pulse: New pulseaudiosink element to handle format changes

This introduces a new bin which wraps around pulsesink and depending on
the formats supported by the sink, plugs in/out a decodebin2 as
required. This allows users to switch sinks on the stream and adapts
accordingly (for example, you could watch a movie in passthrough mode on
your receiver which supports AC3 decode, then plug out and switch to a
non-digital profile to continue uninterrupted on analog output).

The bin is required because doing the same with playbin2/playsink will
require API changes that cannot be made in 0.10. With 0.11/1.0, we
should be able to ask for upstream caps renegotiation to deal with all
this.

https://bugzilla.gnome.org/show_bug.cgi?id=657179

ext/pulse/Makefile.am

@@ -7,12 +7,14 @@ libgstpulse_la_SOURCES = \
 	pulsemixertrack.c \
 	pulseprobe.c \
 	pulsesink.c \
+	pulseaudiosink.c \
 	pulsesrc.c \
 	pulseutil.c
 
 libgstpulse_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(PULSE_CFLAGS)
 libgstpulse_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) \
-	-lgstinterfaces-$(GST_MAJORMINOR) $(GST_BASE_LIBS) $(GST_LIBS) $(PULSE_LIBS)
+	-lgstinterfaces-$(GST_MAJORMINOR) -lgstpbutils-$(GST_MAJORMINOR) \
+	$(GST_BASE_LIBS) $(GST_LIBS) $(PULSE_LIBS)
 libgstpulse_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
 libgstpulse_la_LIBTOOLFLAGS = --tag=disable-static
 

ext/pulse/plugin.c

@@ -49,6 +49,12 @@ plugin_init (GstPlugin * plugin)
           GST_TYPE_PULSESRC))
     return FALSE;
 
+#ifdef HAVE_PULSE_1_0
+  if (!gst_element_register (plugin, "pulseaudiosink", GST_RANK_PRIMARY + 11,
+          GST_TYPE_PULSE_AUDIO_SINK))
+    return FALSE;
+#endif
+
   if (!gst_element_register (plugin, "pulsemixer", GST_RANK_NONE,
           GST_TYPE_PULSEMIXER))
     return FALSE;

