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Updating link to change in openSUSE:Factory/pulseaudio revision 56.0

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OBS-URL: https://build.opensuse.org/package/show/multimedia:libs/pulseaudio?expand=0&rev=a41c182c62abf2480b0be2b95493c211
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OBS User buildservice-autocommit 2010-03-08 17:21:12 +00:00 committed by Git OBS Bridge
parent 026c4958c2
commit 078b914c2d
32 changed files with 2343 additions and 1 deletions
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25
0033-core-Fix-macro-typo-PA_SINK_IS_LINKED-PA_SINK_.patch Normal file
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@ -0,0 +1,25 @@
From a631beeafb4729f8bed1c538846d0bb142cee7bf Mon Sep 17 00:00:00 2001
From: Colin Guthrie <cguthrie@mandriva.org>
Date: Tue, 9 Feb 2010 21:37:32 +0000
Subject: [PATCH] core: Fix macro typo - PA_SINK_IS_LINKED -> PA_SINK_INPUT_IS_LINKED
---
src/pulsecore/sink-input.c | 2 +-
1 files changed, 1 insertions(+), 1 deletions(-)
diff --git a/src/pulsecore/sink-input.c b/src/pulsecore/sink-input.c
index 7acb25f..3c957f1 100644
--- a/src/pulsecore/sink-input.c
+++ b/src/pulsecore/sink-input.c
@@ -1086,7 +1086,7 @@ void pa_sink_input_update_proplist(pa_sink_input *i, pa_update_mode_t mode, pa_p
if (p)
pa_proplist_update(i->proplist, mode, p);
- if (PA_SINK_IS_LINKED(i->state)) {
+ if (PA_SINK_INPUT_IS_LINKED(i->state)) {
pa_hook_fire(&i->core->hooks[PA_CORE_HOOK_SINK_INPUT_PROPLIST_CHANGED], i);
pa_subscription_post(i->core, PA_SUBSCRIPTION_EVENT_SINK_INPUT|PA_SUBSCRIPTION_EVENT_CHANGE, i->index);
}
--
1.6.0.2

292
0034-alsa-cover-Desktop-Speaker-mixer-elements.patch Normal file
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@ -0,0 +1,292 @@
From e8cb96c0103d5bd23fe945afdc658a5f60b70a9d Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Mon, 15 Feb 2010 22:23:39 +0100
Subject: [PATCH] alsa: cover "Desktop Speaker" mixer elements
As exposed by really old Microsoft USB sound systems
---
src/modules/alsa/alsa-mixer.c | 3 +-
.../mixer/paths/analog-output-desktop-speaker.conf | 98 ++++++++++++++++++++
.../mixer/paths/analog-output-headphones-2.conf | 4 +
.../alsa/mixer/paths/analog-output-headphones.conf | 4 +
.../mixer/paths/analog-output-lfe-on-mono.conf | 4 +
.../alsa/mixer/paths/analog-output-mono.conf | 4 +
.../alsa/mixer/paths/analog-output-speaker.conf | 4 +
src/modules/alsa/mixer/paths/analog-output.conf | 4 +
src/modules/alsa/mixer/profile-sets/default.conf | 14 ++--
9 files changed, 131 insertions(+), 8 deletions(-)
create mode 100644 src/modules/alsa/mixer/paths/analog-output-desktop-speaker.conf
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-desktop-speaker.conf b/src/modules/alsa/mixer/paths/analog-output-desktop-speaker.conf
new file mode 100644
index 0000000..9f18b25
--- /dev/null
+++ b/src/modules/alsa/mixer/paths/analog-output-desktop-speaker.conf
@@ -0,0 +1,98 @@
+# This file is part of PulseAudio.
+#
+# PulseAudio is free software; you can redistribute it and/or modify
+# it under the terms of the GNU Lesser General Public License as
+# published by the Free Software Foundation; either version 2.1 of the
+# License, or (at your option) any later version.
+#
+# PulseAudio is distributed in the hope that it will be useful, but
+# WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+# General Public License for more details.
+#
+# You should have received a copy of the GNU Lesser General Public License
+# 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 'Desktop Speaker' control
+;
+; See analog-output.conf.common for an explanation on the directives
+
+[General]
+priority = 100
+
+[Element Hardware Master]
+switch = mute
+volume = merge
+override-map.1 = all
+override-map.2 = all-left,all-right
+
+[Element Master]
+switch = mute
+volume = merge
+override-map.1 = all
+override-map.2 = all-left,all-right
+
+[Element Master Mono]
+switch = off
+volume = off
+
+; 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
+
+[Element Headphone2]
+switch = mute
+volume = zero
+
+[Element Speaker]
+switch = off
+volume = off
+
+[Element Desktop Speaker]
+required = any
+switch = mute
+volume = merge
+override-map.1 = all
+override-map.2 = all-left,all-right
+
+[Element Front]
+switch = mute
+volume = merge
+override-map.1 = all-front
+override-map.2 = front-left,front-right
+
+[Element Rear]
+switch = mute
+volume = merge
+override-map.1 = all-rear
+override-map.2 = rear-left,rear-right
+
+[Element Surround]
+switch = mute
+volume = merge
+override-map.1 = all-rear
+override-map.2 = rear-left,rear-right
+
+[Element Side]
+switch = mute
+volume = merge
+override-map.1 = all-side
+override-map.2 = side-left,side-right
+
+[Element Center]
+switch = mute
+volume = merge
+override-map.1 = all-center
+override-map.2 = all-center,all-center
+
+[Element LFE]
+switch = mute
+volume = merge
+override-map.1 = lfe
+override-map.2 = lfe,lfe
+
+.include analog-output.conf.common
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
--
1.6.0.2

