+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>
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, [
ext/ladspa/Makefile
ext/libcaca/Makefile
ext/libpng/Makefile
+ext/pulse/Makefile
ext/raw1394/Makefile
ext/shout2/Makefile
ext/soup/Makefile
--- /dev/null
+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
+
--- /dev/null
+/*
+ * 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)
--- /dev/null
+/*
+ * 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;
+}
--- /dev/null
+/*
+ * 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__ */
--- /dev/null
+/*
+ * 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);
+}
--- /dev/null
+/*
+ * 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
--- /dev/null
+/*
+ * 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;
+}
--- /dev/null
+/*
+ * 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
--- /dev/null
+/*
+ * 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);
+}
--- /dev/null
+/*
+ * 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
--- /dev/null
+/*
+ * 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;
+}
--- /dev/null
+/*
+ * 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__ */
--- /dev/null
+/*
+ * 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;
+}
--- /dev/null
+/*
+ * 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__ */
--- /dev/null
+/*
+ * 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;
+}
--- /dev/null
+/*
+ * 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
projects / platform / upstream / gst-plugins-good.git / commitdiff