ext/pulse/pulseaudiosink.c

@@ -0,0 +1,927 @@
+/*-*- Mode: C; c-basic-offset: 2 -*-*/
+
+/*  GStreamer pulseaudio plugin
+ *
+ *  Copyright (c) 2011 Intel Corporation
+ *                2011 Collabora
+ *                2011 Arun Raghavan <arun.raghavan@collabora.co.uk>
+ *                2011 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ *
+ *  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.
+ */
+
+/**
+ * SECTION:element-pulseaudiosink
+ * @see_also: pulsesink, pulsesrc, pulsemixer
+ *
+ * This element outputs audio to a
+ * <ulink href="http://www.pulseaudio.org">PulseAudio sound server</ulink> via
+ * the @pulsesink element. It transparently takes care of passing compressed
+ * format as-is if the sink supports it, decoding if necessary, and changes
+ * to supported formats at runtime.
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! pulseaudiosink
+ * ]| Decode and play an Ogg/Vorbis file.
+ * |[
+ * gst-launch -v filesrc location=test.mp3 ! mp3parse ! pulseaudiosink stream-properties="props,media.title=test"
+ * ]| Play an MP3 file on a sink that supports decoding directly, plug in a
+ * decoder if/when required.
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#ifdef HAVE_PULSE_1_0
+
+#include <gst/pbutils/pbutils.h>
+#include <gst/gst-i18n-plugin.h>
+
+#include <gst/audio/gstaudioiec61937.h>
+#include "pulsesink.h"
+
+GST_DEBUG_CATEGORY (pulseaudiosink_debug);
+#define GST_CAT_DEFAULT (pulseaudiosink_debug)
+
+#define GST_PULSE_AUDIO_SINK_LOCK(obj) G_STMT_START {                    \
+    GST_LOG_OBJECT (obj,                                              \
+                    "locking from thread %p",                         \
+                    g_thread_self ());                                \
+    g_mutex_lock (GST_PULSE_AUDIO_SINK_CAST(obj)->lock);                 \
+    GST_LOG_OBJECT (obj,                                              \
+                    "locked from thread %p",                          \
+                    g_thread_self ());                                \
+} G_STMT_END
+
+#define GST_PULSE_AUDIO_SINK_UNLOCK(obj) G_STMT_START {                  \
+    GST_LOG_OBJECT (obj,                                              \
+                    "unlocking from thread %p",                       \
+                    g_thread_self ());                                \
+    g_mutex_unlock (GST_PULSE_AUDIO_SINK_CAST(obj)->lock);               \
+} G_STMT_END
+
+typedef struct
+{
+  GstBin parent;
+  GMutex *lock;
+
+  GstPad *sinkpad;
+  GstPad *sink_proxypad;
+  GstPadEventFunction sinkpad_old_eventfunc;
+  GstPadEventFunction proxypad_old_eventfunc;
+
+  GstPulseSink *psink;
+  GstElement *dbin2;
+
+  GstSegment segment;
+
+  guint event_probe_id;
+  gulong pad_added_id;
+
+  gboolean format_lost;
+} GstPulseAudioSink;
+
+typedef struct
+{
+  GstBinClass parent_class;
+  guint n_prop_own;
+  guint n_prop_total;
+} GstPulseAudioSinkClass;
+
+static void gst_pulse_audio_sink_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+static void gst_pulse_audio_sink_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_pulse_audio_sink_dispose (GObject * object);
+static gboolean gst_pulse_audio_sink_src_event (GstPad * pad, GstEvent * event);
+static gboolean gst_pulse_audio_sink_sink_event (GstPad * pad,
+    GstEvent * event);
+static gboolean gst_pulse_audio_sink_sink_acceptcaps (GstPad * pad,
+    GstCaps * caps);
+static gboolean gst_pulse_audio_sink_sink_setcaps (GstPad * pad,
+    GstCaps * caps);
+static GstStateChangeReturn
+gst_pulse_audio_sink_change_state (GstElement * element,
+    GstStateChange transition);
+
+static void
+gst_pulse_audio_sink_do_init (GType type)
+{
+  GST_DEBUG_CATEGORY_INIT (pulseaudiosink_debug, "pulseaudiosink", 0,
+      "Bin that wraps pulsesink for handling compressed formats");
+}
+
+GST_BOILERPLATE_FULL (GstPulseAudioSink, gst_pulse_audio_sink, GstBin,
+    GST_TYPE_BIN, gst_pulse_audio_sink_do_init);
+
+static GstStaticPadTemplate sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
+    GST_STATIC_CAPS (PULSE_SINK_TEMPLATE_CAPS));
+
+static void
+gst_pulse_audio_sink_base_init (gpointer klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&sink_template));
+
+  gst_element_class_set_details_simple (element_class,
+      "Bin wrapping pulsesink", "Sink/Audio/Bin",
+      "Correctly handles sink changes when streaming compressed formats to "
+      "pulsesink", "Arun Raghavan <arun.raghavan@collabora.co.uk>");
+}
+
+static GParamSpec *
+param_spec_copy (GParamSpec * spec)
+{
+  const char *name, *nick, *blurb;
+  GParamFlags flags;
+
+  name = g_param_spec_get_name (spec);
+  nick = g_param_spec_get_nick (spec);
+  blurb = g_param_spec_get_blurb (spec);
+  flags = spec->flags;
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_BOOLEAN) {
+    return g_param_spec_boolean (name, nick, blurb,
+        G_PARAM_SPEC_BOOLEAN (spec)->default_value, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_BOXED) {
+    return g_param_spec_boxed (name, nick, blurb, spec->value_type, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_CHAR) {
+    GParamSpecChar *cspec = G_PARAM_SPEC_CHAR (spec);
+    return g_param_spec_char (name, nick, blurb, cspec->minimum,
+        cspec->maximum, cspec->default_value, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_DOUBLE) {
+    GParamSpecDouble *dspec = G_PARAM_SPEC_DOUBLE (spec);
+    return g_param_spec_double (name, nick, blurb, dspec->minimum,
+        dspec->maximum, dspec->default_value, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_ENUM) {