49
0035-alsa-cover-Shared-Mic-Line-in-Analog-Source.patch Normal file
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@ -0,0 +1,49 @@
From 7d6bab0d003386baba3ff3fad4cc3caa5839b361 Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Mon, 15 Feb 2010 23:39:12 +0100
Subject: [PATCH] alsa: cover 'Shared Mic/Line in', 'Analog Source'
https://bugzilla.redhat.com/show_bug.cgi?id=558638
---
.../alsa/mixer/paths/analog-input.conf.common | 25 ++++++++++++++++++++
1 files changed, 25 insertions(+), 0 deletions(-)
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]
--
1.6.0.2

214
0036-alsa-cover-Internal-Mic-elements.patch Normal file
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@ -0,0 +1,214 @@
From 0c836aacd8be34bcb5e02581495f7d30a5fbee05 Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Tue, 16 Feb 2010 00:58:49 +0100
Subject: [PATCH] alsa: cover "Internal Mic" elements
https://bugzilla.redhat.com/show_bug.cgi?id=562216
---
src/modules/alsa/mixer/paths/analog-input-aux.conf | 4 +
src/modules/alsa/mixer/paths/analog-input-fm.conf | 4 +
.../mixer/paths/analog-input-internal-mic.conf | 67 ++++++++++++++++++++
.../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 +
9 files changed, 98 insertions(+), 0 deletions(-)
create mode 100644 src/modules/alsa/mixer/paths/analog-input-internal-mic.conf
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-internal-mic.conf b/src/modules/alsa/mixer/paths/analog-input-internal-mic.conf
new file mode 100644
index 0000000..70cd512
--- /dev/null
+++ b/src/modules/alsa/mixer/paths/analog-input-internal-mic.conf
@@ -0,0 +1,67 @@
+# This file is part of PulseAudio.
+#
+# PulseAudio is free software; you can redistribute it and/or modify
+# it under the terms of the GNU Lesser General Public License as
+# published by the Free Software Foundation; either version 2.1 of the
+# License, or (at your option) any later version.
+#
+# PulseAudio is distributed in the hope that it will be useful, but
+# WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+# General Public License for more details.
+#
+# You should have received a copy of the GNU Lesser General Public License
+# 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 'Internal Mic' element exists
+;
+; See analog-output.conf.common for an explanation on the directives
+
+[General]
+priority = 90
+name = analog-input-microphone
+
+[Element Capture]
+switch = mute
+volume = merge
+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
+override-map.1 = all
+override-map.2 = all-left,all-right
+
+[Element Line]
+switch = off
+volume = off
+
+[Element Aux]
+switch = off
+volume = off
+
+[Element Video]
+switch = off
+volume = off
+
+[Element Mic/Line]
+switch = off
+volume = off
+
+[Element TV Tuner]
+switch = off
+volume = off
+
+[Element FM]
+switch = off
+volume = off
+
+.include analog-input.conf.common
+.include analog-input-mic.conf.common
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
--
1.6.0.2

82
0037-alsa-use-default-output-port-names.patch Normal file
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@ -0,0 +1,82 @@
From 307bd142feef8f42d3adcea354b72f4c076470e4 Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Tue, 16 Feb 2010 00:59:24 +0100
Subject: [PATCH] alsa: use default output port names
instead of coming up with pointless aliases, reuse the already established
names, for second headphones, and second speakers.
---
src/modules/alsa/alsa-mixer.c | 2 --
.../mixer/paths/analog-output-desktop-speaker.conf | 3 ++-
.../mixer/paths/analog-output-headphones-2.conf | 1 +
.../alsa/mixer/paths/analog-output-headphones.conf | 1 +
.../alsa/mixer/paths/analog-output-speaker.conf | 1 +
5 files changed, 5 insertions(+), 3 deletions(-)
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
--
1.6.0.2

27
0038-build-sys-add-gobject-to-build-dependencies.patch Normal file
View File

@ -0,0 +1,27 @@
From f9b31fe5443bded4a55b0900f08f04486c3395bd Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Tue, 16 Feb 2010 01:43:00 +0100
Subject: [PATCH] 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.
---
configure.ac | 2 +-
1 files changed, 1 insertions(+), 1 deletions(-)
diff --git a/configure.ac b/configure.ac
index c9ec490..1b80788 100644
--- a/configure.ac
+++ b/configure.ac
@@ -893,7 +893,7 @@ 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
--
1.6.0.2

142
0039-padsp-emulate-dev-audio-too.patch Normal file
View File

@ -0,0 +1,142 @@
From ff2474e5fc9eaa4c7f687e71a39a5bed4f56c259 Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Tue, 16 Feb 2010 02:03:45 +0100
Subject: [PATCH] padsp: emulate /dev/audio, too
https://bugzilla.redhat.com/show_bug.cgi?id=561262
---
src/utils/padsp.c | 55 +++++++++++++++++++---------------------------------
1 files changed, 20 insertions(+), 35 deletions(-)
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);
}
--
1.6.0.2

92
0040-dbus-first-restart-timer-then-dispatch-it.patch Normal file
View File

@ -0,0 +1,92 @@
From 96592c21150e789156153dcebd14e201cd41cbeb Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Tue, 16 Feb 2010 02:49:44 +0100
Subject: [PATCH] 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
---
src/pulsecore/dbus-util.c | 29 ++++++++++++++++-------------
1 files changed, 16 insertions(+), 13 deletions(-)
diff --git a/src/pulsecore/dbus-util.c b/src/pulsecore/dbus-util.c
index 4e6148f..40b64a0 100644
--- a/src/pulsecore/dbus-util.c
+++ b/src/pulsecore/dbus-util.c
@@ -44,17 +44,16 @@ 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 */
@@ -131,13 +130,17 @@ 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);
}
}
@@ -207,7 +210,7 @@ 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);
@@ -236,15 +239,15 @@ 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) {
--
1.6.0.2

