--- /dev/null
+/*-*- 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 */
projects / platform / upstream / gstreamer.git / commitdiff