+    return g_param_spec_enum (name, nick, blurb, spec->value_type,
+        G_PARAM_SPEC_ENUM (spec)->default_value, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_FLAGS) {
+    return g_param_spec_flags (name, nick, blurb, spec->value_type,
+        G_PARAM_SPEC_ENUM (spec)->default_value, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_FLOAT) {
+    GParamSpecFloat *fspec = G_PARAM_SPEC_FLOAT (spec);
+    return g_param_spec_double (name, nick, blurb, fspec->minimum,
+        fspec->maximum, fspec->default_value, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_GTYPE) {
+    return g_param_spec_gtype (name, nick, blurb,
+        G_PARAM_SPEC_GTYPE (spec)->is_a_type, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_INT) {
+    GParamSpecInt *ispec = G_PARAM_SPEC_INT (spec);
+    return g_param_spec_int (name, nick, blurb, ispec->minimum,
+        ispec->maximum, ispec->default_value, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_INT64) {
+    GParamSpecInt64 *ispec = G_PARAM_SPEC_INT64 (spec);
+    return g_param_spec_int64 (name, nick, blurb, ispec->minimum,
+        ispec->maximum, ispec->default_value, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_LONG) {
+    GParamSpecLong *lspec = G_PARAM_SPEC_LONG (spec);
+    return g_param_spec_long (name, nick, blurb, lspec->minimum,
+        lspec->maximum, lspec->default_value, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_OBJECT) {
+    return g_param_spec_object (name, nick, blurb, spec->value_type, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_PARAM) {
+    return g_param_spec_param (name, nick, blurb, spec->value_type, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_POINTER) {
+    return g_param_spec_pointer (name, nick, blurb, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_STRING) {
+    return g_param_spec_string (name, nick, blurb,
+        G_PARAM_SPEC_STRING (spec)->default_value, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_UCHAR) {
+    GParamSpecUChar *cspec = G_PARAM_SPEC_UCHAR (spec);
+    return g_param_spec_uchar (name, nick, blurb, cspec->minimum,
+        cspec->maximum, cspec->default_value, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_UINT) {
+    GParamSpecUInt *ispec = G_PARAM_SPEC_UINT (spec);
+    return g_param_spec_uint (name, nick, blurb, ispec->minimum,
+        ispec->maximum, ispec->default_value, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_UINT64) {
+    GParamSpecUInt64 *ispec = G_PARAM_SPEC_UINT64 (spec);
+    return g_param_spec_uint64 (name, nick, blurb, ispec->minimum,
+        ispec->maximum, ispec->default_value, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_ULONG) {
+    GParamSpecULong *lspec = G_PARAM_SPEC_ULONG (spec);
+    return g_param_spec_ulong (name, nick, blurb, lspec->minimum,
+        lspec->maximum, lspec->default_value, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_UNICHAR) {
+    return g_param_spec_unichar (name, nick, blurb,
+        G_PARAM_SPEC_UNICHAR (spec)->default_value, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == G_TYPE_PARAM_VARIANT) {
+    GParamSpecVariant *vspec = G_PARAM_SPEC_VARIANT (spec);
+    return g_param_spec_variant (name, nick, blurb, vspec->type,
+        vspec->default_value, flags);
+  }
+
+  if (G_PARAM_SPEC_TYPE (spec) == GST_TYPE_PARAM_MINI_OBJECT) {
+    return gst_param_spec_mini_object (name, nick, blurb, spec->value_type,
+        flags);
+  }
+
+  g_warning ("Unknown param type %ld for '%s'",
+      (long) G_PARAM_SPEC_TYPE (spec), name);
+  g_assert_not_reached ();
+}
+
+static void
+gst_pulse_audio_sink_class_init (GstPulseAudioSinkClass * klass)
+{
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstElementClass *element_class = (GstElementClass *) klass;
+  GstPulseSinkClass *psink_class =
+      GST_PULSESINK_CLASS (g_type_class_ref (GST_TYPE_PULSESINK));
+  GParamSpec **specs;
+  guint n, i, j;
+
+  gobject_class->get_property = gst_pulse_audio_sink_get_property;
+  gobject_class->set_property = gst_pulse_audio_sink_set_property;
+  gobject_class->dispose = gst_pulse_audio_sink_dispose;
+  element_class->change_state =
+      GST_DEBUG_FUNCPTR (gst_pulse_audio_sink_change_state);
+
+  /* Find out how many properties we already have */
+  specs = g_object_class_list_properties (gobject_class, &klass->n_prop_own);
+  g_free (specs);
+
+  /* Proxy pulsesink's properties */
+  specs = g_object_class_list_properties (G_OBJECT_CLASS (psink_class), &n);
+  for (i = 0, j = klass->n_prop_own; i < n; i++) {
+    if (g_object_class_find_property (gobject_class,
+            g_param_spec_get_name (specs[i]))) {
+      /* We already inherited this property from a parent, skip */
+      j--;
+    } else {
+      g_object_class_install_property (gobject_class, i + j + 1,
+          param_spec_copy (specs[i]));
+    }
+  }
+
+  klass->n_prop_total = i + j;
+
+  g_free (specs);
+  g_type_class_unref (psink_class);
+}
+
+static GstPad *
+get_proxypad (GstPad * sinkpad)
+{
+  GstIterator *iter = NULL;
+  GstPad *proxypad = NULL;
+
+  iter = gst_pad_iterate_internal_links (sinkpad);
+  if (iter) {
+    if (gst_iterator_next (iter, (gpointer) & proxypad) != GST_ITERATOR_OK)
+      proxypad = NULL;
+    gst_iterator_free (iter);
+  }
+
+  return proxypad;
+}
+
+static void
+post_missing_element_message (GstPulseAudioSink * pbin, const gchar * name)
+{
+  GstMessage *msg;
+
+  msg = gst_missing_element_message_new (GST_ELEMENT_CAST (pbin), name);
+  gst_element_post_message (GST_ELEMENT_CAST (pbin), msg);
+}
+
+static void
+notify_cb (GObject * selector, GParamSpec * pspec, GstPulseAudioSink * pbin)
+{