111
0041-fdsem-be-more-verbose-when-reading-from-eventfd-fai.patch Normal file
View File

@ -0,0 +1,111 @@
From 5e2af2d3f3dea473d46b33a5e90a993ac5da3f35 Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Tue, 16 Feb 2010 03:36:44 +0100
Subject: [PATCH] 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
---
src/pulsecore/fdsem.c | 41 +++++++++++++++++++++++++++++++++++------
1 files changed, 35 insertions(+), 6 deletions(-)
diff --git a/src/pulsecore/fdsem.c b/src/pulsecore/fdsem.c
index 380f34f..00836f9 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
@@ -159,7 +160,12 @@ 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;
@@ -167,7 +173,12 @@ 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;
}
@@ -192,14 +203,22 @@ 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;
}
@@ -228,7 +247,12 @@ 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;
}
@@ -237,7 +261,12 @@ 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;
}
--
1.6.0.2

79
0042-pacat-always-fully-fulfill-write-requests.patch Normal file
View File

@ -0,0 +1,79 @@
From 19fa81bf1375032cb1a27c7715a28a52b238d4cb Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Thu, 18 Feb 2010 01:54:51 +0100
Subject: [PATCH] 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
---
src/utils/pacat.c | 47 ++++++++++++++++++++++++++++++-----------------
1 files changed, 30 insertions(+), 17 deletions(-)
diff --git a/src/utils/pacat.c b/src/utils/pacat.c
index a5d2e9a..d136f6b 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;
+ }
}
}
--
1.6.0.2

60
0043-pacmd-store-away-fd-type.patch Normal file
View File

@ -0,0 +1,60 @@
From 96ec29c6b2a40d3851f46587143a4ba6dce29d0d Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Thu, 18 Feb 2010 02:32:35 +0100
Subject: [PATCH] pacmd: store away fd type
---
src/utils/pacmd.c | 9 +++++----
1 files changed, 5 insertions(+), 4 deletions(-)
diff --git a/src/utils/pacmd.c b/src/utils/pacmd.c
index 5ef57e3..59aae48 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;
}
--
1.6.0.2

257
0044-pacmd-don-t-enter-busy-loop-when-reading-from-stdin.patch Normal file
View File

@ -0,0 +1,257 @@
From 6e064d1d6d0292d230c752b1f41034fd0754487b Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Thu, 18 Feb 2010 03:28:56 +0100
Subject: [PATCH] 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
---
src/utils/pacmd.c | 186 ++++++++++++++++++++++++++++------------------------
1 files changed, 100 insertions(+), 86 deletions(-)
diff --git a/src/utils/pacmd.c b/src/utils/pacmd.c
index 59aae48..87661e3 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 (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 (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;
+ }
}
}
--
1.6.0.2

29
0045-shm-don-t-complain-about-missing-SHM-segments.patch Normal file
View File

@ -0,0 +1,29 @@
From dfe27f623765b20bdaff2306109f89baca67d0cf Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Fri, 19 Feb 2010 03:29:36 +0100
Subject: [PATCH] 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.
---
src/pulsecore/shm.c | 2 +-
1 files changed, 1 insertions(+), 1 deletions(-)
diff --git a/src/pulsecore/shm.c b/src/pulsecore/shm.c
index fbf777a..1e31d54 100644
--- a/src/pulsecore/shm.c
+++ b/src/pulsecore/shm.c
@@ -286,7 +286,7 @@ 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;
}
--
1.6.0.2

25
0046-vala-fix-definition-of-INVALID_INDEX.patch Normal file
View File

@ -0,0 +1,25 @@
From 5ce18c32f6ba8202c9f17f84aaffbdbbd8df071c Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Fri, 19 Feb 2010 03:31:46 +0100
Subject: [PATCH] vala: fix definition of INVALID_INDEX
---
vala/libpulse.vapi | 2 +-
1 files changed, 1 insertions(+), 1 deletions(-)
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")]
--
1.6.0.2

30
0047-vala-fix-definition-of-the-GLib-mainloop-adapter.patch Normal file
View File

@ -0,0 +1,30 @@
From 6bbdd200e89bdbf180071e5281c0feb9617de81d Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Fri, 19 Feb 2010 03:32:00 +0100
Subject: [PATCH] vala: fix definition of the GLib mainloop adapter
---
vala/libpulse.vapi | 4 ++--
1 files changed, 2 insertions(+), 2 deletions(-)
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();
}
--
1.6.0.2

49
0048-Add-missing-profile-and-alsa-mixer-paths-to-src-Make.patch Normal file
View File

@ -0,0 +1,49 @@
From 047e16f2d18c20ea8d9dbb8a24175e1c4bb34d5e Mon Sep 17 00:00:00 2001
From: Daniel T Chen <crimsun@ubuntu.com>
Date: Fri, 19 Feb 2010 00:23:17 -0500
Subject: [PATCH] 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.
---
src/Makefile.am | 3 +++
1 files changed, 3 insertions(+), 0 deletions(-)
diff --git a/src/Makefile.am b/src/Makefile.am
index 598e77f..ffe7558 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -109,6 +109,7 @@ MODULE_LDFLAGS = -module -disable-static -avoid-version $(LDFLAGS_NOUNDEFINED)
ALSA_PROFILES = \
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
@@ -117,6 +118,7 @@ ALSA_PATHS = \
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 \
@@ -126,6 +128,7 @@ ALSA_PATHS = \
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 \
--
1.6.0.2

