "analog-output-mono", N_("Analog Mono Output") },
- { "analog-output-headphones-2", N_("Analog Headphones 2") },
{ "analog-output-speaker", N_("Analog Speakers") }
- { "analog-output-desktop-speaker", N_("Analog Speakers 2") }
};
pa_alsa_element *e;
diff --git a/src/modules/alsa/mixer/paths/analog-output-desktop-speaker.conf b/src/modules/alsa/mixer/paths/analog-output-desktop-speaker.conf
index 9f18b25..dfdecf4 100644
--- a/src/modules/alsa/mixer/paths/analog-output-desktop-speaker.conf
+++ b/src/modules/alsa/mixer/paths/analog-output-desktop-speaker.conf
@@ -19,7 +19,8 @@
; See analog-output.conf.common for an explanation on the directives
[General]
-priority = 100
+priority = 101
+name = analog-output-speaker
[Element Hardware Master]
switch = mute
diff --git a/src/modules/alsa/mixer/paths/analog-output-headphones-2.conf b/src/modules/alsa/mixer/paths/analog-output-headphones-2.conf
index 580c798..e47543f 100644
--- a/src/modules/alsa/mixer/paths/analog-output-headphones-2.conf
+++ b/src/modules/alsa/mixer/paths/analog-output-headphones-2.conf
@@ -20,6 +20,7 @@
[General]
priority = 89
+name = analog-output-headphones
[Element Hardware Master]
switch = mute
diff --git a/src/modules/alsa/mixer/paths/analog-output-headphones.conf b/src/modules/alsa/mixer/paths/analog-output-headphones.conf
index 79ebc7f..1d7bb0b 100644
--- a/src/modules/alsa/mixer/paths/analog-output-headphones.conf
+++ b/src/modules/alsa/mixer/paths/analog-output-headphones.conf
@@ -20,6 +20,7 @@
[General]
priority = 90
+name = analog-output-headphones
[Element Hardware Master]
switch = mute
diff --git a/src/modules/alsa/mixer/paths/analog-output-speaker.conf b/src/modules/alsa/mixer/paths/analog-output-speaker.conf
index b133a2e..c6916d6 100644
--- a/src/modules/alsa/mixer/paths/analog-output-speaker.conf
+++ b/src/modules/alsa/mixer/paths/analog-output-speaker.conf
@@ -20,6 +20,7 @@
[General]
priority = 100
+name = analog-output-speaker
[Element Hardware Master]
switch = mute
commit 80093c94be077d15fda945786bc1291fb864e51c
Author: Lennart Poettering <lennart at poettering.net>
Date: Tue Feb 16 01:43:00 2010 +0100
build-sys: add gobject to build dependencies
Since we call g_object_unref() we need to link against gobject in
compliance of the new Fedora implicit DSO linking policy.
diff --git a/configure.ac b/configure.ac
index 8abaabb..d23e09e 100644
--- a/configure.ac
+++ b/configure.ac
@@ -929,7 +929,7 @@ AC_ARG_ENABLE([gconf],
[gconf=auto])
if test "x${gconf}" != xno ; then
- PKG_CHECK_MODULES(GCONF, [ gconf-2.0 >= 2.4.0 ],
+ PKG_CHECK_MODULES(GCONF, [ gconf-2.0 >= 2.4.0 gobject-2.0 ],
HAVE_GCONF=1,
[
HAVE_GCONF=0
commit 69ddfaf6ed8db98535382413ec4675865053dbc2
Author: Lennart Poettering <lennart at poettering.net>
Date: Tue Feb 16 02:03:45 2010 +0100
padsp: emulate /dev/audio, too
https://bugzilla.redhat.com/show_bug.cgi?id=561262
diff --git a/src/utils/padsp.c b/src/utils/padsp.c
index 2ed0a03..fb756d3 100644
--- a/src/utils/padsp.c
+++ b/src/utils/padsp.c
@@ -1458,11 +1458,11 @@ static int real_open(const char *filename, int flags, mode_t mode) {
return _open(filename, flags, mode);
}
- if (filename && dsp_cloak_enable() && (strcmp(filename, "/dev/dsp") == 0 || strcmp(filename, "/dev/adsp") == 0))
+ if (filename && dsp_cloak_enable() && (pa_streq(filename, "/dev/dsp") || pa_streq(filename, "/dev/adsp") || pa_streq(filename, "/dev/audio")))
r = dsp_open(flags, &_errno);
- else if (filename && mixer_cloak_enable() && strcmp(filename, "/dev/mixer") == 0)
+ else if (filename && mixer_cloak_enable() && pa_streq(filename, "/dev/mixer"))
r = mixer_open(flags, &_errno);
- else if (filename && sndstat_cloak_enable() && strcmp(filename, "/dev/sndstat") == 0)
+ else if (filename && sndstat_cloak_enable() && pa_streq(filename, "/dev/sndstat"))
r = sndstat_open(flags, &_errno);
else {
function_exit();
@@ -2383,15 +2383,21 @@ int close(int fd) {
return 0;
}
+static pa_bool_t is_audio_device_node(const char *path) {
+ return
+ pa_streq(path, "/dev/dsp") ||
+ pa_streq(path, "/dev/adsp") ||
+ pa_streq(path, "/dev/audio") ||
+ pa_streq(path, "/dev/sndstat") ||
+ pa_streq(path, "/dev/mixer");
+}
+
int access(const char *pathname, int mode) {
debug(DEBUG_LEVEL_VERBOSE, __FILE__": access(%s)\n", pathname?pathname:"NULL");
if (!pathname ||
- (strcmp(pathname, "/dev/dsp") != 0 &&
- strcmp(pathname, "/dev/adsp") != 0 &&
- strcmp(pathname, "/dev/sndstat") != 0 &&
- strcmp(pathname, "/dev/mixer") != 0 )) {
+ !is_audio_device_node(pathname)) {
LOAD_ACCESS_FUNC();
return _access(pathname, mode);
}
@@ -2417,10 +2423,7 @@ int stat(const char *pathname, struct stat *buf) {
if (!pathname ||
!buf ||
- ( strcmp(pathname, "/dev/dsp") != 0 &&
- strcmp(pathname, "/dev/adsp") != 0 &&
- strcmp(pathname, "/dev/sndstat") != 0 &&
- strcmp(pathname, "/dev/mixer") != 0 )) {
+ !is_audio_device_node(pathname)) {
debug(DEBUG_LEVEL_VERBOSE, __FILE__": stat(%s)\n", pathname?pathname:"NULL");
LOAD_STAT_FUNC();
return _stat(pathname, buf);
@@ -2474,10 +2477,7 @@ int stat64(const char *pathname, struct stat64 *buf) {
if (!pathname ||
!buf ||
- ( strcmp(pathname, "/dev/dsp") != 0 &&
- strcmp(pathname, "/dev/adsp") != 0 &&
- strcmp(pathname, "/dev/sndstat") != 0 &&
- strcmp(pathname, "/dev/mixer") != 0 )) {
+ !is_audio_device_node(pathname)) {
LOAD_STAT64_FUNC();
return _stat64(pathname, buf);