+  g_object_notify (G_OBJECT (pbin), g_param_spec_get_name (pspec));
+}
+
+static void
+gst_pulse_audio_sink_init (GstPulseAudioSink * pbin,
+    GstPulseAudioSinkClass * klass)
+{
+  GstPad *pad = NULL;
+  GParamSpec **specs;
+  GString *prop;
+  guint i;
+
+  pbin->lock = g_mutex_new ();
+
+  gst_segment_init (&pbin->segment, GST_FORMAT_UNDEFINED);
+
+  pbin->psink = GST_PULSESINK (gst_element_factory_make ("pulsesink",
+          "pulseaudiosink-sink"));
+  g_assert (pbin->psink != NULL);
+
+  if (!gst_bin_add (GST_BIN (pbin), GST_ELEMENT (pbin->psink))) {
+    GST_ERROR_OBJECT (pbin, "Failed to add pulsesink to bin");
+    goto error;
+  }
+
+  pad = gst_element_get_static_pad (GST_ELEMENT (pbin->psink), "sink");
+  pbin->sinkpad = gst_ghost_pad_new_from_template ("sink", pad,
+      gst_static_pad_template_get (&sink_template));
+
+  pbin->sinkpad_old_eventfunc = GST_PAD_EVENTFUNC (pbin->sinkpad);
+  gst_pad_set_event_function (pbin->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_pulse_audio_sink_sink_event));
+  gst_pad_set_setcaps_function (pbin->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_pulse_audio_sink_sink_setcaps));
+  gst_pad_set_acceptcaps_function (pbin->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_pulse_audio_sink_sink_acceptcaps));
+
+  gst_element_add_pad (GST_ELEMENT (pbin), pbin->sinkpad);
+
+  if (!(pbin->sink_proxypad = get_proxypad (pbin->sinkpad)))
+    GST_ERROR_OBJECT (pbin, "Failed to get proxypad of srcpad");
+  else {
+    pbin->proxypad_old_eventfunc = GST_PAD_EVENTFUNC (pbin->sink_proxypad);
+    gst_pad_set_event_function (pbin->sink_proxypad,
+        GST_DEBUG_FUNCPTR (gst_pulse_audio_sink_src_event));
+  }
+
+  /* Now proxy all the notify::* signals */
+  specs = g_object_class_list_properties (G_OBJECT_CLASS (klass), &i);
+  prop = g_string_sized_new (30);
+
+  for (i--; i >= klass->n_prop_own; i--) {
+    g_string_printf (prop, "notify::%s", g_param_spec_get_name (specs[i]));
+    g_signal_connect (pbin->psink, prop->str, G_CALLBACK (notify_cb), pbin);
+  }
+
+  g_string_free (prop, TRUE);
+  g_free (specs);
+
+  pbin->format_lost = FALSE;
+
+out:
+  if (pad)
+    gst_object_unref (pad);
+
+  return;
+
+error:
+  if (pbin->psink)
+    gst_object_unref (pbin->psink);
+  goto out;
+}
+
+static void
+gst_pulse_audio_sink_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (object);
+  GstPulseAudioSinkClass *klass =
+      GST_PULSE_AUDIO_SINK_CLASS (G_OBJECT_GET_CLASS (object));
+
+  g_return_if_fail (prop_id <= klass->n_prop_total);
+
+  g_object_set_property (G_OBJECT (pbin->psink), g_param_spec_get_name (pspec),
+      value);
+}
+
+static void
+gst_pulse_audio_sink_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (object);
+  GstPulseAudioSinkClass *klass =
+      GST_PULSE_AUDIO_SINK_CLASS (G_OBJECT_GET_CLASS (object));
+
+  g_return_if_fail (prop_id <= klass->n_prop_total);
+
+  g_object_get_property (G_OBJECT (pbin->psink), g_param_spec_get_name (pspec),
+      value);
+}
+
+static void
+gst_pulse_audio_sink_free_dbin2 (GstPulseAudioSink * pbin)
+{
+  g_signal_handler_disconnect (pbin->dbin2, pbin->pad_added_id);
+  gst_element_set_state (pbin->dbin2, GST_STATE_NULL);
+
+  gst_bin_remove (GST_BIN (pbin), pbin->dbin2);
+
+  pbin->dbin2 = NULL;
+}
+
+static void
+gst_pulse_audio_sink_dispose (GObject * object)
+{
+  GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (object);
+
+  if (pbin->lock) {
+    g_mutex_free (pbin->lock);
+    pbin->lock = NULL;
+  }
+
+  if (pbin->sink_proxypad) {
+    gst_object_unref (pbin->sink_proxypad);
+    pbin->sink_proxypad = NULL;
+  }
+
+  if (pbin->dbin2) {
+    g_signal_handler_disconnect (pbin->dbin2, pbin->pad_added_id);
+    pbin->dbin2 = NULL;
+  }
+
+  pbin->sinkpad = NULL;
+  pbin->psink = NULL;
+}
+
+static gboolean
+gst_pulse_audio_sink_update_sinkpad (GstPulseAudioSink * pbin, GstPad * sinkpad)
+{
+  gboolean ret;
+
+  ret = gst_ghost_pad_set_target (GST_GHOST_PAD_CAST (pbin->sinkpad), sinkpad);
+
+  if (!ret)
+    GST_WARNING_OBJECT (pbin, "Could not update ghostpad target");
+
+  return ret;
+}
+
+static void
+distribute_running_time (GstElement * element, const GstSegment * segment)
+{
+  GstEvent *event;
+  GstPad *pad;
+
+  pad = gst_element_get_static_pad (element, "sink");
+
+  /* FIXME: Some decoders collect newsegments and send them out at once, making
+   * them lose accumulator events (and thus making dbin2_event_probe() hard to
+   * do right if we're sending these as well. We can get away with not sending
+   * these at the moment, but this should be fixed! */
+#if 0
+  if (segment->accum) {
+    event = gst_event_new_new_segment_full (FALSE, segment->rate,
+        segment->applied_rate, segment->format, 0, segment->accum, 0);
+    gst_pad_send_event (pad, event);
+  }
+#endif
+
+  event = gst_event_new_new_segment_full (FALSE, segment->rate,
+      segment->applied_rate, segment->format,
+      segment->start, segment->stop, segment->time);
+  gst_pad_send_event (pad, event);
+
+  gst_object_unref (pad);
+}
+
+static gboolean
+dbin2_event_probe (GstPad * pad, GstMiniObject * obj, gpointer data)
+{
+  GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (data);
+  GstEvent *event = GST_EVENT (obj);
+
+  if (GST_EVENT_TYPE (event) == GST_EVENT_NEWSEGMENT) {
+    GST_DEBUG_OBJECT (pbin, "Got newsegment - dropping");
+    gst_pad_remove_event_probe (pad, pbin->event_probe_id);
+    gst_object_unref (pbin);
+    return FALSE;
+  }
+
+  return TRUE;
+}
+
+static void
+pad_added_cb (GstElement * dbin2, GstPad * pad, gpointer * data)
+{
+  GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (data);
