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Add pulseaudio GStreamer element from gst-pulse. Development will continue here instead of pulseaudio SVN. Fixes bug ...

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Original commit message from CVS:
* configure.ac:
* ext/pulse/Makefile.am:
* ext/pulse/plugin.c: (plugin_init):
* ext/pulse/pulsemixer.c: (gst_pulsemixer_interface_supported),
(gst_pulsemixer_implements_interface_init),
(gst_pulsemixer_init_interfaces), (gst_pulsemixer_base_init),
(gst_pulsemixer_class_init), (gst_pulsemixer_init),
(gst_pulsemixer_finalize), (gst_pulsemixer_set_property),
(gst_pulsemixer_get_property), (gst_pulsemixer_change_state):
* ext/pulse/pulsemixer.h:
* ext/pulse/pulsemixerctrl.c:
(gst_pulsemixer_ctrl_context_state_cb),
(gst_pulsemixer_ctrl_sink_info_cb),
(gst_pulsemixer_ctrl_source_info_cb),
(gst_pulsemixer_ctrl_subscribe_cb),
(gst_pulsemixer_ctrl_success_cb), (gst_pulsemixer_ctrl_open),
(gst_pulsemixer_ctrl_close), (gst_pulsemixer_ctrl_new),
(gst_pulsemixer_ctrl_free), (gst_pulsemixer_ctrl_list_tracks),
(gst_pulsemixer_ctrl_timeout_event), (restart_time_event),
(gst_pulsemixer_ctrl_set_volume), (gst_pulsemixer_ctrl_get_volume),
(gst_pulsemixer_ctrl_set_record), (gst_pulsemixer_ctrl_set_mute):
* ext/pulse/pulsemixerctrl.h:
* ext/pulse/pulsemixertrack.c: (gst_pulsemixer_track_class_init),
(gst_pulsemixer_track_init), (gst_pulsemixer_track_new):
* ext/pulse/pulsemixertrack.h:
* ext/pulse/pulseprobe.c: (gst_pulseprobe_context_state_cb),
(gst_pulseprobe_sink_info_cb), (gst_pulseprobe_source_info_cb),
(gst_pulseprobe_invalidate), (gst_pulseprobe_open),
(gst_pulseprobe_enumerate), (gst_pulseprobe_close),
(gst_pulseprobe_new), (gst_pulseprobe_free),
(gst_pulseprobe_get_properties), (gst_pulseprobe_needs_probe),
(gst_pulseprobe_probe_property), (gst_pulseprobe_get_values),
(gst_pulseprobe_set_server):
* ext/pulse/pulseprobe.h:
* ext/pulse/pulsesink.c: (gst_pulsesink_base_init),
(gst_pulsesink_class_init), (gst_pulsesink_init),
(gst_pulsesink_destroy_stream), (gst_pulsesink_destroy_context),
(gst_pulsesink_finalize), (gst_pulsesink_dispose),
(gst_pulsesink_set_property), (gst_pulsesink_get_property),
(gst_pulsesink_context_state_cb), (gst_pulsesink_stream_state_cb),
(gst_pulsesink_stream_request_cb),
(gst_pulsesink_stream_latency_update_cb), (gst_pulsesink_open),
(gst_pulsesink_close), (gst_pulsesink_prepare),
(gst_pulsesink_unprepare), (gst_pulsesink_write),
(gst_pulsesink_delay), (gst_pulsesink_success_cb),
(gst_pulsesink_reset), (gst_pulsesink_change_title),
(gst_pulsesink_event), (gst_pulsesink_get_type):
* ext/pulse/pulsesink.h:
* ext/pulse/pulsesrc.c: (gst_pulsesrc_interface_supported),
(gst_pulsesrc_implements_interface_init),
(gst_pulsesrc_init_interfaces), (gst_pulsesrc_base_init),
(gst_pulsesrc_class_init), (gst_pulsesrc_init),
(gst_pulsesrc_destroy_stream), (gst_pulsesrc_destroy_context),
(gst_pulsesrc_finalize), (gst_pulsesrc_dispose),
(gst_pulsesrc_set_property), (gst_pulsesrc_get_property),
(gst_pulsesrc_context_state_cb), (gst_pulsesrc_stream_state_cb),
(gst_pulsesrc_stream_request_cb), (gst_pulsesrc_open),
(gst_pulsesrc_close), (gst_pulsesrc_prepare),
(gst_pulsesrc_unprepare), (gst_pulsesrc_read),
(gst_pulsesrc_delay), (gst_pulsesrc_change_state),
(gst_pulsesrc_get_type):
* ext/pulse/pulsesrc.h:
* ext/pulse/pulseutil.c: (gst_pulse_fill_sample_spec),
(gst_pulse_client_name), (gst_pulse_gst_to_channel_map):
* ext/pulse/pulseutil.h:
Add pulseaudio GStreamer element from gst-pulse. Development will
continue here instead of pulseaudio SVN. Fixes bug #400679.
Only changes over gst-pulse SVN are added copyright to the top of
files and coding style changes.
This commit is contained in:
Sebastian Dröge 2008-06-10 06:45:33 +00:00
parent 660d958685
commit f3b03cd773
18 changed files with 3634 additions and 0 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

72
ChangeLog
View file

@ -1,3 +1,75 @@
2008-06-10 Sebastian Dröge <slomo@circular-chaos.org>
* configure.ac:
* ext/pulse/Makefile.am:
* ext/pulse/plugin.c: (plugin_init):
* ext/pulse/pulsemixer.c: (gst_pulsemixer_interface_supported),
(gst_pulsemixer_implements_interface_init),
(gst_pulsemixer_init_interfaces), (gst_pulsemixer_base_init),
(gst_pulsemixer_class_init), (gst_pulsemixer_init),
(gst_pulsemixer_finalize), (gst_pulsemixer_set_property),
(gst_pulsemixer_get_property), (gst_pulsemixer_change_state):
* ext/pulse/pulsemixer.h:
* ext/pulse/pulsemixerctrl.c:
(gst_pulsemixer_ctrl_context_state_cb),
(gst_pulsemixer_ctrl_sink_info_cb),
(gst_pulsemixer_ctrl_source_info_cb),
(gst_pulsemixer_ctrl_subscribe_cb),
(gst_pulsemixer_ctrl_success_cb), (gst_pulsemixer_ctrl_open),
(gst_pulsemixer_ctrl_close), (gst_pulsemixer_ctrl_new),
(gst_pulsemixer_ctrl_free), (gst_pulsemixer_ctrl_list_tracks),
(gst_pulsemixer_ctrl_timeout_event), (restart_time_event),
(gst_pulsemixer_ctrl_set_volume), (gst_pulsemixer_ctrl_get_volume),
(gst_pulsemixer_ctrl_set_record), (gst_pulsemixer_ctrl_set_mute):
* ext/pulse/pulsemixerctrl.h:
* ext/pulse/pulsemixertrack.c: (gst_pulsemixer_track_class_init),
(gst_pulsemixer_track_init), (gst_pulsemixer_track_new):
* ext/pulse/pulsemixertrack.h:
* ext/pulse/pulseprobe.c: (gst_pulseprobe_context_state_cb),
(gst_pulseprobe_sink_info_cb), (gst_pulseprobe_source_info_cb),
(gst_pulseprobe_invalidate), (gst_pulseprobe_open),
(gst_pulseprobe_enumerate), (gst_pulseprobe_close),
(gst_pulseprobe_new), (gst_pulseprobe_free),
(gst_pulseprobe_get_properties), (gst_pulseprobe_needs_probe),
(gst_pulseprobe_probe_property), (gst_pulseprobe_get_values),
(gst_pulseprobe_set_server):
* ext/pulse/pulseprobe.h:
* ext/pulse/pulsesink.c: (gst_pulsesink_base_init),
(gst_pulsesink_class_init), (gst_pulsesink_init),
(gst_pulsesink_destroy_stream), (gst_pulsesink_destroy_context),
(gst_pulsesink_finalize), (gst_pulsesink_dispose),
(gst_pulsesink_set_property), (gst_pulsesink_get_property),
(gst_pulsesink_context_state_cb), (gst_pulsesink_stream_state_cb),
(gst_pulsesink_stream_request_cb),
(gst_pulsesink_stream_latency_update_cb), (gst_pulsesink_open),
(gst_pulsesink_close), (gst_pulsesink_prepare),
(gst_pulsesink_unprepare), (gst_pulsesink_write),
(gst_pulsesink_delay), (gst_pulsesink_success_cb),
(gst_pulsesink_reset), (gst_pulsesink_change_title),
(gst_pulsesink_event), (gst_pulsesink_get_type):
* ext/pulse/pulsesink.h:
* ext/pulse/pulsesrc.c: (gst_pulsesrc_interface_supported),
(gst_pulsesrc_implements_interface_init),
(gst_pulsesrc_init_interfaces), (gst_pulsesrc_base_init),
(gst_pulsesrc_class_init), (gst_pulsesrc_init),
(gst_pulsesrc_destroy_stream), (gst_pulsesrc_destroy_context),
(gst_pulsesrc_finalize), (gst_pulsesrc_dispose),
(gst_pulsesrc_set_property), (gst_pulsesrc_get_property),
(gst_pulsesrc_context_state_cb), (gst_pulsesrc_stream_state_cb),
(gst_pulsesrc_stream_request_cb), (gst_pulsesrc_open),
(gst_pulsesrc_close), (gst_pulsesrc_prepare),
(gst_pulsesrc_unprepare), (gst_pulsesrc_read),
(gst_pulsesrc_delay), (gst_pulsesrc_change_state),
(gst_pulsesrc_get_type):
* ext/pulse/pulsesrc.h:
* ext/pulse/pulseutil.c: (gst_pulse_fill_sample_spec),
(gst_pulse_client_name), (gst_pulse_gst_to_channel_map):
* ext/pulse/pulseutil.h:
Add pulseaudio GStreamer element from gst-pulse. Development will
continue here instead of pulseaudio SVN. Fixes bug #400679.
Only changes over gst-pulse SVN are added copyright to the top of
files and coding style changes.
2008-06-09 Tim-Philipp Müller <tim.muller at collabora co uk>
Patch by: Benjamin Kampmann <benjamin at fluendo dot com>

7
configure.ac
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@ -776,6 +776,12 @@ AG_GST_CHECK_FEATURE(LIBPNG, [Portable Network Graphics library], png, [
AG_GST_PKG_CHECK_MODULES(LIBPNG, libpng12)
])
dnl *** pulseaudio ***
translit(dnm, m, l) AM_CONDITIONAL(USE_PULSE, true)
AG_GST_CHECK_FEATURE(PULSE, [pulseaudio plug-in], pulseaudio, [
AG_GST_PKG_CHECK_MODULES(PULSE, libpulse >= 0.9.8)
])
dnl *** dv1394 ***
translit(dnm, m, l) AM_CONDITIONAL(USE_DV1394, true)
AG_GST_CHECK_FEATURE(DV1394, [raw1394 and avc1394 library], 1394, [
@ -1073,6 +1079,7 @@ ext/hal/Makefile
ext/ladspa/Makefile
ext/libcaca/Makefile
ext/libpng/Makefile
ext/pulse/Makefile
ext/raw1394/Makefile
ext/shout2/Makefile
ext/soup/Makefile

25
ext/pulse/Makefile.am Normal file
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@ -0,0 +1,25 @@
plugin_LTLIBRARIES = libgstpulse.la
libgstpulse_la_SOURCES = \
plugin.c \
pulsemixer.c \
pulsemixerctrl.c \
pulsemixertrack.c \
pulseprobe.c \
pulsesink.c \
pulsesrc.c \
pulseutil.c
libgstpulse_la_CFLAGS = $(GST_CFLAGS) $(GST_BASE_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) $(PULSE_CFLAGS)
libgstpulse_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) -lgstinterfaces-$(GST_MAJORMINOR) $(PULSE_LIBS)
libgstpulse_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
noinst_HEADERS = \
pulsemixerctrl.h \
pulsemixer.h \
pulsemixertrack.h \
pulseprobe.h \
pulsesink.h \
pulsesrc.h \
pulseutil.h

