diff options
29 files changed, 423 insertions, 155 deletions
diff --git a/configure.ac b/configure.ac index 511e17eb..2ea4d893 100644 --- a/configure.ac +++ b/configure.ac @@ -29,12 +29,12 @@ m4_define(PA_MICRO, [11]) AC_INIT([pulseaudio], PA_MAJOR.PA_MINOR.PA_MICRO,[mzchyfrnhqvb (at) 0pointer (dot) net]) AC_CONFIG_SRCDIR([src/daemon/main.c]) AC_CONFIG_HEADERS([config.h]) -AM_INIT_AUTOMAKE([foreign -Wall]) +AM_INIT_AUTOMAKE([foreign 1.10 -Wall]) AC_SUBST(PA_MAJORMINOR, "PA_MAJOR.PA_MINOR") AC_SUBST(PACKAGE_URL, [http://pulseaudio.org/]) -AC_SUBST(PA_API_VERSION, 11) +AC_SUBST(PA_API_VERSION, 12) AC_SUBST(PA_PROTOCOL_VERSION, 13) # The stable ABI for client applications, for the version info x:y:z @@ -376,7 +376,7 @@ AC_SEARCH_LIBS([connect], [socket]) AC_CHECK_FUNCS([getopt_long], [], [AC_CHECK_LIB([iberty], [getopt_long])]) AC_CHECK_LIB(gdbm, gdbm_open) -AC_CHECK_HEADERS(gdbm.h) +AC_CHECK_HEADERS(gdbm.h, [], [AC_MSG_ERROR([gdbm.h not found])]) #### Check for functions #### @@ -599,7 +599,7 @@ AC_ARG_ENABLE([alsa], [alsa=auto]) if test "x${alsa}" != xno ; then - PKG_CHECK_MODULES(ASOUNDLIB, [ alsa >= 1.0.16 ], + PKG_CHECK_MODULES(ASOUNDLIB, [ alsa >= 1.0.17 ], [ HAVE_ALSA=1 AC_DEFINE([HAVE_ALSA], 1, [Have ALSA?]) @@ -607,7 +607,7 @@ if test "x${alsa}" != xno ; then [ HAVE_ALSA=0 if test "x$alsa" = xyes ; then - AC_MSG_ERROR([*** ALSA support not found]) + AC_MSG_ERROR([*** Needed alsa >= 1.0.17 support not found]) fi ]) else @@ -693,7 +693,7 @@ AC_ARG_ENABLE([gconf], *) AC_MSG_ERROR(bad value ${enableval} for --disable-gconf) ;; esac ], - [glib=auto]) + [gconf=auto]) if test "x${gconf}" != xno ; then PKG_CHECK_MODULES(GCONF, [ gconf-2.0 >= 2.4.0 ], diff --git a/doxygen/.gitignore b/doxygen/.gitignore index ca9b4441..a709d164 100644 --- a/doxygen/.gitignore +++ b/doxygen/.gitignore @@ -1 +1,2 @@ doxygen.conf +html diff --git a/src/.gitignore b/src/.gitignore index f3ed2e2e..6902eb9f 100644 --- a/src/.gitignore +++ b/src/.gitignore @@ -15,6 +15,7 @@ cpulimit-test cpulimit-test2 daemon.conf default.pa +system.pa envelope-test esdcompat flist-test @@ -54,3 +55,4 @@ thread-mainloop-test thread-test utf8-test voltest +start-pulseaudio-x11 diff --git a/src/Makefile.am b/src/Makefile.am index 788857cd..51fd9976 100644 --- a/src/Makefile.am +++ b/src/Makefile.am @@ -96,23 +96,26 @@ EXTRA_DIST = \ pulse/version.h.in \ daemon/daemon.conf.in \ daemon/default.pa.in \ + daemon/system.pa.in \ daemon/default.pa.win32 \ depmod.py \ daemon/esdcompat.in \ + daemon/start-pulseaudio-x11.in \ utils/padsp \ modules/module-defs.h.m4 \ - daemon/pulseaudio-module-xsmp.desktop \ + daemon/pulseaudio.desktop \ map-file \ daemon/org.pulseaudio.policy pulseconf_DATA = \ default.pa \ + system.pa \ daemon.conf \ client.conf if HAVE_X11 xdgautostart_DATA = \ - daemon/pulseaudio-module-xsmp.desktop + daemon/pulseaudio.desktop endif BUILT_SOURCES = \ @@ -184,7 +187,7 @@ if HAVE_AVAHI bin_PROGRAMS += pabrowse endif -bin_SCRIPTS = esdcompat +bin_SCRIPTS = esdcompat start-pulseaudio-x11 pacat_SOURCES = utils/pacat.c pacat_LDADD = $(AM_LDADD) libpulse.la @@ -585,7 +588,7 @@ endif libpulse_la_CFLAGS = $(AM_CFLAGS) libpulse_la_LDFLAGS = -version-info $(LIBPULSE_VERSION_INFO) -Wl,-version-script=$(srcdir)/map-file -libpulse_la_LIBADD = $(AM_LIBADD) $(WINSOCK_LIBS) $(LIBICONV) +libpulse_la_LIBADD = $(AM_LIBADD) $(WINSOCK_LIBS) $(LTLIBICONV) if HAVE_X11 libpulse_la_CFLAGS += $(X_CFLAGS) @@ -801,7 +804,7 @@ endif libpulsecore_la_CPPFLAGS = $(AM_CPPFLAGS) $(LIBOIL_CFLAGS) libpulsecore_la_LDFLAGS = -version-info $(LIBPULSECORE_VERSION_INFO) -libpulsecore_la_LIBADD = $(AM_LIBADD) $(LIBLTDL) $(LIBSAMPLERATE_LIBS) $(LIBSNDFILE_LIBS) $(LIBSPEEX_LIBS) $(WINSOCK_LIBS) $(LIBOIL_LIBS) $(LIBICONV) libffmpeg-resampler.la +libpulsecore_la_LIBADD = $(AM_LIBADD) $(LIBLTDL) $(LIBSAMPLERATE_LIBS) $(LIBSNDFILE_LIBS) $(LIBSPEEX_LIBS) $(WINSOCK_LIBS) $(LIBOIL_LIBS) $(LTLIBICONV) libffmpeg-resampler.la ################################### # Plug-in support libraries # @@ -1542,7 +1545,7 @@ suid: pulseaudio .libs/lt-pulseaudio chown root $^ chmod u+s $^ -CLEANFILES = esdcompat client.conf default.pa daemon.conf +CLEANFILES = esdcompat client.conf default.pa system.pa daemon.conf start-pulseaudio-x11 esdcompat: daemon/esdcompat.in Makefile sed -e 's,@PACKAGE_VERSION\@,$(PACKAGE_VERSION),g' \ @@ -1550,17 +1553,28 @@ esdcompat: daemon/esdcompat.in Makefile -e 's,@PA_BINARY\@,$(PA_BINARY),g' < $< > $@ chmod +x esdcompat +start-pulseaudio-x11: daemon/start-pulseaudio-x11.in Makefile + sed -e 's,@PA_BINARY\@,$(PA_BINARY),g' \ + -e 's,@PACTL_BINARY\@,$(bindir)/pactl,g' < $< > $@ + chmod +x start-pulseaudio-x11 + client.conf: pulse/client.conf.in Makefile sed -e 's,@PA_BINARY\@,$(PA_BINARY),g' < $< > $@ if OS_IS_WIN32 default.pa: daemon/default.pa.win32 cp $< $@ +system.pa: daemon/default.pa.win32 + cp $< $@ else default.pa: daemon/default.pa.in Makefile sed -e 's,@PA_BINARY\@,$(PA_BINARY),g' \ -e 's,@PA_DLSEARCHPATH\@,$(modlibexecdir),g' \ -e 's,@PA_SOEXT\@,.so,g' < $< > $@ +system.pa: daemon/system.pa.in Makefile + sed -e 's,@PA_BINARY\@,$(PA_BINARY),g' \ + -e 's,@PA_DLSEARCHPATH\@,$(modlibexecdir),g' \ + -e 's,@PA_SOEXT\@,.so,g' < $< > $@ endif daemon.conf: daemon/daemon.conf.in Makefile diff --git a/src/daemon/daemon-conf.c b/src/daemon/daemon-conf.c index 9ac40901..50b812dc 100644 --- a/src/daemon/daemon-conf.c +++ b/src/daemon/daemon-conf.c @@ -62,7 +62,7 @@ static const pa_daemon_conf default_conf = { .realtime_scheduling = FALSE, .realtime_priority = 5, /* Half of JACK's default rtprio */ .disallow_module_loading = FALSE, - .exit_idle_time = -1, + .exit_idle_time = 20, .module_idle_time = 20, .scache_idle_time = 20, .auto_log_target = 1, diff --git a/src/daemon/daemon.conf.in b/src/daemon/daemon.conf.in index dfabcfb2..41c26e2a 100644 --- a/src/daemon/daemon.conf.in +++ b/src/daemon/daemon.conf.in @@ -32,7 +32,7 @@ ; realtime-scheduling = no ; realtime-priority = 5 -; exit-idle-time = -1 +; exit-idle-time = 20 ; module-idle-time = 20 ; scache-idle-time = 20 diff --git a/src/daemon/default.pa.in b/src/daemon/default.pa.in index aad9f5d6..917fc309 100755 --- a/src/daemon/default.pa.in +++ b/src/daemon/default.pa.in @@ -16,6 +16,9 @@ # along with PulseAudio; if not, write to the Free Software Foundation, # Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. +# This startup script is used only if PulseAudio is started per-user +# (i.e. not in system mode) + .nofail ### Load something into the sample cache @@ -80,14 +83,6 @@ load-module module-always-sink ### Automatically suspend sinks/sources that become idle for too long load-module module-suspend-on-idle -### Load X11 bell module -#load-module module-x11-bell sample=bell-windowing-system - -### Register ourselves in the X11 session manager -# Deactivated by default, to avoid deadlock when PA is started as esd from gnome-session -# Instead we load this via /etc/xdg/autostart/ and "pactl load-module" now -#load-module module-x11-xsmp - ### If autoexit on idle is enabled we want to make sure we only quit ### when no local session needs us anymore. load-module module-console-kit @@ -104,12 +99,21 @@ load-module module-gconf .fail .endif +# X11 modules should not be started from default.pa so that one daemon +# can be shared by multiple sessions. + +### Load X11 bell module +#load-module module-x11-bell sample=bell-windowing-system + +### Register ourselves in the X11 session manager +#load-module module-x11-xsmp + ### Publish connection data in the X11 root window -.ifexists module-x11-publish@PA_SOEXT@ -.nofail -load-module module-x11-publish -.fail -.endif +#.ifexists module-x11-publish@PA_SOEXT@ +#.nofail +#load-module module-x11-publish +#.fail +#.endif ### Make some devices default #set-default-sink output diff --git a/src/daemon/main.c b/src/daemon/main.c index 14594416..5fc9f01c 100644 --- a/src/daemon/main.c +++ b/src/daemon/main.c @@ -496,7 +496,7 @@ int main(int argc, char *argv[]) { if (conf->high_priority && (conf->cmd == PA_CMD_DAEMON || conf->cmd == PA_CMD_START)) pa_raise_priority(conf->nice_level); - if (pa_have_caps()) { + if (!real_root && pa_have_caps()) { pa_bool_t drop; drop = (conf->cmd != PA_CMD_DAEMON && conf->cmd != PA_CMD_START) || !conf->realtime_scheduling; @@ -627,6 +627,11 @@ int main(int argc, char *argv[]) { goto finish; } + if (conf->cmd == PA_CMD_START && conf->system_instance) { + pa_log("--start not supported for system instances."); + goto finish; + } + if (conf->cmd == PA_CMD_START) { /* If we shall start PA only when it is not running yet, we * first take the autospawn lock to make things @@ -762,8 +767,6 @@ int main(int argc, char *argv[]) { /* If we are already running and with are run in * --start mode, then let's return this as success. */ - pa_log_info("z=%i rock!", z); - retval = 0; goto finish; } diff --git a/src/daemon/pulseaudio-module-xsmp.desktop b/src/daemon/pulseaudio-module-xsmp.desktop deleted file mode 100644 index fa719a73..00000000 --- a/src/daemon/pulseaudio-module-xsmp.desktop +++ /dev/null @@ -1,10 +0,0 @@ -[Desktop Entry] -Version=1.0 -Encoding=UTF-8 -Name=PulseAudio Session Management -Comment=Load module-x11-xsmp into PulseAudio -Exec=pactl load-module module-x11-xsmp -Terminal=false -Type=Application -Categories= -GenericName= diff --git a/src/daemon/pulseaudio.desktop b/src/daemon/pulseaudio.desktop new file mode 100644 index 00000000..57a7a6e4 --- /dev/null +++ b/src/daemon/pulseaudio.desktop @@ -0,0 +1,10 @@ +[Desktop Entry] +Version=1.0 +Encoding=UTF-8 +Name=PulseAudio Sound System +Comment=Start the PulseAudio Sound System +Exec=start-pulseaudio-x11 +Terminal=false +Type=Application +Categories= +GenericName= diff --git a/src/daemon/start-pulseaudio-x11.in b/src/daemon/start-pulseaudio-x11.in new file mode 100755 index 00000000..3cccc4dc --- /dev/null +++ b/src/daemon/start-pulseaudio-x11.in @@ -0,0 +1,31 @@ +#!/bin/sh + +# 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 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. + +set -e + +@PA_BINARY@ --start "$@" + +if [ x"$DISPLAY" != x ] ; then + + @PACTL_BINARY@ load-module module-x11-publish "display=$DISPLAY" > /dev/null + + if [ x"$SESSION_MANAGER" != x ] ; then + @PACTL_BINARY@ load-module module-x11-xsmp "display=$DISPLAY session_manager=$SESSION_MANAGER" > /dev/null + fi +fi diff --git a/src/daemon/system.pa.in b/src/daemon/system.pa.in new file mode 100755 index 00000000..f6052c4b --- /dev/null +++ b/src/daemon/system.pa.in @@ -0,0 +1,54 @@ +#!