/* -*- Mode: C; tab-width: 8; indent-tabs-mode: t; c-basic-offset: 8 -*-
 *
 *  Copyright (C) 2006,2010   Jonathan Matthew  <jonathan@d14n.org>
 *
 *  This program is free software; you can redistribute it and/or modify
 *  it under the terms of the GNU General Public License as published by
 *  the Free Software Foundation; either version 2 of the License, or
 *  (at your option) any later version.
 *
 *  The Rhythmbox authors hereby grant permission for non-GPL compatible
 *  GStreamer plugins to be used and distributed together with GStreamer
 *  and Rhythmbox. This permission is above and beyond the permissions granted
 *  by the GPL license by which Rhythmbox is covered. If you modify this code
 *  you may extend this exception to your version of the code, but you are not
 *  obligated to do so. If you do not wish to do so, delete this exception
 *  statement from your version.
 *
 *  This program is distributed in the hope that it will be useful,
 *  but WITHOUT ANY WARRANTY; without even the implied warranty of
 *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 *  GNU General Public License for more details.
 *
 *  You should have received a copy of the GNU General Public License
 *  along with this program; if not, write to the Free Software
 *  Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301  USA.
 *
 */

/*
 * GStreamer player backend with crossfading and gaplessness and trees and
 * flowers and bunnies.
 */

/*
 * basic design:
 *
 * we have a single output bin, beginning with an adder.
 * connected to this are a number of stream bins, consisting of a
 * source, decodebin, audio convert/resample, and a volume element used
 * for fading in and out.  (might be interesting to replace those with
 * high/low pass filter elements?)
 *
 * stream bins only stay connected to the adder while actually playing.
 * when not playing (prerolling or paused), the stream bin's source pad
 * is blocked so no data can flow.
 *
 * streams go through a number of states:
 *
 * when a stream is created (in rb_player_open()), it starts in PREROLLING
 * state.  from there:
 *
 * - rb_player_play():  -> PREROLL_PLAY
 * - preroll finishes:  -> WAITING
 *
 * from WAITING:
 *
 * - rb_player_play(), _AFTER_EOS, other stream playing:  -> WAITING_EOS
 * - rb_player_play(), _CROSSFADE, other stream playing:   -> FADING IN, link to adder, unblock
 *      + fade out existing stream
 * - rb_player_play(), _REPLACE, other stream playing:   -> PLAYING, link to adder, unblock
 *      + stop existing stream
 * - rb_player_play(), existing stream paused:  -> PLAYING, link to adder, unblock
 *      + stop existing stream
 * - rb_player_play(), nothing already playing:  -> PLAYING, link to adder, unblock
 *
 * from PREROLL_PLAY:
 *
 * - preroll finishes, _AFTER_EOS, other stream playing:  -> WAITING_EOS
 * - preroll finishes, _CROSSFADE, other stream playing:  -> FADING_IN, link to adder, unblock
 *   	+ fade out existing stream
 * - preroll finishes, _REPLACE, other stream playing:  -> PLAYING, link to adder, unblock
 *      + stop existing stream
 * - preroll finishes, existing stream paused:  -> PLAYING, link to adder, unblock
 *      + stop existing stream
 * - preroll finishes, nothing already playing:  -> PLAYING, link to adder, unblock
 *
 * from WAITING_EOS:
 *
 * - EOS received for another stream:  -> PLAYING, link to adder, unblock
 *
 * from FADING_IN:
 *
 * - fade in completes:  -> PLAYING
 * - another stream starts fading in:  -> FADING_OUT
 * - rb_player_pause():  -> PAUSED, block, unlink from adder
 * - stopped for another stream:  -> PENDING_REMOVE
 * - rb_player_set_time():  -> SEEKING, block, unlink
 * - reused for another stream:  -> REUSING; block, unlink
 *
 * from PLAYING:
 *
 * - rb_player_pause(): -> FADING_OUT_PAUSE, fade out (short fade)
 * - EOS:  -> PENDING_REMOVE
 * - another stream starts fading in:  -> FADING_OUT
 * - stopped for another stream:  -> PENDING_REMOVE
 * - rb_player_set_time():  -> SEEKING, block, unlink
 * - reused for another stream:  -> REUSING; block, unlink
 *
 * from SEEKING:
 * - rb_player_pause():  -> SEEKING_PAUSED
 * - blocked:  perform seek, link, unblock -> PLAYING | FADING_IN
 *
 * from SEEKING_PAUSED:
 * - blocked:  perform seek, -> PAUSED
 * - rb_player_play():   -> SEEKING
 *
 * from PAUSED:
 *
 * - rb_player_play():    -> FADING IN, link to adder, unblock (short fade)
 * - stopped for another stream:  -> PENDING_REMOVE
 * - rb_player_set_time(): -> perform seek
 *
 * from FADING_OUT:
 *
 * - fade out finishes:  -> PENDING_REMOVE
 * - EOS:  -> PENDING_REMOVE
 * - reused for another stream:  -> REUSING; block, unlink
 *
 * from FADING_OUT_PAUSED:
 *
 * - fade out finishes: -> SEEKING_PAUSED, block, unlink
 * - EOS: -> PENDING_REMOVE
 * - reused for another stream: -> REUSING, block, unlink
 * - rb_player_set_time():  -> SEEKING_PAUSED, block, unlink
 *
 * from PENDING_REMOVE:
 * - rb_player_set_time():  -> block, seek, -> SEEKING_EOS
 * - reap_streams idle handler called:  -> unlink from adder, stream destroyed
 *
 * from SEEKING_EOS:
 * - block completes -> link, unblock, -> PLAYING
 * - rb_player_pause() -> SEEKING_PAUSED
 *
 * from REUSING:
 *  - EOS: emit reuse-stream, -> PLAYING
 *  - rb_player_play(): -> block, unlink
 *  - blocked:  emit reuse-stream, link -> PLAYING
 */

#include "config.h"
#include <math.h>

#include <glib/gi18n.h>
#include <gst/gst.h>
#include <gst/controller/gstinterpolationcontrolsource.h>
#include <gst/controller/gstdirectcontrolbinding.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/streamvolume.h>
#include <gst/audio/gstaudiodecoder.h>
#include <gst/pbutils/pbutils.h>

#include "rb-player.h"
#include "rb-player-gst-xfade.h"
#include "rb-debug.h"
#include "rb-file-helpers.h"
#include "rb-util.h"
#include "rb-marshal.h"
#include "rb-player-gst-tee.h"
#include "rb-player-gst-filter.h"
#include "rb-player-gst-helper.h"

static void rb_player_init (RBPlayerIface *iface);
static void rb_player_gst_tee_init (RBPlayerGstTeeIface *iface);
static void rb_player_gst_filter_init (RBPlayerGstFilterIface *iface);
static void rb_player_gst_xfade_dispose (GObject *object);
static void rb_player_gst_xfade_finalize (GObject *object);

static gboolean rb_player_gst_xfade_open (RBPlayer *player,
					  const char *uri,
					  gpointer stream_data,
					  GDestroyNotify stream_data_destroy,
					  GError **error);
static gboolean rb_player_gst_xfade_opened (RBPlayer *player);
static gboolean rb_player_gst_xfade_close (RBPlayer *player, const char *uri, GError **error);
static gboolean rb_player_gst_xfade_play (RBPlayer *player, RBPlayerPlayType play_type, gint64 crossfade, GError **error);
static void rb_player_gst_xfade_pause (RBPlayer *player);
static gboolean rb_player_gst_xfade_playing (RBPlayer *player);
static gboolean rb_player_gst_xfade_seekable (RBPlayer *player);
static void rb_player_gst_xfade_set_time (RBPlayer *player, gint64 time);
static gint64 rb_player_gst_xfade_get_time (RBPlayer *player);
static void rb_player_gst_xfade_set_volume (RBPlayer *player, float volume);
static float rb_player_gst_xfade_get_volume (RBPlayer *player);
static gboolean rb_player_gst_xfade_add_tee (RBPlayerGstTee *player, GstElement *element);
static gboolean rb_player_gst_xfade_add_filter (RBPlayerGstFilter *player, GstElement *element);
static gboolean rb_player_gst_xfade_remove_tee (RBPlayerGstTee *player, GstElement *element);
static gboolean rb_player_gst_xfade_remove_filter (RBPlayerGstFilter *player, GstElement *element);

static gboolean create_sink (RBPlayerGstXFade *player, GError **error);
static gboolean start_sink (RBPlayerGstXFade *player, GError **error);
static gboolean stop_sink (RBPlayerGstXFade *player);
static void maybe_stop_sink (RBPlayerGstXFade *player);

static gboolean silencesrc_push (RBPlayerGstXFade *player);

GType rb_xfade_stream_get_type (void);
GType rb_xfade_stream_bin_get_type (void);

G_DEFINE_TYPE_WITH_CODE(RBPlayerGstXFade, rb_player_gst_xfade, G_TYPE_OBJECT,
			G_IMPLEMENT_INTERFACE(RB_TYPE_PLAYER,
					      rb_player_init)
			G_IMPLEMENT_INTERFACE(RB_TYPE_PLAYER_GST_TEE,
					      rb_player_gst_tee_init)
			G_IMPLEMENT_INTERFACE(RB_TYPE_PLAYER_GST_FILTER,
					      rb_player_gst_filter_init))