28
0049-channelmap-Use-Subwoofer-as-pretty-name-for-LFE.patch Normal file
View File

@ -0,0 +1,28 @@
From 3f44bf0afaba9c227f09a475f4886c14813afc2f Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Sun, 21 Feb 2010 16:19:40 +0100
Subject: [PATCH] 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.
---
src/pulse/channelmap.c | 2 +-
1 files changed, 1 insertions(+), 1 deletions(-)
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"),
--
1.6.0.2

34
0050-vala-fix-wrapping-of-port-setting-calls.patch Normal file
View File

@ -0,0 +1,34 @@
From 117c99de3af9a19104d946852821f7b2f5646526 Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Sun, 21 Feb 2010 16:21:13 +0100
Subject: [PATCH] vala: fix wrapping of port setting calls
---
vala/libpulse.vapi | 4 ++--
1 files changed, 2 insertions(+), 2 deletions(-)
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);
--
1.6.0.2

25
0051-proplist-explicitly-mention-a-role-test.patch Normal file
View File

@ -0,0 +1,25 @@
From ddabaa3b0f7b95e2cf5279f5be74769905af8496 Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Sun, 21 Feb 2010 16:21:43 +0100
Subject: [PATCH] proplist: explicitly mention a role 'test'
---
src/pulse/proplist.h | 2 +-
1 files changed, 1 insertions(+), 1 deletions(-)
diff --git a/src/pulse/proplist.h b/src/pulse/proplist.h
index 8dff8df..6283177 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") */
--
1.6.0.2

25
0052-stream-restore-be-a-little-bit-more-verbose-why-we.patch Normal file
View File

@ -0,0 +1,25 @@
From 8adf536426a86e986fb5bd4ad31ba2d50054c485 Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Sun, 21 Feb 2010 16:22:17 +0100
Subject: [PATCH] stream-restore: be a little bit more verbose why we don't reastore a sink
---
src/modules/module-stream-restore.c | 2 +-
1 files changed, 1 insertions(+), 1 deletions(-)
diff --git a/src/modules/module-stream-restore.c b/src/modules/module-stream-restore.c
index 9b6f914..00ce5ab 100644
--- a/src/modules/module-stream-restore.c
+++ b/src/modules/module-stream-restore.c
@@ -394,7 +394,7 @@ 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;
--
1.6.0.2

26
0053-sample-cache-use-the-sample-name-as-unmodified-fall.patch Normal file
View File

@ -0,0 +1,26 @@
From 180589511dab1b5d0388b58b41a0e9843ea1a35a Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Sun, 21 Feb 2010 16:23:25 +0100
Subject: [PATCH] sample-cache: use the sample name as unmodified fallback in the properties when playing a sample from the cache
---
src/pulsecore/core-scache.c | 3 ++-
1 files changed, 2 insertions(+), 1 deletions(-)
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;
--
1.6.0.2

153
0054-scache-when-playing-a-sample-from-the-cache-make-su.patch Normal file
View File

@ -0,0 +1,153 @@
From f9b9579cf4fe65d3619629b5bc28ce02ceca305e Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Sun, 21 Feb 2010 17:35:05 +0100
Subject: [PATCH] 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.
---
src/pulsecore/core-scache.c | 7 ++++++-
src/pulsecore/play-memblockq.c | 7 +++++--
src/pulsecore/play-memblockq.h | 18 ++++++++++--------
src/pulsecore/play-memchunk.c | 3 ++-
src/pulsecore/play-memchunk.h | 15 ++++++++-------
5 files changed, 31 insertions(+), 19 deletions(-)
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
--
1.6.0.2

28
0055-pacat-pass-buffer_attr-to-recording-streams-too.patch Normal file
View File

@ -0,0 +1,28 @@
From b2e9fb6f6e12a3eab8a41c67017507e60d616e2a Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Sun, 21 Feb 2010 21:09:26 +0100
Subject: [PATCH] pacat: pass buffer_attr to recording streams too
Pointed out by Colin Guthrie.
https://tango.0pointer.de/pipermail/pulseaudio-discuss/2010-February/006698.html
---
src/utils/pacat.c | 2 +-
1 files changed, 1 insertions(+), 1 deletions(-)
diff --git a/src/utils/pacat.c b/src/utils/pacat.c
index d136f6b..394cfbf 100644
--- a/src/utils/pacat.c
+++ b/src/utils/pacat.c
@@ -473,7 +473,7 @@ static void context_state_callback(pa_context *c, void *userdata) {
}
} else {
- if (pa_stream_connect_record(stream, device, latency > 0 ? &buffer_attr : NULL, flags) < 0) {
+ if (pa_stream_connect_record(stream, device, &buffer_attr, flags) < 0) {
pa_log(_("pa_stream_connect_record() failed: %s"), pa_strerror(pa_context_errno(c)));
goto fail;
}
--
1.6.0.2

47
0056-suspend-on-idle-resume-audio-device-even-for-initia.patch Normal file
View File