56
ext/pulse/plugin.c Normal file
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@ -0,0 +1,56 @@
/*
* GStreamer pulseaudio plugin
*
* Copyright (c) 2004-2008 Lennart Poettering
*
* gst-pulse 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.
*
* gst-pulse 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with gst-pulse; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "pulsesink.h"
#include "pulsesrc.h"
#include "pulsemixer.h"
GST_DEBUG_CATEGORY (pulse_debug);
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "pulsesink", GST_RANK_PRIMARY,
GST_TYPE_PULSESINK))
return FALSE;
if (!gst_element_register (plugin, "pulsesrc", GST_RANK_PRIMARY,
GST_TYPE_PULSESRC))
return FALSE;
if (!gst_element_register (plugin, "pulsemixer", GST_RANK_NONE,
GST_TYPE_PULSEMIXER))
return FALSE;
GST_DEBUG_CATEGORY_INIT (pulse_debug, "pulse", 0, "PulseAudio elements");
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"pulseaudio",
"PulseAudio Elements Plugin",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)

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ext/pulse/pulsemixer.c Normal file
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@ -0,0 +1,277 @@
/*
* GStreamer pulseaudio plugin
*
* Copyright (c) 2004-2008 Lennart Poettering
*
* gst-pulse 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.
*
* gst-pulse 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with gst-pulse; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <stdio.h>
#include "pulsemixer.h"
enum
{
PROP_SERVER = 1,
PROP_DEVICE,
PROP_DEVICE_NAME
};
GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
#define GST_CAT_DEFAULT pulse_debug
static void gst_pulsemixer_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_pulsemixer_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_pulsemixer_finalize (GObject * object);
static GstStateChangeReturn gst_pulsemixer_change_state (GstElement * element,
GstStateChange transition);
static void gst_pulsemixer_init_interfaces (GType type);
GST_IMPLEMENT_PULSEMIXER_CTRL_METHODS (GstPulseMixer, gst_pulsemixer);
GST_IMPLEMENT_PULSEPROBE_METHODS (GstPulseMixer, gst_pulsemixer);
GST_BOILERPLATE_FULL (GstPulseMixer, gst_pulsemixer, GstElement,
GST_TYPE_ELEMENT, gst_pulsemixer_init_interfaces);
static gboolean
gst_pulsemixer_interface_supported (GstImplementsInterface
* iface, GType interface_type)
{
GstPulseMixer *this = GST_PULSEMIXER (iface);
if (interface_type == GST_TYPE_MIXER && this->mixer)
return TRUE;
if (interface_type == GST_TYPE_PROPERTY_PROBE && this->probe)
return TRUE;
return FALSE;
}
static void
gst_pulsemixer_implements_interface_init (GstImplementsInterfaceClass * klass)
{
klass->supported = gst_pulsemixer_interface_supported;
}
static void
gst_pulsemixer_init_interfaces (GType type)
{
static const GInterfaceInfo implements_iface_info = {
(GInterfaceInitFunc) gst_pulsemixer_implements_interface_init,
NULL,
NULL,
};
static const GInterfaceInfo mixer_iface_info = {
(GInterfaceInitFunc) gst_pulsemixer_mixer_interface_init,
NULL,
NULL,
};
static const GInterfaceInfo probe_iface_info = {
(GInterfaceInitFunc) gst_pulsemixer_property_probe_interface_init,
NULL,
NULL,
};
g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
&implements_iface_info);
g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
g_type_add_interface_static (type, GST_TYPE_PROPERTY_PROBE,
&probe_iface_info);
}
static void
gst_pulsemixer_base_init (gpointer g_class)
{
static const GstElementDetails details =
GST_ELEMENT_DETAILS ("PulseAudio Mixer",
"Generic/Audio",
"Control sound input and output levels for PulseAudio",
"Lennart Poettering");
gst_element_class_set_details (GST_ELEMENT_CLASS (g_class), &details);
}
static void
gst_pulsemixer_class_init (GstPulseMixerClass * g_class)
{
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
GObjectClass *gobject_class = G_OBJECT_CLASS (g_class);
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_pulsemixer_change_state);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_pulsemixer_finalize);
gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_pulsemixer_get_property);
gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_pulsemixer_set_property);
g_object_class_install_property (gobject_class,
PROP_SERVER,
g_param_spec_string ("server", "Server",
"The PulseAudio server to connect to", NULL, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class,
PROP_DEVICE,
g_param_spec_string ("device", "Sink/Source",
"The PulseAudio sink or source to control", NULL, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class,
PROP_DEVICE_NAME,
g_param_spec_string ("device-name", "Device name",
"Human-readable name of the sound device", NULL, G_PARAM_READABLE));
}
static void
gst_pulsemixer_init (GstPulseMixer * this, GstPulseMixerClass * g_class)
{
this->mixer = NULL;
this->server = NULL;
this->device = NULL;
this->probe =
gst_pulseprobe_new (G_OBJECT_GET_CLASS (this), PROP_DEVICE, this->device,
TRUE, TRUE);
}
static void
gst_pulsemixer_finalize (GObject * object)
{
GstPulseMixer *this = GST_PULSEMIXER (object);
g_free (this->server);
g_free (this->device);
if (this->mixer) {
gst_pulsemixer_ctrl_free (this->mixer);
this->mixer = NULL;
}
if (this->probe) {
gst_pulseprobe_free (this->probe);
this->probe = NULL;
}
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_pulsemixer_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstPulseMixer *this = GST_PULSEMIXER (object);
switch (prop_id) {
case PROP_SERVER:
g_free (this->server);
this->server = g_value_dup_string (value);
break;
case PROP_DEVICE:
g_free (this->device);
this->device = g_value_dup_string (value);
if (this->probe)
gst_pulseprobe_set_server (this->probe, this->device);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_pulsemixer_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstPulseMixer *this = GST_PULSEMIXER (object);
switch (prop_id) {
case PROP_SERVER:
g_value_set_string (value, this->server);
break;
case PROP_DEVICE:
g_value_set_string (value, this->device);
break;
case PROP_DEVICE_NAME:
if (this->mixer) {
char *t = g_strdup_printf ("%s: %s",
this->mixer->type == GST_PULSEMIXER_SINK ? "Playback" : "Capture",
this->mixer->description);
g_value_set_string (value, t);
g_free (t);
} else
g_value_set_string (value, NULL);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_pulsemixer_change_state (GstElement * element, GstStateChange transition)
{
GstPulseMixer *this = GST_PULSEMIXER (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!this->mixer)
this->mixer =
gst_pulsemixer_ctrl_new (this->server, this->device,
GST_PULSEMIXER_UNKNOWN);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
if (this->mixer) {
gst_pulsemixer_ctrl_free (this->mixer);
this->mixer = NULL;
}
break;
default:
;
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
return GST_STATE_CHANGE_SUCCESS;
}

68
ext/pulse/pulsemixer.h Normal file
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@ -0,0 +1,68 @@
/*
* GStreamer pulseaudio plugin
*
* Copyright (c) 2004-2008 Lennart Poettering
*
* gst-pulse 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.
*
* gst-pulse 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with gst-pulse; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA.
*/
#ifndef __GST_PULSEMIXER_H__
#define __GST_PULSEMIXER_H__
#include <gst/gst.h>
#include <pulse/pulseaudio.h>
#include <pulse/thread-mainloop.h>
#include "pulsemixerctrl.h"
#include "pulseprobe.h"
G_BEGIN_DECLS
#define GST_TYPE_PULSEMIXER \
(gst_pulsemixer_get_type())
#define GST_PULSEMIXER(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_PULSEMIXER,GstPulseMixer))
#define GST_PULSEMIXER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_PULSEMIXER,GstPulseMixerClass))
#define GST_IS_PULSEMIXER(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_PULSEMIXER))
#define GST_IS_PULSEMIXER_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_PULSEMIXER))
typedef struct _GstPulseMixer GstPulseMixer;
typedef struct _GstPulseMixerClass GstPulseMixerClass;
struct _GstPulseMixer
{
GstElement parent;
gchar *server, *device;
GstPulseMixerCtrl *mixer;
GstPulseProbe *probe;
};
struct _GstPulseMixerClass
{
GstElementClass parent_class;
};
GType gst_pulsemixer_get_type (void);
G_END_DECLS
#endif /* __GST_PULSEMIXER_H__ */