#define GET_PRIVATE(o) (G_TYPE_INSTANCE_GET_PRIVATE ((o), RB_TYPE_PLAYER_GST_XFADE, RBPlayerGstXFadePrivate))

#define RB_PLAYER_GST_XFADE_TICK_HZ 5

#define EPSILON			(0.001)
#define STREAM_PLAYING_MESSAGE	"rb-stream-playing"
#define FADE_OUT_DONE_MESSAGE	"rb-fade-out-done"
#define FADE_IN_DONE_MESSAGE	"rb-fade-in-done"
#define STREAM_EOS_MESSAGE	"rb-stream-eos"

#define STREAM_URI_TAG		"rb-stream-uri"

#define PAUSE_FADE_LENGTH	(GST_SECOND / 2)

enum
{
	PROP_0,
	PROP_BUS
};

enum
{
	PREPARE_SOURCE,
	CAN_REUSE_STREAM,
	REUSE_STREAM,
	MISSING_PLUGINS,
	GET_STREAM_FILTERS,
	LAST_SIGNAL
};

/* copied from gsturidecodebin.c:stream_uris */
static const char *stream_schemes[] = {
	"http", "https", "mms", "mmsh", "mmsu", "mmst", "ssh", "ftp", "sftp"
};

static guint signals[LAST_SIGNAL] = { 0 };

struct _RBPlayerGstXFadePrivate
{
	/* probably don't need to store pointers to these either */
	GstElement *pipeline;
	GstElement *outputbin;
	GstElement *silencesrc;
	GstElement *silencebin;
	GstElement *adder;
	GstElement *capsfilter;
	GstElement *volume;
	GstElement *sink;
	GstElement *tee;
	GstElement *filterbin;
	GstElement *filteridentity;
	GstElement *filterconvert;
	GstElement *volume_handler;
	enum {
		SINK_NULL,
		SINK_STOPPED,
		SINK_PLAYING
	} sink_state;
	GRecMutex sink_lock;

	GList *waiting_tees;
	GList *waiting_filters;

	GRecMutex stream_list_lock;
	GList *streams;
	gint linked_streams;

	int volume_changed;
	int volume_applied;
	float cur_volume;

	guint tick_timeout_id;

	guint stream_reap_id;
	guint stop_sink_id;
	guint bus_watch_id;

	guint bus_idle_id;
	GList *idle_messages;

	char silence_buffer[1024];
	guint silence_idle_id;
};


/* these aren't actually used to construct bitmasks,
 * but we search the list that way.
 */
typedef enum
{
	/* stable states */
	WAITING = 1,
	PLAYING = 2,
	PAUSED = 4,

	/* transition states */
	REUSING = 8,
	PREROLLING = 16,
	PREROLL_PLAY = 32,
	FADING_IN = 64,
	SEEKING = 128,
	SEEKING_PAUSED = 256,
	SEEKING_EOS = 512, 
	WAITING_EOS = 1024,
	FADING_OUT = 2048,
	FADING_OUT_PAUSED = 4096,
	PENDING_REMOVE = 8192
} StreamState;

typedef struct {
	GstBinClass bin_class;
} RBXFadeStreamClass;


typedef struct
{
	GstBin parent;
	RBPlayerGstXFade *player;

	GMutex lock;

	char *uri;
	gpointer stream_data;
	GDestroyNotify stream_data_destroy;

	/* stream reuse data */
	char *new_uri;
	gpointer new_stream_data;
	GDestroyNotify new_stream_data_destroy;

	/* probably don't need to store pointers to all of these.. */
	GstElement *decoder;
	GstElement *volume;
	GstElement *audioconvert;
	GstElement *audioresample;
	GstElement *capsfilter;
	GstElement *preroll;
	GstElement *identity;
	gboolean decoder_linked;
	gboolean emitted_playing;
	gboolean emitted_image;
	gboolean emitted_fake_playing;

	GstPad *decoder_pad;
	GstPad *src_pad;
	GstPad *ghost_pad;
	GstPad *adder_pad;
	gboolean src_blocked;
	gboolean needs_unlink;
	GstClockTime base_time;

	gint64 seek_target;

	GstTimedValueControlSource *fader;
	StreamState state;
	RBPlayerPlayType play_type;
	gint64 crossfade;
	gboolean fading;
	gboolean starting_eos;
	gboolean use_buffering;

	gulong adjust_probe_id;
	gulong block_probe_id;

	double fade_end;

	gboolean emitted_error;
	gulong error_idle_id;
	GError *error;

	GSList *missing_plugins;
	gulong  emit_missing_plugins_id;

	GList *tags;
} RBXFadeStream;

#define RB_TYPE_XFADE_STREAM 	(rb_xfade_stream_get_type ())
#define RB_XFADE_STREAM(obj)	(G_TYPE_CHECK_INSTANCE_CAST ((obj), RB_TYPE_XFADE_STREAM, RBXFadeStream))
#define RB_IS_XFADE_STREAM(obj)	(G_TYPE_CHECK_INSTANCE_TYPE ((obj), RB_TYPE_XFADE_STREAM))

static void adjust_stream_base_time (RBXFadeStream *stream);
static gboolean actually_start_stream (RBXFadeStream *stream, GError **error);

static void rb_xfade_stream_class_init (RBXFadeStreamClass *klass);

G_DEFINE_TYPE(RBXFadeStream, rb_xfade_stream, GST_TYPE_BIN)

static gboolean
rb_xfade_stream_send_event (GstElement *element, GstEvent *event)
{
	GstPad *pad;
	GstPad *ghost_pad;
	gboolean ret;

	/* just send the event to the element that provides the src pad */
	ghost_pad = gst_element_get_static_pad (element, "src");
	pad = gst_ghost_pad_get_target (GST_GHOST_PAD (ghost_pad));

	ret = gst_element_send_event (GST_PAD_PARENT (pad), event);

	gst_object_unref (pad);
	gst_object_unref (ghost_pad);

	return ret;
}

static void
rb_xfade_stream_init (RBXFadeStream *stream)
{
	g_mutex_init (&stream->lock);
}

static void
rb_xfade_stream_dispose_stream_data (RBXFadeStream *stream)
{
	if (stream->stream_data && stream->stream_data_destroy) {
		stream->stream_data_destroy (stream->stream_data);
	}
	stream->stream_data = NULL;
	stream->stream_data_destroy = NULL;
}

static void
rb_xfade_stream_dispose (GObject *object)
{
	RBXFadeStream *sd = RB_XFADE_STREAM (object);

	rb_debug ("disposing stream %s", sd->uri);

	if (sd->decoder != NULL) {
		gst_object_unref (sd->decoder);
		sd->decoder = NULL;
	}

	if (sd->volume != NULL) {
		gst_object_unref (sd->volume);
		sd->volume = NULL;
	}

	if (sd->fader != NULL) {
		gst_object_unref (sd->fader);
		sd->fader = NULL;
	}

	if (sd->audioconvert != NULL) {
		gst_object_unref (sd->audioconvert);
		sd->audioconvert = NULL;
	}

	if (sd->audioresample != NULL) {
		gst_object_unref (sd->audioresample);
		sd->audioresample = NULL;
	}

	if (sd->player != NULL) {
		g_object_unref (sd->player);
		sd->player = NULL;
	}

	if (sd->tags != NULL) {
		rb_list_destroy_free (sd->tags, (GDestroyNotify) gst_tag_list_unref);
		sd->tags = NULL;
	}

	rb_xfade_stream_dispose_stream_data (sd);

	G_OBJECT_CLASS (rb_xfade_stream_parent_class)->dispose (object);
}

static void
rb_xfade_stream_finalize (GObject *object)
{
	RBXFadeStream *sd = RB_XFADE_STREAM (object);
	
	g_free (sd->uri);

	if (sd->error != NULL) {
		g_error_free (sd->error);
	}

	G_OBJECT_CLASS (rb_xfade_stream_parent_class)->finalize (object);
}

static void
rb_xfade_stream_class_init (RBXFadeStreamClass *klass)
{
	GObjectClass *object_class = G_OBJECT_CLASS (klass);
	GstElementClass *element_class = GST_ELEMENT_CLASS (klass);

	object_class->dispose = rb_xfade_stream_dispose;
	object_class->finalize = rb_xfade_stream_finalize;

	element_class->send_event = rb_xfade_stream_send_event;
}

/* caller must hold stream list lock */
static void
dump_stream_list (RBPlayerGstXFade *player)
{
	GList *l;
	if (player->priv->streams == NULL) {
		rb_debug ("stream list is empty");
	} else {
		rb_debug ("current stream list:");
		for (l = player->priv->streams; l != NULL; l = l->next) {
			RBXFadeStream *stream = (RBXFadeStream *)l->data;
			const char *statename = "<wtf>";
			switch (stream->state) {
			case WAITING:	 	statename = "waiting";		break;
			case PLAYING:	 	statename = "playing";		break;
			case PAUSED:	 	statename = "paused";		break;

			case REUSING:		statename = "reusing";		break;
			case PREROLLING: 	statename = "prerolling"; 	break;
			case PREROLL_PLAY: 	statename = "preroll->play"; 	break;
			case FADING_IN: 	statename = "fading in"; 	break;
			case SEEKING:		statename = "seeking";		break;
			case SEEKING_PAUSED:	statename = "seeking->paused";	break;
			case SEEKING_EOS:	statename = "seeking post EOS"; break;
			case WAITING_EOS: 	statename = "waiting for EOS"; 	break;
			case FADING_OUT: 	statename = "fading out"; 	break;
			case FADING_OUT_PAUSED: statename = "fading->paused";   break;

			case PENDING_REMOVE:	statename = "pending remove";	break;
			}

			rb_debug ("[%s] %s", statename, stream->uri);
		}
	}
}

/* caller must hold stream list lock */
static RBXFadeStream *
find_stream_by_uri (RBPlayerGstXFade *player, const char *uri)
{
	GList *i;
	if (uri == NULL)
		return NULL;

	for (i = player->priv->streams; i != NULL; i = i->next) {
		RBXFadeStream *stream = (RBXFadeStream *)i->data;
		if (strcmp (uri, stream->uri) == 0)
			return g_object_ref (stream);
	}
	return NULL;
}