@ -0,0 +1,47 @@
From a469d44e6993c4e9e7a53ac91ed53eacb500e279 Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Mon, 22 Feb 2010 03:22:28 +0100
Subject: [PATCH] suspend-on-idle: resume audio device even for initially corked streams
We need to resume audio devices even for streams that are created in
corked stat, so that the latency ranges of the audio device are known
during the initial latency negotiation. If we don't the latency
negotiation will be based on placeholder data and changed later on which
clients do not expect.
This should fix issues with Skype.
https://bugzilla.redhat.com/show_bug.cgi?id=554929
---
src/modules/module-suspend-on-idle.c | 8 +++-----
1 files changed, 3 insertions(+), 5 deletions(-)
diff --git a/src/modules/module-suspend-on-idle.c b/src/modules/module-suspend-on-idle.c
index 7adaa0b..cfb7879 100644
--- a/src/modules/module-suspend-on-idle.c
+++ b/src/modules/module-suspend-on-idle.c
@@ -145,8 +145,9 @@ static pa_hook_result_t sink_input_fixate_hook_cb(pa_core *c, pa_sink_input_new_
pa_assert(data);
pa_assert(u);
- if (data->flags & PA_SINK_INPUT_START_CORKED)
- return PA_HOOK_OK;
+ /* We need to resume the audio device here even for
+ * PA_SINK_INPUT_START_CORKED, since we need the device parameters
+ * to be fully available while the stream is set up. */
if ((d = pa_hashmap_get(u->device_infos, data->sink)))
resume(d);
@@ -161,9 +162,6 @@ static pa_hook_result_t source_output_fixate_hook_cb(pa_core *c, pa_source_outpu
pa_assert(data);
pa_assert(u);
- if (data->flags & PA_SOURCE_OUTPUT_START_CORKED)
- return PA_HOOK_OK;
-
if (data->source->monitor_of)
d = pa_hashmap_get(u->device_infos, data->source->monitor_of);
else
--
1.6.0.2

48
0057-native-improve-logging-for-buffer_attrs.patch Normal file
View File

@ -0,0 +1,48 @@
From 4a32103d84102ee91368eed1f30613de6da12f6a Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Mon, 22 Feb 2010 03:40:29 +0100
Subject: [PATCH] native: improve logging for buffer_attrs
---
src/pulsecore/protocol-native.c | 18 ++++++++++++++++++
1 files changed, 18 insertions(+), 0 deletions(-)
diff --git a/src/pulsecore/protocol-native.c b/src/pulsecore/protocol-native.c
index f80b0c2..337869d 100644
--- a/src/pulsecore/protocol-native.c
+++ b/src/pulsecore/protocol-native.c
@@ -858,6 +858,18 @@ static void fix_playback_buffer_attr(playback_stream *s) {
pa_assert(s);
+ /* pa_log("Client requested: maxlength=%li bytes tlength=%li bytes minreq=%li bytes prebuf=%li bytes", */
+ /* (long) s->buffer_attr.maxlength, */
+ /* (long) s->buffer_attr.tlength, */
+ /* (long) s->buffer_attr.minreq, */
+ /* (long) s->buffer_attr.prebuf); */
+
+ /* pa_log("Client requested: maxlength=%lu ms tlength=%lu ms minreq=%lu ms prebuf=%lu ms", */
+ /* (unsigned long) (pa_bytes_to_usec(s->buffer_attr.maxlength, &s->sink_input->sample_spec) / PA_USEC_PER_MSEC), */
+ /* (unsigned long) (pa_bytes_to_usec(s->buffer_attr.tlength, &s->sink_input->sample_spec) / PA_USEC_PER_MSEC), */
+ /* (unsigned long) (pa_bytes_to_usec(s->buffer_attr.minreq, &s->sink_input->sample_spec) / PA_USEC_PER_MSEC), */
+ /* (unsigned long) (pa_bytes_to_usec(s->buffer_attr.prebuf, &s->sink_input->sample_spec) / PA_USEC_PER_MSEC)); */
+
/* This function will be called from the main thread, before as
* well as after the sink input has been activated using
* pa_sink_input_put()! That means it may not touch any
@@ -984,6 +996,12 @@ static void fix_playback_buffer_attr(playback_stream *s) {
if (s->buffer_attr.prebuf == (uint32_t) -1 ||
s->buffer_attr.prebuf > max_prebuf)
s->buffer_attr.prebuf = max_prebuf;
+
+ /* pa_log("Client accepted: maxlength=%lu ms tlength=%lu ms minreq=%lu ms prebuf=%lu ms", */
+ /* (unsigned long) (pa_bytes_to_usec(s->buffer_attr.maxlength, &s->sink_input->sample_spec) / PA_USEC_PER_MSEC), */
+ /* (unsigned long) (pa_bytes_to_usec(s->buffer_attr.tlength, &s->sink_input->sample_spec) / PA_USEC_PER_MSEC), */
+ /* (unsigned long) (pa_bytes_to_usec(s->buffer_attr.minreq, &s->sink_input->sample_spec) / PA_USEC_PER_MSEC), */
+ /* (unsigned long) (pa_bytes_to_usec(s->buffer_attr.prebuf, &s->sink_input->sample_spec) / PA_USEC_PER_MSEC)); */
}
/* Called from main context */
--
1.6.0.2

53
0058-alsa-util-strip-spaces-from-ALSA-card-pcm-names.patch Normal file
View File