583
ext/pulse/pulsemixerctrl.c Normal file
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@ -0,0 +1,583 @@
/*
* GStreamer pulseaudio plugin
*
* Copyright (c) 2004-2008 Lennart Poettering
*
* gst-pulse 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.
*
* gst-pulse 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with gst-pulse; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include "pulsemixerctrl.h"
#include "pulsemixertrack.h"
#include "pulseutil.h"
GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
#define GST_CAT_DEFAULT pulse_debug
static void
gst_pulsemixer_ctrl_context_state_cb (pa_context * context, void *userdata)
{
GstPulseMixerCtrl *c = GST_PULSEMIXER_CTRL (userdata);
/* Called from the background thread! */
switch (pa_context_get_state (context)) {
case PA_CONTEXT_READY:
case PA_CONTEXT_TERMINATED:
case PA_CONTEXT_FAILED:
pa_threaded_mainloop_signal (c->mainloop, 0);
break;
case PA_CONTEXT_UNCONNECTED:
case PA_CONTEXT_CONNECTING:
case PA_CONTEXT_AUTHORIZING:
case PA_CONTEXT_SETTING_NAME:
break;
}
}
static void
gst_pulsemixer_ctrl_sink_info_cb (pa_context * context, const pa_sink_info * i,
int eol, void *userdata)
{
GstPulseMixerCtrl *c = userdata;
/* Called from the background thread! */
if (c->outstandig_queries > 0)
c->outstandig_queries--;
if (c->ignore_queries > 0 || c->time_event) {
if (c->ignore_queries > 0)
c->ignore_queries--;
return;
}
if (!i && eol < 0) {
c->operation_success = 0;
pa_threaded_mainloop_signal (c->mainloop, 0);
return;
}
if (eol)
return;
g_free (c->name);
g_free (c->description);
c->name = g_strdup (i->name);
c->description = g_strdup (i->description);
c->index = i->index;
c->channel_map = i->channel_map;
c->volume = i->volume;
c->muted = i->mute;
c->type = GST_PULSEMIXER_SINK;
if (c->track) {
int i = g_atomic_int_get (&c->track->flags);
i = (i & ~GST_MIXER_TRACK_MUTE) | (c->muted ? GST_MIXER_TRACK_MUTE : 0);
g_atomic_int_set (&c->track->flags, i);
}
c->operation_success = 1;
pa_threaded_mainloop_signal (c->mainloop, 0);
}
static void
gst_pulsemixer_ctrl_source_info_cb (pa_context * context,
const pa_source_info * i, int eol, void *userdata)
{
GstPulseMixerCtrl *c = userdata;
/* Called from the background thread! */
if (c->outstandig_queries > 0)
c->outstandig_queries--;
if (c->ignore_queries > 0 || c->time_event) {
if (c->ignore_queries > 0)
c->ignore_queries--;
return;
}
if (!i && eol < 0) {
c->operation_success = 0;
pa_threaded_mainloop_signal (c->mainloop, 0);
return;
}
if (eol)
return;
g_free (c->name);
g_free (c->description);
c->name = g_strdup (i->name);
c->description = g_strdup (i->description);
c->index = i->index;
c->channel_map = i->channel_map;
c->volume = i->volume;
c->muted = i->mute;
c->type = GST_PULSEMIXER_SOURCE;
if (c->track) {
int i = g_atomic_int_get (&c->track->flags);
i = (i & ~GST_MIXER_TRACK_MUTE) | (c->muted ? GST_MIXER_TRACK_MUTE : 0);
g_atomic_int_set (&c->track->flags, i);
}
c->operation_success = 1;
pa_threaded_mainloop_signal (c->mainloop, 0);
}
static void
gst_pulsemixer_ctrl_subscribe_cb (pa_context * context,
pa_subscription_event_type_t t, uint32_t idx, void *userdata)
{
GstPulseMixerCtrl *c = GST_PULSEMIXER_CTRL (userdata);
pa_operation *o = NULL;
/* Called from the background thread! */
if (c->index != idx)
return;
if ((t & PA_SUBSCRIPTION_EVENT_TYPE_MASK) != PA_SUBSCRIPTION_EVENT_CHANGE)
return;
if (c->type == GST_PULSEMIXER_SINK)
o = pa_context_get_sink_info_by_index (c->context, c->index,
gst_pulsemixer_ctrl_sink_info_cb, c);
else
o = pa_context_get_source_info_by_index (c->context, c->index,
gst_pulsemixer_ctrl_source_info_cb, c);
if (!o) {
GST_WARNING ("Failed to get sink info: %s",
pa_strerror (pa_context_errno (c->context)));
return;
}
pa_operation_unref (o);
c->outstandig_queries++;
}
static void
gst_pulsemixer_ctrl_success_cb (pa_context * context, int success,
void *userdata)
{
GstPulseMixerCtrl *c = (GstPulseMixerCtrl *) userdata;
c->operation_success = success;
pa_threaded_mainloop_signal (c->mainloop, 0);
}
#define CHECK_DEAD_GOTO(c, label) do { \
if (!(c)->context || pa_context_get_state((c)->context) != PA_CONTEXT_READY) { \
GST_WARNING("Not connected: %s", (c)->context ? pa_strerror(pa_context_errno((c)->context)) : "NULL"); \
goto label; \
} \
} while(0);
static gboolean
gst_pulsemixer_ctrl_open (GstPulseMixerCtrl * c)
{
int e;
gchar *name = gst_pulse_client_name ();
pa_operation *o = NULL;
g_assert (c);
c->mainloop = pa_threaded_mainloop_new ();
g_assert (c->mainloop);
e = pa_threaded_mainloop_start (c->mainloop);
g_assert (e == 0);
pa_threaded_mainloop_lock (c->mainloop);
if (!(c->context =
pa_context_new (pa_threaded_mainloop_get_api (c->mainloop), name))) {
GST_WARNING ("Failed to create context");
goto unlock_and_fail;
}
pa_context_set_state_callback (c->context,
gst_pulsemixer_ctrl_context_state_cb, c);
pa_context_set_subscribe_callback (c->context,
gst_pulsemixer_ctrl_subscribe_cb, c);
if (pa_context_connect (c->context, c->server, 0, NULL) < 0) {
GST_WARNING ("Failed to connect context: %s",
pa_strerror (pa_context_errno (c->context)));
goto unlock_and_fail;
}
/* Wait until the context is ready */
pa_threaded_mainloop_wait (c->mainloop);
if (pa_context_get_state (c->context) != PA_CONTEXT_READY) {
GST_WARNING ("Failed to connect context: %s",
pa_strerror (pa_context_errno (c->context)));
goto unlock_and_fail;
}
/* Subscribe to events */
if (!(o =
pa_context_subscribe (c->context,
PA_SUBSCRIPTION_MASK_SINK | PA_SUBSCRIPTION_MASK_SOURCE,
gst_pulsemixer_ctrl_success_cb, c))) {
GST_WARNING ("Failed to subscribe to events: %s",
pa_strerror (pa_context_errno (c->context)));
goto unlock_and_fail;
}
c->operation_success = 0;
while (pa_operation_get_state (o) != PA_OPERATION_DONE) {
pa_threaded_mainloop_wait (c->mainloop);
CHECK_DEAD_GOTO (c, unlock_and_fail);
}
if (!c->operation_success) {
GST_WARNING ("Failed to subscribe to events: %s",
pa_strerror (pa_context_errno (c->context)));
goto unlock_and_fail;
}
/* Get sink info */
if (c->type == GST_PULSEMIXER_UNKNOWN || c->type == GST_PULSEMIXER_SINK) {
if (!(o =
pa_context_get_sink_info_by_name (c->context, c->device,
gst_pulsemixer_ctrl_sink_info_cb, c))) {
GST_WARNING ("Failed to get sink info: %s",
pa_strerror (pa_context_errno (c->context)));
goto unlock_and_fail;
}
c->operation_success = 0;
while (pa_operation_get_state (o) != PA_OPERATION_DONE) {
pa_threaded_mainloop_wait (c->mainloop);
CHECK_DEAD_GOTO (c, unlock_and_fail);
}
pa_operation_unref (o);
o = NULL;
if (!c->operation_success && (c->type == GST_PULSEMIXER_SINK
|| pa_context_errno (c->context) != PA_ERR_NOENTITY)) {
GST_WARNING ("Failed to get sink info: %s",
pa_strerror (pa_context_errno (c->context)));
goto unlock_and_fail;
}
}
if (c->type == GST_PULSEMIXER_UNKNOWN || c->type == GST_PULSEMIXER_SOURCE) {
if (!(o =
pa_context_get_source_info_by_name (c->context, c->device,
gst_pulsemixer_ctrl_source_info_cb, c))) {
GST_WARNING ("Failed to get source info: %s",
pa_strerror (pa_context_errno (c->context)));
goto unlock_and_fail;
}
c->operation_success = 0;
while (pa_operation_get_state (o) != PA_OPERATION_DONE) {
pa_threaded_mainloop_wait (c->mainloop);
CHECK_DEAD_GOTO (c, unlock_and_fail);
}
pa_operation_unref (o);
o = NULL;
if (!c->operation_success) {
GST_WARNING ("Failed to get source info: %s",
pa_strerror (pa_context_errno (c->context)));
goto unlock_and_fail;
}
}
g_assert (c->type != GST_PULSEMIXER_UNKNOWN);
c->track = gst_pulsemixer_track_new (c);
c->tracklist = g_list_append (c->tracklist, c->track);
pa_threaded_mainloop_unlock (c->mainloop);
g_free (name);
return TRUE;
unlock_and_fail:
if (o)
pa_operation_unref (o);
if (c->mainloop)
pa_threaded_mainloop_unlock (c->mainloop);
g_free (name);
return FALSE;
}
static void
gst_pulsemixer_ctrl_close (GstPulseMixerCtrl * c)
{
g_assert (c);
if (c->mainloop)
pa_threaded_mainloop_stop (c->mainloop);
if (c->context) {
pa_context_disconnect (c->context);
pa_context_unref (c->context);
c->context = NULL;
}
if (c->mainloop) {
pa_threaded_mainloop_free (c->mainloop);
c->mainloop = NULL;
c->time_event = NULL;
}
if (c->tracklist) {
g_list_free (c->tracklist);
c->tracklist = NULL;
}
if (c->track) {
GST_PULSEMIXER_TRACK (c->track)->control = NULL;
g_object_unref (c->track);
c->track = NULL;
}
}
GstPulseMixerCtrl *
gst_pulsemixer_ctrl_new (const gchar * server, const gchar * device,
GstPulseMixerType type)
{
GstPulseMixerCtrl *c = NULL;
c = g_new (GstPulseMixerCtrl, 1);
c->tracklist = NULL;
c->server = g_strdup (server);
c->device = g_strdup (device);
c->mainloop = NULL;
c->context = NULL;
c->track = NULL;
c->ignore_queries = c->outstandig_queries = 0;
pa_cvolume_mute (&c->volume, PA_CHANNELS_MAX);
pa_channel_map_init (&c->channel_map);
c->muted = 0;
c->index = PA_INVALID_INDEX;
c->type = type;
c->name = NULL;
c->description = NULL;
c->time_event = NULL;
c->update_volume = c->update_mute = FALSE;
if (!(gst_pulsemixer_ctrl_open (c))) {
gst_pulsemixer_ctrl_free (c);
return NULL;
}
return c;
}
void
gst_pulsemixer_ctrl_free (GstPulseMixerCtrl * c)
{
g_assert (c);
gst_pulsemixer_ctrl_close (c);
g_free (c->server);
g_free (c->device);
g_free (c->name);
g_free (c->description);
g_free (c);
}
const GList *
gst_pulsemixer_ctrl_list_tracks (GstPulseMixerCtrl * c)
{
g_assert (c);
return c->tracklist;
}
static void
gst_pulsemixer_ctrl_timeout_event (pa_mainloop_api * a, pa_time_event * e,
const struct timeval *tv, void *userdata)
{
pa_operation *o;
GstPulseMixerCtrl *c = GST_PULSEMIXER_CTRL (userdata);
if (c->update_volume) {
if (c->type == GST_PULSEMIXER_SINK)
o = pa_context_set_sink_volume_by_index (c->context, c->index, &c->volume,
NULL, NULL);
else
o = pa_context_set_source_volume_by_index (c->context, c->index,
&c->volume, NULL, NULL);
if (!o)
GST_WARNING ("Failed to set device volume: %s",
pa_strerror (pa_context_errno (c->context)));
else
pa_operation_unref (o);
c->update_volume = FALSE;
}
if (c->update_mute) {
if (c->type == GST_PULSEMIXER_SINK)
o = pa_context_set_sink_mute_by_index (c->context, c->index, !!c->muted,
NULL, NULL);
else
o = pa_context_set_source_mute_by_index (c->context, c->index, !!c->muted,
NULL, NULL);
if (!o)
GST_WARNING ("Failed to set device mute: %s",
pa_strerror (pa_context_errno (c->context)));
else
pa_operation_unref (o);
c->update_mute = FALSE;
}
/* Make sure that all outstanding queries are being ignored */
c->ignore_queries = c->outstandig_queries;
g_assert (e == c->time_event);
a->time_free (e);
c->time_event = NULL;
}
#define UPDATE_DELAY 50000
static void
restart_time_event (GstPulseMixerCtrl * c)
{
g_assert (c);
if (c->time_event)
return;
/* Updating the volume too often will cause a lot of traffic
* when accessing a networked server. Therefore we make sure
* to update the volume only once every 50ms */
struct timeval tv;
pa_mainloop_api *api = pa_threaded_mainloop_get_api (c->mainloop);
c->time_event =
api->time_new (api, pa_timeval_add (pa_gettimeofday (&tv), UPDATE_DELAY),
gst_pulsemixer_ctrl_timeout_event, c);
}
void
gst_pulsemixer_ctrl_set_volume (GstPulseMixerCtrl * c, GstMixerTrack * track,
gint * volumes)
{
pa_cvolume v;
int i;
g_assert (c);
g_assert (track == c->track);
pa_threaded_mainloop_lock (c->mainloop);
for (i = 0; i < c->channel_map.channels; i++)
v.values[i] = (pa_volume_t) volumes[i];
v.channels = c->channel_map.channels;
c->volume = v;
c->update_volume = TRUE;
restart_time_event (c);
pa_threaded_mainloop_unlock (c->mainloop);
}
void
gst_pulsemixer_ctrl_get_volume (GstPulseMixerCtrl * c, GstMixerTrack * track,
gint * volumes)
{
int i;
g_assert (c);
g_assert (track == c->track);
pa_threaded_mainloop_lock (c->mainloop);
for (i = 0; i < c->channel_map.channels; i++)
volumes[i] = c->volume.values[i];
pa_threaded_mainloop_unlock (c->mainloop);
}
void
gst_pulsemixer_ctrl_set_record (GstPulseMixerCtrl * c, GstMixerTrack * track,
gboolean record)
{
g_assert (c);
g_assert (track == c->track);
}
void
gst_pulsemixer_ctrl_set_mute (GstPulseMixerCtrl * c, GstMixerTrack * track,
gboolean mute)
{
g_assert (c);
g_assert (track == c->track);
pa_threaded_mainloop_lock (c->mainloop);
c->muted = !!mute;
c->update_mute = TRUE;
if (c->track) {
int i = g_atomic_int_get (&c->track->flags);
i = (i & ~GST_MIXER_TRACK_MUTE) | (c->muted ? GST_MIXER_TRACK_MUTE : 0);
g_atomic_int_set (&c->track->flags, i);
}
restart_time_event (c);
pa_threaded_mainloop_unlock (c->mainloop);
}