/* caller must hold stream list lock */
static RBXFadeStream *
find_stream_by_element (RBPlayerGstXFade *player, GstElement *element)
{
	GList *i;

	for (i = player->priv->streams; i != NULL; i = i->next) {
		RBXFadeStream *stream;
		GstElement *e;

		stream = (RBXFadeStream *)i->data;
		e = element;
		while (e != NULL) {
			if (e == GST_ELEMENT (stream))
				return g_object_ref (stream);

			e = GST_ELEMENT_PARENT (e);
		}
	}

	return NULL;
}

/* caller must hold stream list lock */
static RBXFadeStream *
find_stream_by_state (RBPlayerGstXFade *player, gint state_mask)
{
	GList *i;

	for (i = player->priv->streams; i != NULL; i = i->next) {
		RBXFadeStream *stream;

		stream = (RBXFadeStream *)i->data;
		if ((stream->state & state_mask) != 0) {
			return g_object_ref (stream);
		}
	}

	return NULL;
}

static RBXFadeStream *
find_stream_for_message (RBPlayerGstXFade *player, GstMessage *message)
{
	GstObject *message_src;
	RBXFadeStream *stream;

	/* first see if the message comes from an element in the stream bin */
	message_src = GST_MESSAGE_SRC (message);
	if (GST_IS_PAD (message_src)) {
		message_src = GST_OBJECT_PARENT (message_src);
	}
	stream = find_stream_by_element (player, GST_ELEMENT (message_src));
	if (stream != NULL)
		return stream;

	/* tag messages are emitted by the sink, so we can't find the message
	 * source in a stream bin.  instead, we add a tag to the stream containing
	 * the stream uri and use that to locate the stream.
	 */
	if (GST_MESSAGE_TYPE (message) == GST_MESSAGE_TAG) {
		GstTagList *tags;
		char *loc;
		gst_message_parse_tag (message, &tags);

		if (gst_tag_list_get_string (tags, STREAM_URI_TAG, &loc)) {
			stream = find_stream_by_uri (player, loc);
			g_free (loc);
			if (stream)
				return stream;
		}
	}

	return NULL;
}

static void
rb_player_gst_xfade_get_property (GObject *object,
				  guint prop_id,
				  GValue *value,
				  GParamSpec *pspec)
{
	RBPlayerGstXFade *player = RB_PLAYER_GST_XFADE (object);

	switch (prop_id) {
	case PROP_BUS:
		if (player->priv->pipeline) {
			GstBus *bus;
			bus = gst_element_get_bus (player->priv->pipeline);
			g_value_set_object (value, bus);
			gst_object_unref (bus);
		}
		break;
	default:
		G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
		break;
	}
}

static void
rb_player_gst_xfade_set_property (GObject *object,
				  guint prop_id,
				  const GValue *value,
				  GParamSpec *pspec)
{
	/*RBPlayerGstXFade *player = RB_PLAYER_GST_XFADE (object);*/

	switch (prop_id) {
	default:
		G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
		break;
	}
}

static void
rb_player_gst_xfade_class_init (RBPlayerGstXFadeClass *klass)
{
	GObjectClass *object_class = G_OBJECT_CLASS (klass);

	object_class->dispose = rb_player_gst_xfade_dispose;
	object_class->finalize = rb_player_gst_xfade_finalize;
	object_class->set_property = rb_player_gst_xfade_set_property;
	object_class->get_property = rb_player_gst_xfade_get_property;

	g_object_class_install_property (object_class,
					 PROP_BUS,
					 g_param_spec_object ("bus",
							      "bus",
							      "GStreamer message bus",
							      GST_TYPE_BUS,
							      G_PARAM_READABLE));

	signals[PREPARE_SOURCE] =
		g_signal_new ("prepare-source",
			      G_OBJECT_CLASS_TYPE (object_class),
			      G_SIGNAL_RUN_LAST,
			      G_STRUCT_OFFSET (RBPlayerGstXFadeClass, prepare_source),
			      NULL, NULL,
			      rb_marshal_VOID__STRING_OBJECT,
			      G_TYPE_NONE,
			      2,
			      G_TYPE_STRING, GST_TYPE_ELEMENT);
	signals[CAN_REUSE_STREAM] =
		g_signal_new ("can-reuse-stream",
			      G_OBJECT_CLASS_TYPE (object_class),
			      G_SIGNAL_RUN_LAST,
			      G_STRUCT_OFFSET (RBPlayerGstXFadeClass, can_reuse_stream),
			      NULL, NULL,
			      rb_marshal_BOOLEAN__STRING_STRING_OBJECT,
			      G_TYPE_BOOLEAN,
			      3,
			      G_TYPE_STRING, G_TYPE_STRING, GST_TYPE_ELEMENT);
	signals[REUSE_STREAM] =
		g_signal_new ("reuse-stream",
			      G_OBJECT_CLASS_TYPE (object_class),
			      G_SIGNAL_RUN_LAST,
			      G_STRUCT_OFFSET (RBPlayerGstXFadeClass, reuse_stream),
			      NULL, NULL,
			      rb_marshal_VOID__STRING_STRING_OBJECT,
			      G_TYPE_NONE,
			      3,
			      G_TYPE_STRING, G_TYPE_STRING, GST_TYPE_ELEMENT);
	signals[MISSING_PLUGINS] =
		g_signal_new ("missing-plugins",
			      G_OBJECT_CLASS_TYPE (object_class),
			      G_SIGNAL_RUN_LAST,
			      0,	/* no point handling this internally */
			      NULL, NULL,
			      rb_marshal_VOID__POINTER_POINTER_POINTER,
			      G_TYPE_NONE,
			      3,
			      G_TYPE_POINTER, G_TYPE_STRV, G_TYPE_STRV);
	signals[GET_STREAM_FILTERS] =
		g_signal_new ("get-stream-filters",
			      G_OBJECT_CLASS_TYPE (object_class),
			      G_SIGNAL_RUN_LAST,
			      0,
			      rb_signal_accumulator_value_array, NULL,
			      rb_marshal_BOXED__STRING,
			      G_TYPE_ARRAY,
			      1,
			      G_TYPE_STRING);

	g_type_class_add_private (klass, sizeof (RBPlayerGstXFadePrivate));

	gst_tag_register_static (STREAM_URI_TAG, GST_TAG_FLAG_META, G_TYPE_STRING, "rb stream uri", "rb stream uri", gst_tag_merge_use_first);
}

static void
rb_player_init (RBPlayerIface *iface)
{
	iface->open = rb_player_gst_xfade_open;
	iface->opened = rb_player_gst_xfade_opened;
	iface->close = rb_player_gst_xfade_close;
	iface->play = rb_player_gst_xfade_play;
	iface->pause = rb_player_gst_xfade_pause;
	iface->playing = rb_player_gst_xfade_playing;
	iface->set_volume = rb_player_gst_xfade_set_volume;
	iface->get_volume = rb_player_gst_xfade_get_volume;
	iface->seekable = rb_player_gst_xfade_seekable;
	iface->set_time = rb_player_gst_xfade_set_time;
	iface->get_time = rb_player_gst_xfade_get_time;
	iface->multiple_open = (RBPlayerFeatureFunc) rb_true_function;
}

static void
rb_player_gst_tee_init (RBPlayerGstTeeIface *iface)
{
	iface->add_tee = rb_player_gst_xfade_add_tee;
	iface->remove_tee = rb_player_gst_xfade_remove_tee;
}

static void
rb_player_gst_filter_init (RBPlayerGstFilterIface *iface)
{
	iface->add_filter = rb_player_gst_xfade_add_filter;
	iface->remove_filter = rb_player_gst_xfade_remove_filter;
}


static void
rb_player_gst_xfade_init (RBPlayerGstXFade *player)
{
	player->priv = GET_PRIVATE (player);

	g_rec_mutex_init (&player->priv->stream_list_lock);
	g_rec_mutex_init (&player->priv->sink_lock);
	player->priv->cur_volume = 1.0f;
}

static void
rb_player_gst_xfade_dispose (GObject *object)
{
	RBPlayerGstXFade *player;
	GList *l;

	g_return_if_fail (RB_IS_PLAYER_GST_XFADE (object));
	player = RB_PLAYER_GST_XFADE (object);

	/* clean up streams */
	g_rec_mutex_lock (&player->priv->stream_list_lock);
	for (l = player->priv->streams; l != NULL; l = l->next) {
		RBXFadeStream *stream = (RBXFadeStream *)l->data;