@ -0,0 +1,53 @@
From 522c446ddc2bf2f4467f9de16f8618fc544a94fe Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Sun, 21 Feb 2010 22:00:49 +0100
Subject: [PATCH] alsa-util: strip spaces from ALSA card/pcm names
http://pulseaudio.org/ticket/778
---
src/modules/alsa/alsa-util.c | 12 +++++++-----
1 files changed, 7 insertions(+), 5 deletions(-)
diff --git a/src/modules/alsa/alsa-util.c b/src/modules/alsa/alsa-util.c
index 52f1259..1cbb3f3 100644
--- a/src/modules/alsa/alsa-util.c
+++ b/src/modules/alsa/alsa-util.c
@@ -874,12 +874,12 @@ void pa_alsa_init_proplist_card(pa_core *c, pa_proplist *p, int card) {
pa_proplist_setf(p, "alsa.card", "%i", card);
if (snd_card_get_name(card, &cn) >= 0) {
- pa_proplist_sets(p, "alsa.card_name", cn);
+ pa_proplist_sets(p, "alsa.card_name", pa_strip(cn));
free(cn);
}
if (snd_card_get_longname(card, &lcn) >= 0) {
- pa_proplist_sets(p, "alsa.long_card_name", lcn);
+ pa_proplist_sets(p, "alsa.long_card_name", pa_strip(lcn));
free(lcn);
}
@@ -937,8 +937,11 @@ void pa_alsa_init_proplist_pcm_info(pa_core *c, pa_proplist *p, snd_pcm_info_t *
if (alsa_subclass_table[subclass])
pa_proplist_sets(p, "alsa.subclass", alsa_subclass_table[subclass]);
- if ((n = snd_pcm_info_get_name(pcm_info)))
- pa_proplist_sets(p, "alsa.name", n);
+ if ((n = snd_pcm_info_get_name(pcm_info))) {
+ char *t = pa_xstrdup(n);
+ pa_proplist_sets(p, "alsa.name", pa_strip(t));
+ pa_xfree(t);
+ }
if ((id = snd_pcm_info_get_id(pcm_info)))
pa_proplist_sets(p, "alsa.id", id);
@@ -1331,6 +1334,5 @@ pa_bool_t pa_alsa_may_tsched(pa_bool_t want) {
return FALSE;
}
-
return TRUE;
}
--
1.6.0.2

48
0059-alsa-reset-max_rewind-max_request-while-suspending.patch Normal file
View File

@ -0,0 +1,48 @@
From dd682d6f96fe7133b866c97b72edcb6136a9737c Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Mon, 22 Feb 2010 04:54:24 +0100
Subject: [PATCH] alsa: reset max_rewind/max_request while suspending
---
src/modules/alsa/alsa-sink.c | 11 ++++++++++-
1 files changed, 10 insertions(+), 1 deletions(-)
diff --git a/src/modules/alsa/alsa-sink.c b/src/modules/alsa/alsa-sink.c
index 690480f..60c10c4 100644
--- a/src/modules/alsa/alsa-sink.c
+++ b/src/modules/alsa/alsa-sink.c
@@ -79,7 +79,7 @@
#define TSCHED_MIN_WAKEUP_USEC (4*PA_USEC_PER_MSEC) /* 4ms -- Wakeup at least this long before the buffer runs empty*/
#define SMOOTHER_MIN_INTERVAL (2*PA_USEC_PER_MSEC) /* 2ms -- min smoother update interval */
-#define SMOOTHER_MAX_INTERVAL (200*PA_USEC_PER_MSEC) /* 200ms -- max smoother update inteval */
+#define SMOOTHER_MAX_INTERVAL (200*PA_USEC_PER_MSEC) /* 200ms -- max smoother update interval */
#define VOLUME_ACCURACY (PA_VOLUME_NORM/100) /* don't require volume adjustments to be perfectly correct. don't necessarily extend granularity in software unless the differences get greater than this level */
@@ -876,6 +876,14 @@ static int suspend(struct userdata *u) {
u->alsa_rtpoll_item = NULL;
}
+ /* We reset max_rewind/max_request here to make sure that while we
+ * are suspended the old max_request/max_rewind values set before
+ * the suspend can influence the per-stream buffer of newly
+ * created streams, without their requirements having any
+ * influence on them. */
+ pa_sink_set_max_rewind_within_thread(u->sink, 0);
+ pa_sink_set_max_request_within_thread(u->sink, 0);
+
pa_log_info("Device suspended...");
return 0;
@@ -933,6 +941,7 @@ static int update_sw_params(struct userdata *u) {
}
pa_sink_set_max_request_within_thread(u->sink, u->hwbuf_size - u->hwbuf_unused);
+ pa_sink_set_max_rewind_within_thread(u->sink, u->hwbuf_size);
return 0;
}
--
1.6.0.2

78
0060-core-util-introduce-generic-function-pa_strip.patch Normal file
View File