@PA_BINARY@ -nF +# +# 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 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. + +# This startup script is used only if PulseAudio is started in system +# mode. + +### Automatically load driver modules depending on the hardware available +.ifexists module-hal-detect@PA_SOEXT@ +load-module module-hal-detect +.else +### Alternatively use the static hardware detection module (for systems that +### lack HAL support) +load-module module-detect +.endif + +### Load several protocols +.ifexists module-esound-protocol-unix@PA_SOEXT@ +load-module module-esound-protocol-unix +.endif +load-module module-native-protocol-unix + +### Automatically restore the volume of playback streams +load-module module-volume-restore + +### Automatically restore the default sink/source when changed by the user during runtime +load-module module-default-device-restore + +### Automatically move streams to the default sink if the sink they are +### connected to dies, similar for sources +load-module module-rescue-streams + +### Make sure we always have a sink around, even if it is a null sink. +load-module module-always-sink + +### Automatically suspend sinks/sources that become idle for too long +load-module module-suspend-on-idle + +### Enable positioned event sounds +load-module module-position-event-sounds diff --git a/src/modules/module-bt-proximity.c b/src/modules/module-bt-proximity.c index 77b95868..f924c3cb 100644 --- a/src/modules/module-bt-proximity.c +++ b/src/modules/module-bt-proximity.c @@ -358,7 +358,6 @@ static int add_matches(struct userdata *u, pa_bool_t add) { } else dbus_bus_remove_match(pa_dbus_connection_get(u->dbus_connection), filter2, &e); - if (add) pa_assert_se(dbus_connection_add_filter(pa_dbus_connection_get(u->dbus_connection), filter_func, u, NULL)); else diff --git a/src/modules/module-console-kit.c b/src/modules/module-console-kit.c index 3adee99e..e1933c28 100644 --- a/src/modules/module-console-kit.c +++ b/src/modules/module-console-kit.c @@ -71,7 +71,7 @@ struct userdata { static void add_session(struct userdata *u, const char *id) { DBusError error; DBusMessage *m = NULL, *reply = NULL; - int32_t uid; + uint32_t uid; struct session *session; char *t; @@ -92,10 +92,14 @@ static void add_session(struct userdata *u, const char *id) { goto fail; } - /* FIXME: Why is this in int32? and not an uint32? */ - if (!dbus_message_get_args(reply, &error, DBUS_TYPE_INT32, &uid, DBUS_TYPE_INVALID)) { - pa_log("Failed to parse GetUnixUser() result: %s: %s", error.name, error.message); - goto fail; + /* CK 0.3 this changed from int32 to uint32 */ + if (!dbus_message_get_args(reply, &error, DBUS_TYPE_UINT32, &uid, DBUS_TYPE_INVALID)) { + dbus_error_free(&error); + + if (!dbus_message_get_args(reply, &error, DBUS_TYPE_INT32, &uid, DBUS_TYPE_INVALID)) { + pa_log("Failed to parse GetUnixUser() result: %s: %s", error.name, error.message); + goto fail; + } } /* We only care about our own sessions */ @@ -163,27 +167,39 @@ static DBusHandlerResult filter_cb(DBusConnection *bus, DBusMessage *message, vo if (dbus_message_is_signal(message, "org.freedesktop.ConsoleKit.Seat", "SessionAdded")) { - if (!dbus_message_get_args(message, &error, DBUS_TYPE_STRING, &path, DBUS_TYPE_INVALID) || dbus_error_is_set(&error)) { - pa_log_error("Failed to parse SessionAdded message: %s: %s", error.name, error.message); - goto finish; + /* CK API changed to match spec in 0.3 */ + if (!dbus_message_get_args(message, &error, DBUS_TYPE_OBJECT_PATH, &path, DBUS_TYPE_INVALID)) { + dbus_error_free(&error); + + if (!dbus_message_get_args(message, &error, DBUS_TYPE_STRING, &path, DBUS_TYPE_INVALID)) { + pa_log_error("Failed to parse SessionAdded message: %s: %s", error.name, error.message); + goto finish; + } } add_session(u, path); + return DBUS_HANDLER_RESULT_HANDLED; } else if (dbus_message_is_signal(message, "org.freedesktop.ConsoleKit.Seat", "SessionRemoved")) { - if (!dbus_message_get_args(message, &error, DBUS_TYPE_STRING, &path, DBUS_TYPE_INVALID) || dbus_error_is_set(&error)) { - pa_log_error("Failed to parse SessionRemoved message: %s: %s", error.name, error.message); - goto finish; + /* CK API changed to match spec in 0.3 */ + if (!dbus_message_get_args(message, &error, DBUS_TYPE_OBJECT_PATH, &path, DBUS_TYPE_INVALID)) { + dbus_error_free(&error); + + if (!dbus_message_get_args(message, &error, DBUS_TYPE_STRING, &path, DBUS_TYPE_INVALID)) { + pa_log_error("Failed to parse SessionRemoved message: %s: %s", error.name, error.message); + goto finish; + } } remove_session(u, path); + return DBUS_HANDLER_RESULT_HANDLED; } finish: dbus_error_free(&error); - return DBUS_HANDLER_RESULT_HANDLED; + return DBUS_HANDLER_RESULT_NOT_YET_HANDLED; } static int get_session_list(struct userdata *u) { @@ -327,8 +343,17 @@ void pa__done(pa_module *m) { pa_hashmap_free(u->sessions, NULL, NULL); } - if (u->connection) + if (u->connection) { + DBusError error; + dbus_error_init(&error); + + dbus_bus_remove_match(pa_dbus_connection_get(u->connection), "type='signal',sender='org.freedesktop.ConsoleKit', interface='org.freedesktop.ConsoleKit.Seat'", &error); + dbus_error_free(&error); + + dbus_connection_remove_filter(pa_dbus_connection_get(u->connection), filter_cb, u); + pa_dbus_connection_unref(u->connection); + } pa_xfree(u); } diff --git a/src/modules/module-hal-detect.c b/src/modules/module-hal-detect.c index 19430a3d..ab5b206f 100644 --- a/src/modules/module-hal-detect.c +++ b/src/modules/module-hal-detect.c @@ -615,6 +615,8 @@ static DBusHandlerResult filter_cb(DBusConnection *bus, DBusMessage *message, vo