+  GstPad *sinkpad = NULL;
+
+  pbin = GST_PULSE_AUDIO_SINK (data);
+  sinkpad = gst_element_get_static_pad (GST_ELEMENT (pbin->psink), "sink");
+
+  GST_PULSE_AUDIO_SINK_LOCK (pbin);
+  if (gst_pad_link (pad, sinkpad) != GST_PAD_LINK_OK)
+    GST_ERROR_OBJECT (pbin, "Failed to link decodebin2 to pulsesink");
+  else
+    GST_DEBUG_OBJECT (pbin, "Linked new pad to pulsesink");
+  GST_PULSE_AUDIO_SINK_UNLOCK (pbin);
+
+  gst_object_unref (sinkpad);
+}
+
+/* Called with pbin lock held */
+static void
+gst_pulse_audio_sink_add_dbin2 (GstPulseAudioSink * pbin)
+{
+  GstPad *sinkpad = NULL;
+
+  g_assert (pbin->dbin2 == NULL);
+
+  pbin->dbin2 = gst_element_factory_make ("decodebin2", "pulseaudiosink-dbin2");
+
+  if (!pbin->dbin2) {
+    post_missing_element_message (pbin, "decodebin2");
+    GST_ELEMENT_WARNING (pbin, CORE, MISSING_PLUGIN,
+        (_("Missing element '%s' - check your GStreamer installation."),
+            "decodebin2"), ("audio playback might fail"));
+    goto out;
+  }
+
+  if (!gst_bin_add (GST_BIN (pbin), pbin->dbin2)) {
+    GST_ERROR_OBJECT (pbin, "Failed to add decodebin2 to bin");
+    goto out;
+  }
+
+  pbin->pad_added_id = g_signal_connect (pbin->dbin2, "pad-added",
+      G_CALLBACK (pad_added_cb), pbin);
+
+  if (!gst_element_sync_state_with_parent (pbin->dbin2)) {
+    GST_ERROR_OBJECT (pbin, "Failed to set decodebin2 to parent state");
+    goto out;
+  }
+
+  /* Trap the newsegment events that we feed the decodebin and discard them */
+  sinkpad = gst_element_get_static_pad (GST_ELEMENT (pbin->psink), "sink");
+  pbin->event_probe_id = gst_pad_add_event_probe (sinkpad,
+      G_CALLBACK (dbin2_event_probe), gst_object_ref (pbin));
+  gst_object_unref (sinkpad);
+  sinkpad = NULL;
+
+  GST_DEBUG_OBJECT (pbin, "Distributing running time to decodebin");
+  distribute_running_time (pbin->dbin2, &pbin->segment);
+
+  sinkpad = gst_element_get_static_pad (pbin->dbin2, "sink");
+
+  gst_pulse_audio_sink_update_sinkpad (pbin, sinkpad);
+
+out:
+  if (sinkpad)
+    gst_object_unref (sinkpad);
+}
+
+static void
+update_eac3_alignment (GstPulseAudioSink * pbin)
+{
+  GstCaps *caps = gst_pad_peer_get_caps_reffed (pbin->sinkpad);
+  GstStructure *st;
+
+  if (!caps)
+    return;
+
+  st = gst_caps_get_structure (caps, 0);
+
+  if (g_str_equal (gst_structure_get_name (st), "audio/x-eac3")) {
+    GstStructure *event_st = gst_structure_new ("ac3parse-set-alignment",
+        "alignment", G_TYPE_STRING, pbin->dbin2 ? "frame" : "iec61937", NULL);
+
+    if (!gst_pad_push_event (pbin->sinkpad,
+            gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, event_st)))
+      GST_WARNING_OBJECT (pbin->sinkpad, "Could not update alignment");
+  }
+
+  gst_caps_unref (caps);
+}
+
+static void
+proxypad_blocked_cb (GstPad * pad, gboolean blocked, gpointer data)
+{
+  GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (data);
+  GstCaps *caps;
+  GstPad *sinkpad = NULL;
+
+  if (!blocked) {
+    /* Unblocked, don't need to do anything */
+    GST_DEBUG_OBJECT (pbin, "unblocked");
+    return;
+  }
+
+  GST_DEBUG_OBJECT (pbin, "blocked");
+
+  GST_PULSE_AUDIO_SINK_LOCK (pbin);
+
+  if (!pbin->format_lost) {
+    sinkpad = gst_element_get_static_pad (GST_ELEMENT (pbin->psink), "sink");
+    caps = gst_pad_get_caps_reffed (pad);
+
+    if (gst_pad_accept_caps (sinkpad, caps)) {
+      if (pbin->dbin2) {
+        GST_DEBUG_OBJECT (pbin, "Removing decodebin");
+        gst_pulse_audio_sink_free_dbin2 (pbin);
+        gst_pulse_audio_sink_update_sinkpad (pbin, sinkpad);
+      } else
+        GST_DEBUG_OBJECT (pbin, "Doing nothing");
+
+      gst_caps_unref (caps);
+      gst_object_unref (sinkpad);
+      goto done;
+    }
+    /* pulsesink doesn't accept the incoming caps, so add a decodebin
+     * (potentially after removing the existing once, since decodebin2 can't
+     * renegotiate). */
+  } else {
+    /* Format lost, proceed to try plugging a decodebin */
+    pbin->format_lost = FALSE;
+  }
+
+  if (pbin->dbin2 != NULL) {
+    /* decodebin2 doesn't support reconfiguration, so throw this one away and
+     * create a new one. */
+    gst_pulse_audio_sink_free_dbin2 (pbin);
+  }
+
+  GST_DEBUG_OBJECT (pbin, "Adding decodebin");
+  gst_pulse_audio_sink_add_dbin2 (pbin);
+
+done:
+  update_eac3_alignment (pbin);
+
+  gst_pad_set_blocked_async_full (pad, FALSE, proxypad_blocked_cb,
+      gst_object_ref (pbin), (GDestroyNotify) gst_object_unref);
+
+  GST_PULSE_AUDIO_SINK_UNLOCK (pbin);
+}
+
+static gboolean
+gst_pulse_audio_sink_src_event (GstPad * pad, GstEvent * event)
+{
+  GstPulseAudioSink *pbin = NULL;
+  GstPad *ghostpad = NULL;
+  gboolean ret = FALSE;
+
+  ghostpad = GST_PAD_CAST (gst_pad_get_parent (pad));
+  if (G_UNLIKELY (!ghostpad)) {
+    GST_WARNING_OBJECT (pad, "Could not get ghostpad");
+    goto out;
+  }
+
+  pbin = GST_PULSE_AUDIO_SINK (gst_pad_get_parent (ghostpad));
+  if (G_UNLIKELY (!pbin)) {
+    GST_WARNING_OBJECT (pad, "Could not get pulseaudiosink");
+    goto out;
+  }
+
+  if (G_UNLIKELY (GST_EVENT_TYPE (event) == GST_EVENT_CUSTOM_UPSTREAM) &&
+      (gst_event_has_name (event, "pulse-format-lost") ||
+          gst_event_has_name (event, "pulse-sink-changed"))) {
+    g_return_val_if_fail (pad->mode != GST_ACTIVATE_PULL, FALSE);
+
+    GST_PULSE_AUDIO_SINK_LOCK (pbin);
+    if (gst_event_has_name (event, "pulse-format-lost"))
+      pbin->format_lost = TRUE;
+
+    if (!gst_pad_is_blocked (pad))
+      gst_pad_set_blocked_async_full (pad, TRUE, proxypad_blocked_cb,
+          gst_object_ref (pbin), (GDestroyNotify) gst_object_unref);
+    GST_PULSE_AUDIO_SINK_UNLOCK (pbin);
+