@ -0,0 +1,78 @@
From 195069c1815a1483b2c44f8cc72e2dd25e9b8fb1 Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Sun, 21 Feb 2010 21:59:53 +0100
Subject: [PATCH] core-util: introduce generic function pa_strip()
---
src/pulsecore/core-util.c | 28 ++++++++++++++++++++++++----
src/pulsecore/core-util.h | 1 +
2 files changed, 25 insertions(+), 4 deletions(-)
diff --git a/src/pulsecore/core-util.c b/src/pulsecore/core-util.c
index 258e8ee..d6017b9 100644
--- a/src/pulsecore/core-util.c
+++ b/src/pulsecore/core-util.c
@@ -126,6 +126,9 @@
#define MSG_NOSIGNAL 0
#endif
+#define NEWLINE "\r\n"
+#define WHITESPACE "\n\r \t"
+
static pa_strlist *recorded_env = NULL;
#ifdef OS_IS_WIN32
@@ -830,9 +833,6 @@ char *pa_split(const char *c, const char *delimiter, const char**state) {
return pa_xstrndup(current, l);
}
-/* What is interpreted as whitespace? */
-#define WHITESPACE " \t\n"
-
/* Split a string into words. Otherwise similar to pa_split(). */
char *pa_split_spaces(const char *c, const char **state) {
const char *current = *state ? *state : c;
@@ -1189,7 +1189,27 @@ int pa_lock_fd(int fd, int b) {
char* pa_strip_nl(char *s) {
pa_assert(s);
- s[strcspn(s, "\r\n")] = 0;
+ s[strcspn(s, NEWLINE)] = 0;
+ return s;
+}
+
+char *pa_strip(char *s) {
+ char *e, *l = NULL;
+
+ /* Drops trailing whitespace. Modifies the string in
+ * place. Returns pointer to first non-space character */
+
+ s += strspn(s, WHITESPACE);
+
+ for (e = s; *e; e++)
+ if (!strchr(WHITESPACE, *e))
+ l = e;
+
+ if (l)
+ *(l+1) = 0;
+ else
+ *s = 0;
+
return s;
}
diff --git a/src/pulsecore/core-util.h b/src/pulsecore/core-util.h
index eba1b40..71be9de 100644
--- a/src/pulsecore/core-util.h
+++ b/src/pulsecore/core-util.h
@@ -102,6 +102,7 @@ char *pa_split(const char *c, const char*delimiters, const char **state);
char *pa_split_spaces(const char *c, const char **state);
char *pa_strip_nl(char *s);
+char *pa_strip(char *s);
const char *pa_sig2str(int sig) PA_GCC_PURE;
--
1.6.0.2

46
0061-esd-simple-use-pa_memblockq_pop_missing.patch Normal file
View File

@ -0,0 +1,46 @@
From 30f28ebf3619a86b49009e8dbce154233f597dbb Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Tue, 23 Feb 2010 00:48:35 +0100
Subject: [PATCH] esd,simple: use pa_memblockq_pop_missing()
We need to use pa_memblockq_pop_missing() for all request handling,
including the initial request, because otherwise the counters will be
stay off during the entire runtime.
This should fix:
https://bugzilla.redhat.com/show_bug.cgi?id=559467
---
src/pulsecore/protocol-esound.c | 2 +-
src/pulsecore/protocol-simple.c | 2 +-
2 files changed, 2 insertions(+), 2 deletions(-)
diff --git a/src/pulsecore/protocol-esound.c b/src/pulsecore/protocol-esound.c
index a89f327..1e16334 100644
--- a/src/pulsecore/protocol-esound.c
+++ b/src/pulsecore/protocol-esound.c
@@ -462,7 +462,7 @@ static int esd_proto_stream_play(connection *c, esd_proto_t request, const void
c->protocol->n_player++;
- pa_atomic_store(&c->playback.missing, (int) pa_memblockq_missing(c->input_memblockq));
+ pa_atomic_store(&c->playback.missing, (int) pa_memblockq_pop_missing(c->input_memblockq));
pa_sink_input_put(c->sink_input);
diff --git a/src/pulsecore/protocol-simple.c b/src/pulsecore/protocol-simple.c
index fb2e564..77277e1 100644
--- a/src/pulsecore/protocol-simple.c
+++ b/src/pulsecore/protocol-simple.c
@@ -574,7 +574,7 @@ void pa_simple_protocol_connect(pa_simple_protocol *p, pa_iochannel *io, pa_simp
pa_iochannel_socket_set_rcvbuf(io, l);
- pa_atomic_store(&c->playback.missing, (int) pa_memblockq_missing(c->input_memblockq));
+ pa_atomic_store(&c->playback.missing, (int) pa_memblockq_pop_missing(c->input_memblockq));
pa_sink_input_put(c->sink_input);
}
--
1.6.0.2

72
0062-core-rework-how-stream-volumes-affect-sink-volumes.patch Normal file
View File

@ -0,0 +1,72 @@
From 0d1154d078f8f149e2ab9a67b09835a829416e49 Mon Sep 17 00:00:00 2001
From: Lennart Poettering <lennart@poettering.net>
Date: Tue, 23 Feb 2010 18:41:18 +0100
Subject: [PATCH] core: rework how stream volumes affect sink volumes
---
src/pulsecore/sink.c | 42 +++++++++++++++++++++++++++++++++++++++++-
1 files changed, 41 insertions(+), 1 deletions(-)
diff --git a/src/pulsecore/sink.c b/src/pulsecore/sink.c
index e6d718f..3a92f67 100644
--- a/src/pulsecore/sink.c
+++ b/src/pulsecore/sink.c
@@ -1162,6 +1162,46 @@ pa_usec_t pa_sink_get_latency_within_thread(pa_sink *s) {
return usec;
}
+static pa_cvolume* cvolume_remap_minimal_impact(
+ pa_cvolume *v,
+ const pa_cvolume *template,
+ const pa_channel_map *from,
+ const pa_channel_map *to) {
+
+ pa_cvolume t;
+
+ pa_assert(v);
+ pa_assert(template);
+ pa_assert(from);
+ pa_assert(to);
+
+ pa_return_val_if_fail(pa_cvolume_compatible_with_channel_map(v, from), NULL);
+ pa_return_val_if_fail(pa_cvolume_compatible_with_channel_map(template, to), NULL);
+
+ /* Much like pa_cvolume_remap(), but tries to minimize impact when
+ * mapping from sink input to sink volumes:
+ *
+ * If template is a possible remapping from v it is used instead
+ * of remapping anew.
+ *
+ * If the channel maps don't match we set an all-channel volume on
+ * the sink to ensure that changing a volume on one stream has no
+ * effect that cannot be compensated for in another stream that
+ * does not have the same channel map as the sink. */
+
+ if (pa_channel_map_equal(from, to))
+ return v;
+
+ t = *template;
+ if (pa_cvolume_equal(pa_cvolume_remap(&t, to, from), v)) {
+ *v = *template;
+ return v;
+ }
+
+ pa_cvolume_set(v, to->channels, pa_cvolume_max(v));
+ return v;
+}
+
/* Called from main context */
static void compute_reference_ratios(pa_sink *s) {
uint32_t idx;
@@ -1289,7 +1329,7 @@ static void compute_real_volume(pa_sink *s) {
pa_cvolume remapped;
remapped = i->volume;
- pa_cvolume_remap(&remapped, &i->channel_map, &s->channel_map);
+ cvolume_remap_minimal_impact(&remapped, &s->real_volume, &i->channel_map, &s->channel_map);
pa_cvolume_merge(&s->real_volume, &s->real_volume, &remapped);
}
--
1.6.0.2