		/* unlink instead? */
		gst_element_set_state (GST_ELEMENT (stream), GST_STATE_NULL);

		g_object_unref (stream);
	}
	g_list_free (player->priv->streams);
	player->priv->streams = NULL;
	g_rec_mutex_unlock (&player->priv->stream_list_lock);

	if (player->priv->volume_handler) {
		g_object_unref (player->priv->volume_handler);
		player->priv->volume_handler = NULL;
	}

	g_rec_mutex_lock (&player->priv->sink_lock);
	stop_sink (player);

	if (player->priv->bus_idle_id != 0) {
		g_source_remove (player->priv->bus_idle_id);
		player->priv->bus_idle_id = 0;

		rb_list_destroy_free (player->priv->idle_messages, (GDestroyNotify) gst_mini_object_unref);
		player->priv->idle_messages = NULL;
	}
	g_rec_mutex_unlock (&player->priv->sink_lock);

	if (player->priv->pipeline != NULL) {
		/* maybe we should keep references to the adder, sink, etc.? */
		gst_element_set_state (player->priv->pipeline, GST_STATE_NULL);
		g_object_unref (player->priv->pipeline);
		player->priv->pipeline = NULL;
	}

	G_OBJECT_CLASS (rb_player_gst_xfade_parent_class)->dispose (object);
}

static void
rb_player_gst_xfade_finalize (GObject *object)
{
	RBPlayerGstXFade *player;

	player = RB_PLAYER_GST_XFADE (object);

	if (player->priv->waiting_tees) {
		g_list_foreach (player->priv->waiting_tees, (GFunc)gst_object_ref_sink, NULL);
	}
	g_list_free (player->priv->waiting_tees);

	if (player->priv->waiting_filters) {
		g_list_foreach (player->priv->waiting_filters, (GFunc)gst_object_ref_sink, NULL);
	}
	g_list_free (player->priv->waiting_filters);

	G_OBJECT_CLASS (rb_player_gst_xfade_parent_class)->finalize (object);
}

RBPlayer *
rb_player_gst_xfade_new (GError **error)
{
	RBPlayer *mp;

	mp = RB_PLAYER (g_object_new (RB_TYPE_PLAYER_GST_XFADE, NULL, NULL));

	return mp;
}

static gboolean
emit_stream_error_cb (RBXFadeStream *stream)
{
	stream->error_idle_id = 0;
	_rb_player_emit_error (RB_PLAYER (stream->player),
			       stream->stream_data,
			       stream->error);
	g_error_free (stream->error);
	stream->error = NULL;

	return FALSE;
}

static void
emit_stream_error (RBXFadeStream *stream, GError *error)
{
	if (stream->error_idle_id != 0) {
		g_error_free (error);
	} else {
		stream->error = error;
		stream->error_idle_id = g_idle_add ((GSourceFunc) emit_stream_error_cb,
						    stream);
	}
}

static void
post_stream_playing_message (RBXFadeStream *stream, gboolean fake)
{
	GstMessage *msg;
	GstStructure *s;

	if (stream->emitted_playing) {
		return;
	}

	rb_debug ("posting " STREAM_PLAYING_MESSAGE " message for stream %s", stream->uri);
	s = gst_structure_new_empty (STREAM_PLAYING_MESSAGE);
	msg = gst_message_new_application (GST_OBJECT (stream), s);
	gst_element_post_message (GST_ELEMENT (stream), msg);

	if (fake == FALSE) {
		stream->emitted_playing = TRUE;
	} else {
		stream->emitted_fake_playing = TRUE;
	}
}

static GstPadProbeReturn
adjust_base_time_probe_cb (GstPad *pad, GstPadProbeInfo *info, RBXFadeStream *stream)
{
	rb_debug ("attempting to adjust base time for stream %s", stream->uri);
	g_mutex_lock (&stream->lock);
	adjust_stream_base_time (stream);
	g_mutex_unlock (&stream->lock);
	return GST_PAD_PROBE_OK;
}

/* updates a stream's base time so its position is reported correctly */
static void
adjust_stream_base_time (RBXFadeStream *stream)
{
	gint64 output_pos = -1;
	gint64 stream_pos = -1;

	if (stream->adder_pad == NULL) {
		rb_debug ("stream isn't linked, can't adjust base time");
		return;
	}

	gst_element_query_position (GST_PAD_PARENT (stream->adder_pad), GST_FORMAT_TIME, &output_pos);
	if (output_pos != -1) {
		stream->base_time = output_pos;
	}

	/* offset the base position to account for the current stream position */
	gst_element_query_position (stream->volume, GST_FORMAT_TIME, &stream_pos);
	if (stream_pos != -1) {
		rb_debug ("adjusting base time: %" G_GINT64_FORMAT
		    " - %" G_GINT64_FORMAT " => %" G_GINT64_FORMAT,
		    stream->base_time, stream_pos,
		    stream->base_time - stream_pos);
		stream->base_time -= stream_pos;

		/* once we've successfully adjusted the base time, we don't need the data probe */
		if (stream->adjust_probe_id != 0) {
			gst_pad_remove_probe (stream->ghost_pad, stream->adjust_probe_id);
			stream->adjust_probe_id = 0;
		}
	} else {
		rb_debug ("unable to adjust base time as position query failed");

		/* add a pad probe to attempt to adjust when the next buffer goes out */
		if (stream->adjust_probe_id == 0) {
			stream->adjust_probe_id =
				gst_pad_add_probe (stream->ghost_pad,
						   GST_PAD_PROBE_TYPE_BUFFER,
						   (GstPadProbeCallback) adjust_base_time_probe_cb,
						   stream,
						   NULL);
		}
	}
}

/* called on a streaming thread when the volume level for a stream changes. */
static void
volume_changed_cb (GObject *object, GParamSpec *pspec, RBPlayerGstXFade *player)
{
	RBXFadeStream *stream;
	gdouble vol;
	char *message = NULL;

	/* post app messages on the bus when fades complete.
	 * our bus callback will handle them on the main thread.
	 */

	g_rec_mutex_lock (&player->priv->stream_list_lock);
	stream = find_stream_by_element (player, GST_ELEMENT (object));
	g_rec_mutex_unlock (&player->priv->stream_list_lock);

	if (stream == NULL) {
		rb_debug ("got volume change for unknown stream");
		return;
	}

	g_mutex_lock (&stream->lock);

	/* check if the fade is complete */
	g_object_get (stream->volume, "volume", &vol, NULL);
	switch (stream->state) {
	case FADING_IN:
		if (vol > (stream->fade_end - EPSILON) && stream->fading) {
			rb_debug ("stream %s fully faded in (at %f) -> PLAYING state", stream->uri, vol);
			message = FADE_IN_DONE_MESSAGE;
			stream->fading = FALSE;
			stream->state = PLAYING;
			
		/*} else {
			rb_debug ("fading %s in: %f", stream->uri, (float)vol);*/
		}
		break;
	case FADING_OUT:
	case FADING_OUT_PAUSED:
		if (vol < (stream->fade_end + EPSILON)) {
			rb_debug ("stream %s fully faded out (at %f)", stream->uri, vol);
			if (stream->fading) {
				message = FADE_OUT_DONE_MESSAGE;
				stream->fading = FALSE;
			}
		} else {
			/*rb_debug ("fading %s out: %f", stream->uri, (float)vol);*/
			/* force the volume element out of passthrough mode so it
			 * continues to update the controller (otherwise, if the
			 * fade out starts at 1.0, it never gets anywhere)
			 */
			gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (stream->volume), FALSE);
		}
		break;
	default:
		/*rb_debug ("unexpectedly got a volume change for stream %s to %f (not fading)", stream->uri, (float)vol);*/
		break;
	}
	
	g_mutex_unlock (&stream->lock);

	if (message != NULL) {
		GstMessage *msg;
		GstStructure *s;

		rb_debug ("posting %s message for stream %s", message, stream->uri);
		s = gst_structure_new_empty (message);
		msg = gst_message_new_application (GST_OBJECT (object), s);
		gst_element_post_message (GST_ELEMENT (object), msg);
	}

	g_object_unref (stream);
}

/* starts a volume slide on a stream.
 * volume_changed_cb watches the volume change
 * and posts a message on the bus when the slide
 * is done.
 */
static void
start_stream_fade (RBXFadeStream *stream, double start, double end, gint64 time)
{
	gint64 pos = -1;

	/* hmm, can we take the stream lock safely here?  probably should.. */

	gst_element_query_position (stream->volume, GST_FORMAT_TIME, &pos);
	if (pos < 0) {
		/* probably means we haven't actually started the stream yet.
		 * we also get (weird) negative results with some decoders
		 * (mad but not flump3dec, for instance) immediately after prerolling.
		 * the controller doesn't seem to work if we give it a 0 timestamp
		 * here, but something unnoticeably later does work.
		 */
		pos = 100000;
	}

	rb_debug ("fading stream %s: [%f, %" G_GINT64_FORMAT "] to [%f, %" G_GINT64_FORMAT "]",
		  stream->uri,
		  (float)start, pos,
		  (float)end, pos + time);

	g_signal_handlers_block_by_func (stream->volume, volume_changed_cb, stream->player);