device_added_cb(u->context, udi); } + return DBUS_HANDLER_RESULT_HANDLED; + } else if (dbus_message_is_signal(message, "org.pulseaudio.Server", "DirtyGiveUpMessage")) { /* We use this message to avoid a dirty race condition when we get an ACLAdded message before the previously owning PA @@ -661,12 +663,14 @@ static DBusHandlerResult filter_cb(DBusConnection *bus, DBusMessage *message, vo } else /* Yes, we don't check the UDI for validity, but hopefully HAL will */ device_added_cb(u->context, udi); + + return DBUS_HANDLER_RESULT_HANDLED; } finish: dbus_error_free(&error); - return DBUS_HANDLER_RESULT_HANDLED; + return DBUS_HANDLER_RESULT_NOT_YET_HANDLED; } static void hal_context_free(LibHalContext* hal_context) { @@ -842,8 +846,20 @@ void pa__done(pa_module *m) { if (u->devices) pa_hashmap_free(u->devices, hal_device_free_cb, NULL); - if (u->connection) + if (u->connection) { + DBusError error; + dbus_error_init(&error); + + dbus_bus_remove_match(pa_dbus_connection_get(u->connection), "type='signal',sender='org.freedesktop.Hal', interface='org.freedesktop.Hal.Device.AccessControl'", &error); + dbus_error_free(&error); + + dbus_bus_remove_match(pa_dbus_connection_get(u->connection), "type='signal',interface='org.pulseaudio.Server'", &error); + dbus_error_free(&error); + + dbus_connection_remove_filter(pa_dbus_connection_get(u->connection), filter_cb, u); + pa_dbus_connection_unref(u->connection); + } pa_xfree(u); } diff --git a/src/modules/module-x11-xsmp.c b/src/modules/module-x11-xsmp.c index 696826d8..0b2e375a 100644 --- a/src/modules/module-x11-xsmp.c +++ b/src/modules/module-x11-xsmp.c @@ -47,7 +47,7 @@ PA_MODULE_AUTHOR("Lennart Poettering"); PA_MODULE_DESCRIPTION("X11 session management"); PA_MODULE_VERSION(PACKAGE_VERSION); -PA_MODULE_LOAD_ONCE(TRUE); +PA_MODULE_LOAD_ONCE(FALSE); PA_MODULE_USAGE("session_manager=<session manager string> display=<X11 display>"); static pa_bool_t ice_in_use = FALSE; diff --git a/src/pulse/channelmap.h b/src/pulse/channelmap.h index 2551eae9..7c32b868 100644 --- a/src/pulse/channelmap.h +++ b/src/pulse/channelmap.h @@ -47,8 +47,10 @@ * * \li pa_channel_map_init_mono() - Create a channel map with only mono audio. * \li pa_channel_map_init_stereo() - Create a standard stereo mapping. - * \li pa_channel_map_init_auto() - Create a standard channel map for up to - * six channels. + * \li pa_channel_map_init_auto() - Create a standard channel map for a specific number of channels + * \li pa_channel_map_init_extend() - Similar to + * pa_channel_map_init_auto() but synthesize a channel map if noone + * predefined one is known for the specified number of channels. * * \section conv_sec Convenience Functions * diff --git a/src/pulse/client-conf.c b/src/pulse/client-conf.c index 2ead871f..915d0ccb 100644 --- a/src/pulse/client-conf.c +++ b/src/pulse/client-conf.c @@ -66,7 +66,7 @@ pa_client_conf *pa_client_conf_new(void) { pa_client_conf *c = pa_xmemdup(&default_conf, sizeof(default_conf)); c->daemon_binary = pa_xstrdup(PA_BINARY); - c->extra_arguments = pa_xstrdup("--log-target=syslog --exit-idle-time=5"); + c->extra_arguments = pa_xstrdup("--log-target=syslog"); c->cookie_file = pa_xstrdup(PA_NATIVE_COOKIE_FILE); return c; diff --git a/src/pulse/client.conf.in b/src/pulse/client.conf.in index 749e9688..8339d651 100644 --- a/src/pulse/client.conf.in +++ b/src/pulse/client.conf.in @@ -25,7 +25,7 @@ ; autospawn = yes ; daemon-binary = @PA_BINARY@ -; extra-arguments = --log-target=syslog --exit-idle-time=5 +; extra-arguments = --log-target=syslog ; cookie-file = diff --git a/src/pulse/context.h b/src/pulse/context.h index 8dff7642..3b513976 100644 --- a/src/pulse/context.h +++ b/src/pulse/context.h @@ -35,7 +35,7 @@ * \section overv_sec Overview * * The asynchronous API is the native interface to the PulseAudio library. - * It allows full access to all available functions. This also means that + * It allows full access to all available functionality. This however means that * it is rather complex and can take some time to fully master. * * \section mainloop_sec Main Loop Abstraction @@ -64,8 +64,7 @@ * implementation where all of PulseAudio's * internal handling runs in a separate * thread. - * \li \subpage glib-mainloop - A wrapper around GLIB's main loop. Available - * for both GLIB 1.2 and GLIB 2.x. + * \li \subpage glib-mainloop - A wrapper around GLib's main loop. * * UNIX signals may be hooked to a main loop using the functions from * \ref mainloop-signal.h. These rely only on the main loop abstraction diff --git a/src/pulse/def.h b/src/pulse/def.h index 8407ddeb..95071978 100644 --- a/src/pulse/def.h +++ b/src/pulse/def.h @@ -233,12 +233,11 @@ typedef enum pa_stream_flags { PA_STREAM_START_MUTED = 4096, /**< Create in muted state. \since 0.9.11 */ - PA_STREAM_ADJUST_LATENCY = 8192, /**< Try to adjust the latency of * the sink/source based on the * requested buffer metrics and * adjust buffer metrics - * accordingly. \since 0.9.11 */ + * accordingly. See pa_buffer_attr \since 0.9.11 */ } pa_stream_flags_t; @@ -248,53 +247,86 @@ typedef enum pa_stream_flags { /** Playback and record buffer metrics */ typedef struct pa_buffer_attr { uint32_t maxlength; /**< Maximum length of the - * buffer. Setting this to 0 will + * buffer. Setting this to (uint32_t) -1 will * initialize this to the maximum value * supported by server, which is * recommended. */ uint32_t