+    ret = TRUE;
+  } else if (pbin->proxypad_old_eventfunc) {
+    ret = pbin->proxypad_old_eventfunc (pad, event);
+    event = NULL;
+  }
+
+out:
+  if (ghostpad)
+    gst_object_unref (ghostpad);
+  if (pbin)
+    gst_object_unref (pbin);
+  if (event)
+    gst_event_unref (event);
+
+  return ret;
+}
+
+static gboolean
+gst_pulse_audio_sink_sink_event (GstPad * pad, GstEvent * event)
+{
+  GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (gst_pad_get_parent (pad));
+  gboolean ret;
+
+  ret = pbin->sinkpad_old_eventfunc (pad, gst_event_ref (event));
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_NEWSEGMENT:
+    {
+      GstFormat format;
+      gdouble rate, arate;
+      gint64 start, stop, time;
+      gboolean update;
+
+      GST_PULSE_AUDIO_SINK_LOCK (pbin);
+      gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
+          &start, &stop, &time);
+
+      GST_DEBUG_OBJECT (pbin,
+          "newsegment: update %d, rate %g, arate %g, start %" GST_TIME_FORMAT
+          ", stop %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT,
+          update, rate, arate, GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
+          GST_TIME_ARGS (time));
+
+      if (format == GST_FORMAT_TIME) {
+        /* Store the values for feeding to sub-elements */
+        gst_segment_set_newsegment_full (&pbin->segment, update,
+            rate, arate, format, start, stop, time);
+      } else {
+        GST_WARNING_OBJECT (pbin, "Got a non-TIME format segment");
+        gst_segment_init (&pbin->segment, GST_FORMAT_TIME);
+      }
+      GST_PULSE_AUDIO_SINK_UNLOCK (pbin);
+
+      break;
+    }
+
+    case GST_EVENT_FLUSH_STOP:
+      GST_PULSE_AUDIO_SINK_LOCK (pbin);
+      gst_segment_init (&pbin->segment, GST_FORMAT_UNDEFINED);
+      GST_PULSE_AUDIO_SINK_UNLOCK (pbin);
+      break;
+
+    default:
+      break;
+  }
+
+  gst_object_unref (pbin);
+  gst_event_unref (event);
+
+  return ret;
+}
+
+/* The bin's acceptcaps should be exactly equivalent to a pulsesink that is
+ * connected to a sink that supports all the formats in template caps. This
+ * means that upstream will have to have everything possibly upto a parser
+ * plugged and we plugin a decoder whenever required. */
+static gboolean
+gst_pulse_audio_sink_sink_acceptcaps (GstPad * pad, GstCaps * caps)
+{
+  GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (gst_pad_get_parent (pad));
+  GstRingBufferSpec spec = { 0 };
+  const GstStructure *st;
+  GstCaps *pad_caps = NULL;
+  gboolean ret = FALSE;
+
+  pad_caps = gst_pad_get_caps_reffed (pad);
+  if (!pad_caps || !gst_caps_can_intersect (pad_caps, caps))
+    goto out;
+
+  /* If we've not got fixed caps, creating a stream might fail, so let's just
+   * return from here with default acceptcaps behaviour */
+  if (!gst_caps_is_fixed (caps))
+    goto out;
+
+  spec.latency_time = GST_BASE_AUDIO_SINK (pbin->psink)->latency_time;
+  if (!gst_ring_buffer_parse_caps (&spec, caps))
+    goto out;
+
+  /* Make sure non-raw input is framed (one frame per buffer) and can be
+   * payloaded */
+  st = gst_caps_get_structure (caps, 0);
+
+  if (!g_str_has_prefix (gst_structure_get_name (st), "audio/x-raw")) {
+    gboolean framed = FALSE, parsed = FALSE;
+
+    gst_structure_get_boolean (st, "framed", &framed);
+    gst_structure_get_boolean (st, "parsed", &parsed);
+    if ((!framed && !parsed) || gst_audio_iec61937_frame_size (&spec) <= 0)
+      goto out;
+  }
+
+  ret = TRUE;
+
+out:
+  if (pad_caps)
+    gst_caps_unref (pad_caps);
+
+  gst_object_unref (pbin);
+
+  return ret;
+}
+
+static gboolean
+gst_pulse_audio_sink_sink_setcaps (GstPad * pad, GstCaps * caps)
+{
+  GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (gst_pad_get_parent (pad));
+  gboolean ret = TRUE;
+
+  GST_PULSE_AUDIO_SINK_LOCK (pbin);
+
+  if (!gst_pad_is_blocked (pbin->sinkpad))
+    gst_pad_set_blocked_async_full (pbin->sink_proxypad, TRUE,
+        proxypad_blocked_cb, gst_object_ref (pbin),
+        (GDestroyNotify) gst_object_unref);
+
+  GST_PULSE_AUDIO_SINK_UNLOCK (pbin);
+
+  gst_object_unref (pbin);
+
+  return ret;
+}
+
+static GstStateChangeReturn
+gst_pulse_audio_sink_change_state (GstElement * element,
+    GstStateChange transition)
+{
+  GstPulseAudioSink *pbin = GST_PULSE_AUDIO_SINK (element);
+  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
+
+  /* Nothing to do for upward transitions */
+  switch (transition) {
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      GST_PULSE_AUDIO_SINK_LOCK (pbin);
+      if (gst_pad_is_blocked (pbin->sinkpad)) {
+        gst_pad_set_blocked_async_full (pbin->sink_proxypad, FALSE,
+            proxypad_blocked_cb, gst_object_ref (pbin),
+            (GDestroyNotify) gst_object_unref);
+      }
+      GST_PULSE_AUDIO_SINK_UNLOCK (pbin);
+      break;
+
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+  if (ret != GST_STATE_CHANGE_SUCCESS) {
+    GST_DEBUG_OBJECT (pbin, "Base class returned %d on state change", ret);
+    goto out;
+  }
+
+  switch (transition) {
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      GST_PULSE_AUDIO_SINK_LOCK (pbin);
+      gst_segment_init (&pbin->segment, GST_FORMAT_UNDEFINED);
+
+      if (pbin->dbin2) {
+        GstPad *pad = gst_element_get_static_pad (GST_ELEMENT (pbin->psink),
+            "sink");
+
+        gst_pulse_audio_sink_free_dbin2 (pbin);
+        gst_pulse_audio_sink_update_sinkpad (pbin, pad);
+
+        gst_object_unref (pad);
+
+      }
+      GST_PULSE_AUDIO_SINK_UNLOCK (pbin);
+
+      break;
+
+    default:
+      break;
+  }
+
+out:
+  return ret;
+}
+
+#endif /* HAVE_PULSE_1_0 */