}
@@ -2519,10 +2519,7 @@ int open64(const char *filename, int flags, ...) {
}
if (!filename ||
- ( strcmp(filename, "/dev/dsp") != 0 &&
- strcmp(filename, "/dev/adsp") != 0 &&
- strcmp(filename, "/dev/sndstat") != 0 &&
- strcmp(filename, "/dev/mixer") != 0 )) {
+ !is_audio_device_node(filename)) {
LOAD_OPEN64_FUNC();
return _open64(filename, flags, mode);
}
@@ -2539,10 +2536,7 @@ int __xstat(int ver, const char *pathname, struct stat *buf) {
if (!pathname ||
!buf ||
- ( strcmp(pathname, "/dev/dsp") != 0 &&
- strcmp(pathname, "/dev/adsp") != 0 &&
- strcmp(pathname, "/dev/sndstat") != 0 &&
- strcmp(pathname, "/dev/mixer") != 0 )) {
+ !is_audio_device_node(pathname)) {
LOAD_XSTAT_FUNC();
return ___xstat(ver, pathname, buf);
}
@@ -2562,10 +2556,7 @@ int __xstat64(int ver, const char *pathname, struct stat64 *buf) {
if (!pathname ||
!buf ||
- ( strcmp(pathname, "/dev/dsp") != 0 &&
- strcmp(pathname, "/dev/adsp") != 0 &&
- strcmp(pathname, "/dev/sndstat") != 0 &&
- strcmp(pathname, "/dev/mixer") != 0 )) {
+ !is_audio_device_node(pathname)) {
LOAD_XSTAT64_FUNC();
return ___xstat64(ver, pathname, buf);
}
@@ -2591,10 +2582,7 @@ FILE* fopen(const char *filename, const char *mode) {
if (!filename ||
!mode ||
- ( strcmp(filename, "/dev/dsp") != 0 &&
- strcmp(filename, "/dev/adsp") != 0 &&
- strcmp(filename, "/dev/sndstat") != 0 &&
- strcmp(filename, "/dev/mixer") != 0 )) {
+ !is_audio_device_node(filename)) {
LOAD_FOPEN_FUNC();
return _fopen(filename, mode);
}
@@ -2634,10 +2622,7 @@ FILE *fopen64(const char *filename, const char *mode) {
if (!filename ||
!mode ||
- ( strcmp(filename, "/dev/dsp") != 0 &&
- strcmp(filename, "/dev/adsp") != 0 &&
- strcmp(filename, "/dev/sndstat") != 0 &&
- strcmp(filename, "/dev/mixer") != 0 )) {
+ !is_audio_device_node(filename)) {
LOAD_FOPEN64_FUNC();
return _fopen64(filename, mode);
}
commit ac76dbde6312b8052b80ce0239ef9443cfb27e1f
Author: Lennart Poettering <lennart at poettering.net>
Date: Tue Feb 16 02:49:44 2010 +0100
dbus: first restart timer, then dispatch it
This makes sure that we don't access the timer after it might have been
destroyed already from the dbus timeout callback.
https://bugzilla.redhat.com/attachment.cgi?id=389952
diff --git a/src/pulsecore/dbus-util.c b/src/pulsecore/dbus-util.c
index e3700ea..09ab071 100644
--- a/src/pulsecore/dbus-util.c
+++ b/src/pulsecore/dbus-util.c
@@ -45,17 +45,16 @@ struct pa_dbus_wrap_connection {
};
struct timeout_data {
- pa_dbus_wrap_connection *c;
+ pa_dbus_wrap_connection *connection;
DBusTimeout *timeout;
};
static void dispatch_cb(pa_mainloop_api *ea, pa_defer_event *ev, void *userdata) {
DBusConnection *conn = userdata;
- if (dbus_connection_dispatch(conn) == DBUS_DISPATCH_COMPLETE) {
+ if (dbus_connection_dispatch(conn) == DBUS_DISPATCH_COMPLETE)
/* no more data to process, disable the deferred */
ea->defer_enable(ev, 0);
- }
}
/* DBusDispatchStatusFunction callback for the pa mainloop */
@@ -132,13 +131,17 @@ static void handle_time_event(pa_mainloop_api *ea, pa_time_event* e, const struc
struct timeout_data *d = userdata;
pa_assert(d);
- pa_assert(d->c);
+ pa_assert(d->connection);
if (dbus_timeout_get_enabled(d->timeout)) {
- dbus_timeout_handle(d->timeout);
+ /* Restart it for the next scheduled time. We do this before
+ * calling dbus_timeout_handle() to make sure that the time
+ * event is still around. */
+ ea->time_restart(e, pa_timeval_rtstore(&tv,
+ pa_timeval_load(t) + dbus_timeout_get_interval(d->timeout) * PA_USEC_PER_MSEC,
+ d->connection->use_rtclock));
- /* restart it for the next scheduled time */
- ea->time_restart(e, pa_timeval_rtstore(&tv, pa_timeval_load(t) + dbus_timeout_get_interval(d->timeout) * PA_USEC_PER_MSEC, d->c->use_rtclock));
+ dbus_timeout_handle(d->timeout);
}
}
@@ -208,7 +211,7 @@ static dbus_bool_t add_timeout(DBusTimeout *timeout, void *data) {
return FALSE;
d = pa_xnew(struct timeout_data, 1);
- d->c = c;
+ d->connection = c;
d->timeout = timeout;
ev = c->mainloop->time_new(c->mainloop, pa_timeval_rtstore(&tv, pa_rtclock_now() + dbus_timeout_get_interval(timeout) * PA_USEC_PER_MSEC, c->use_rtclock), handle_time_event, d);
c->mainloop->time_set_destroy(ev, time_event_destroy_cb);
@@ -237,15 +240,15 @@ static void toggle_timeout(DBusTimeout *timeout, void *data) {
struct timeval tv;
pa_assert(d);
- pa_assert(d->c);
+ pa_assert(d->connection);
pa_assert(timeout);
pa_assert_se(ev = dbus_timeout_get_data(timeout));
- if (dbus_timeout_get_enabled(timeout)) {
- d->c->mainloop->time_restart(ev, pa_timeval_rtstore(&tv, pa_rtclock_now() + dbus_timeout_get_interval(timeout) * PA_USEC_PER_MSEC, d->c->use_rtclock));
- } else
- d->c->mainloop->time_restart(ev, pa_timeval_rtstore(&tv, PA_USEC_INVALID, d->c->use_rtclock));
+ if (dbus_timeout_get_enabled(timeout))
+ d->connection->mainloop->time_restart(ev, pa_timeval_rtstore(&tv, pa_rtclock_now() + dbus_timeout_get_interval(timeout) * PA_USEC_PER_MSEC, d->connection->use_rtclock));
+ else
+ d->connection->mainloop->time_restart(ev, pa_timeval_rtstore(&tv, PA_USEC_INVALID, d->connection->use_rtclock));
}
static void wakeup_main(void *userdata) {
commit b9bcc7c733d0e877da24686f413fac10defcfa3c
Author: Lennart Poettering <lennart at poettering.net>
Date: Tue Feb 16 03:36:44 2010 +0100
fdsem: be more verbose when reading from eventfd fails.