tlength; /**< Playback only: target length of the * buffer. The server tries to assure * that at least tlength bytes are always - * available in the buffer. It is - * recommended to set this to 0, which - * will initialize this to a value that - * is deemed sensible by the + * available in the per-stream + * server-side playback buffer. It is + * recommended to set this to (uint32_t) + * -1, which will initialize this to a + * value that is deemed sensible by the * server. However, this value will * default to something like 2s, i.e. for * applications that have specific * latency requirements this value should * be set to the maximum latency that the - * application can deal with. */ + * application can deal with. When + * PA_STREAM_ADJUST_LATENCY is not set + * this value will influence only the + * per-stream playback buffer size. When + * PA_STREAM_ADJUST_LATENCY is set the + * overall latency of the sink plus the + * playback buffer size is configured to + * this value. Set + * PA_STREAM_ADJUST_LATENCY if you are + * interested in adjusting the overall + * latency. Don't set it if you are + * interested in configuring the + * server-sider per-stream playback + * buffer size. */ uint32_t prebuf; /**< Playback only: pre-buffering. The * server does not start with playback * before at least prebug bytes are * available in the buffer. It is - * recommended to set this to 0, which - * will initialize this to the same value - * as tlength, whatever that may be. */ + * recommended to set this to (uint32_t) + * -1, which will initialize this to the + * same value as tlength, whatever that + * may be. Initialize to 0 to enable + * manual start/stop control of the + * stream. This means that playback will + * not stop on underrun and playback will + * not start automatically. Instead + * pa_stream_corked() needs to be called + * explicitly. If you set this value to 0 + * you should also set + * PA_STREAM_START_CORKED. */ uint32_t minreq; /**< Playback only: minimum request. The * server does not request less than * minreq bytes from the client, instead * waits until the buffer is free enough * to request more bytes at once. It is - * recommended to set this to 0, which - * will initialize this to a value that - * is deemed sensible by the server. */ + * recommended to set this to (uint32_t) + * -1, which will initialize this to a + * value that is deemed sensible by the + * server. This should be set to a value + * that gives PulseAudio enough time to + * move the data from the per-stream + * playback buffer into the hardware + * playback buffer. */ uint32_t fragsize; /**< Recording only: fragment size. The * server sends data in blocks of * fragsize bytes size. Large values * deminish interactivity with other * operations on the connection context * but decrease control overhead. It is - * recommended to set this to 0, which - * will initialize this to a value that - * is deemed sensible by the + * recommended to set this to (uint32_t) + * -1, which will initialize this to a + * value that is deemed sensible by the * server. However, this value will * default to something like 2s, i.e. for * applications that have specific * latency requirements this value should * be set to the maximum latency that the - * application can deal with. */ + * application can deal with. If + * PA_STREAM_ADJUST_LATENCY is set the + * overall source latency will be + * adjusted according to this value. If + * it is not set the source latency is + * left unmodified. */ } pa_buffer_attr; /** Error values as used by pa_context_errno(). Use pa_strerror() to convert these values to human readable strings */ @@ -431,9 +463,9 @@ typedef struct pa_timing_info { * PA_SEEK_RELATIVE_ON_READ * instead. */ - pa_usec_t configured_sink_usec; /**< The static configured latency for + pa_usec_t configured_sink_usec; /**< The configured latency for * the sink. \since 0.9.11 */ - pa_usec_t configured_source_usec; /**< The static configured latency for + pa_usec_t configured_source_usec; /**< The configured latency for * the source. \since 0.9.11 */ int64_t since_underrun; /**< Bytes that were handed to the sink diff --git a/src/pulse/introspect.h b/src/pulse/introspect.h index c8c13a75..6a6755e7 100644 --- a/src/pulse/introspect.h +++ b/src/pulse/introspect.h @@ -130,8 +130,10 @@ * * \subsection autoload_subsec Autoload Entries * - * Modules can be autoloaded as a result of a client requesting a certain - * sink or source. This mapping between sink/source names and modules can be + * Modules can be autoloaded as a result of a client requesting a + * certain sink or source. Please note that autoloading is deprecated + * in 0.9.11. and is likely to be removed from the API in a later + * version. This mapping between sink/source names and modules can be * queried from the server: * * \li By index - pa_context_get_autoload_info_by_index() @@ -191,7 +193,9 @@ * * New module autoloading rules can be added, and existing can be removed * using pa_context_add_autoload() and pa_context_remove_autoload_by_index() - * / pa_context_remove_autoload_by_name(). + * / pa_context_remove_autoload_by_name(). Please note that autoloading is deprecated + * in 0.9.11. and is likely to be removed from the API in a later + * version. * * \subsection client_subsec Clients * diff --git a/src/pulse/proplist.h b/src/pulse/proplist.h