	/* apparently we need to set the starting volume, otherwise fading in doesn't work. */
	stream->fade_end = end;
	g_object_set (stream->volume, "volume", start, NULL);

	gst_timed_value_control_source_unset_all (stream->fader);

	if (gst_timed_value_control_source_set (stream->fader, pos, start/10.0) == FALSE) {
		rb_debug ("controller didn't like our start point");
	}
	if (gst_timed_value_control_source_set (stream->fader, 0, start/10.0) == FALSE) {
		rb_debug ("controller didn't like our 0 start point");
	}

	if (gst_timed_value_control_source_set (stream->fader, pos + time, end/10.0) == FALSE) {
		rb_debug ("controller didn't like our end point");
	}

	g_signal_handlers_unblock_by_func (stream->volume, volume_changed_cb, stream->player);

	stream->fading = TRUE;

	/* tiny hack:  if the controlled element is in passthrough mode, the
	 * controller won't get updated.
	 */
	gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (stream->volume), FALSE);
}


/* links a stream bin to the adder
 * - adds the bin to the pipeline
 * - links to a new adder pad
 * - unblocks the stream if it's blocked
 */
static gboolean
link_and_unblock_stream (RBXFadeStream *stream, GError **error)
{
	GstPadLinkReturn plr;
	GstStateChangeReturn scr;
	RBPlayerGstXFade *player = stream->player;
	gboolean result;
	
	if (start_sink (player, error) == FALSE) {
		rb_debug ("sink didn't start, so we're not going to link the stream");
		return FALSE;
	}

	if (stream->adder_pad != NULL) {
		rb_debug ("stream %s is already linked", stream->uri);
		return TRUE;
	}
	stream->needs_unlink = FALSE;

	rb_debug ("linking stream %s", stream->uri);
	if (GST_ELEMENT_PARENT (GST_ELEMENT (stream)) == NULL)
		gst_bin_add (GST_BIN (player->priv->pipeline), GST_ELEMENT (stream));

	stream->adder_pad = gst_element_get_request_pad (player->priv->adder, "sink_%u");
	if (stream->adder_pad == NULL) {
		/* this error message kind of sucks */
		rb_debug ("couldn't get adder pad to link in new stream");
		g_set_error (error,
			     RB_PLAYER_ERROR,
			     RB_PLAYER_ERROR_GENERAL,
			     _("Failed to link new stream into GStreamer pipeline"));
		return FALSE;
	}

	plr = gst_pad_link (stream->ghost_pad, stream->adder_pad);
	if (GST_PAD_LINK_FAILED (plr)) {
		gst_element_release_request_pad (player->priv->adder, stream->adder_pad);
		stream->adder_pad = NULL;

		/* this error message kind of sucks */
		rb_debug ("linking stream pad to adder pad failed: %d", plr);
		g_set_error (error,
			     RB_PLAYER_ERROR,
			     RB_PLAYER_ERROR_GENERAL,
			     _("Failed to link new stream into GStreamer pipeline"));
		return FALSE;
	}


	g_atomic_int_inc (&player->priv->linked_streams);
	rb_debug ("now have %d linked streams", player->priv->linked_streams);

	result = TRUE;
	g_mutex_lock (&stream->lock);
	if (stream->src_blocked) {
		GstStateChangeReturn state_ret;

		gst_pad_remove_probe (stream->src_pad, stream->block_probe_id);
		stream->block_probe_id = 0;

		rb_debug ("stream %s is unblocked -> FADING_IN | PLAYING", stream->uri);
		stream->src_blocked = FALSE;
		if (stream->fading)
			stream->state = FADING_IN;
		else
			stream->state = PLAYING;
		
		adjust_stream_base_time (stream);

		/* should handle state change failures here.. */
		state_ret = gst_element_set_state (GST_ELEMENT (stream), GST_STATE_PLAYING);
		rb_debug ("stream %s state change returned: %s", stream->uri,
			  gst_element_state_change_return_get_name (state_ret));

		post_stream_playing_message (stream, FALSE);
	} else {
		rb_debug ("??? stream %s is already unblocked -> PLAYING", stream->uri);
		stream->state = PLAYING;
		adjust_stream_base_time (stream);

		scr = gst_element_set_state (GST_ELEMENT (stream), GST_STATE_PLAYING);

		post_stream_playing_message (stream, FALSE);

		if (scr == GST_STATE_CHANGE_FAILURE) {
			g_set_error (error,
				     RB_PLAYER_ERROR,
				     RB_PLAYER_ERROR_GENERAL,
				     _("Failed to start new stream"));
			result = FALSE;
		}
	}
	g_mutex_unlock (&stream->lock);
	return result;
}

/*
 * reuses a stream.  the stream reuse signal is handled by some
 * external code somewhere.
 */
static void
reuse_stream (RBXFadeStream *stream)
{
	g_signal_emit (stream->player,
		       signals[REUSE_STREAM], 0,
		       stream->new_uri, stream->uri, GST_ELEMENT (stream));

	/* replace URI and stream data */
	g_free (stream->uri);
	stream->uri = stream->new_uri;

	rb_xfade_stream_dispose_stream_data (stream);
	stream->stream_data = stream->new_stream_data;
	stream->stream_data_destroy = stream->new_stream_data_destroy;

	stream->new_uri = NULL;
	stream->new_stream_data = NULL;
	stream->new_stream_data_destroy = NULL;

	stream->emitted_playing = FALSE;
	stream->emitted_image = FALSE;
}


/*
 * performs a seek on an unlinked and blocked stream.
 * if the stream is still in FADING_IN or PLAYING state,
 * relinks and unblocks the stream.
 */
static void
perform_seek (RBXFadeStream *stream)
{
	GstEvent *event;

	rb_debug ("sending seek event..");

	event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
				    GST_SEEK_FLAG_FLUSH,
				    GST_SEEK_TYPE_SET, stream->seek_target,
				    GST_SEEK_TYPE_NONE, -1);
	gst_pad_send_event (stream->src_pad, event);

	g_mutex_lock (&stream->lock);
	switch (stream->state) {
	case SEEKING:
		stream->state = PLAYING;
		break;
	case SEEKING_PAUSED:
		rb_debug ("leaving paused stream %s unlinked", stream->uri);
		stream->state = PAUSED;
		break;
	case SEEKING_EOS:
		rb_debug ("waiting for pad block to complete for %s before unlinking", stream->uri);
		break;
	default:
		break;
	}

	g_mutex_unlock (&stream->lock);
}

static gboolean
perform_seek_idle (RBXFadeStream *stream)
{
	perform_seek (stream);
	g_object_unref (stream);
	return FALSE;
}

/*
 * called when a stream doing a post-EOS seek is blocked.  this indicates
 * that the seek has completed (that's the only way data can flow out of
 * the stream bin), so the stream can be linked and unblocked.
 */
static GstPadProbeReturn
post_eos_seek_blocked_cb (GstPad *pad, GstPadProbeInfo *info, RBXFadeStream *stream)
{
	GError *error = NULL;

	g_mutex_lock (&stream->lock);
	rb_debug ("stream %s is blocked; linking and unblocking", stream->uri);
	stream->src_blocked = TRUE;
	g_mutex_unlock (&stream->lock);

	if (link_and_unblock_stream (stream, &error) == FALSE) {
		emit_stream_error (stream, error);
	}


	return GST_PAD_PROBE_REMOVE;
}

/*
 * called when a src pad for a stream is blocked during reuse.
 * we don't need to do anything here.
 */
static GstPadProbeReturn
unlink_reuse_blocked_cb (GstPad *pad, GstPadProbeInfo *info, RBXFadeStream *stream)
{
	return GST_PAD_PROBE_OK;
}

static void
unlink_reuse_relink (RBPlayerGstXFade *player, RBXFadeStream *stream)
{
	GError *error = NULL;

	g_mutex_lock (&stream->lock);

	if (stream->adder_pad == NULL) {
		rb_debug ("stream %s doesn't need to be unlinked.. weird.", stream->uri);
	} else {
		rb_debug ("unlinking stream %s for reuse", stream->uri);

		if (gst_pad_unlink (stream->ghost_pad, stream->adder_pad) == FALSE) {
			g_warning ("Couldn't unlink stream %s: this is going to suck.", stream->uri);
		}

		gst_element_release_request_pad (player->priv->adder, stream->adder_pad);
		stream->adder_pad = NULL;

		(void) g_atomic_int_dec_and_test (&player->priv->linked_streams);
		rb_debug ("%d linked streams left", player->priv->linked_streams);
	}

	stream->needs_unlink = FALSE;
	stream->emitted_playing = FALSE;
	stream->emitted_image = FALSE;

	g_mutex_unlock (&stream->lock);

	/* block the src pad so we don't get not-linked errors if it pushes a buffer
	 * before we get around to relinking
	 */
	gst_pad_add_probe (stream->src_pad,
			   GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM,
			   (GstPadProbeCallback) unlink_reuse_blocked_cb,
			   stream,
			   NULL);
	stream->src_blocked = TRUE;

	reuse_stream (stream);
	if (link_and_unblock_stream (stream, &error) == FALSE) {
		emit_stream_error (stream, error);
	}
}