Apperently reading from an eventfd can fail, which results in an assert
to be hit. I am not sure about the reason for the failure, but in
attempt to track down the issue the next time is hit this prints a more
useful log message.
https://bugzilla.redhat.com/attachment.cgi?id=386380
diff --git a/src/pulsecore/fdsem.c b/src/pulsecore/fdsem.c
index ea14e8a..5bf9947 100644
--- a/src/pulsecore/fdsem.c
+++ b/src/pulsecore/fdsem.c
@@ -35,6 +35,7 @@
#include <pulsecore/thread.h>
#include <pulsecore/macro.h>
#include <pulsecore/core-util.h>
+#include <pulsecore/core-error.h>
#include <pulse/xmalloc.h>
#ifndef HAVE_PIPE
@@ -154,7 +155,12 @@ static void flush(pa_fdsem *f) {
uint64_t u;
if ((r = read(f->efd, &u, sizeof(u))) != sizeof(u)) {
- pa_assert(r < 0 && errno == EINTR);
+
+ if (r >= 0 || errno != EINTR) {
+ pa_log_error("Invalid read from eventfd: %s", r < 0 ? pa_cstrerror(errno) : "EOF");
+ pa_assert_not_reached();
+ }
+
continue;
}
r = (ssize_t) u;
@@ -162,7 +168,12 @@ static void flush(pa_fdsem *f) {
#endif
if ((r = read(f->fds[0], &x, sizeof(x))) <= 0) {
- pa_assert(r < 0 && errno == EINTR);
+
+ if (r >= 0 || errno != EINTR) {
+ pa_log_error("Invalid read from pipe: %s", r < 0 ? pa_cstrerror(errno) : "EOF");
+ pa_assert_not_reached();
+ }
+
continue;
}
@@ -187,14 +198,22 @@ void pa_fdsem_post(pa_fdsem *f) {
uint64_t u = 1;
if ((r = write(f->efd, &u, sizeof(u))) != sizeof(u)) {
- pa_assert(r < 0 && errno == EINTR);
+ if (r >= 0 || errno != EINTR) {
+ pa_log_error("Invalid read from pipe: %s", r < 0 ? pa_cstrerror(errno) : "EOF");
+ pa_assert_not_reached();
+ }
+
continue;
}
} else
#endif
if ((r = write(f->fds[1], &x, 1)) != 1) {
- pa_assert(r < 0 && errno == EINTR);
+ if (r >= 0 || errno != EINTR) {
+ pa_log_error("Invalid read from pipe: %s", r < 0 ? pa_cstrerror(errno) : "EOF");
+ pa_assert_not_reached();
+ }
+
continue;
}
@@ -223,7 +242,12 @@ void pa_fdsem_wait(pa_fdsem *f) {
uint64_t u;
if ((r = read(f->efd, &u, sizeof(u))) != sizeof(u)) {
- pa_assert(r < 0 && errno == EINTR);
+
+ if (r >= 0 || errno != EINTR) {
+ pa_log_error("Invalid read from pipe: %s", r < 0 ? pa_cstrerror(errno) : "EOF");
+ pa_assert_not_reached();
+ }
+
continue;
}
@@ -232,7 +256,12 @@ void pa_fdsem_wait(pa_fdsem *f) {
#endif
if ((r = read(f->fds[0], &x, sizeof(x))) <= 0) {
- pa_assert(r < 0 && errno == EINTR);
+
+ if (r >= 0 || errno != EINTR) {
+ pa_log_error("Invalid read from pipe: %s", r < 0 ? pa_cstrerror(errno) : "EOF");
+ pa_assert_not_reached();
+ }
+
continue;
}
commit d57ba824145f9e42610c0aa21740f219ba671041
Author: Lennart Poettering <lennart at poettering.net>
Date: Thu Feb 18 01:54:51 2010 +0100
pacat: always fully fulfill write requests
Make sure we always fulfill write requests from the server. If we don't
the server won't ask us again and playback will stay stuck.