index f75cca54..39d53303 100644 --- a/src/pulse/proplist.h +++ b/src/pulse/proplist.h @@ -36,19 +36,20 @@ PA_C_DECL_BEGIN * media.artist "Guns'N'Roses" * media.language "de_DE" * media.filename - * media.icon - * media.icon_name + * media.icon Binary blob containing PNG icon data + * media.icon_name Name from XDG icon naming spec * media.role video, music, game, event, phone, production, filter, abstract, stream - * event.id button-click, session-login + * event.id Name from XDG sound naming spec + * event.description "Button blabla clicked" for a11y * event.mouse.x * event.mouse.y - * event.mouse.hpos - * event.mouse.vpos - * event.mouse.button + * event.mouse.hpos Float formatted as string in range 0..1 + * event.mouse.vpos Float formatted as string in range 0..1 + * event.mouse.button Button number following X11 ordering * window.name - * window.id - * window.icon - * window.icon_name + * window.id "org.gnome.rhytmbox.MainWindow" + * window.icon Binary blob containing PNG icon data + * window.icon_name Name from XDG icon naming spec * window.x11.display * window.x11.screen * window.x11.monitor @@ -56,23 +57,23 @@ PA_C_DECL_BEGIN * application.name "Rhythmbox Media Player" * application.id "org.gnome.rhythmbox" * application.version - * application.icon - * application.icon_name + * application.icon Binary blob containing PNG icon data + * application.icon_name Name from XDG icon naming spec * application.language * application.process.id * application.process.binary * application.process.user * application.process.host * device.string - * device.api oss, alsa, sunaudio + * device.api oss, alsa, sunaudio * device.description * device.bus_path * device.serial * device.vendor_product_id - * device.class sound, modem, monitor, filter, abstract - * device.form_factor laptop-speakers, external-speakers, telephone, tv-capture, webcam-capture, microphone-capture, headset - * device.connector isa, pci, usb, firewire, bluetooth - * device.access_mode mmap, mmap_rewrite, serial + * device.class sound, modem, monitor, filter, abstract + * device.form_factor laptop-speakers, external-speakers, telephone, tv-capture, webcam-capture, microphone-capture, headset + * device.connector isa, pci, usb, firewire, bluetooth + * device.access_mode mmap, mmap_rewrite, serial * device.master_device * device.bufferin.buffer_size * device.bufferin.fragment_size @@ -86,6 +87,7 @@ PA_C_DECL_BEGIN #define PA_PROP_MEDIA_ICON_NAME "media.icon_name" #define PA_PROP_MEDIA_ROLE "media.role" #define PA_PROP_EVENT_ID "event.id" +#define PA_PROP_EVENT_DESCRIPTION "event.description" #define PA_PROP_EVENT_MOUSE_X "event.mouse.x" #define PA_PROP_EVENT_MOUSE_Y "event.mouse.y" #define PA_PROP_EVENT_MOUSE_HPOS "event.mouse.hpos" diff --git a/src/pulse/pulseaudio.h b/src/pulse/pulseaudio.h index 4a4531e6..e09cacaf 100644 --- a/src/pulse/pulseaudio.h +++ b/src/pulse/pulseaudio.h @@ -89,17 +89,17 @@ * * \section thread_sec Threads * - * The PulseAudio client libraries are not designed to be used in a - * heavily threaded environment. They are however designed to be reentrant - * safe. + * The PulseAudio client libraries are not designed to be directly + * thread-safe. They are however designed to be reentrant and + * threads-aware. * - * To use a the libraries in a threaded environment, you must assure that + * To use the libraries in a threaded environment, you must assure that * all objects are only used in one thread at a time. Normally, this means * that all objects belonging to a single context must be accessed from the * same thread. * * The included main loop implementation is also not thread safe. Take care - * to make sure event lists are not manipulated when any other code is + * to make sure event objects are not manipulated when any other code is * using the main loop. * * \section pkgconfig pkg-config diff --git a/src/pulse/sample.h b/src/pulse/sample.h index 1ba3f871..2680cf7e 100644 --- a/src/pulse/sample.h +++ b/src/pulse/sample.h @@ -52,7 +52,7 @@ * \li PA_SAMPLE_S32LE - Signed 32 bit integer PCM, little endian. * \li PA_SAMPLE_S32BE - Signed 32 bit integer PCM, big endian. * - * The floating point sample formats have the range from -1 to 1. + * The floating point sample formats have the range from -1.0 to 1.0. * * The sample formats that are sensitive to endianness have convenience * macros for native endian (NE), and reverse endian (RE). diff --git a/src/pulse/stream.h b/src/pulse/stream.h index fbf0ae44..2a8f7a8b 100644 --- a/src/pulse/stream.h +++ b/src/pulse/stream.h @@ -70,35 +70,85 @@ * * \subsection bufattr_subsec Buffer Attributes * - * Playback and record streams always have a server side buffer as - * part of the data flow. The size of this buffer strikes a - * compromise between low latency and sensitivity for buffer + * Playback and record streams always have a server-side buffer as + * part of the data flow. The size of this buffer needs to be chosen + * in a compromise between low latency and sensitivity for buffer * overflows/underruns. * * The buffer metrics may be controlled by the application. They are * described with a pa_buffer_attr structure which contains a number * of fields: * - * \li maxlength - The absolute maximum number of bytes that can be stored in - * the buffer. If this value is exceeded then data will be - * lost. - * \li tlength - The target length of a