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/*
* GStreamer pulseaudio plugin
*
* Copyright (c) 2004-2008 Lennart Poettering
*
* gst-pulse 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.
*
* gst-pulse 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with gst-pulse; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA.
*/
#ifndef __GST_PULSEMIXERCTRL_H__
#define __GST_PULSEMIXERCTRL_H__
#include <gst/gst.h>
#include <gst/interfaces/mixer.h>
#include <pulse/pulseaudio.h>
#include <pulse/thread-mainloop.h>
G_BEGIN_DECLS
#define GST_PULSEMIXER_CTRL(obj) ((GstPulseMixerCtrl*)(obj))
typedef struct _GstPulseMixerCtrl GstPulseMixerCtrl;
typedef enum
{
GST_PULSEMIXER_UNKNOWN,
GST_PULSEMIXER_SINK,
GST_PULSEMIXER_SOURCE
} GstPulseMixerType;
struct _GstPulseMixerCtrl
{
GList *tracklist;
gchar *server, *device;
pa_threaded_mainloop *mainloop;
pa_context *context;
gchar *name, *description;
pa_channel_map channel_map;
pa_cvolume volume;
int muted;
guint32 index;
GstPulseMixerType type;
int operation_success;
GstMixerTrack *track;
pa_time_event *time_event;
int outstandig_queries;
int ignore_queries;
gboolean update_volume, update_mute;
};
GstPulseMixerCtrl *gst_pulsemixer_ctrl_new (const gchar * server,
const gchar * device, GstPulseMixerType type);
void gst_pulsemixer_ctrl_free (GstPulseMixerCtrl * mixer);
const GList *gst_pulsemixer_ctrl_list_tracks (GstPulseMixerCtrl * mixer);
void gst_pulsemixer_ctrl_set_volume (GstPulseMixerCtrl * mixer,
GstMixerTrack * track, gint * volumes);
void gst_pulsemixer_ctrl_get_volume (GstPulseMixerCtrl * mixer,
GstMixerTrack * track, gint * volumes);
void gst_pulsemixer_ctrl_set_mute (GstPulseMixerCtrl * mixer,
GstMixerTrack * track, gboolean mute);
void gst_pulsemixer_ctrl_set_record (GstPulseMixerCtrl * mixer,
GstMixerTrack * track, gboolean record);
#define GST_IMPLEMENT_PULSEMIXER_CTRL_METHODS(Type, interface_as_function) \
static const GList* \
interface_as_function ## _list_tracks (GstMixer * mixer) \
{ \
Type *this = (Type*) mixer; \
\
g_return_val_if_fail (this != NULL, NULL); \
g_return_val_if_fail (this->mixer != NULL, NULL); \
\
return gst_pulsemixer_ctrl_list_tracks (this->mixer); \
} \
static void \
interface_as_function ## _set_volume (GstMixer * mixer, GstMixerTrack * track, \
gint * volumes) \
{ \
Type *this = (Type*) mixer; \
\
g_return_if_fail (this != NULL); \
g_return_if_fail (this->mixer != NULL); \
\
gst_pulsemixer_ctrl_set_volume (this->mixer, track, volumes); \
} \
static void \
interface_as_function ## _get_volume (GstMixer * mixer, GstMixerTrack * track, \
gint * volumes) \
{ \
Type *this = (Type*) mixer; \
\
g_return_if_fail (this != NULL); \
g_return_if_fail (this->mixer != NULL); \
\
gst_pulsemixer_ctrl_get_volume (this->mixer, track, volumes); \
} \
static void \
interface_as_function ## _set_record (GstMixer * mixer, GstMixerTrack * track, \
gboolean record) \
{ \
Type *this = (Type*) mixer; \
\
g_return_if_fail (this != NULL); \
g_return_if_fail (this->mixer != NULL); \
\
gst_pulsemixer_ctrl_set_record (this->mixer, track, record); \
} \
static void \
interface_as_function ## _set_mute (GstMixer * mixer, GstMixerTrack * track, \
gboolean mute) \
{ \
Type *this = (Type*) mixer; \
\
g_return_if_fail (this != NULL); \
g_return_if_fail (this->mixer != NULL); \
\
gst_pulsemixer_ctrl_set_mute (this->mixer, track, mute); \
} \
static void \
interface_as_function ## _mixer_interface_init (GstMixerClass * klass) \
{ \
GST_MIXER_TYPE (klass) = GST_MIXER_HARDWARE; \
\
klass->list_tracks = interface_as_function ## _list_tracks; \
klass->set_volume = interface_as_function ## _set_volume; \
klass->get_volume = interface_as_function ## _get_volume; \
klass->set_mute = interface_as_function ## _set_mute; \
klass->set_record = interface_as_function ## _set_record; \
}
G_END_DECLS
#endif

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/*
* GStreamer pulseaudio plugin
*
* Copyright (c) 2004-2008 Lennart Poettering
*
* gst-pulse 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.
*
* gst-pulse 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with gst-pulse; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include "pulsemixertrack.h"
GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
#define GST_CAT_DEFAULT pulse_debug
G_DEFINE_TYPE (GstPulseMixerTrack, gst_pulsemixer_track, GST_TYPE_MIXER_TRACK);
static void
gst_pulsemixer_track_class_init (GstPulseMixerTrackClass * klass)
{
}
static void
gst_pulsemixer_track_init (GstPulseMixerTrack * track)
{
track->control = NULL;
}
GstMixerTrack *
gst_pulsemixer_track_new (GstPulseMixerCtrl * control)
{
GstPulseMixerTrack *pulsetrack;
GstMixerTrack *track;
pulsetrack = g_object_new (GST_TYPE_PULSEMIXER_TRACK, NULL);
pulsetrack->control = control;
track = GST_MIXER_TRACK (pulsetrack);
track->label = g_strdup ("Master");
track->num_channels = control->channel_map.channels;
track->flags =
(control->type ==
GST_PULSEMIXER_SINK ? GST_MIXER_TRACK_OUTPUT | GST_MIXER_TRACK_MASTER :
GST_MIXER_TRACK_INPUT | GST_MIXER_TRACK_RECORD) | (control->muted ?
GST_MIXER_TRACK_MUTE : 0);
track->min_volume = PA_VOLUME_MUTED;
track->max_volume = PA_VOLUME_NORM;
return track;
}

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/*
* GStreamer pulseaudio plugin
*
* Copyright (c) 2004-2008 Lennart Poettering
*
* gst-pulse 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.
*
* gst-pulse 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with gst-pulse; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA.
*/
#ifndef __GST_PULSEMIXERTRACK_H__
#define __GST_PULSEMIXERTRACK_H__
#include <gst/gst.h>
#include "pulsemixerctrl.h"
G_BEGIN_DECLS
#define GST_TYPE_PULSEMIXER_TRACK \
(gst_pulsemixer_track_get_type())
#define GST_PULSEMIXER_TRACK(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj), GST_TYPE_PULSEMIXER_TRACK, GstPulseMixerTrack))
#define GST_PULSEMIXER_TRACK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass), GST_TYPE_PULSEMIXER_TRACK, GstPulseMixerTrackClass))
#define GST_IS_PULSEMIXER_TRACK(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj), GST_TYPE_PULSEMIXER_TRACK))
#define GST_IS_PULSEMIXER_TRACK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass), GST_TYPE_PULSEMIXER_TRACK))
typedef struct _GstPulseMixerTrack
{
GstMixerTrack parent;
GstPulseMixerCtrl *control;
} GstPulseMixerTrack;
typedef struct _GstPulseMixerTrackClass
{
GstMixerTrackClass parent;
} GstPulseMixerTrackClass;
GType gst_pulsemixer_track_get_type (void);
GstMixerTrack *gst_pulsemixer_track_new (GstPulseMixerCtrl * control);
G_END_DECLS
#endif

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/*
* GStreamer pulseaudio plugin
*
* Copyright (c) 2004-2008 Lennart Poettering
*
* gst-pulse 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.
*
* gst-pulse 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with gst-pulse; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "pulseprobe.h"
#include "pulseutil.h"
GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
#define GST_CAT_DEFAULT pulse_debug
static void
gst_pulseprobe_context_state_cb (pa_context * context, void *userdata)
{
GstPulseProbe *c = (GstPulseProbe *) userdata;
/* Called from the background thread! */
switch (pa_context_get_state (context)) {
case PA_CONTEXT_READY:
case PA_CONTEXT_TERMINATED:
case PA_CONTEXT_FAILED:
pa_threaded_mainloop_signal (c->mainloop, 0);
break;
case PA_CONTEXT_UNCONNECTED:
case PA_CONTEXT_CONNECTING:
case PA_CONTEXT_AUTHORIZING:
case PA_CONTEXT_SETTING_NAME:
break;
}
}
static void
gst_pulseprobe_sink_info_cb (pa_context * context, const pa_sink_info * i,
int eol, void *userdata)
{
GstPulseProbe *c = (GstPulseProbe *) userdata;
/* Called from the background thread! */
if (eol || !i) {
c->operation_success = eol > 0;
pa_threaded_mainloop_signal (c->mainloop, 0);
}
if (i)
c->devices = g_list_append (c->devices, g_strdup (i->name));
}
static void
gst_pulseprobe_source_info_cb (pa_context * context, const pa_source_info * i,
int eol, void *userdata)
{
GstPulseProbe *c = (GstPulseProbe *) userdata;
/* Called from the background thread! */
if (eol || !i) {
c->operation_success = eol > 0;
pa_threaded_mainloop_signal (c->mainloop, 0);
}
if (i)
c->devices = g_list_append (c->devices, g_strdup (i->name));
}
static void
gst_pulseprobe_invalidate (GstPulseProbe * c)
{
g_list_foreach (c->devices, (GFunc) g_free, NULL);
g_list_free (c->devices);
c->devices = NULL;
c->devices_valid = 0;
}
static gboolean
gst_pulseprobe_open (GstPulseProbe * c)
{
int e;
gchar *name = gst_pulse_client_name ();
g_assert (c);
c->mainloop = pa_threaded_mainloop_new ();
g_assert (c->mainloop);
e = pa_threaded_mainloop_start (c->mainloop);
g_assert (e == 0);
pa_threaded_mainloop_lock (c->mainloop);
if (!(c->context =
pa_context_new (pa_threaded_mainloop_get_api (c->mainloop), name))) {
GST_WARNING ("Failed to create context");
goto unlock_and_fail;
}
pa_context_set_state_callback (c->context, gst_pulseprobe_context_state_cb,
c);
if (pa_context_connect (c->context, c->server, 0, NULL) < 0) {
GST_WARNING ("Failed to connect context: %s",
pa_strerror (pa_context_errno (c->context)));
goto unlock_and_fail;
}
/* Wait until the context is ready */
pa_threaded_mainloop_wait (c->mainloop);
if (pa_context_get_state (c->context) != PA_CONTEXT_READY) {
GST_WARNING ("Failed to connect context: %s",
pa_strerror (pa_context_errno (c->context)));
goto unlock_and_fail;
}
pa_threaded_mainloop_unlock (c->mainloop);
g_free (name);
gst_pulseprobe_invalidate (c);
return TRUE;
unlock_and_fail:
if (c->mainloop)
pa_threaded_mainloop_unlock (c->mainloop);
g_free (name);
return FALSE;
}
#define CHECK_DEAD_GOTO(c, label) do { \
if (!(c)->context || pa_context_get_state((c)->context) != PA_CONTEXT_READY) { \
GST_WARNING("Not connected: %s", (c)->context ? pa_strerror(pa_context_errno((c)->context)) : "NULL"); \
goto label; \
} \
} while(0);
static gboolean
gst_pulseprobe_enumerate (GstPulseProbe * c)
{
pa_operation *o = NULL;
pa_threaded_mainloop_lock (c->mainloop);
if (c->enumerate_sinks) {
/* Get sink info */
if (!(o =
pa_context_get_sink_info_list (c->context,
gst_pulseprobe_sink_info_cb, c))) {
GST_WARNING ("Failed to get sink info: %s",
pa_strerror (pa_context_errno (c->context)));
goto unlock_and_fail;
}
c->operation_success = 0;
while (pa_operation_get_state (o) != PA_OPERATION_DONE) {
pa_threaded_mainloop_wait (c->mainloop);
CHECK_DEAD_GOTO (c, unlock_and_fail);
}
if (!c->operation_success) {
GST_WARNING ("Failed to get sink info: %s",
pa_strerror (pa_context_errno (c->context)));
goto unlock_and_fail;
}
pa_operation_unref (o);
o = NULL;
}
if (c->enumerate_sources) {
/* Get source info */
if (!(o =
pa_context_get_source_info_list (c->context,
gst_pulseprobe_source_info_cb, c))) {
GST_WARNING ("Failed to get source info: %s",
pa_strerror (pa_context_errno (c->context)));
goto unlock_and_fail;
}
c->operation_success = 0;
while (pa_operation_get_state (o) != PA_OPERATION_DONE) {
pa_threaded_mainloop_wait (c->mainloop);
CHECK_DEAD_GOTO (c, unlock_and_fail);
}
if (!c->operation_success) {
GST_WARNING ("Failed to get sink info: %s",
pa_strerror (pa_context_errno (c->context)));
goto unlock_and_fail;
}
pa_operation_unref (o);
o = NULL;
}
c->devices_valid = 1;
pa_threaded_mainloop_unlock (c->mainloop);
return TRUE;
unlock_and_fail:
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (c->mainloop);
return FALSE;
}
static void
gst_pulseprobe_close (GstPulseProbe * c)
{
g_assert (c);
if (c->mainloop)
pa_threaded_mainloop_stop (c->mainloop);
if (c->context) {
pa_context_disconnect (c->context);
pa_context_unref (c->context);
c->context = NULL;
}
if (c->mainloop) {
pa_threaded_mainloop_free (c->mainloop);
c->mainloop = NULL;
}
}
GstPulseProbe *
gst_pulseprobe_new (GObjectClass * klass, guint prop_id, const gchar * server,
gboolean sinks, gboolean sources)
{
GstPulseProbe *c = NULL;
c = g_new (GstPulseProbe, 1);
c->server = g_strdup (server);
c->enumerate_sinks = sinks;
c->enumerate_sources = sources;
c->mainloop = NULL;
c->context = NULL;
c->prop_id = prop_id;
c->properties =
g_list_append (NULL, g_object_class_find_property (klass, "device"));
c->devices = NULL;
c->devices_valid = 0;
return c;
}
void
gst_pulseprobe_free (GstPulseProbe * c)
{
g_assert (c);
gst_pulseprobe_close (c);
g_list_free (c->properties);
g_free (c->server);
g_list_foreach (c->devices, (GFunc) g_free, NULL);
g_list_free (c->devices);
g_free (c);
}
const GList *
gst_pulseprobe_get_properties (GstPulseProbe * c)
{
return c->properties;
}
gboolean
gst_pulseprobe_needs_probe (GstPulseProbe * c, guint prop_id,
const GParamSpec * pspec)
{
if (prop_id == c->prop_id)
return !c->devices_valid;
G_OBJECT_WARN_INVALID_PROPERTY_ID (c, prop_id, pspec);
return FALSE;
}
void
gst_pulseprobe_probe_property (GstPulseProbe * c, guint prop_id,
const GParamSpec * pspec)
{
if (prop_id != c->prop_id) {
G_OBJECT_WARN_INVALID_PROPERTY_ID (c, prop_id, pspec);
return;
}
if (gst_pulseprobe_open (c)) {
gst_pulseprobe_enumerate (c);
gst_pulseprobe_close (c);
}
}
GValueArray *
gst_pulseprobe_get_values (GstPulseProbe * c, guint prop_id,
const GParamSpec * pspec)
{
GValueArray *array;
GValue value = { 0 };
GList *item;
if (prop_id != c->prop_id) {
G_OBJECT_WARN_INVALID_PROPERTY_ID (c, prop_id, pspec);
return NULL;
}
if (!c->devices_valid)
return NULL;
array = g_value_array_new (g_list_length (c->devices));
g_value_init (&value, G_TYPE_STRING);
for (item = c->devices; item != NULL; item = item->next) {
GST_WARNING ("device found: %s", (const gchar *) item->data);
g_value_set_string (&value, (const gchar *) item->data);
g_value_array_append (array, &value);
}
g_value_unset (&value);
return array;
}
void
gst_pulseprobe_set_server (GstPulseProbe * c, const gchar * server)
{
g_assert (c);
gst_pulseprobe_invalidate (c);
g_free (c->server);
c->server = g_strdup (server);
}