https://tango.0pointer.de/pipermail/pulseaudio-discuss/2010-February/006611.html
diff --git a/src/utils/pacat.c b/src/utils/pacat.c
index 1036af6..75ccd9a 100644
--- a/src/utils/pacat.c
+++ b/src/utils/pacat.c
@@ -195,28 +195,41 @@ static void stream_write_callback(pa_stream *s, size_t length, void *userdata) {
pa_assert(sndfile);
- if (pa_stream_begin_write(s, &data, &length) < 0) {
- pa_log(_("pa_stream_begin_write() failed: %s"), pa_strerror(pa_context_errno(context)));
- quit(1);
- return;
- }
+ for (;;) {
+ size_t data_length = length;
- if (readf_function) {
- size_t k = pa_frame_size(&sample_spec);
+ if (pa_stream_begin_write(s, &data, &data_length) < 0) {
+ pa_log(_("pa_stream_begin_write() failed: %s"), pa_strerror(pa_context_errno(context)));
+ quit(1);
+ return;
+ }
- if ((bytes = readf_function(sndfile, data, (sf_count_t) (length/k))) > 0)
- bytes *= (sf_count_t) k;
+ if (readf_function) {
+ size_t k = pa_frame_size(&sample_spec);
- } else
- bytes = sf_read_raw(sndfile, data, (sf_count_t) length);
+ if ((bytes = readf_function(sndfile, data, (sf_count_t) (data_length/k))) > 0)
+ bytes *= (sf_count_t) k;
- if (bytes > 0)
- pa_stream_write(s, data, (size_t) bytes, NULL, 0, PA_SEEK_RELATIVE);
- else
- pa_stream_cancel_write(s);
+ } else
+ bytes = sf_read_raw(sndfile, data, (sf_count_t) data_length);
- if (bytes < (sf_count_t) length)
- start_drain();
+ if (bytes > 0)
+ pa_stream_write(s, data, (size_t) bytes, NULL, 0, PA_SEEK_RELATIVE);
+ else
+ pa_stream_cancel_write(s);
+
+ /* EOF? */
+ if (bytes < (sf_count_t) data_length) {
+ start_drain();
+ break;
+ }
+
+ /* Request fulfilled */
+ if ((size_t) bytes >= length)
+ break;
+
+ length -= bytes;
+ }
}
}
commit f9f2d6f3e0f9dfa310c7cd29bb06b8b52a0c8a1c
Author: Lennart Poettering <lennart at poettering.net>
Date: Thu Feb 18 02:32:35 2010 +0100
pacmd: store away fd type
diff --git a/src/utils/pacmd.c b/src/utils/pacmd.c
index 6ffe94f..c173ccc 100644
--- a/src/utils/pacmd.c
+++ b/src/utils/pacmd.c
@@ -61,6 +61,7 @@ int main(int argc, char*argv[]) {
char *cli;
pa_bool_t ibuf_eof, obuf_eof, ibuf_closed, obuf_closed;
struct pollfd pollfd[N_WATCH];
+ int stdin_type = 0, stdout_type = 0, fd_type = 0;
setlocale(LC_ALL, "");
bindtextdomain(GETTEXT_PACKAGE, PULSE_LOCALEDIR);
@@ -166,7 +167,7 @@ int main(int argc, char*argv[]) {
ssize_t r;
pa_assert(!ibuf_length);
- if ((r = pa_read(STDIN_FILENO, ibuf, sizeof(ibuf), NULL)) <= 0) {
+ if ((r = pa_read(STDIN_FILENO, ibuf, sizeof(ibuf), &stdin_type)) <= 0) {
if (r < 0) {
pa_log(_("read(): %s"), strerror(errno));
goto fail;
@@ -183,7 +184,7 @@ int main(int argc, char*argv[]) {
ssize_t r;
pa_assert(!obuf_length);
- if ((r = pa_read(fd, obuf, sizeof(obuf), NULL)) <= 0) {
+ if ((r = pa_read(fd, obuf, sizeof(obuf), &fd_type)) <= 0) {
if (r < 0) {
pa_log(_("read(): %s"), strerror(errno));
goto fail;
@@ -203,7 +204,7 @@ int main(int argc, char*argv[]) {
ssize_t r;
pa_assert(obuf_length);
- if ((r = pa_write(STDOUT_FILENO, obuf + obuf_index, obuf_length, NULL)) < 0) {
+ if ((r = pa_write(STDOUT_FILENO, obuf + obuf_index, obuf_length, &stdout_type)) < 0) {
pa_log(_("write(): %s"), strerror(errno));
goto fail;
}
@@ -219,7 +220,7 @@ int main(int argc, char*argv[]) {
ssize_t r;
pa_assert(ibuf_length);
- if ((r = pa_write(fd, ibuf + ibuf_index, ibuf_length, NULL)) < 0) {
+ if ((r = pa_write(fd, ibuf + ibuf_index, ibuf_length, &fd_type)) < 0) {
pa_log(_("write(): %s"), strerror(errno));
goto fail;
}
commit c2079d792c0eb8d2cbb8fdfcfc58302842a0a2d2
Author: Lennart Poettering <lennart at poettering.net>
Date: Fri Feb 19 03:29:36 2010 +0100
shm: don't complain about missing SHM segments
If two clients try to cleanup the SHM directory at the same time, they
might want to open and then delete the same segment at the same time, in
which case one client might win, the other one lose. In this case, don't
warn about ENOENT.
diff --git a/src/pulsecore/shm.c b/src/pulsecore/shm.c
index 29849d5..5d5d85a 100644
--- a/src/pulsecore/shm.c
+++ b/src/pulsecore/shm.c
@@ -290,7 +290,7 @@ int pa_shm_attach_ro(pa_shm *m, unsigned id) {
segment_name(fn, sizeof(fn), m->id = id);
if ((fd = shm_open(fn, O_RDONLY, 0)) < 0) {
- if (errno != EACCES)
+ if (errno != EACCES && errno != ENOENT)
pa_log("shm_open() failed: %s", pa_cstrerror(errno));
goto fail;
}
commit 57e0e2cdab5ad59f570d2c476b7329047c03ae85
Author: Lennart Poettering <lennart at poettering.net>
Date: Fri Feb 19 03:31:46 2010 +0100
vala: fix definition of INVALID_INDEX
diff --git a/vala/libpulse.vapi b/vala/libpulse.vapi
index 9e05b14..b02bb56 100644
--- a/vala/libpulse.vapi
+++ b/vala/libpulse.vapi
@@ -46,7 +46,7 @@ namespace PulseAudio {
[CCode (cname="PA_CHECK_VERSION")]
public bool CHECK_VERSION(int major, int minor, int micro);
- [CCode (cname="INVALID_INDEX")]
+ [CCode (cname="PA_INVALID_INDEX")]
public const uint32 INVALID_INDEX;
[CCode (cname="pa_free_cb_t")]
commit c25665ed10d6cebaa3d0c6d0ce1e620b01ff8ae9
Author: Lennart Poettering <lennart at poettering.net>
Date: Fri Feb 19 03:32:00 2010 +0100
vala: fix definition of the GLib mainloop adapter
diff --git a/vala/libpulse.vapi b/vala/libpulse.vapi
index b02bb56..d47ef73 100644
--- a/vala/libpulse.vapi
+++ b/vala/libpulse.vapi
@@ -879,11 +879,11 @@ namespace PulseAudio {
}
[Compact]
- [CCode (cname="pa_glib_mainloop", cprefix="pa_glib_mainloop_", free_function="pa_glib_mainloop_free")]
+ [CCode (cheader_filename="pulse/glib-mainloop.h", cname="pa_glib_mainloop", cprefix="pa_glib_mainloop_", free_function="pa_glib_mainloop_free")]
public class GLibMainLoop {
[CCode (cname="pa_glib_mainloop_new")]
- public GLibMainLoop();
+ public GLibMainLoop(MainContext? c = null);
public unowned MainLoopApi get_api();
}
commit 27427c114cb93f52bcea3bb8e7900d39b286a271
Author: Lennart Poettering <lennart at poettering.net>
Date: Sun Feb 21 16:19:40 2010 +0100
channelmap: Use "Subwoofer" as pretty name for LFE
The pretty name is suspposed to be understandable by non-technical
folks, and they are generally more used to the term "Subwoofer" than
"Low Frequency Emitter", so let's change the name here.