playback buffer. The server will only - * send requests for more data as long as the buffer has less - * than this number of bytes of data. + * \li maxlength - The absolute maximum number of bytes that can be + * stored in the buffer. If this value is exceeded + * then data will be lost. It is recommended to pass + * (uint32_t) -1 here which will cause the server to + * fill in the maximum possible value. + * + * \li tlength - The target fill level of the playback buffer. The + * server will only send requests for more data as long + * as the buffer has less than this number of bytes of + * data. If you pass (uint32_t) -1 (which is + * recommended) here the server will choose the longest + * target buffer fill level possible to minimize the + * number of necessary wakeups and maximize drop-out + * safety. This can exceed 2s of buffering. For + * low-latency applications or applications where + * latency matters you should pass a proper value here. + * * \li prebuf - Number of bytes that need to be in the buffer before - * playback will commence. Start of playback can be forced using - * pa_stream_trigger() even though the prebuffer size hasn't been - * reached. If a buffer underrun occurs, this prebuffering will be - * again enabled. If the playback shall never stop in case of a buffer - * underrun, this value should be set to 0. In that case the read - * index of the output buffer overtakes the write index, and hence the - * fill level of the buffer is negative. - * \li minreq - Minimum free number of the bytes in the playback buffer before - * the server will request more data. - * \li fragsize - Maximum number of bytes that the server will push in one - * chunk for record streams. - * - * The server side playback buffers are indexed by a write and a read + * playback will commence. Start of playback can be + * forced using pa_stream_trigger() even though the + * prebuffer size hasn't been reached. If a buffer + * underrun occurs, this prebuffering will be again + * enabled. If the playback shall never stop in case of a + * buffer underrun, this value should be set to 0. In + * that case the read index of the output buffer + * overtakes the write index, and hence the fill level of + * the buffer is negative. If you pass (uint32_t) -1 here + * (which is recommended) the server will choose the same + * value as tlength here. + * + * \li minreq - Minimum free number of the bytes in the playback + * buffer before the server will request more data. It is + * recommended to fill in (uint32_t) -1 here. This value + * influences how much time the sound server has to move + * data from the per-stream server-side playback buffer + * to the hardware playback buffer. + * + * \li fragsize - Maximum number of bytes that the server will push in + * one chunk for record streams. If you pass (uint32_t) + * -1 (which is recommended) here, the server will + * choose the longest fragment setting possible to + * minimize the number of necessary wakeups and + * maximize drop-out safety. This can exceed 2s of + * buffering. For low-latency applications or + * applications where latency matters you should pass a + * proper value here. + * + * If PA_STREAM_ADJUST_LATENCY is set, then the tlength/fragsize + * parameters will be interpreted slightly differently than described + * above when passed to pa_stream_connect_record() and + * pa_stream_connect_playback(): the overall latency that is comprised + * of both the server side playback buffer length, the hardware + * playback buffer length and additional latencies will be adjusted in + * a way that it matches tlength resp. fragsize. Set + * PA_STREAM_ADJUST_LATENCY if you want to control the overall + * playback latency for your stream. Unset it if you want to control + * only the latency induced by the server-side, rewritable playback + * buffer. The server will try to fulfill the clients latency requests + * as good as possible. However if the underlying hardware cannot + * change the hardware buffer length or only in a limited range, the + * actually resulting latency might be different from what the client + * requested. Thus, for synchronization clients always need to check + * the actual measured latency via pa_stream_get_latency() or a + * similar call, and not make any assumptions. about the latency + * available. The function pa_stream_get_buffer_attr() will always + * return the actual size of the server-side per-stream buffer in + * tlength/fragsize, regardless whether PA_STREAM_ADJUST_LATENCY is + * set or not. + * + * The server-side per-stream playback buffers are indexed by a write and a read * index. The application writes to the write index and the sound * device reads from the read index. The read index is increased * monotonically, while the write index may be freely controlled by @@ -196,10 +246,10 @@ * accordingly. * * The raw timing data in the pa_timing_info structure is usually hard - * to deal with. Therefore a more simplistic interface is available: + * to deal with. Therefore a simpler interface is available: * you can call pa_stream_get_time() or pa_stream_get_latency(). The * former will return the current playback time of the hardware since - * the stream has been started. The latter returns the time a sample + * the stream has been started. The latter returns the overall time a sample * that you write now