ext/pulse/pulsesink.c

@@ -420,6 +420,27 @@ gst_pulsering_context_subscribe_cb (pa_context * c,
     if (idx != pa_stream_get_index (pbuf->stream))
       continue;
 
+#ifdef HAVE_PULSE_1_0
+    if (psink->device && pa_format_info_is_pcm (pbuf->format) &&
+        !g_str_equal (psink->device,
+            pa_stream_get_device_name (pbuf->stream))) {
+      /* Underlying sink changed. And this is not a passthrough stream. Let's
+       * see if someone upstream wants to try to renegotiate. */
+      GstEvent *renego;
+
+      g_free (psink->device);
+      psink->device = g_strdup (pa_stream_get_device_name (pbuf->stream));
+
+      GST_INFO_OBJECT (psink, "emitting sink-changed");
+
+      renego = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
+          gst_structure_new ("pulse-sink-changed", NULL));
+
+      if (!gst_pad_push_event (GST_BASE_SINK (psink)->sinkpad, renego))
+        GST_DEBUG_OBJECT (psink, "Emitted sink-changed - nobody was listening");
+    }
+#endif
+
     /* Actually this event is also triggered when other properties of
      * the stream change that are unrelated to the volume. However it is
      * probably cheaper to signal the change here and check for the
@@ -1719,12 +1740,6 @@ static GstStateChangeReturn gst_pulsesink_change_state (GstElement * element,
 
 static void gst_pulsesink_init_interfaces (GType type);
 
-#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
-# define ENDIANNESS   "LITTLE_ENDIAN, BIG_ENDIAN"
-#else
-# define ENDIANNESS   "BIG_ENDIAN, LITTLE_ENDIAN"
-#endif
-
 GST_IMPLEMENT_PULSEPROBE_METHODS (GstPulseSink, gst_pulsesink);
 
 #define _do_init(type) \
@@ -1784,57 +1799,7 @@ 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, "
+      GST_STATIC_CAPS (PULSE_SINK_TEMPLATE_CAPS));
-          "endianness = (int) { " ENDIANNESS " }, "
-          "signed = (boolean) TRUE, "
-          "width = (int) 16, "
-          "depth = (int) 16, "
-          "rate = (int) [ 1, MAX ], "
-          "channels = (int) [ 1, 32 ];"
-          "audio/x-raw-float, "
-          "endianness = (int) { " ENDIANNESS " }, "
-          "width = (int) 32, "
-          "rate = (int) [ 1, MAX ], "
-          "channels = (int) [ 1, 32 ];"
-          "audio/x-raw-int, "
-          "endianness = (int) { " ENDIANNESS " }, "
-          "signed = (boolean) TRUE, "
-          "width = (int) 32, "
-          "depth = (int) 32, "
-          "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 32 ];"
-          "audio/x-raw-int, "
-          "endianness = (int) { " ENDIANNESS " }, "
-          "signed = (boolean) TRUE, "
-          "width = (int) 24, "
-          "depth = (int) 24, "
-          "rate = (int) [ 1, MAX ], "
-          "channels = (int) [ 1, 32 ];"
-          "audio/x-raw-int, "
-          "endianness = (int) { " ENDIANNESS " }, "
-          "signed = (boolean) TRUE, "
-          "width = (int) 32, "
-          "depth = (int) 24, "
-          "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 32 ];"
-          "audio/x-raw-int, "
-          "signed = (boolean) FALSE, "
-          "width = (int) 8, "
-          "depth = (int) 8, "
-          "rate = (int) [ 1, MAX ], "
-          "channels = (int) [ 1, 32 ];"
-          "audio/x-alaw, "
-          "rate = (int) [ 1, MAX], "
-          "channels = (int) [ 1, 32 ];"
-          "audio/x-mulaw, "
-          "rate = (int) [ 1, MAX], " "channels = (int) [ 1, 32 ];"
-#ifdef HAVE_PULSE_1_0
-          "audio/x-ac3, framed = (boolean) true;"
-          "audio/x-eac3, framed = (boolean) true; "
-          "audio/x-dts, framed = (boolean) true, "
-          "  block_size = (int) { 512, 1024, 2048 }; "
-          "audio/mpeg, mpegversion = (int)1, "
-          "  mpegaudioversion = (int) [ 1, 2 ], parsed = (boolean) true; "
-#endif
-      ));
 
   GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
 
@@ -2075,6 +2040,8 @@ done:
 }
 
 #ifdef HAVE_PULSE_1_0
+/* NOTE: If you're making changes here, see if pulseaudiosink acceptcaps also
+ * needs to be changed accordingly. */
 static gboolean
 gst_pulsesink_pad_acceptcaps (GstPad * pad, GstCaps * caps)
 {

ext/pulse/pulsesink.h

@@ -24,6 +24,10 @@
 #ifndef __GST_PULSESINK_H__
 #define __GST_PULSESINK_H__
 
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
 #include <gst/gst.h>
 #include <gst/audio/gstaudiosink.h>
 
@@ -88,6 +92,91 @@ struct _GstPulseSinkClass
 
 GType gst_pulsesink_get_type (void);
 
+#if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
+# define ENDIANNESS   "LITTLE_ENDIAN, BIG_ENDIAN"
+#else
+# define ENDIANNESS   "BIG_ENDIAN, LITTLE_ENDIAN"
+#endif
+
+#define _PULSE_SINK_CAPS_COMMON \
+    "audio/x-raw-int, " \
+      "endianness = (int) { " ENDIANNESS " }, " \
+      "signed = (boolean) TRUE, " \
+      "width = (int) 16, " \
+      "depth = (int) 16, " \
+      "rate = (int) [ 1, MAX ], " \
+      "channels = (int) [ 1, 32 ];" \
+    "audio/x-raw-float, " \
+      "endianness = (int) { " ENDIANNESS " }, " \
+      "width = (int) 32, " \
+      "rate = (int) [ 1, MAX ], " \
+      "channels = (int) [ 1, 32 ];" \
+    "audio/x-raw-int, " \
+      "endianness = (int) { " ENDIANNESS " }, " \
+      "signed = (boolean) TRUE, " \
+      "width = (int) 32, " \
+      "depth = (int) 32, " \
+      "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 32 ];" \
+    "audio/x-raw-int, " \
+      "signed = (boolean) FALSE, " \
+      "width = (int) 8, " \
+      "depth = (int) 8, " \
+      "rate = (int) [ 1, MAX ], " \
+      "channels = (int) [ 1, 32 ];" \
+    "audio/x-alaw, " \
+      "rate = (int) [ 1, MAX], " \
+      "channels = (int) [ 1, 32 ];" \
+    "audio/x-mulaw, " \
+      "rate = (int) [ 1, MAX], " "channels = (int) [ 1, 32 ];" \
+    "audio/x-raw-int, " \
+      "endianness = (int) { " ENDIANNESS " }, " \
+      "signed = (boolean) TRUE, " \
+      "width = (int) 24, " \
+      "depth = (int) 24, " \
+      "rate = (int) [ 1, MAX ], " \
+      "channels = (int) [ 1, 32 ];" \
+    "audio/x-raw-int, " \
+      "endianness = (int) { " ENDIANNESS " }, " \
+      "signed = (boolean) TRUE, " \
+      "width = (int) 32, " \
+      "depth = (int) 24, " \
+      "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 32 ];"
+
+#ifdef HAVE_PULSE_1_0
+#define _PULSE_SINK_CAPS_1_0 \
+    "audio/x-ac3, framed = (boolean) true;" \
+    "audio/x-eac3, framed = (boolean) true; " \
+    "audio/x-dts, framed = (boolean) true, " \
+      "block-size = (int) { 512, 1024, 2048 }; " \
+    "audio/mpeg, mpegversion = (int) 1, " \
+      "mpegaudioversion = (int) [ 1, 2 ], parsed = (boolean) true;"
+#else
+#define _PULSE_SINK_CAPS_1_0 ""
+#endif
+
+#define PULSE_SINK_TEMPLATE_CAPS \
+  _PULSE_SINK_CAPS_COMMON \
+  _PULSE_SINK_CAPS_1_0
+
+#ifdef HAVE_PULSE_1_0
+
+#define GST_TYPE_PULSE_AUDIO_SINK \
+  (gst_pulse_audio_sink_get_type())
+#define GST_PULSE_AUDIO_SINK(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_PULSE_AUDIO_SINK,GstPulseAudioSink))
+#define GST_PULSE_AUDIO_SINK_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_PULSE_AUDIO_SINK,GstPulseAudioSinkClass))
+#define GST_IS_PULSE_AUDIO_SINK(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_PULSE_AUDIO_SINK))
+#define GST_IS_PULSE_AUDIO_SINK_CLASS(obj) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_PULSE_AUDIO_SINK))
+#define GST_PULSE_AUDIO_SINK_CAST(obj) \
+  ((GstPulseAudioSink *)(obj))
+
+GType gst_pulse_audio_sink_get_type (void);
+
+#endif /* HAVE_PULSE_1_0 */
+
 G_END_DECLS
 
 #endif /* __GST_PULSESINK_H__ */

ext/pulse/pulseutil.h

@@ -22,6 +22,10 @@
 #ifndef __GST_PULSEUTIL_H__
 #define __GST_PULSEUTIL_H__
 
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
 #include <gst/gst.h>
 #include <pulse/pulseaudio.h>
 #include <gst/audio/gstaudiosink.h>