121
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/*
* GStreamer pulseaudio plugin
*
* Copyright (c) 2004-2008 Lennart Poettering
*
* gst-pulse 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.
*
* gst-pulse 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with gst-pulse; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA.
*/
#ifndef __GST_PULSEPROBE_H__
#define __GST_PULSEPROBE_H__
#include <gst/gst.h>
G_BEGIN_DECLS
#include <gst/interfaces/propertyprobe.h>
#include <pulse/pulseaudio.h>
#include <pulse/thread-mainloop.h>
typedef struct _GstPulseProbe GstPulseProbe;
struct _GstPulseProbe
{
gchar *server;
GList *devices;
int devices_valid;
pa_threaded_mainloop *mainloop;
pa_context *context;
GList *properties;
guint prop_id;
int enumerate_sinks, enumerate_sources;
int operation_success;
};
GstPulseProbe *gst_pulseprobe_new (GObjectClass * klass, guint prop_id,
const gchar * server, gboolean sinks, gboolean sources);
void gst_pulseprobe_free (GstPulseProbe * probe);
const GList *gst_pulseprobe_get_properties (GstPulseProbe * probe);
gboolean gst_pulseprobe_needs_probe (GstPulseProbe * probe, guint prop_id,
const GParamSpec * pspec);
void gst_pulseprobe_probe_property (GstPulseProbe * probe, guint prop_id,
const GParamSpec * pspec);
GValueArray *gst_pulseprobe_get_values (GstPulseProbe * probe, guint prop_id,
const GParamSpec * pspec);
void gst_pulseprobe_set_server (GstPulseProbe * c, const gchar * server);
#define GST_IMPLEMENT_PULSEPROBE_METHODS(Type, interface_as_function) \
static const GList* \
interface_as_function ## _get_properties(GstPropertyProbe * probe) \
{ \
Type *this = (Type*) probe; \
\
g_return_val_if_fail(this != NULL, NULL); \
g_return_val_if_fail(this->probe != NULL, NULL); \
\
return gst_pulseprobe_get_properties(this->probe); \
} \
static gboolean \
interface_as_function ## _needs_probe(GstPropertyProbe *probe, guint prop_id, \
const GParamSpec *pspec) \
{ \
Type *this = (Type*) probe; \
\
g_return_val_if_fail(this != NULL, FALSE); \
g_return_val_if_fail(this->probe != NULL, FALSE); \
\
return gst_pulseprobe_needs_probe(this->probe, prop_id, pspec); \
} \
static void \
interface_as_function ## _probe_property(GstPropertyProbe *probe, \
guint prop_id, const GParamSpec *pspec) \
{ \
Type *this = (Type*) probe; \
\
g_return_if_fail(this != NULL); \
g_return_if_fail(this->probe != NULL); \
\
gst_pulseprobe_probe_property(this->probe, prop_id, pspec); \
} \
static GValueArray* \
interface_as_function ## _get_values(GstPropertyProbe *probe, guint prop_id, \
const GParamSpec *pspec) \
{ \
Type *this = (Type*) probe; \
\
g_return_val_if_fail(this != NULL, NULL); \
g_return_val_if_fail(this->probe != NULL, NULL); \
\
return gst_pulseprobe_get_values(this->probe, prop_id, pspec); \
} \
static void \
interface_as_function ## _property_probe_interface_init(GstPropertyProbeInterface *iface)\
{ \
iface->get_properties = interface_as_function ## _get_properties; \
iface->needs_probe = interface_as_function ## _needs_probe; \
iface->probe_property = interface_as_function ## _probe_property; \
iface->get_values = interface_as_function ## _get_values; \
}
G_END_DECLS
#endif