/* called when a stream's source pad is blocked, so it can be unlinked
 * from the pipeline.
 */
static GstPadProbeReturn
unlink_blocked_cb (GstPad *pad, GstPadProbeInfo *info, RBXFadeStream *stream)
{
	int stream_state;
	gboolean last;
	RBPlayerGstXFade *player;
	GError *error = NULL;

	g_mutex_lock (&stream->lock);

	if (stream->needs_unlink == FALSE || stream->adder_pad == NULL) {
		g_mutex_unlock (&stream->lock);
		return GST_PAD_PROBE_OK;
	}

	rb_debug ("stream %s is blocked; unlinking", stream->uri);

	if (gst_pad_unlink (stream->ghost_pad, stream->adder_pad) == FALSE) {
		g_warning ("Couldn't unlink stream %s: things will probably go quite badly from here on", stream->uri);
	}
	stream->needs_unlink = FALSE;

	gst_element_release_request_pad (GST_PAD_PARENT (stream->adder_pad), stream->adder_pad);
	stream->adder_pad = NULL;

	stream->src_blocked = TRUE;
	stream->emitted_playing = FALSE;
	stream->emitted_image = FALSE;

	stream_state = stream->state;
	player = stream->player;

	g_mutex_unlock (&stream->lock);

	/* might want a stream-paused signal here? */

	last = g_atomic_int_dec_and_test (&player->priv->linked_streams);
	rb_debug ("%d linked streams left", player->priv->linked_streams);

	/* handle unlinks for seeking and stream reuse */
	switch (stream_state) {
	case REUSING:
		reuse_stream (stream);
		if (link_and_unblock_stream (stream, &error) == FALSE) {
			emit_stream_error (stream, error);
		}
		break;

	case SEEKING_PAUSED:
		g_idle_add ((GSourceFunc) perform_seek_idle, g_object_ref (stream));
		/* fall through.  this only happens when pausing, so it's OK
		 * to stop the sink here.
		 */
	default:
		/* consider pausing the sink if this is the linked last stream */
		if (last) {
			maybe_stop_sink (player);
		}

		break;
	}

	return GST_PAD_PROBE_OK;
}

/*
 * blocks and unlinks a stream.  this is the only way we can pause a stream -
 * if the stream is linked to the adder and the audio sink is in PLAYING, the
 * stream will play.
 */
static void
unlink_and_block_stream (RBXFadeStream *stream)
{
	g_mutex_lock (&stream->lock);
	if (stream->adder_pad == NULL) {
		rb_debug ("stream %s is not linked", stream->uri);
		g_mutex_unlock (&stream->lock);
		return;
	}

	stream->needs_unlink = TRUE;
	if (stream->src_blocked) {
		/* probably shouldn't happen, but we'll handle it anyway */
		g_mutex_unlock (&stream->lock);
		unlink_blocked_cb (stream->src_pad, NULL, stream);
		return;
	}

	if (stream->block_probe_id == 0) {
		stream->block_probe_id = gst_pad_add_probe (stream->src_pad,
							    GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM,
							    (GstPadProbeCallback) unlink_blocked_cb,
							    stream,
							    NULL);
	} else {
		rb_debug ("already unlinking");
	}
	g_mutex_unlock (&stream->lock);
}

/*
 * sets a stream to NULL state, unlinks it from the adder,
 * removes it from the pipeline, removes it from the
 * stream list, and frees it (hopefully).
 *
 * must not be called on a streaming thread.
 */
static void
unlink_and_dispose_stream (RBPlayerGstXFade *player, RBXFadeStream *stream)
{
	GstStateChangeReturn sr;
	gboolean was_linked = FALSE;
	gboolean was_in_pipeline = FALSE;

	/* seems to be too much locking in here.. */


	rb_debug ("stopping stream %s", stream->uri);
	sr = gst_element_set_state (GST_ELEMENT (stream), GST_STATE_NULL);
	if (sr == GST_STATE_CHANGE_ASYNC) {
		/* downward state transitions aren't supposed to return ASYNC.. */
		rb_debug ("!!! stream %s isn't cooperating", stream->uri);
		gst_element_get_state (GST_ELEMENT (stream), NULL, NULL, GST_CLOCK_TIME_NONE);
	}
	
	g_mutex_lock (&stream->lock);

	if (stream->adder_pad != NULL) {
		rb_debug ("unlinking stream %s", stream->uri);
		if (gst_pad_unlink (stream->ghost_pad, stream->adder_pad) == FALSE) {
			g_warning ("Couldn't unlink stream %s: things will probably go quite badly from here on", stream->uri);
		}

		gst_element_release_request_pad (GST_PAD_PARENT (stream->adder_pad), stream->adder_pad);
		stream->adder_pad = NULL;

		was_linked = TRUE;
	}

	was_in_pipeline = (GST_ELEMENT_PARENT (GST_ELEMENT (stream)) == player->priv->pipeline);
	
	g_mutex_unlock (&stream->lock);

	if (was_in_pipeline)
		gst_bin_remove (GST_BIN (player->priv->pipeline), GST_ELEMENT (stream));

	if (was_linked) {
		gboolean last;

		last = g_atomic_int_dec_and_test (&player->priv->linked_streams);
		rb_debug ("now have %d linked streams", player->priv->linked_streams);

		if (last) {
			maybe_stop_sink (player);
		}
	}

	g_rec_mutex_lock (&player->priv->stream_list_lock);
	player->priv->streams = g_list_remove (player->priv->streams, stream);
	dump_stream_list (player);
	g_rec_mutex_unlock (&player->priv->stream_list_lock);

	g_object_unref (stream);
}

/* idle handler used to clean up finished streams */
static gboolean
reap_streams (RBPlayerGstXFade *player)
{
	GList *t;
	GList *reap = NULL;

	g_rec_mutex_lock (&player->priv->stream_list_lock);
	player->priv->stream_reap_id = 0;
	dump_stream_list (player);
	for (t = player->priv->streams; t != NULL; t = t->next) {
		RBXFadeStream *stream = (RBXFadeStream *)t->data;

		if (stream->state == PENDING_REMOVE) {
			reap = g_list_prepend (reap, stream);
		}
	}
	g_rec_mutex_unlock (&player->priv->stream_list_lock);

	for (t = reap; t != NULL; t = t->next) {
		RBXFadeStream *stream = (RBXFadeStream *)t->data;
		rb_debug ("reaping stream %s", stream->uri);
		unlink_and_dispose_stream (player, stream);
	}
	g_list_free (reap);

	return FALSE;
}

/* schedules a call to reap_streams */
static void
schedule_stream_reap (RBPlayerGstXFade *player)
{
	g_rec_mutex_lock (&player->priv->stream_list_lock);

	if (player->priv->stream_reap_id == 0) {
		dump_stream_list (player);
		player->priv->stream_reap_id = g_idle_add ((GSourceFunc) reap_streams, player);
	}
	
	g_rec_mutex_unlock (&player->priv->stream_list_lock);
}

/* emits a tag signal from the player, maybe */
static void
process_tag (const GstTagList *list, const gchar *tag, RBXFadeStream *stream)
{
	RBMetaDataField field;
	GValue value = {0,};

	/* process embedded images */
	if (!g_strcmp0 (tag, GST_TAG_IMAGE) || !g_strcmp0 (tag, GST_TAG_PREVIEW_IMAGE)) {
		if (stream->emitted_playing == FALSE || stream->emitted_image == FALSE) {
			GdkPixbuf *pixbuf;
			pixbuf = rb_gst_process_embedded_image (list, tag);
			if (pixbuf != NULL) {
				_rb_player_emit_image (RB_PLAYER (stream->player),
						       stream->stream_data,
						       pixbuf);
				g_object_unref (pixbuf);
				stream->emitted_image = TRUE;
				rb_debug ("emitting tag %s (p %d)", tag, stream->emitted_playing);
			}
		}
	} else if (rb_gst_process_tag_string (list, tag, &field, &value)) {
		rb_debug ("emitting info field %d", field);
		_rb_player_emit_info (RB_PLAYER (stream->player),
				      stream->stream_data,
				      field,
				      &value);
		g_value_unset (&value);
	}
}


static gboolean
emit_missing_plugins (RBXFadeStream *stream)
{
	char **details;
	char **descriptions;
	int count;
	GSList *t;
	int i;

	stream->emit_missing_plugins_id = 0;
	count = g_slist_length (stream->missing_plugins);

	details = g_new0 (char *, count + 1);
	descriptions = g_new0 (char *, count + 1);
	i = 0;
	for (t = stream->missing_plugins; t != NULL; t = t->next) {
		GstMessage *msg = GST_MESSAGE (t->data);
		char *detail;
		char *description;

		detail = gst_missing_plugin_message_get_installer_detail (msg);
		description = gst_missing_plugin_message_get_description (msg);
		details[i] = g_strdup (detail);
		descriptions[i] = g_strdup (description);
		i++;

		gst_message_unref (msg);
	}

	g_signal_emit (stream->player, signals[MISSING_PLUGINS], 0, stream->stream_data, details, descriptions);
	g_strfreev (details);
	g_strfreev (descriptions);

	g_slist_free (stream->missing_plugins);
	stream->missing_plugins = NULL;

	return FALSE;
}


static void
rb_player_gst_xfade_handle_missing_plugin_message (RBPlayerGstXFade *player, RBXFadeStream *stream, GstMessage *message)
{
	if (stream == NULL) {
		rb_debug ("got missing-plugin message from unknown stream");
		return;
	}

	rb_debug ("got missing-plugin message from %s: %s",
		  stream->uri,
		  gst_missing_plugin_message_get_installer_detail (message));