diff --git a/src/pulse/channelmap.c b/src/pulse/channelmap.c
index 9b51626..83fdb43 100644
--- a/src/pulse/channelmap.c
+++ b/src/pulse/channelmap.c
@@ -112,7 +112,7 @@ const char *const pretty_table[PA_CHANNEL_POSITION_MAX] = {
[PA_CHANNEL_POSITION_REAR_LEFT] = N_("Rear Left"),
[PA_CHANNEL_POSITION_REAR_RIGHT] = N_("Rear Right"),
- [PA_CHANNEL_POSITION_LFE] = N_("Low Frequency Emmiter"),
+ [PA_CHANNEL_POSITION_LFE] = N_("Subwoofer"),
[PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER] = N_("Front Left-of-center"),
[PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER] = N_("Front Right-of-center"),
commit 534def2d2b5d62c3ce85672d82d58223c7ea054d
Author: Lennart Poettering <lennart at poettering.net>
Date: Sun Feb 21 16:21:13 2010 +0100
vala: fix wrapping of port setting calls
diff --git a/vala/libpulse.vapi b/vala/libpulse.vapi
index d47ef73..21c0f7c 100644
--- a/vala/libpulse.vapi
+++ b/vala/libpulse.vapi
@@ -1024,7 +1024,7 @@ namespace PulseAudio {
public Operation? suspend_sink_by_index(uint32 idx, bool suspend, SuccessCb? cb = null);
public Operation? set_sink_port_by_name(string name, string port, SuccessCb? cb = null);
- public Operation? set_sink_port_by_index(string idx, string port, SuccessCb? cb = null);
+ public Operation? set_sink_port_by_index(uint32 idx, string port, SuccessCb? cb = null);
public Operation? get_source_info_by_name(string name, SourceInfoCb cb);
public Operation? get_source_info_by_index(uint32 idx, SourceInfoCb cb);
@@ -1039,7 +1039,7 @@ namespace PulseAudio {
public Operation? suspend_source_by_index(uint32 idx, bool suspend, SuccessCb? cb = null);
public Operation? set_source_port_by_name(string name, string port, SuccessCb? cb = null);
- public Operation? set_source_port_by_index(string idx, string port, SuccessCb? cb = null);
+ public Operation? set_source_port_by_index(uint32 idx, string port, SuccessCb? cb = null);
public Operation? get_server_info(ServerInfoCb cb);
commit 7098d60b302c94f560ea1014b5079a346ba843b2
Author: Lennart Poettering <lennart at poettering.net>
Date: Sun Feb 21 16:21:43 2010 +0100
proplist: explicitly mention a role 'test'
diff --git a/src/pulse/proplist.h b/src/pulse/proplist.h
index e3f0af0..5db3ada 100644
--- a/src/pulse/proplist.h
+++ b/src/pulse/proplist.h
@@ -59,7 +59,7 @@ PA_C_DECL_BEGIN
/** For streams: an XDG icon name for the media. e.g. "audio-x-mp3" */
#define PA_PROP_MEDIA_ICON_NAME "media.icon_name"
-/** For streams: logic role of this media. One of the strings "video", "music", "game", "event", "phone", "animation", "production", "a11y" */
+/** For streams: logic role of this media. One of the strings "video", "music", "game", "event", "phone", "animation", "production", "a11y", "test" */
#define PA_PROP_MEDIA_ROLE "media.role"
/** For event sound streams: XDG event sound name. e.g. "message-new-email" (Event sound streams are those with media.role set to "event") */
commit 857ff2c160c19699ecc541767ac443fa61456cb5
Author: Lennart Poettering <lennart at poettering.net>
Date: Sun Feb 21 16:22:17 2010 +0100
stream-restore: be a little bit more verbose why we don't reastore a sink
diff --git a/src/modules/module-stream-restore.c b/src/modules/module-stream-restore.c
index 64a7fd3..c16a74c 100644
--- a/src/modules/module-stream-restore.c
+++ b/src/modules/module-stream-restore.c
@@ -1278,7 +1278,7 @@ static pa_hook_result_t sink_input_new_hook_callback(pa_core *c, pa_sink_input_n
return PA_HOOK_OK;
if (new_data->sink)
- pa_log_debug("Not restoring device for stream %s, because already set.", name);
+ pa_log_debug("Not restoring device for stream %s, because already set to '%s'.", name, new_data->sink->name);
else if ((e = read_entry(u, name))) {
pa_sink *s = NULL;
commit 3f1c90b9d7fa434ae90b45018255d086b0fa6e8a
Author: Lennart Poettering <lennart at poettering.net>
Date: Sun Feb 21 16:23:25 2010 +0100
sample-cache: use the sample name as unmodified fallback in the properties when playing a sample from the cache
diff --git a/src/pulsecore/core-scache.c b/src/pulsecore/core-scache.c
index 1fb81d0..95aaa10 100644
--- a/src/pulsecore/core-scache.c
+++ b/src/pulsecore/core-scache.c
@@ -310,7 +310,8 @@ int pa_scache_play_item(pa_core *c, const char *name, pa_sink *sink, pa_volume_t
return -1;
merged = pa_proplist_new();
- pa_proplist_setf(merged, PA_PROP_MEDIA_NAME, "Sample %s", name);
+ pa_proplist_sets(merged, PA_PROP_MEDIA_NAME, name);
+ pa_proplist_sets(merged, PA_PROP_EVENT_ID, name);
if (e->lazy && !e->memchunk.memblock) {
pa_channel_map old_channel_map = e->channel_map;
commit ea29b11097b9d4180582924ca5df142fae4b6400
Author: Lennart Poettering <lennart at poettering.net>
Date: Sun Feb 21 17:35:05 2010 +0100
scache: when playing a sample from the cache make sure not queue them up when the sink is suspended
libcanberra already sets the appropriate flags for uncached sample
streams, we now need to make sure to set them for cached samples too.