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ext/pulse/pulsesink.c Normal file
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/*
* GStreamer pulseaudio plugin
*
* Copyright (c) 2004-2008 Lennart Poettering
*
* gst-pulse 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.
*
* gst-pulse 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with gst-pulse; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <stdio.h>
#include <gst/base/gstbasesink.h>
#include <gst/gsttaglist.h>
#include "pulsesink.h"
#include "pulseutil.h"
GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
#define GST_CAT_DEFAULT pulse_debug
enum
{
PROP_SERVER = 1,
PROP_DEVICE,
};
static GstAudioSinkClass *parent_class = NULL;
static void gst_pulsesink_destroy_stream (GstPulseSink * pulsesink);
static void gst_pulsesink_destroy_context (GstPulseSink * pulsesink);
static void gst_pulsesink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_pulsesink_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_pulsesink_finalize (GObject * object);
static void gst_pulsesink_dispose (GObject * object);
static gboolean gst_pulsesink_open (GstAudioSink * asink);
static gboolean gst_pulsesink_close (GstAudioSink * asink);
static gboolean gst_pulsesink_prepare (GstAudioSink * asink,
GstRingBufferSpec * spec);
static gboolean gst_pulsesink_unprepare (GstAudioSink * asink);
static guint gst_pulsesink_write (GstAudioSink * asink, gpointer data,
guint length);
static guint gst_pulsesink_delay (GstAudioSink * asink);
static void gst_pulsesink_reset (GstAudioSink * asink);
static gboolean gst_pulsesink_event (GstBaseSink * sink, GstEvent * event);
#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
# define ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
#else
# define ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
#endif
static void
gst_pulsesink_base_init (gpointer g_class)
{
static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) { " ENDIANNESS " }, "
"signed = (boolean) TRUE, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, 16 ];"
"audio/x-raw-float, "
"endianness = (int) { " ENDIANNESS " }, "
"width = (int) 32, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, 16 ];"
"audio/x-raw-int, "
"endianness = (int) { " ENDIANNESS " }, "
"signed = (boolean) TRUE, "
"width = (int) 32, "
"depth = (int) 32, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, 16 ];"
"audio/x-raw-int, "
"signed = (boolean) FALSE, "
"width = (int) 8, "
"depth = (int) 8, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, 16 ];"
"audio/x-alaw, "
"rate = (int) [ 1, MAX], "
"channels = (int) [ 1, 16 ];"
"audio/x-mulaw, "
"rate = (int) [ 1, MAX], " "channels = (int) [ 1, 16 ]")
);
static const GstElementDetails details =
GST_ELEMENT_DETAILS ("PulseAudio Audio Sink",
"Sink/Audio",
"Plays audio to a PulseAudio server",
"Lennart Poettering");
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details (element_class, &details);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&pad_template));
}
static void
gst_pulsesink_class_init (gpointer g_class, gpointer class_data)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (g_class);
GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (g_class);
GstAudioSinkClass *gstaudiosink_class = GST_AUDIO_SINK_CLASS (g_class);
parent_class = g_type_class_peek_parent (g_class);
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_pulsesink_dispose);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_pulsesink_finalize);
gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_pulsesink_set_property);
gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_pulsesink_get_property);
gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_pulsesink_event);
gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_pulsesink_open);
gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_pulsesink_close);
gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_pulsesink_prepare);
gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_pulsesink_unprepare);
gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_pulsesink_write);
gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_pulsesink_delay);
gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_pulsesink_reset);
/* Overwrite GObject fields */
g_object_class_install_property (gobject_class,
PROP_SERVER,
g_param_spec_string ("server", "Server",
"The PulseAudio server to connect to", NULL, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_DEVICE,
g_param_spec_string ("device", "Sink",
"The PulseAudio sink device to connect to", NULL, G_PARAM_READWRITE));
}
static void
gst_pulsesink_init (GTypeInstance * instance, gpointer g_class)
{
GstPulseSink *pulsesink = GST_PULSESINK (instance);
int e;
pulsesink->server = pulsesink->device = pulsesink->stream_name = NULL;
pulsesink->context = NULL;
pulsesink->stream = NULL;
pulsesink->mainloop = pa_threaded_mainloop_new ();
g_assert (pulsesink->mainloop);
e = pa_threaded_mainloop_start (pulsesink->mainloop);
g_assert (e == 0);
}
static void
gst_pulsesink_destroy_stream (GstPulseSink * pulsesink)
{
if (pulsesink->stream) {
pa_stream_disconnect (pulsesink->stream);
pa_stream_unref (pulsesink->stream);
pulsesink->stream = NULL;
}
g_free (pulsesink->stream_name);
pulsesink->stream_name = NULL;
}
static void
gst_pulsesink_destroy_context (GstPulseSink * pulsesink)
{
gst_pulsesink_destroy_stream (pulsesink);
if (pulsesink->context) {
pa_context_disconnect (pulsesink->context);
pa_context_unref (pulsesink->context);
pulsesink->context = NULL;
}
}
static void
gst_pulsesink_finalize (GObject * object)
{
GstPulseSink *pulsesink = GST_PULSESINK (object);
pa_threaded_mainloop_stop (pulsesink->mainloop);
gst_pulsesink_destroy_context (pulsesink);
g_free (pulsesink->server);
g_free (pulsesink->device);
g_free (pulsesink->stream_name);
pa_threaded_mainloop_free (pulsesink->mainloop);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_pulsesink_dispose (GObject * object)
{
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_pulsesink_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstPulseSink *pulsesink = GST_PULSESINK (object);
switch (prop_id) {
case PROP_SERVER:
g_free (pulsesink->server);
pulsesink->server = g_value_dup_string (value);
break;
case PROP_DEVICE:
g_free (pulsesink->device);
pulsesink->device = g_value_dup_string (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_pulsesink_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstPulseSink *pulsesink = GST_PULSESINK (object);
switch (prop_id) {
case PROP_SERVER:
g_value_set_string (value, pulsesink->server);
break;
case PROP_DEVICE:
g_value_set_string (value, pulsesink->device);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_pulsesink_context_state_cb (pa_context * c, void *userdata)
{
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
switch (pa_context_get_state (c)) {
case PA_CONTEXT_READY:
case PA_CONTEXT_TERMINATED:
case PA_CONTEXT_FAILED:
pa_threaded_mainloop_signal (pulsesink->mainloop, 0);
break;
case PA_CONTEXT_UNCONNECTED:
case PA_CONTEXT_CONNECTING:
case PA_CONTEXT_AUTHORIZING:
case PA_CONTEXT_SETTING_NAME:
break;
}
}
static void
gst_pulsesink_stream_state_cb (pa_stream * s, void *userdata)
{
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
switch (pa_stream_get_state (s)) {
case PA_STREAM_READY:
case PA_STREAM_FAILED:
case PA_STREAM_TERMINATED:
pa_threaded_mainloop_signal (pulsesink->mainloop, 0);
break;
case PA_STREAM_UNCONNECTED:
case PA_STREAM_CREATING:
break;
}
}
static void
gst_pulsesink_stream_request_cb (pa_stream * s, size_t length, void *userdata)
{
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
pa_threaded_mainloop_signal (pulsesink->mainloop, 0);
}
static void
gst_pulsesink_stream_latency_update_cb (pa_stream * s, void *userdata)
{
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
pa_threaded_mainloop_signal (pulsesink->mainloop, 0);
}
static gboolean
gst_pulsesink_open (GstAudioSink * asink)
{
GstPulseSink *pulsesink = GST_PULSESINK (asink);
gchar *name = gst_pulse_client_name ();
pa_threaded_mainloop_lock (pulsesink->mainloop);
if (!(pulsesink->context =
pa_context_new (pa_threaded_mainloop_get_api (pulsesink->mainloop),
name))) {
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
("Failed to create context"), (NULL));
goto unlock_and_fail;
}
pa_context_set_state_callback (pulsesink->context,
gst_pulsesink_context_state_cb, pulsesink);
if (pa_context_connect (pulsesink->context, pulsesink->server, 0, NULL) < 0) {
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED, ("Failed to connect: %s",
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
goto unlock_and_fail;
}
/* Wait until the context is ready */
pa_threaded_mainloop_wait (pulsesink->mainloop);
if (pa_context_get_state (pulsesink->context) != PA_CONTEXT_READY) {
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED, ("Failed to connect: %s",
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
goto unlock_and_fail;
}
pa_threaded_mainloop_unlock (pulsesink->mainloop);
g_free (name);
return TRUE;
unlock_and_fail:
pa_threaded_mainloop_unlock (pulsesink->mainloop);
g_free (name);
return FALSE;
}
static gboolean
gst_pulsesink_close (GstAudioSink * asink)
{
GstPulseSink *pulsesink = GST_PULSESINK (asink);
pa_threaded_mainloop_lock (pulsesink->mainloop);
gst_pulsesink_destroy_context (pulsesink);
pa_threaded_mainloop_unlock (pulsesink->mainloop);
return TRUE;
}
static gboolean
gst_pulsesink_prepare (GstAudioSink * asink, GstRingBufferSpec * spec)
{
pa_buffer_attr buf_attr;
pa_channel_map channel_map;
GstPulseSink *pulsesink = GST_PULSESINK (asink);
if (!gst_pulse_fill_sample_spec (spec, &pulsesink->sample_spec)) {
GST_ELEMENT_ERROR (pulsesink, RESOURCE, SETTINGS,
("Invalid sample specification."), (NULL));
goto unlock_and_fail;
}
pa_threaded_mainloop_lock (pulsesink->mainloop);
if (!pulsesink->context
|| pa_context_get_state (pulsesink->context) != PA_CONTEXT_READY) {
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED, ("Bad context state: %s",
pulsesink->context ? pa_strerror (pa_context_errno (pulsesink->
context)) : NULL), (NULL));
goto unlock_and_fail;
}
if (!(pulsesink->stream = pa_stream_new (pulsesink->context,
pulsesink->stream_name ? pulsesink->
stream_name : "Playback Stream", &pulsesink->sample_spec,
gst_pulse_gst_to_channel_map (&channel_map, spec)))) {
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
("Failed to create stream: %s",
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
goto unlock_and_fail;
}
pa_stream_set_state_callback (pulsesink->stream,
gst_pulsesink_stream_state_cb, pulsesink);
pa_stream_set_write_callback (pulsesink->stream,
gst_pulsesink_stream_request_cb, pulsesink);
pa_stream_set_latency_update_callback (pulsesink->stream,
gst_pulsesink_stream_latency_update_cb, pulsesink);
memset (&buf_attr, 0, sizeof (buf_attr));
buf_attr.tlength = spec->segtotal * spec->segsize;
buf_attr.maxlength = buf_attr.tlength * 2;
buf_attr.prebuf = buf_attr.tlength - spec->segsize;
buf_attr.minreq = spec->segsize;
if (pa_stream_connect_playback (pulsesink->stream, pulsesink->device,
&buf_attr,
PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
PA_STREAM_NOT_MONOTONOUS, NULL, NULL) < 0) {
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
goto unlock_and_fail;
}
/* Wait until the stream is ready */
pa_threaded_mainloop_wait (pulsesink->mainloop);
if (pa_stream_get_state (pulsesink->stream) != PA_STREAM_READY) {
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
goto unlock_and_fail;
}
pa_threaded_mainloop_unlock (pulsesink->mainloop);
spec->bytes_per_sample = pa_frame_size (&pulsesink->sample_spec);
memset (spec->silence_sample, 0, spec->bytes_per_sample);
return TRUE;
unlock_and_fail:
pa_threaded_mainloop_unlock (pulsesink->mainloop);
return FALSE;
}
static gboolean
gst_pulsesink_unprepare (GstAudioSink * asink)
{
GstPulseSink *pulsesink = GST_PULSESINK (asink);
pa_threaded_mainloop_lock (pulsesink->mainloop);
gst_pulsesink_destroy_stream (pulsesink);
pa_threaded_mainloop_unlock (pulsesink->mainloop);
return TRUE;
}
#define CHECK_DEAD_GOTO(pulsesink, label) \
if (!(pulsesink)->context || pa_context_get_state((pulsesink)->context) != PA_CONTEXT_READY || \
!(pulsesink)->stream || pa_stream_get_state((pulsesink)->stream) != PA_STREAM_READY) { \
GST_ELEMENT_ERROR((pulsesink), RESOURCE, FAILED, ("Disconnected: %s", (pulsesink)->context ? pa_strerror(pa_context_errno((pulsesink)->context)) : NULL), (NULL)); \
goto label; \
}
static guint
gst_pulsesink_write (GstAudioSink * asink, gpointer data, guint length)
{
GstPulseSink *pulsesink = GST_PULSESINK (asink);
size_t sum = 0;
pa_threaded_mainloop_lock (pulsesink->mainloop);
while (length > 0) {
size_t l;
for (;;) {
CHECK_DEAD_GOTO (pulsesink, unlock_and_fail);
if ((l = pa_stream_writable_size (pulsesink->stream)) == (size_t) - 1) {
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
("pa_stream_writable_size() failed: %s",
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
goto unlock_and_fail;
}
if (l > 0)
break;
pa_threaded_mainloop_wait (pulsesink->mainloop);
}
if (l > length)
l = length;
if (pa_stream_write (pulsesink->stream, data, l, NULL, 0,
PA_SEEK_RELATIVE) < 0) {
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
("pa_stream_write() failed: %s",
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
goto unlock_and_fail;
}
data = (guint8 *) data + l;
length -= l;
sum += l;
}
pa_threaded_mainloop_unlock (pulsesink->mainloop);
return sum;
unlock_and_fail:
pa_threaded_mainloop_unlock (pulsesink->mainloop);
return 0;
}
static guint
gst_pulsesink_delay (GstAudioSink * asink)
{
GstPulseSink *pulsesink = GST_PULSESINK (asink);
pa_usec_t t;
pa_threaded_mainloop_lock (pulsesink->mainloop);
for (;;) {
CHECK_DEAD_GOTO (pulsesink, unlock_and_fail);
if (pa_stream_get_latency (pulsesink->stream, &t, NULL) >= 0)
break;
if (pa_context_errno (pulsesink->context) != PA_ERR_NODATA) {
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
("pa_stream_get_latency() failed: %s",
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
goto unlock_and_fail;
}
pa_threaded_mainloop_wait (pulsesink->mainloop);
}
pa_threaded_mainloop_unlock (pulsesink->mainloop);
return gst_util_uint64_scale_int (t, pulsesink->sample_spec.rate, 1000000LL);
unlock_and_fail:
pa_threaded_mainloop_unlock (pulsesink->mainloop);
return 0;
}
static void
gst_pulsesink_success_cb (pa_stream * s, int success, void *userdata)
{
GstPulseSink *pulsesink = GST_PULSESINK (userdata);
pulsesink->operation_success = success;
pa_threaded_mainloop_signal (pulsesink->mainloop, 0);
}
static void
gst_pulsesink_reset (GstAudioSink * asink)
{
GstPulseSink *pulsesink = GST_PULSESINK (asink);
pa_operation *o = NULL;
pa_threaded_mainloop_lock (pulsesink->mainloop);
CHECK_DEAD_GOTO (pulsesink, unlock_and_fail);
if (!(o =
pa_stream_flush (pulsesink->stream, gst_pulsesink_success_cb,
pulsesink))) {
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
("pa_stream_flush() failed: %s",
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
goto unlock_and_fail;
}
pulsesink->operation_success = 0;
while (pa_operation_get_state (o) != PA_OPERATION_DONE) {
CHECK_DEAD_GOTO (pulsesink, unlock_and_fail);
pa_threaded_mainloop_wait (pulsesink->mainloop);
}
if (!pulsesink->operation_success) {
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED, ("Flush failed: %s",
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
goto unlock_and_fail;
}
unlock_and_fail:
if (o) {
pa_operation_cancel (o);
pa_operation_unref (o);
}
pa_threaded_mainloop_unlock (pulsesink->mainloop);
}
static void
gst_pulsesink_change_title (GstPulseSink * pulsesink, const gchar * t)
{
pa_operation *o = NULL;
pa_threaded_mainloop_lock (pulsesink->mainloop);
g_free (pulsesink->stream_name);
pulsesink->stream_name = g_strdup (t);
if (!(pulsesink)->context
|| pa_context_get_state ((pulsesink)->context) != PA_CONTEXT_READY
|| !(pulsesink)->stream
|| pa_stream_get_state ((pulsesink)->stream) != PA_STREAM_READY) {
goto unlock_and_fail;
}
if (!(o =
pa_stream_set_name (pulsesink->stream, pulsesink->stream_name, NULL,
pulsesink))) {
GST_ELEMENT_ERROR (pulsesink, RESOURCE, FAILED,
("pa_stream_set_name() failed: %s",
pa_strerror (pa_context_errno (pulsesink->context))), (NULL));
goto unlock_and_fail;
}
/* We're not interested if this operation failed or not */
unlock_and_fail:
if (o)
pa_operation_unref (o);
pa_threaded_mainloop_unlock (pulsesink->mainloop);
}
static gboolean
gst_pulsesink_event (GstBaseSink * sink, GstEvent * event)
{
GstPulseSink *pulsesink = GST_PULSESINK (sink);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_TAG:{
gchar *title = NULL, *artist = NULL, *location = NULL, *description =
NULL, *t = NULL, *buf = NULL;
GstTagList *l;
gst_event_parse_tag (event, &l);
gst_tag_list_get_string (l, GST_TAG_TITLE, &title);
gst_tag_list_get_string (l, GST_TAG_ARTIST, &artist);
gst_tag_list_get_string (l, GST_TAG_LOCATION, &location);
gst_tag_list_get_string (l, GST_TAG_DESCRIPTION, &description);
if (title && artist)
t = buf =
g_strdup_printf ("'%s' by '%s'", g_strstrip (title),
g_strstrip (artist));
else if (title)
t = g_strstrip (title);
else if (description)
t = g_strstrip (description);
else if (location)
t = g_strstrip (location);
if (t)
gst_pulsesink_change_title (pulsesink, t);
g_free (title);
g_free (artist);
g_free (location);
g_free (description);
g_free (buf);
break;
}
default:
;
}
return GST_BASE_SINK_CLASS (parent_class)->event (sink, event);
}
GType
gst_pulsesink_get_type (void)
{
static GType pulsesink_type = 0;
if (!pulsesink_type) {
static const GTypeInfo pulsesink_info = {
sizeof (GstPulseSinkClass),
gst_pulsesink_base_init,
NULL,
gst_pulsesink_class_init,
NULL,
NULL,
sizeof (GstPulseSink),
0,
gst_pulsesink_init,
};
pulsesink_type = g_type_register_static (GST_TYPE_AUDIO_SINK,
"GstPulseSink", &pulsesink_info, 0);
}
return pulsesink_type;
}