takes to be played by the hardware. These two * functions base their calculations on the same data that is returned * by pa_stream_get_timing_info(). Hence the same rules for keeping @@ -512,9 +562,14 @@ const pa_sample_spec* pa_stream_get_sample_spec(pa_stream *s); /** Return a pointer to the stream's channel map. */ const pa_channel_map* pa_stream_get_channel_map(pa_stream *s); -/** Return the buffer metrics of the stream. Only valid after the - * stream has been connected successfuly and if the server is at least - * PulseAudio 0.9. \since 0.9.0 */ +/** Return the per-stream server-side buffer metrics of the + * stream. Only valid after the stream has been connected successfuly + * and if the server is at least PulseAudio 0.9. This will return the + * actual configured buffering metrics, which may differ from what was + * requested during pa_stream_connect_record() or + * pa_stream_connect_playback(). This call will always return the + * actually per-stream server-side buffer metrics, regardless whether + * PA_STREAM_ADJUST_LATENCY is set or not. \since 0.9.0 */ const pa_buffer_attr* pa_stream_get_buffer_attr(pa_stream *s); /** Change the buffer metrics of the stream during playback. The @@ -522,7 +577,9 @@ const pa_buffer_attr* pa_stream_get_buffer_attr(pa_stream *s); * requested. The selected metrics may be queried with * pa_stream_get_buffer_attr() as soon as the callback is called. Only * valid after the stream has been connected successfully and if the - * server is at least PulseAudio 0.9.8. \since 0.9.8 */ + * server is at least PulseAudio 0.9.8. Please be aware of the + * slightly different semantics of the call depending whether + * PA_STREAM_ADJUST_LATENCY is set or not. \since 0.9.8 */ pa_operation *pa_stream_set_buffer_attr(pa_stream *s, const pa_buffer_attr *attr, pa_stream_success_cb_t cb, void *userdata); /** Change the stream sampling rate during playback. You need to pass diff --git a/src/pulsecore/core-util.c b/src/pulsecore/core-util.c index d259fb16..c4818e39 100644 --- a/src/pulsecore/core-util.c +++ b/src/pulsecore/core-util.c @@ -177,7 +177,7 @@ int pa_make_secure_dir(const char* dir, mode_t m, uid_t uid, gid_t gid) { #else { mode_t u; - u = umask(~m); + u = umask((~m) & 0777); r = mkdir(dir, m); umask(u); } @@ -1887,17 +1887,21 @@ int pa_close_allv(const int except_fds[]) { if (getrlimit(RLIMIT_NOFILE, &rl) < 0) return -1; - for (fd = 0; fd < (int) rl.rlim_max; fd++) { + for (fd = 3; fd < (int) rl.rlim_max; fd++) { int i; + pa_bool_t found; - if (fd <= 3) - continue; - + found = FALSE; for (i = 0; except_fds[i] >= 0; i++) - if (except_fds[i] == fd) - continue; + if (except_fds[i] == fd) { + found = TRUE; + break; + } - if (close(fd) < 0 && errno != EBADF) + if (found) + continue; + + if (pa_close(fd) < 0 && errno != EBADF) return -1; } @@ -1972,10 +1976,11 @@ int pa_reset_sigs(int except, ...) { i = 0; if (except >= 1) { + int sig; p[i++] = except; - while ((p[i++] = va_arg(ap, int)) >= 0) - ; + while ((sig = va_arg(ap, int)) >= 0) + sig = p[i++]; } p[i] = -1; diff --git a/src/pulsecore/sink.c b/src/pulsecore/sink.c index 74971035..4102f31d 100644 --- a/src/pulsecore/sink.c +++ b/src/pulsecore/sink.c @@ -261,22 +261,31 @@ pa_sink* pa_sink_new( static int sink_set_state(pa_sink *s, pa_sink_state_t state) { int ret; pa_bool_t suspend_change; + pa_sink_state_t original_state; pa_assert(s); if (s->state == state) return 0; + original_state = s->state; + suspend_change = - (s->state == PA_SINK_SUSPENDED && PA_SINK_IS_OPENED(state)) || - (PA_SINK_IS_OPENED(s->state) && state == PA_SINK_SUSPENDED); + (original_state == PA_SINK_SUSPENDED && PA_SINK_IS_OPENED(state)) || + (PA_SINK_IS_OPENED(original_state) && state == PA_SINK_SUSPENDED); if (s->set_state) if ((ret = s->set_state(s, state)) < 0) - return -1; + return ret; if (s->asyncmsgq) - pa_assert_se(pa_asyncmsgq_send(s->asyncmsgq, PA_MSGOBJECT(s), PA_SINK_MESSAGE_SET_STATE, PA_UINT_TO_PTR(state), 0, NULL) == 0); + if ((ret = pa_asyncmsgq_send(s->asyncmsgq, PA_MSGOBJECT(s), PA_SINK_MESSAGE_SET_STATE, PA_UINT_TO_PTR(state), 0, NULL)) < 0) { + + if (s->set_state) + s->set_state(s, original_state); + + return ret; + } s->state = state; diff --git a/src/pulsecore/source.c b/src/pulsecore/source.c index e62c6c24..95007af4 100644 --- a/src/pulsecore/source.c +++ b/src/pulsecore/source.c @@ -224,22 +224,31 @@ pa_source* pa_source_new( static int source_set_state(pa_source *s, pa_source_state_t state) { int ret; pa_bool_t suspend_change; + pa_source_state_t original_state; pa_assert(s); if (s->state == state) return 0; + original_state = s->state; + suspend_change = - (s->state == PA_SOURCE_SUSPENDED && PA_SOURCE_IS_OPENED(state)) || - (PA_SOURCE_IS_OPENED(s->state) && state == PA_SOURCE_SUSPENDED); + (original_state == PA_SOURCE_SUSPENDED && PA_SOURCE_IS_OPENED(state)) || + (PA_SOURCE_IS_OPENED(original_state) && state == PA_SOURCE_SUSPENDED); if (s->set_state) if ((ret = s->set_state(s, state)) < 0) - return -1; + return ret; if (s->asyncmsgq) - pa_assert_se(pa_asyncmsgq_send(s->asyncmsgq, PA_MSGOBJECT(s), PA_SOURCE_MESSAGE_SET_STATE, PA_UINT_TO_PTR(state), 0, NULL) == 0); + if ((ret = pa_asyncmsgq_send(s->asyncmsgq, PA_MSGOBJECT(s), PA_SOURCE_MESSAGE_SET_STATE, PA_UINT_TO_PTR(state), 0, NULL)) < 0) { + + if (s->set_state) + s->set_state(s, original_state); + + return ret; + } s->state = state; |