diff --git a/src/pulsecore/core-scache.c b/src/pulsecore/core-scache.c
index 95aaa10..0172419 100644
--- a/src/pulsecore/core-scache.c
+++ b/src/pulsecore/core-scache.c
@@ -351,7 +351,12 @@ int pa_scache_play_item(pa_core *c, const char *name, pa_sink *sink, pa_volume_t
if (p)
pa_proplist_update(merged, PA_UPDATE_REPLACE, p);
- if (pa_play_memchunk(sink, &e->sample_spec, &e->channel_map, &e->memchunk, pass_volume ? &r : NULL, merged, sink_input_idx) < 0)
+ if (pa_play_memchunk(sink,
+ &e->sample_spec, &e->channel_map,
+ &e->memchunk,
+ pass_volume ? &r : NULL,
+ merged,
+ PA_SINK_INPUT_NO_CREATE_ON_SUSPEND|PA_SINK_INPUT_KILL_ON_SUSPEND, sink_input_idx) < 0)
goto fail;
pa_proplist_free(merged);
diff --git a/src/pulsecore/play-memblockq.c b/src/pulsecore/play-memblockq.c
index f528c49..0d6da3e 100644
--- a/src/pulsecore/play-memblockq.c
+++ b/src/pulsecore/play-memblockq.c
@@ -173,7 +173,8 @@ pa_sink_input* pa_memblockq_sink_input_new(
const pa_channel_map *map,
pa_memblockq *q,
pa_cvolume *volume,
- pa_proplist *p) {
+ pa_proplist *p,
+ pa_sink_input_flags_t flags) {
memblockq_stream *u = NULL;
pa_sink_input_new_data data;
@@ -198,6 +199,7 @@ pa_sink_input* pa_memblockq_sink_input_new(
pa_sink_input_new_data_set_channel_map(&data, map);
pa_sink_input_new_data_set_volume(&data, volume);
pa_proplist_update(data.proplist, PA_UPDATE_REPLACE, p);
+ data.flags |= flags;
pa_sink_input_new(&u->sink_input, sink->core, &data);
pa_sink_input_new_data_done(&data);
@@ -237,6 +239,7 @@ int pa_play_memblockq(
pa_memblockq *q,
pa_cvolume *volume,
pa_proplist *p,
+ pa_sink_input_flags_t flags,
uint32_t *sink_input_index) {
pa_sink_input *i;
@@ -245,7 +248,7 @@ int pa_play_memblockq(
pa_assert(ss);
pa_assert(q);
- if (!(i = pa_memblockq_sink_input_new(sink, ss, map, q, volume, p)))
+ if (!(i = pa_memblockq_sink_input_new(sink, ss, map, q, volume, p, flags)))
return -1;
pa_sink_input_put(i);
diff --git a/src/pulsecore/play-memblockq.h b/src/pulsecore/play-memblockq.h
index 9d5f40f..a55fed0 100644
--- a/src/pulsecore/play-memblockq.h
+++ b/src/pulsecore/play-memblockq.h
@@ -31,17 +31,19 @@ pa_sink_input* pa_memblockq_sink_input_new(
const pa_channel_map *map,
pa_memblockq *q,
pa_cvolume *volume,
- pa_proplist *p);
+ pa_proplist *p,
+ pa_sink_input_flags_t flags);
void pa_memblockq_sink_input_set_queue(pa_sink_input *i, pa_memblockq *q);
int pa_play_memblockq(
- pa_sink *sink,
- const pa_sample_spec *ss,
- const pa_channel_map *map,
- pa_memblockq *q,
- pa_cvolume *cvolume,
- pa_proplist *p,
- uint32_t *sink_input_index);
+ pa_sink *sink,
+ const pa_sample_spec *ss,
+ const pa_channel_map *map,
+ pa_memblockq *q,
+ pa_cvolume *cvolume,
+ pa_proplist *p,
+ pa_sink_input_flags_t flags,
+ uint32_t *sink_input_index);
#endif
diff --git a/src/pulsecore/play-memchunk.c b/src/pulsecore/play-memchunk.c
index 1a3bd5b..6fb8902 100644
--- a/src/pulsecore/play-memchunk.c
+++ b/src/pulsecore/play-memchunk.c
@@ -43,6 +43,7 @@ int pa_play_memchunk(
const pa_memchunk *chunk,
pa_cvolume *volume,
pa_proplist *p,
+ pa_sink_input_flags_t flags,
uint32_t *sink_input_index) {
pa_memblockq *q;
@@ -59,7 +60,7 @@ int pa_play_memchunk(
pa_assert_se(pa_memblockq_push(q, chunk) >= 0);
- if ((r = pa_play_memblockq(sink, ss, map, q, volume, p, sink_input_index)) < 0) {
+ if ((r = pa_play_memblockq(sink, ss, map, q, volume, p, flags, sink_input_index)) < 0) {
pa_memblockq_free(q);
return r;
}
diff --git a/src/pulsecore/play-memchunk.h b/src/pulsecore/play-memchunk.h
index c813611..068add1 100644
--- a/src/pulsecore/play-memchunk.h
+++ b/src/pulsecore/play-memchunk.h
@@ -26,12 +26,13 @@
#include <pulsecore/memchunk.h>
int pa_play_memchunk(
- pa_sink *sink,
- const pa_sample_spec *ss,
- const pa_channel_map *map,
- const pa_memchunk *chunk,
- pa_cvolume *cvolume,
- pa_proplist *p,
- uint32_t *sink_input_index);
+ pa_sink *sink,
+ const pa_sample_spec *ss,
+ const pa_channel_map *map,
+ const pa_memchunk *chunk,
+ pa_cvolume *cvolume,
+ pa_proplist *p,
+ pa_sink_input_flags_t flags,
+ uint32_t *sink_input_index);
#endif
commit 83eb8afc5f9faaa4d17fce7dbde58b1863d9173c
Author: Lennart Poettering <lennart at poettering.net>
Date: Thu Jan 14 02:33:49 2010 +0100
daemon: first take name on the bus, then return in starter process
http://pulseaudio.org/ticket/748
diff --git a/src/daemon/main.c b/src/daemon/main.c
index 5ccda77..9bea2ae 100644
--- a/src/daemon/main.c
+++ b/src/daemon/main.c
@@ -1027,15 +1027,6 @@ int main(int argc, char *argv[]) {
* from now on, if requested */
c->disallow_module_loading = !!conf->disallow_module_loading;
-#ifdef HAVE_FORK
- if (daemon_pipe[1] >= 0) {
- int ok = 0;
- pa_loop_write(daemon_pipe[1], &ok, sizeof(ok), NULL);
- pa_close(daemon_pipe[1]);