72
ext/pulse/pulsesink.h Normal file
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@ -0,0 +1,72 @@
/*
* GStreamer pulseaudio plugin
*
* Copyright (c) 2004-2008 Lennart Poettering
*
* gst-pulse 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.
*
* gst-pulse 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with gst-pulse; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA.
*/
#ifndef __GST_PULSESINK_H__
#define __GST_PULSESINK_H__
#include <gst/gst.h>
#include <gst/audio/gstaudiosink.h>
#include <pulse/pulseaudio.h>
#include <pulse/thread-mainloop.h>
G_BEGIN_DECLS
#define GST_TYPE_PULSESINK \
(gst_pulsesink_get_type())
#define GST_PULSESINK(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_PULSESINK,GstPulseSink))
#define GST_PULSESINK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_PULSESINK,GstPulseSinkClass))
#define GST_IS_PULSESINK(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_PULSESINK))
#define GST_IS_PULSESINK_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_PULSESINK))
typedef struct _GstPulseSink GstPulseSink;
typedef struct _GstPulseSinkClass GstPulseSinkClass;
struct _GstPulseSink
{
GstAudioSink sink;
gchar *server, *device, *stream_name;
pa_threaded_mainloop *mainloop;
pa_context *context;
pa_stream *stream;
pa_sample_spec sample_spec;
int operation_success;
};
struct _GstPulseSinkClass
{
GstAudioSinkClass parent_class;
};
GType gst_pulsesink_get_type (void);
G_END_DECLS
#endif /* __GST_PULSESINK_H__ */

703
ext/pulse/pulsesrc.c Normal file
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@ -0,0 +1,703 @@
/*
* GStreamer pulseaudio plugin
*
* Copyright (c) 2004-2008 Lennart Poettering
*
* gst-pulse 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.
*
* gst-pulse 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with gst-pulse; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <stdio.h>
#include <gst/base/gstbasesrc.h>
#include <gst/gsttaglist.h>
#include "pulsesrc.h"
#include "pulseutil.h"
#include "pulsemixerctrl.h"
GST_DEBUG_CATEGORY_EXTERN (pulse_debug);
#define GST_CAT_DEFAULT pulse_debug
enum
{
PROP_SERVER = 1,
PROP_DEVICE
};
static GstAudioSrcClass *parent_class = NULL;
GST_IMPLEMENT_PULSEMIXER_CTRL_METHODS (GstPulseSrc, gst_pulsesrc)
static void gst_pulsesrc_destroy_stream (GstPulseSrc * pulsesrc);
static void gst_pulsesrc_destroy_context (GstPulseSrc * pulsesrc);
static void gst_pulsesrc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_pulsesrc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_pulsesrc_finalize (GObject * object);
static void gst_pulsesrc_dispose (GObject * object);
static gboolean gst_pulsesrc_open (GstAudioSrc * asrc);
static gboolean gst_pulsesrc_close (GstAudioSrc * asrc);
static gboolean gst_pulsesrc_prepare (GstAudioSrc * asrc,
GstRingBufferSpec * spec);
static gboolean gst_pulsesrc_unprepare (GstAudioSrc * asrc);
static guint gst_pulsesrc_read (GstAudioSrc * asrc, gpointer data,
guint length);
static guint gst_pulsesrc_delay (GstAudioSrc * asrc);
static GstStateChangeReturn gst_pulsesrc_change_state (GstElement *
element, GstStateChange transition);
#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
# define ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN"
#else
# define ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN"
#endif
static gboolean gst_pulsesrc_interface_supported (GstImplementsInterface *
iface, GType interface_type)
{
GstPulseSrc *this = GST_PULSESRC (iface);
if (interface_type == GST_TYPE_MIXER && this->mixer)
return TRUE;
return FALSE;
}
static void
gst_pulsesrc_implements_interface_init (GstImplementsInterfaceClass * klass)
{
klass->supported = gst_pulsesrc_interface_supported;
}
static void
gst_pulsesrc_init_interfaces (GType type)
{
static const GInterfaceInfo implements_iface_info = {
(GInterfaceInitFunc) gst_pulsesrc_implements_interface_init,
NULL,
NULL,
};
static const GInterfaceInfo mixer_iface_info = {
(GInterfaceInitFunc) gst_pulsesrc_mixer_interface_init,
NULL,
NULL,
};
g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE,
&implements_iface_info);
g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info);
}
static void
gst_pulsesrc_base_init (gpointer g_class)
{
static GstStaticPadTemplate pad_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, "
"endianness = (int) { " ENDIANNESS " }, "
"signed = (boolean) TRUE, "
"width = (int) 16, "
"depth = (int) 16, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, 16 ];"
"audio/x-raw-int, "
"endianness = (int) { " ENDIANNESS " }, "
"signed = (boolean) TRUE, "
"width = (int) 32, "
"depth = (int) 32, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, 16 ];"
"audio/x-raw-float, "
"endianness = (int) { " ENDIANNESS " }, "
"width = (int) 32, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, 16 ];"
"audio/x-raw-int, "
"signed = (boolean) FALSE, "
"width = (int) 8, "
"depth = (int) 8, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, 16 ];"
"audio/x-alaw, "
"rate = (int) [ 1, MAX], "
"channels = (int) [ 1, 16 ];"
"audio/x-mulaw, "
"rate = (int) [ 1, MAX], " "channels = (int) [ 1, 16 ]")
);
static const GstElementDetails details =
GST_ELEMENT_DETAILS ("PulseAudio Audio Source",
"Source/Audio",
"Captures audio from a PulseAudio server",
"Lennart Poettering");
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_set_details (element_class, &details);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&pad_template));
}
static void
gst_pulsesrc_class_init (gpointer g_class, gpointer class_data)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (g_class);
GstAudioSrcClass *gstaudiosrc_class = GST_AUDIO_SRC_CLASS (g_class);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class);
parent_class = g_type_class_peek_parent (g_class);
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_pulsesrc_change_state);
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_pulsesrc_dispose);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_pulsesrc_finalize);
gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_pulsesrc_set_property);
gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_pulsesrc_get_property);
gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_pulsesrc_open);
gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_pulsesrc_close);
gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_pulsesrc_prepare);
gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_pulsesrc_unprepare);
gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_pulsesrc_read);
gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_pulsesrc_delay);
/* Overwrite GObject fields */
g_object_class_install_property (gobject_class,
PROP_SERVER,
g_param_spec_string ("server", "Server",
"The PulseAudio server to connect to", NULL, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_DEVICE,
g_param_spec_string ("device", "Source",
"The PulseAudio source device to connect to", NULL,
G_PARAM_READWRITE));
}
static void
gst_pulsesrc_init (GTypeInstance * instance, gpointer g_class)
{
GstPulseSrc *pulsesrc = GST_PULSESRC (instance);
int e;
pulsesrc->server = pulsesrc->device = NULL;
pulsesrc->context = NULL;
pulsesrc->stream = NULL;
pulsesrc->read_buffer = NULL;
pulsesrc->read_buffer_length = 0;
pulsesrc->mainloop = pa_threaded_mainloop_new ();
g_assert (pulsesrc->mainloop);
e = pa_threaded_mainloop_start (pulsesrc->mainloop);
g_assert (e == 0);
pulsesrc->mixer = NULL;
}
static void
gst_pulsesrc_destroy_stream (GstPulseSrc * pulsesrc)
{
if (pulsesrc->stream) {
pa_stream_disconnect (pulsesrc->stream);
pa_stream_unref (pulsesrc->stream);
pulsesrc->stream = NULL;
}
}
static void
gst_pulsesrc_destroy_context (GstPulseSrc * pulsesrc)
{
gst_pulsesrc_destroy_stream (pulsesrc);
if (pulsesrc->context) {
pa_context_disconnect (pulsesrc->context);
pa_context_unref (pulsesrc->context);
pulsesrc->context = NULL;
}
}
static void
gst_pulsesrc_finalize (GObject * object)
{
GstPulseSrc *pulsesrc = GST_PULSESRC (object);
pa_threaded_mainloop_stop (pulsesrc->mainloop);
gst_pulsesrc_destroy_context (pulsesrc);
g_free (pulsesrc->server);
g_free (pulsesrc->device);
pa_threaded_mainloop_free (pulsesrc->mainloop);
if (pulsesrc->mixer)
gst_pulsemixer_ctrl_free (pulsesrc->mixer);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_pulsesrc_dispose (GObject * object)
{
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_pulsesrc_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstPulseSrc *pulsesrc = GST_PULSESRC (object);
switch (prop_id) {
case PROP_SERVER:
g_free (pulsesrc->server);
pulsesrc->server = g_value_dup_string (value);
break;
case PROP_DEVICE:
g_free (pulsesrc->device);
pulsesrc->device = g_value_dup_string (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_pulsesrc_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstPulseSrc *pulsesrc = GST_PULSESRC (object);
switch (prop_id) {
case PROP_SERVER:
g_value_set_string (value, pulsesrc->server);
break;
case PROP_DEVICE:
g_value_set_string (value, pulsesrc->device);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_pulsesrc_context_state_cb (pa_context * c, void *userdata)
{
GstPulseSrc *pulsesrc = GST_PULSESRC (userdata);
switch (pa_context_get_state (c)) {
case PA_CONTEXT_READY:
case PA_CONTEXT_TERMINATED:
case PA_CONTEXT_FAILED:
pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
break;
case PA_CONTEXT_UNCONNECTED:
case PA_CONTEXT_CONNECTING:
case PA_CONTEXT_AUTHORIZING:
case PA_CONTEXT_SETTING_NAME:
break;
}
}
static void
gst_pulsesrc_stream_state_cb (pa_stream * s, void *userdata)
{
GstPulseSrc *pulsesrc = GST_PULSESRC (userdata);
switch (pa_stream_get_state (s)) {
case PA_STREAM_READY:
case PA_STREAM_FAILED:
case PA_STREAM_TERMINATED:
pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
break;
case PA_STREAM_UNCONNECTED:
case PA_STREAM_CREATING:
break;
}
}
static void
gst_pulsesrc_stream_request_cb (pa_stream * s, size_t length, void *userdata)
{
GstPulseSrc *pulsesrc = GST_PULSESRC (userdata);
pa_threaded_mainloop_signal (pulsesrc->mainloop, 0);
}
static gboolean
gst_pulsesrc_open (GstAudioSrc * asrc)
{
GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);
gchar *name = gst_pulse_client_name ();
pa_threaded_mainloop_lock (pulsesrc->mainloop);
if (!(pulsesrc->context =
pa_context_new (pa_threaded_mainloop_get_api (pulsesrc->mainloop),
name))) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to create context"),
(NULL));
goto unlock_and_fail;
}
pa_context_set_state_callback (pulsesrc->context,
gst_pulsesrc_context_state_cb, pulsesrc);
if (pa_context_connect (pulsesrc->context, pulsesrc->server, 0, NULL) < 0) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to connect: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
/* Wait until the context is ready */
pa_threaded_mainloop_wait (pulsesrc->mainloop);
if (pa_context_get_state (pulsesrc->context) != PA_CONTEXT_READY) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Failed to connect: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
g_free (name);
return TRUE;
unlock_and_fail:
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
g_free (name);
return FALSE;
}
static gboolean
gst_pulsesrc_close (GstAudioSrc * asrc)
{
GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);
pa_threaded_mainloop_lock (pulsesrc->mainloop);
gst_pulsesrc_destroy_context (pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return TRUE;
}
static gboolean
gst_pulsesrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
{
pa_buffer_attr buf_attr;
pa_channel_map channel_map;
GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);
if (!gst_pulse_fill_sample_spec (spec, &pulsesrc->sample_spec)) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, SETTINGS,
("Invalid sample specification."), (NULL));
goto unlock_and_fail;
}
pa_threaded_mainloop_lock (pulsesrc->mainloop);
if (!pulsesrc->context
|| pa_context_get_state (pulsesrc->context) != PA_CONTEXT_READY) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED, ("Bad context state: %s",
pulsesrc->context ? pa_strerror (pa_context_errno (pulsesrc->
context)) : NULL), (NULL));
goto unlock_and_fail;
}
if (!(pulsesrc->stream = pa_stream_new (pulsesrc->context,
"Record Stream",
&pulsesrc->sample_spec,
gst_pulse_gst_to_channel_map (&channel_map, spec)))) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to create stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
pa_stream_set_state_callback (pulsesrc->stream, gst_pulsesrc_stream_state_cb,
pulsesrc);
pa_stream_set_read_callback (pulsesrc->stream, gst_pulsesrc_stream_request_cb,
pulsesrc);
memset (&buf_attr, 0, sizeof (buf_attr));
buf_attr.maxlength = spec->segtotal * spec->segsize * 2;
buf_attr.fragsize = spec->segsize;
if (pa_stream_connect_record (pulsesrc->stream, pulsesrc->device, &buf_attr,
PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_AUTO_TIMING_UPDATE |
PA_STREAM_NOT_MONOTONOUS) < 0) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
/* Wait until the stream is ready */
pa_threaded_mainloop_wait (pulsesrc->mainloop);
if (pa_stream_get_state (pulsesrc->stream) != PA_STREAM_READY) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("Failed to connect stream: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
spec->bytes_per_sample = pa_frame_size (&pulsesrc->sample_spec);
memset (spec->silence_sample, 0, spec->bytes_per_sample);
return TRUE;
unlock_and_fail:
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return FALSE;
}
static gboolean
gst_pulsesrc_unprepare (GstAudioSrc * asrc)
{
GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);
pa_threaded_mainloop_lock (pulsesrc->mainloop);
gst_pulsesrc_destroy_stream (pulsesrc);
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
pulsesrc->read_buffer = NULL;
pulsesrc->read_buffer_length = 0;
return TRUE;
}
#define CHECK_DEAD_GOTO(pulsesrc, label) \
if (!(pulsesrc)->context || pa_context_get_state((pulsesrc)->context) != PA_CONTEXT_READY || \
!(pulsesrc)->stream || pa_stream_get_state((pulsesrc)->stream) != PA_STREAM_READY) { \
GST_ELEMENT_ERROR((pulsesrc), RESOURCE, FAILED, ("Disconnected: %s", (pulsesrc)->context ? pa_strerror(pa_context_errno((pulsesrc)->context)) : NULL), (NULL)); \
goto label; \
}
static guint
gst_pulsesrc_read (GstAudioSrc * asrc, gpointer data, guint length)
{
GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);
size_t sum = 0;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
CHECK_DEAD_GOTO (pulsesrc, unlock_and_fail);
while (length > 0) {
size_t l;
if (!pulsesrc->read_buffer) {
for (;;) {
if (pa_stream_peek (pulsesrc->stream, &pulsesrc->read_buffer,
&pulsesrc->read_buffer_length) < 0) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_stream_peek() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
if (pulsesrc->read_buffer)
break;
pa_threaded_mainloop_wait (pulsesrc->mainloop);
CHECK_DEAD_GOTO (pulsesrc, unlock_and_fail);
}
}
g_assert (pulsesrc->read_buffer && pulsesrc->read_buffer_length);
l = pulsesrc->read_buffer_length >
length ? length : pulsesrc->read_buffer_length;
memcpy (data, pulsesrc->read_buffer, l);
pulsesrc->read_buffer = (const guint8 *) pulsesrc->read_buffer + l;
pulsesrc->read_buffer_length -= l;
data = (guint8 *) data + l;
length -= l;
sum += l;
if (pulsesrc->read_buffer_length <= 0) {
if (pa_stream_drop (pulsesrc->stream) < 0) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_stream_drop() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
pulsesrc->read_buffer = NULL;
pulsesrc->read_buffer_length = 0;
}
}
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return sum;
unlock_and_fail:
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return 0;
}
static guint
gst_pulsesrc_delay (GstAudioSrc * asrc)
{
GstPulseSrc *pulsesrc = GST_PULSESRC (asrc);
pa_usec_t t;
int negative;
pa_threaded_mainloop_lock (pulsesrc->mainloop);
CHECK_DEAD_GOTO (pulsesrc, unlock_and_fail);
if (pa_stream_get_latency (pulsesrc->stream, &t, &negative) < 0) {
if (pa_context_errno (pulsesrc->context) != PA_ERR_NODATA) {
GST_ELEMENT_ERROR (pulsesrc, RESOURCE, FAILED,
("pa_stream_get_latency() failed: %s",
pa_strerror (pa_context_errno (pulsesrc->context))), (NULL));
goto unlock_and_fail;
}
GST_WARNING ("Not data while querying latency");
t = 0;
} else if (negative)
t = 0;
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return (guint) ((t * pulsesrc->sample_spec.rate) / 1000000LL);
unlock_and_fail:
pa_threaded_mainloop_unlock (pulsesrc->mainloop);
return 0;
}
static GstStateChangeReturn
gst_pulsesrc_change_state (GstElement * element, GstStateChange transition)
{
GstPulseSrc *this = GST_PULSESRC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!this->mixer)
this->mixer =
gst_pulsemixer_ctrl_new (this->server, this->device,
GST_PULSEMIXER_SOURCE);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
if (this->mixer) {
gst_pulsemixer_ctrl_free (this->mixer);
this->mixer = NULL;
}
break;
default:
;
}
if (GST_ELEMENT_CLASS (parent_class)->change_state)
return GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
return GST_STATE_CHANGE_SUCCESS;
}
GType
gst_pulsesrc_get_type (void)
{
static GType pulsesrc_type = 0;
if (!pulsesrc_type) {
static const GTypeInfo pulsesrc_info = {
sizeof (GstPulseSrcClass),
gst_pulsesrc_base_init,
NULL,
gst_pulsesrc_class_init,
NULL,
NULL,
sizeof (GstPulseSrc),
0,
gst_pulsesrc_init,
};
pulsesrc_type = g_type_register_static (GST_TYPE_AUDIO_SRC,
"GstPulseSrc", &pulsesrc_info, 0);
gst_pulsesrc_init_interfaces (pulsesrc_type);
}
return pulsesrc_type;
}