7
pulseaudio.changes
View File

@ -1,3 +1,10 @@
-------------------------------------------------------------------
Mon Mar 1 21:26:48 UTC 2010 - sreeves@novell.com
- Add another set of patches backported from upstream git
[0033 - 0062].patch
Fixes for latency settings, cpu utilization, stuck playback,...
-------------------------------------------------------------------
Tue Jan 19 22:13:22 UTC 2010 - sreeves@novell.com

63
pulseaudio.spec
View File

@ -22,7 +22,7 @@
Name: pulseaudio
Summary: A Networked Sound Server
Version: 0.9.21
Release: 2
Release: 3
License: LGPLv2.1+
Group: System/Sound Daemons
Source: %{name}-%{version}.tar.bz2
@ -62,6 +62,36 @@ Patch29: 0029-More-src-pulsecore-cpu-arm.c-FTBFS-fixes.patch
Patch30: 0030-Fix-the-following-warnings-which-now-cause-buildd-f.patch
Patch31: 0031-libpulse-Store-pa_stream-pointers-to-hashmaps-inste.patch
Patch32: 0032-native-rework-handling-of-seeks-that-depend-on-vari.patch
Patch33: 0033-core-Fix-macro-typo-PA_SINK_IS_LINKED-PA_SINK_.patch
Patch34: 0034-alsa-cover-Desktop-Speaker-mixer-elements.patch
Patch35: 0035-alsa-cover-Shared-Mic-Line-in-Analog-Source.patch
Patch36: 0036-alsa-cover-Internal-Mic-elements.patch
Patch37: 0037-alsa-use-default-output-port-names.patch
Patch38: 0038-build-sys-add-gobject-to-build-dependencies.patch
Patch39: 0039-padsp-emulate-dev-audio-too.patch
Patch40: 0040-dbus-first-restart-timer-then-dispatch-it.patch
Patch41: 0041-fdsem-be-more-verbose-when-reading-from-eventfd-fai.patch
Patch42: 0042-pacat-always-fully-fulfill-write-requests.patch
Patch43: 0043-pacmd-store-away-fd-type.patch
Patch44: 0044-pacmd-don-t-enter-busy-loop-when-reading-from-stdin.patch
Patch45: 0045-shm-don-t-complain-about-missing-SHM-segments.patch
Patch46: 0046-vala-fix-definition-of-INVALID_INDEX.patch
Patch47: 0047-vala-fix-definition-of-the-GLib-mainloop-adapter.patch
Patch48: 0048-Add-missing-profile-and-alsa-mixer-paths-to-src-Make.patch
Patch49: 0049-channelmap-Use-Subwoofer-as-pretty-name-for-LFE.patch
Patch50: 0050-vala-fix-wrapping-of-port-setting-calls.patch
Patch51: 0051-proplist-explicitly-mention-a-role-test.patch
Patch52: 0052-stream-restore-be-a-little-bit-more-verbose-why-we.patch
Patch53: 0053-sample-cache-use-the-sample-name-as-unmodified-fall.patch
Patch54: 0054-scache-when-playing-a-sample-from-the-cache-make-su.patch
Patch55: 0055-pacat-pass-buffer_attr-to-recording-streams-too.patch
Patch56: 0056-suspend-on-idle-resume-audio-device-even-for-initia.patch
Patch57: 0057-native-improve-logging-for-buffer_attrs.patch
Patch58: 0058-alsa-util-strip-spaces-from-ALSA-card-pcm-names.patch
Patch59: 0059-alsa-reset-max_rewind-max_request-while-suspending.patch
Patch60: 0060-core-util-introduce-generic-function-pa_strip.patch
Patch61: 0061-esd-simple-use-pa_memblockq_pop_missing.patch
Patch62: 0062-core-rework-how-stream-volumes-affect-sink-volumes.patch
Url: http://pulseaudio.org
BuildRoot: %{_tmppath}/%{name}-%{version}-build
BuildRequires: alsa-devel
@ -304,8 +334,39 @@ This package contains GDM integration hooks for the PulseAudio sound server.
%patch30 -p1
%patch31 -p1
%patch32 -p1
%patch33 -p1
%patch34 -p1
%patch35 -p1
%patch36 -p1
%patch37 -p1
%patch38 -p1
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%patch56 -p1
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%patch58 -p1
%patch59 -p1
%patch60 -p1
%patch61 -p1
%patch62 -p1
%build
autoreconf
export CFLAGS="$RPM_OPT_FLAGS -fno-strict-aliasing"
# libpulse and libpulsecommon need each other - no way with as-needed
export SUSE_ASNEEDED=0