- daemon_pipe[1] = -1;
- }
-#endif
-
#ifdef HAVE_DBUS
if (!conf->system_instance) {
if (!(server_lookup = pa_dbusobj_server_lookup_new(c)))
@@ -1048,6 +1039,15 @@ int main(int argc, char *argv[]) {
goto finish;
#endif
+#ifdef HAVE_FORK
+ if (daemon_pipe[1] >= 0) {
+ int ok = 0;
+ pa_loop_write(daemon_pipe[1], &ok, sizeof(ok), NULL);
+ pa_close(daemon_pipe[1]);
+ daemon_pipe[1] = -1;
+ }
+#endif
+
pa_log_info(_("Daemon startup complete."));
retval = 0;
commit 609a005b77b5dc691271f48351bd39e3eecb7dc9
Author: Lennart Poettering <lennart at poettering.net>
Date: Thu Feb 18 03:28:56 2010 +0100
pacmd: don't enter busy loop when reading from stdin very early
pointed out in:
https://tango.0pointer.de/pipermail/pulseaudio-discuss/2010-January/006365.html
diff --git a/src/utils/pacmd.c b/src/utils/pacmd.c
index c173ccc..143a948 100644
--- a/src/utils/pacmd.c
+++ b/src/utils/pacmd.c
@@ -45,13 +45,6 @@
int main(int argc, char*argv[]) {
- enum {
- WATCH_STDIN,
- WATCH_STDOUT,
- WATCH_SOCKET,
- N_WATCH
- };
-
pid_t pid ;
int fd = -1;
int ret = 1, i;
@@ -60,7 +53,9 @@ int main(int argc, char*argv[]) {
size_t ibuf_index, ibuf_length, obuf_index, obuf_length;
char *cli;
pa_bool_t ibuf_eof, obuf_eof, ibuf_closed, obuf_closed;
- struct pollfd pollfd[N_WATCH];
+ struct pollfd pollfd[3];
+ struct pollfd *watch_socket, *watch_stdin, *watch_stdout;
+
int stdin_type = 0, stdout_type = 0, fd_type = 0;
setlocale(LC_ALL, "");
@@ -129,32 +124,53 @@ int main(int argc, char*argv[]) {
ibuf_eof = TRUE;
}
- pa_zero(pollfd);
-
- pollfd[WATCH_STDIN].fd = STDIN_FILENO;
- pollfd[WATCH_STDOUT].fd = STDOUT_FILENO;
- pollfd[WATCH_SOCKET].fd = fd;
-
for (;;) {
+ struct pollfd *p;
+
if (ibuf_eof &&
obuf_eof &&
ibuf_length <= 0 &&
obuf_length <= 0)
break;
- pollfd[WATCH_STDIN].events = pollfd[WATCH_STDOUT].events = pollfd[WATCH_SOCKET].events = 0;
+ if (ibuf_length <= 0 && ibuf_eof && !ibuf_closed) {
+ shutdown(fd, SHUT_WR);
+ ibuf_closed = TRUE;
+ }
- if (obuf_length > 0)
- pollfd[WATCH_STDOUT].events |= POLLOUT;
- else if (!obuf_eof)
- pollfd[WATCH_SOCKET].events |= POLLIN;
+ if (obuf_length <= 0 && obuf_eof && !obuf_closed) {
+ shutdown(fd, SHUT_RD);
+ obuf_closed = TRUE;
+ }
+
+ pa_zero(pollfd);
+
+ p = pollfd;
- if (ibuf_length > 0)
- pollfd[WATCH_SOCKET].events |= POLLOUT;
- else if (!ibuf_eof)
- pollfd[WATCH_STDIN].events |= POLLIN;
+ if (ibuf_length > 0 || (!obuf_eof && obuf_length <= 0)) {
+ watch_socket = p++;
+ watch_socket->fd = fd;
+ watch_socket->events =
+ (ibuf_length > 0 ? POLLOUT : 0) |
+ (!obuf_eof && obuf_length <= 0 ? POLLIN : 0);
+ } else
+ watch_socket = NULL;
- if (pa_poll(pollfd, N_WATCH, -1) < 0) {
+ if (!ibuf_eof && ibuf_length <= 0) {
+ watch_stdin = p++;
+ watch_stdin->fd = STDIN_FILENO;
+ watch_stdin->events = POLLIN;
+ } else
+ watch_stdin = NULL;
+
+ if (obuf_length > 0) {
+ watch_stdout = p++;
+ watch_stdout->fd = STDOUT_FILENO;
+ watch_stdout->events = POLLOUT;
+ } else
+ watch_stdout = NULL;
+
+ if (pa_poll(pollfd, p-pollfd, -1) < 0) {
if (errno == EINTR)
continue;
@@ -163,82 +179,80 @@ int main(int argc, char*argv[]) {
goto fail;
}
- if (pollfd[WATCH_STDIN].revents & POLLIN) {
- ssize_t r;
- pa_assert(!ibuf_length);
-
- if ((r = pa_read(STDIN_FILENO, ibuf, sizeof(ibuf), &stdin_type)) <= 0) {
- if (r < 0) {
- pa_log(_("read(): %s"), strerror(errno));
- goto fail;
+ if (watch_stdin) {
+ if (watch_stdin->revents & POLLIN) {
+ ssize_t r;
+ pa_assert(ibuf_length <= 0);
+
+ if ((r = pa_read(STDIN_FILENO, ibuf, sizeof(ibuf), &stdin_type)) <= 0) {
+ if (r < 0) {
+ pa_log(_("read(): %s"), strerror(errno));
+ goto fail;
+ }
+
+ ibuf_eof = TRUE;
+ } else {
+ ibuf_length = (size_t) r;
+ ibuf_index = 0;
}
-
+ } else if (watch_stdin->revents & POLLHUP)
ibuf_eof = TRUE;
- } else {
- ibuf_length = (size_t) r;
- ibuf_index = 0;
- }
}
- if (pollfd[WATCH_SOCKET].revents & POLLIN) {
- ssize_t r;
- pa_assert(!obuf_length);
-
- if ((r = pa_read(fd, obuf, sizeof(obuf), &fd_type)) <= 0) {
- if (r < 0) {
- pa_log(_("read(): %s"), strerror(errno));
- goto fail;
+ if (watch_socket) {
+ if (watch_socket->revents & POLLIN) {
+ ssize_t r;
+ pa_assert(obuf_length <= 0);
+
+ if ((r = pa_read(fd, obuf, sizeof(obuf), &fd_type)) <= 0) {
+ if (r < 0) {
+ pa_log(_("read(): %s"), strerror(errno));
+ goto fail;
+ }
+
+ obuf_eof = TRUE;
+ } else {
+ obuf_length = (size_t) r;
+ obuf_index = 0;
}
-
+ } else if (watch_socket->revents & POLLHUP)
obuf_eof = TRUE;
- } else {
- obuf_length = (size_t) r;
- obuf_index = 0;
- }
}
- if (pollfd[WATCH_STDOUT].revents & POLLHUP) {
- obuf_eof = TRUE;
- obuf_length = 0;
- } else if (pollfd[WATCH_STDOUT].revents & POLLOUT) {
- ssize_t r;
- pa_assert(obuf_length);
-
- if ((r = pa_write(STDOUT_FILENO, obuf + obuf_index, obuf_length, &stdout_type)) < 0) {