77
ext/pulse/pulsesrc.h Normal file
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@ -0,0 +1,77 @@
/*
* GStreamer pulseaudio plugin
*
* Copyright (c) 2004-2008 Lennart Poettering
*
* gst-pulse 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.
*
* gst-pulse 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with gst-pulse; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA.
*/
#ifndef __GST_PULSESRC_H__
#define __GST_PULSESRC_H__
#include <gst/gst.h>
#include <gst/audio/gstaudiosrc.h>
#include <pulse/pulseaudio.h>
#include <pulse/thread-mainloop.h>
#include "pulsemixerctrl.h"
G_BEGIN_DECLS
#define GST_TYPE_PULSESRC \
(gst_pulsesrc_get_type())
#define GST_PULSESRC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_PULSESRC,GstPulseSrc))
#define GST_PULSESRC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_PULSESRC,GstPulseSrcClass))
#define GST_IS_PULSESRC(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_PULSESRC))
#define GST_IS_PULSESRC_CLASS(obj) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_PULSESRC))
typedef struct _GstPulseSrc GstPulseSrc;
typedef struct _GstPulseSrcClass GstPulseSrcClass;
struct _GstPulseSrc
{
GstAudioSrc src;
gchar *server, *device;
pa_threaded_mainloop *mainloop;
pa_context *context;
pa_stream *stream;
pa_sample_spec sample_spec;
const void *read_buffer;
size_t read_buffer_length;
GstPulseMixerCtrl *mixer;
};
struct _GstPulseSrcClass
{
GstAudioSrcClass parent_class;
};
GType gst_pulsesrc_get_type (void);
G_END_DECLS
#endif /* __GST_PULSESRC_H__ */

138
ext/pulse/pulseutil.c Normal file
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/*
* GStreamer pulseaudio plugin
*
* Copyright (c) 2004-2008 Lennart Poettering
*
* gst-pulse 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.
*
* gst-pulse 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with gst-pulse; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "pulseutil.h"
#include <gst/audio/multichannel.h>
static const pa_channel_position_t gst_pos_to_pa[GST_AUDIO_CHANNEL_POSITION_NUM]
= {
[GST_AUDIO_CHANNEL_POSITION_FRONT_MONO] = PA_CHANNEL_POSITION_MONO,
[GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT] = PA_CHANNEL_POSITION_FRONT_LEFT,
[GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT] = PA_CHANNEL_POSITION_FRONT_RIGHT,
[GST_AUDIO_CHANNEL_POSITION_REAR_CENTER] = PA_CHANNEL_POSITION_REAR_CENTER,
[GST_AUDIO_CHANNEL_POSITION_REAR_LEFT] = PA_CHANNEL_POSITION_REAR_LEFT,
[GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT] = PA_CHANNEL_POSITION_REAR_RIGHT,
[GST_AUDIO_CHANNEL_POSITION_LFE] = PA_CHANNEL_POSITION_LFE,
[GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER] = PA_CHANNEL_POSITION_FRONT_CENTER,
[GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER] =
PA_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER,
[GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER] =
PA_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER,
[GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT] = PA_CHANNEL_POSITION_SIDE_LEFT,
[GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT] = PA_CHANNEL_POSITION_SIDE_RIGHT,
[GST_AUDIO_CHANNEL_POSITION_NONE] = PA_CHANNEL_POSITION_INVALID
};
gboolean
gst_pulse_fill_sample_spec (GstRingBufferSpec * spec, pa_sample_spec * ss)
{
if (spec->format == GST_MU_LAW && spec->width == 8)
ss->format = PA_SAMPLE_ULAW;
else if (spec->format == GST_A_LAW && spec->width == 8)
ss->format = PA_SAMPLE_ALAW;
else if (spec->format == GST_U8 && spec->width == 8)
ss->format = PA_SAMPLE_U8;
else if (spec->format == GST_S16_LE && spec->width == 16)
ss->format = PA_SAMPLE_S16LE;
else if (spec->format == GST_S16_BE && spec->width == 16)
ss->format = PA_SAMPLE_S16BE;
else if (spec->format == GST_FLOAT32_LE && spec->width == 32)
ss->format = PA_SAMPLE_FLOAT32LE;
else if (spec->format == GST_FLOAT32_BE && spec->width == 32)
ss->format = PA_SAMPLE_FLOAT32BE;
else if (spec->format == GST_S32_LE && spec->width == 32)
ss->format = PA_SAMPLE_S32LE;
else if (spec->format == GST_S32_BE && spec->width == 32)
ss->format = PA_SAMPLE_S32BE;
else
return FALSE;
ss->channels = spec->channels;
ss->rate = spec->rate;
if (!pa_sample_spec_valid (ss))
return FALSE;
return TRUE;
}
gchar *
gst_pulse_client_name (void)
{
gchar buf[PATH_MAX];
const char *c;
if ((c = g_get_application_name ()))
return g_strdup_printf ("%s", c);
else if (pa_get_binary_name (buf, sizeof (buf)))
return g_strdup_printf ("%s", buf);
else
return g_strdup ("GStreamer");
}
pa_channel_map *
gst_pulse_gst_to_channel_map (pa_channel_map * map, GstRingBufferSpec * spec)
{
int i;
GstAudioChannelPosition *pos;
pa_channel_map_init (map);
if (!(pos =
gst_audio_get_channel_positions (gst_caps_get_structure (spec->caps,
0)))) {
/* g_debug("%s: No channel positions!\n", G_STRFUNC); */
return NULL;
}
/* g_debug("%s: Got channel positions:\n", G_STRFUNC); */
for (i = 0; i < spec->channels; i++) {
if (pos[i] == GST_AUDIO_CHANNEL_POSITION_NONE) {
/* no valid mappings for these channels */
g_free (pos);
return NULL;
} else if (pos[i] < GST_AUDIO_CHANNEL_POSITION_NUM)
map->map[i] = gst_pos_to_pa[pos[i]];
else
map->map[i] = PA_CHANNEL_POSITION_INVALID;
/*g_debug(" channel %d: gst: %d pulse: %d\n", i, pos[i], map->map[i]); */
}
g_free (pos);
map->channels = spec->channels;
if (!pa_channel_map_valid (map)) {
/* g_debug("generated invalid map!\n"); */
return NULL;
}
return map;
}

37
ext/pulse/pulseutil.h Normal file
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@ -0,0 +1,37 @@
/*
* GStreamer pulseaudio plugin
*
* Copyright (c) 2004-2008 Lennart Poettering
*
* gst-pulse 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.
*
* gst-pulse 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with gst-pulse; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
* USA.
*/
#ifndef __GST_PULSEUTIL_H__
#define __GST_PULSEUTIL_H__
#include <gst/gst.h>
#include <pulse/pulseaudio.h>
#include <gst/audio/gstaudiosink.h>
gboolean gst_pulse_fill_sample_spec (GstRingBufferSpec * spec,
pa_sample_spec * ss);
gchar *gst_pulse_client_name (void);
pa_channel_map *gst_pulse_gst_to_channel_map (pa_channel_map * map,
GstRingBufferSpec * spec);
#endif