	/* can only handle missing-plugins while prerolling */
	switch (stream->state) {
	case PREROLLING:
	case PREROLL_PLAY:
		stream->missing_plugins = g_slist_prepend (stream->missing_plugins,
							   gst_message_ref (message));
		if (stream->emit_missing_plugins_id == 0) {
			stream->emit_missing_plugins_id =
				g_idle_add ((GSourceFunc) emit_missing_plugins,
					    g_object_ref (stream));
		}

		/* what do we do now?  if we're missing the decoder
		 * or something, it'll never preroll..
		 */
		break;

	default:
		rb_debug ("can't process missing-plugin messages for this stream now");
		break;
	}
}

static void
start_waiting_eos_streams (RBPlayerGstXFade *player)
{
	GList *l;
	GList *to_start = NULL;

	g_rec_mutex_lock (&player->priv->stream_list_lock);
	for (l = player->priv->streams; l != NULL; l = l->next) {
		RBXFadeStream *pstream = l->data;
		if (pstream->state == WAITING_EOS && pstream->starting_eos == FALSE) {
			pstream->starting_eos = TRUE;
			to_start = g_list_prepend (to_start, g_object_ref (pstream));
		}
	}
	g_rec_mutex_unlock (&player->priv->stream_list_lock);

	for (l = to_start; l != NULL; l = l->next) {
		RBXFadeStream *pstream = l->data;
		GError *error = NULL;

		rb_debug ("starting stream %s on EOS from previous", pstream->uri);
		if (link_and_unblock_stream (pstream, &error) == FALSE) {
			emit_stream_error (pstream, error);
		}

		g_object_unref (pstream);
	}
	g_list_free (to_start);
}

/* gstreamer message bus callback */
static gboolean
rb_player_gst_xfade_bus_cb (GstBus *bus, GstMessage *message, RBPlayerGstXFade *player)
{
	RBXFadeStream *stream;

	g_return_val_if_fail (player != NULL, FALSE);

	g_rec_mutex_lock (&player->priv->stream_list_lock);

	stream = find_stream_for_message (player, message);
	g_rec_mutex_unlock (&player->priv->stream_list_lock);

	switch (GST_MESSAGE_TYPE (message)) {
	case GST_MESSAGE_ERROR:
	{
		char *debug;
		GError *error, *sig_error;
		int code;
		gboolean emit = TRUE;

		gst_message_parse_error (message, &error, &debug);

		if (stream == NULL) {
			rb_debug ("Couldn't find stream for error \"%s\": %s", error->message, debug);
			g_error_free (error);
			g_free (debug);
			break;
		}

		/* If we've already got an error, ignore 'internal data flow error'
		 * type messages, as they're too generic to be helpful.
		 */
		if (stream->emitted_error &&
		    error->domain == GST_STREAM_ERROR &&
		    error->code == GST_STREAM_ERROR_FAILED) {
			rb_debug ("Ignoring generic error \"%s\"", error->message);
			emit = FALSE;
		}

		code = rb_gst_error_get_error_code (error);

		if (emit) {
			rb_debug ("emitting error %s for stream %s", error->message, stream->uri);
			sig_error = g_error_new_literal (RB_PLAYER_ERROR,
							 code,
							 error->message);
			stream->emitted_error = TRUE;
			if (stream->emitted_playing == FALSE) {
				_rb_player_emit_playing_stream (RB_PLAYER (player), stream->stream_data);
			}
			_rb_player_emit_error (RB_PLAYER (player), stream->stream_data, sig_error);
		}

		g_error_free (error);
		g_free (debug);
		break;
	}
	case GST_MESSAGE_TAG:
		if (stream == NULL) {
			rb_debug ("got tag message for unknown stream");
		} else {
			GstTagList *tags;
			gst_message_parse_tag (message, &tags);

			g_mutex_lock (&stream->lock);
			if (stream->emitted_playing) {
				gst_tag_list_foreach (tags, (GstTagForeachFunc) process_tag, stream);
				gst_tag_list_free (tags);
			} else {
				stream->tags = g_list_append (stream->tags, tags);
			}
			g_mutex_unlock (&stream->lock);
		}
		break;

	case GST_MESSAGE_DURATION:
		if (stream == NULL) {
			rb_debug ("got duration message for unknown stream");
		} else {
			gint64 duration;
			GstFormat format;
			gst_message_parse_duration (message, &format, &duration);
			rb_debug ("got duration %" G_GINT64_FORMAT
			    " for stream %s", duration, stream->uri);
		}
		break;

	case GST_MESSAGE_APPLICATION:
	{
		/* process fade messages and emit signals for
		 * other stuff.
		 */
		const GstStructure *structure;
		const char *name;

		structure = gst_message_get_structure (message);
		name = gst_structure_get_name (structure);
		if (stream == NULL) {
			rb_debug ("got application message %s for unknown stream", name);
		} else if (strcmp (name, STREAM_PLAYING_MESSAGE) == 0) {
			GList *l;
			GList *t;

			rb_debug ("got stream playing message for %s", stream->uri);
			_rb_player_emit_playing_stream (RB_PLAYER (player), stream->stream_data);

			/* process any buffered tag lists we received while prerolling the stream */
			g_mutex_lock (&stream->lock);
			l = stream->tags;
			stream->tags = NULL;
			g_mutex_unlock (&stream->lock);

			for (t = l; t != NULL; t = t->next) {
				GstTagList *tags;

				tags = (GstTagList *)t->data;
				rb_debug ("processing buffered taglist");
				gst_tag_list_foreach (tags, (GstTagForeachFunc) process_tag, stream);
				gst_tag_list_free (tags);
			}
			g_list_free (l);

		} else if (strcmp (name, FADE_IN_DONE_MESSAGE) == 0) {
			gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (stream->volume), TRUE);
		} else if (strcmp (name, FADE_OUT_DONE_MESSAGE) == 0) {
			switch (stream->state) {
			case FADING_OUT:
				/* stop the stream and dispose of it */
				rb_debug ("got fade-out-done for stream %s -> PENDING_REMOVE", stream->uri);
				g_mutex_lock (&stream->lock);
				stream->state = PENDING_REMOVE;
				g_mutex_unlock (&stream->lock);
				schedule_stream_reap (player);
				break;

			case FADING_OUT_PAUSED:
				{
					/* try to seek back a bit to account for the fade */
					gint64 pos = -1;

					g_mutex_lock (&stream->lock);
					gst_element_query_position (stream->volume, GST_FORMAT_TIME, &pos);
					if (pos != -1) {
						stream->seek_target = pos > PAUSE_FADE_LENGTH ? pos - PAUSE_FADE_LENGTH : 0;
						stream->state = SEEKING_PAUSED;
						rb_debug ("got fade-out-done for stream %s -> SEEKING_PAUSED [%" G_GINT64_FORMAT "]",
							  stream->uri, stream->seek_target);
					} else {
						stream->state = PAUSED;
						rb_debug ("got fade-out-done for stream %s -> PAUSED (position query failed)",
							  stream->uri);
					}
					g_mutex_unlock (&stream->lock);
				}
				unlink_and_block_stream (stream);
				break;

			default:
				g_assert_not_reached ();
			}
		} else if (strcmp (name, STREAM_EOS_MESSAGE) == 0) {
			/* emit EOS (if we aren't already reusing the stream), then unlink it.
			 * the stream stay around so we can seek back in it.
			 */
			stream->needs_unlink = TRUE;
			if (stream->state != REUSING) {
				rb_debug ("got EOS message for stream %s -> PENDING_REMOVE", stream->uri);
				_rb_player_emit_eos (RB_PLAYER (player), stream->stream_data, FALSE);
				stream->state = PENDING_REMOVE;

				unlink_blocked_cb (stream->src_pad, NULL, stream);

				/* start playing any streams that were waiting on an EOS
				 * if they finished preroll between when we posted the EOS
				 * message on the stream thread and now.
				 */
				start_waiting_eos_streams (player);
			} else {
				/* no need to emit EOS here, we already know what to do next */
				rb_debug ("got EOS message for stream %s in REUSING state", stream->uri);

				unlink_reuse_relink (player, stream);
			}

		} else {
			_rb_player_emit_event (RB_PLAYER (player), stream->stream_data, name, NULL);
		}

		break;
	}
	case GST_MESSAGE_BUFFERING:
	{
		const GstStructure *s;
		gint progress;

		s = gst_message_get_structure (message);
		if (!gst_structure_get_int (s, "buffer-percent", &progress)) {
			g_warning ("Could not get value from BUFFERING message");
			break;
		}
		if (stream == NULL) {
			rb_debug ("got buffering message for unknown stream (%d)", progress);
			break;
		}

		if (progress >= 100) {
			GError *error = NULL;
			switch (stream->state) {
			case PREROLLING:
				rb_debug ("stream %s is buffered, now waiting", stream->uri);
				stream->state = WAITING;
				break;

			case WAITING_EOS:
				/* hmm, not sure */
				break;

			case PREROLL_PLAY:
				rb_debug ("stream %s is buffered, now playing", stream->uri);
				if (actually_start_stream (stream, &error) == FALSE) {
					emit_stream_error (stream, error);
				}
				break;

			case PAUSED:
				rb_debug ("stream %s is buffered, leaving paused", stream->uri);
				break;

			default:
				rb_debug ("stream %s is buffered, resuming", stream->uri);
				link_and_unblock_stream (stream, &error);
				if (error) {
					g_warning ("couldn't restart newly buffered stream: %s", error->message);
					g_clear_error (&error);
				}
				break;
			}
		} else {
			switch (stream->state) {
			case PREROLLING:
			case WAITING:
				rb_debug ("still buffering, %d", progress);
				stream->state = PREROLLING;
				break;

			case WAITING_EOS:
				/* not sure */
				break;

			case FADING_OUT:
				rb_debug ("fading out stream is buffering, abandoning it");
				stream->state = PENDING_REMOVE;
				schedule_stream_reap (player);
				/* might need to remove it immediately to avoid stalling adder? */