- pa_log(_("write(): %s"), strerror(errno));
- goto fail;
- }
-
- obuf_length -= (size_t) r;
- obuf_index += obuf_index;
- }
+ if (watch_stdout) {
+ if (watch_stdout->revents & POLLHUP) {
+ obuf_eof = TRUE;
+ obuf_length = 0;
+ } else if (watch_stdout->revents & POLLOUT) {
+ ssize_t r;
+ pa_assert(obuf_length > 0);
- if (pollfd[WATCH_SOCKET].revents & POLLHUP) {
- ibuf_eof = TRUE;
- ibuf_length = 0;
- } if (pollfd[WATCH_SOCKET].revents & POLLOUT) {
- ssize_t r;
- pa_assert(ibuf_length);
+ if ((r = pa_write(STDOUT_FILENO, obuf + obuf_index, obuf_length, &stdout_type)) < 0) {
+ pa_log(_("write(): %s"), strerror(errno));
+ goto fail;
+ }
- if ((r = pa_write(fd, ibuf + ibuf_index, ibuf_length, &fd_type)) < 0) {
- pa_log(_("write(): %s"), strerror(errno));
- goto fail;
+ obuf_length -= (size_t) r;
+ obuf_index += obuf_index;
}
-
- ibuf_length -= (size_t) r;
- ibuf_index += obuf_index;
}
- if (ibuf_length <= 0 && ibuf_eof && !ibuf_closed) {
- pa_close(STDIN_FILENO);
- shutdown(fd, SHUT_WR);
- ibuf_closed = TRUE;
- }
+ if (watch_socket) {
+ if (watch_socket->revents & POLLHUP) {
+ ibuf_eof = TRUE;
+ ibuf_length = 0;
+ } if (watch_socket->revents & POLLOUT) {
+ ssize_t r;
+ pa_assert(ibuf_length > 0);
- if (obuf_length <= 0 && obuf_eof && !obuf_closed) {
- shutdown(fd, SHUT_RD);
- pa_close(STDOUT_FILENO);
- obuf_closed = TRUE;
+ if ((r = pa_write(fd, ibuf + ibuf_index, ibuf_length, &fd_type)) < 0) {
+ pa_log(_("write(): %s"), strerror(errno));
+ goto fail;
+ }
+
+ ibuf_length -= (size_t) r;
+ ibuf_index += obuf_index;
+ }
}
}
commit 2b2149dc9d6b2010a9acd4d05ec4e331e710fd25
Author: Daniel T Chen <crimsun at ubuntu.com>
Date: Fri Feb 19 00:23:17 2010 -0500
build-sys: Add missing profile and alsa-mixer/paths to src/Makefile.am
Commits e8cb96c and 0c836aa created mixer paths but did not update
src/Makefile.am. Building a snapshot containing these two commits
therefore results in the user being unable to adjust the volume or
(un)mute through PA. Fix this by adding the two new mixer paths
files to src/Makefile.am.
Likewise, commit 66e1a2d created a profile for the M-Audio FastTrack
Pro USB but did not update src/Makefile.am. Fix this by adding the
profile to src/Makefile.am.
diff --git a/src/Makefile.am b/src/Makefile.am
index 3268137..01ec1a5 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -1095,6 +1095,7 @@ modlibexec_LTLIBRARIES += \
dist_alsaprofilesets_DATA = \
modules/alsa/mixer/profile-sets/default.conf \
+ modules/alsa/mixer/profile-sets/maudio-fasttrack-pro.conf \
modules/alsa/mixer/profile-sets/native-instruments-audio4dj.conf \
modules/alsa/mixer/profile-sets/native-instruments-audio8dj.conf
@@ -1108,6 +1109,7 @@ dist_alsapaths_DATA = \
modules/alsa/mixer/paths/analog-input.conf \
modules/alsa/mixer/paths/analog-input.conf.common \
modules/alsa/mixer/paths/analog-input-fm.conf \
+ modules/alsa/mixer/paths/analog-input-internal-mic.conf \
modules/alsa/mixer/paths/analog-input-linein.conf \
modules/alsa/mixer/paths/analog-input-mic.conf \
modules/alsa/mixer/paths/analog-input-mic.conf.common \
@@ -1117,6 +1119,7 @@ dist_alsapaths_DATA = \
modules/alsa/mixer/paths/analog-output.conf \
modules/alsa/mixer/paths/analog-output-speaker.conf \
modules/alsa/mixer/paths/analog-output.conf.common \
+ modules/alsa/mixer/paths/analog-output-desktop-speaker.conf \
modules/alsa/mixer/paths/analog-output-headphones.conf \
modules/alsa/mixer/paths/analog-output-headphones-2.conf \
modules/alsa/mixer/paths/analog-output-lfe-on-mono.conf \
commit 323f0ad4034bf37b8c2002a8ee8efab22db5b140
Author: Lennart Poettering <lennart at poettering.net>
Date: Sun Feb 21 20:11:23 2010 +0100
build-sys: fix check for pthread_setaffinity_np()
diff --git a/configure.ac b/configure.ac
index d23e09e..c277f6e 100644
--- a/configure.ac
+++ b/configure.ac
@@ -402,6 +402,7 @@ AC_SEARCH_LIBS([dlopen], [dl])
AC_SEARCH_LIBS([shm_open], [rt])
AC_SEARCH_LIBS([inet_ntop], [nsl])
AC_SEARCH_LIBS([timer_create], [rt])
+AC_SEARCH_LIBS([pthread_setaffinity_np], [pthread])
# BSD
AC_SEARCH_LIBS([connect], [socket])
diff --git a/src/tests/once-test.c b/src/tests/once-test.c
index 5b55532..37f3ac3 100644
--- a/src/tests/once-test.c
+++ b/src/tests/once-test.c
@@ -36,7 +36,6 @@ static pa_once once = PA_ONCE_INIT;
static volatile unsigned n_run = 0;
static const char * volatile ran_by = NULL;
static pthread_barrier_t barrier;
-static pa_atomic_t i_cpu = PA_ATOMIC_INIT(0);
static unsigned n_cpu;
#define N_ITERATIONS 500
@@ -51,6 +50,7 @@ static void thread_func(void *data) {
int r;
#ifdef HAVE_PTHREAD_SETAFFINITY_NP
+ static pa_atomic_t i_cpu = PA_ATOMIC_INIT(0);
cpu_set_t mask;
CPU_ZERO(&mask);
--
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