				/* since we're abandoning this stream, pretend it's not buffering */
				progress = 100;
				break;
			default:
				rb_debug ("stream buffering, stopping playback");
				unlink_and_block_stream (stream);
				break;
			}
		}

		_rb_player_emit_buffering (RB_PLAYER (player), stream->stream_data, progress);
		break;
	}
	case GST_MESSAGE_ELEMENT:
	{
		const GstStructure *s;
		const char *name;

		if (gst_is_missing_plugin_message (message)) {
			rb_player_gst_xfade_handle_missing_plugin_message (player, stream, message);
			break;
		}

		s = gst_message_get_structure (message);
		name = gst_structure_get_name (s);
		if ((strcmp (name, "imperfect-timestamp") == 0) ||
		    (strcmp (name, "imperfect-offset") == 0)) {
			char *details;
			const char *uri = "unknown-stream";

			if (stream != NULL) {
				uri = stream->uri;
			}

			details = gst_structure_to_string (s);
			rb_debug_realf ("check-imperfect", __FILE__, __LINE__, TRUE, "%s: %s", uri, details);
			g_free (details);
		} else if (strcmp (name, "redirect") == 0) {
			const char *uri = gst_structure_get_string (s, "new-location");
			if (stream != NULL) {
				_rb_player_emit_redirect (RB_PLAYER (player), stream->stream_data, uri);
			} else {
				rb_debug ("got redirect to %s, but no active stream found", uri);
			}
		}
		break;
	}
	default:
		break;
	}

	if (stream != NULL)
		g_object_unref (stream);

	/* emit message signals too, so plugins can process bus messages */
	gst_bus_async_signal_func (bus, message, NULL);

	return TRUE;
}

static gboolean
bus_idle_cb (RBPlayerGstXFade *player)
{
	GList *messages, *l;
	GstBus *bus;

	g_rec_mutex_lock (&player->priv->sink_lock);
	messages = player->priv->idle_messages;
	player->priv->idle_messages = NULL;
	player->priv->bus_idle_id = 0;
	g_rec_mutex_unlock (&player->priv->sink_lock);

	bus = gst_element_get_bus (GST_ELEMENT (player->priv->pipeline));
	for (l = messages; l != NULL; l = l->next)
		rb_player_gst_xfade_bus_cb (bus, l->data, player);

	rb_list_destroy_free (messages, (GDestroyNotify) gst_mini_object_unref);
	return FALSE;
}

static void
stream_source_setup_cb (GstElement *decoder, GstElement *source, RBXFadeStream *stream)
{
	rb_debug ("got source notification for stream %s", stream->uri);
	g_signal_emit (stream->player, signals[PREPARE_SOURCE], 0, stream->uri, source);
}

static GstPadProbeReturn
drop_events (GstPad *pad, GstPadProbeInfo *info, gpointer data)
{
	return GST_PAD_PROBE_DROP;
}

static void
add_stream_uri_tag (GstPad *pad, RBXFadeStream *stream)
{
	GstTagList *t;
	GstElement *e;
	GstPad *target;
	GstPad *t2;
	GstPad *sink;
	gulong probe_id;

	t = gst_tag_list_new (STREAM_URI_TAG, stream->uri, NULL);
	gst_tag_list_set_scope (t, GST_TAG_SCOPE_STREAM);

	/* uridecodebin src -> decodebin src */
	t2 = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
	if (GST_IS_GHOST_PAD (t2) == FALSE) {
		/* raw sources get exposed directly */
		rb_debug ("not setting stream uri for raw stream");
		gst_object_unref (t2);
		gst_tag_list_unref (t);
		return;
	}

	/* decodebin src -> actual decoder src */
	target = gst_ghost_pad_get_target (GST_GHOST_PAD (t2));
	probe_id = gst_pad_add_probe (target, GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM, drop_events, NULL, NULL);

	/*
	 * if the decoder is not a GstAudioDecoder, it may send the tag event
	 * directly through its sink pad, which would cause deadlock.  since
	 * there are few examples of decoders that do not use the
	 * GstAudioDecoder base class, and the most notable one (modplug)
	 * doesn't provide interesting tag events, not having stream uri tags
	 * won't be a problem.
	 */

	e = GST_ELEMENT (gst_pad_get_parent (target));
	if (GST_IS_AUDIO_DECODER (e)) {
		sink = gst_element_get_static_pad (e, "sink");
		gst_pad_send_event (sink, gst_event_new_tag (t));
		gst_object_unref (sink);
	} else {
		rb_debug ("not setting stream uri tag for %s", GST_OBJECT_NAME (e));
		gst_tag_list_unref (t);
	}
	gst_object_unref (e);

	gst_pad_remove_probe (target, probe_id);

	gst_object_unref (target);
	gst_object_unref (t2);
}

/* links uridecodebin src pads to the rest of the output pipeline */
static void
stream_pad_added_cb (GstElement *decoder, GstPad *pad, RBXFadeStream *stream)
{
	GstCaps *caps;
	GstStructure *structure;
	const char *mediatype;
	GstPad *vpad;

	/* make sure this is an audio pad */
	caps = gst_pad_query_caps (pad, NULL);
	if (caps == NULL) {
		caps = gst_pad_query_caps (pad, NULL);
	}

	if (gst_caps_is_empty (caps) || gst_caps_is_any (caps)) {
		rb_debug ("got empty/any decoded caps.  hmm?");
		gst_caps_unref (caps);
		return;
	}

	structure = gst_caps_get_structure (caps, 0);
	mediatype = gst_structure_get_name (structure);
	if (g_str_has_prefix (mediatype, "audio/x-raw") == FALSE) {
		rb_debug ("got non-audio decoded caps: %s", mediatype);
	} else if (stream->decoder_linked) {
		/* probably should never happen */
		rb_debug ("hmm, decoder is already linked");
	} else {
		/* push a location tag through the decoder so we can use it to identify the stream later */
		add_stream_uri_tag (pad, stream);

		rb_debug ("got decoded audio pad for stream %s", stream->uri);
		vpad = gst_element_get_static_pad (stream->identity, "sink");
		gst_pad_link (pad, vpad);
		gst_object_unref (vpad);
		stream->decoder_linked = TRUE;

		stream->decoder_pad = gst_object_ref (pad);
	}
	
	gst_caps_unref (caps);
}

static void
stream_pad_removed_cb (GstElement *decoder, GstPad *pad, RBXFadeStream *stream)
{
	if (pad == stream->decoder_pad) {
		rb_debug ("active output pad for stream %s removed", stream->uri);
		stream->decoder_linked = FALSE;

		gst_object_unref (stream->decoder_pad);
		stream->decoder_pad = NULL;
	}
}

/* handles EOS events on stream bins.  since the pipeline as a whole
 * never goes EOS, we don't get an EOS bus message, so we have to handle
 * it here.
 *
 * when an EOS event is received, a bus message is posted, and any streams
 * in the WAITING_EOS state are started.
 *
 * when a new segment event is received, the stream base time is updated
 * (mostly for seeking)
 *
 * flush events are dropped, as they're only relevant inside the stream bin.
 * flushing the adder or the output bin mostly just breaks everything.
 */
static GstPadProbeReturn
stream_src_event_cb (GstPad *pad, GstPadProbeInfo *info, RBXFadeStream *stream)
{
	GstMessage *msg;
	GstEvent *event;
	GstStructure *s;

	event = GST_EVENT (info->data);

	switch (GST_EVENT_TYPE (event)) {
	case GST_EVENT_EOS:
		rb_debug ("posting EOS message for stream %s", stream->uri);
		s = gst_structure_new_empty (STREAM_EOS_MESSAGE);
		msg = gst_message_new_application (GST_OBJECT (stream), s);
		gst_element_post_message (GST_ELEMENT (stream), msg);

		/* start playing any streams that were waiting on an EOS
		 * (are we really allowed to do this on a stream thread?)
		 */
		start_waiting_eos_streams (stream->player);
		break;

	case GST_EVENT_SEGMENT:
		rb_debug ("got new segment for stream %s", stream->uri);
		g_mutex_lock (&stream->lock);
		adjust_stream_base_time (stream);
		g_mutex_unlock (&stream->lock);
		break;

	case GST_EVENT_FLUSH_STOP:
	case GST_EVENT_FLUSH_START:
		rb_debug ("dropping %s event for stream %s", GST_EVENT_TYPE_NAME (event), stream->uri);
		return GST_PAD_PROBE_DROP;

	default:
		rb_debug ("got %s event for stream %s", GST_EVENT_TYPE_NAME (event), stream->uri);
		break;
	}

	return GST_PAD_PROBE_OK;
}

/*
 * stream playback bin:
 *
 * src [ ! queue ] ! decodebin ! audioconvert ! audioresample ! caps ! queue ! volume
 *
 * the first queue is only added for non-local streams.  the thresholds
 * and such are probably going to be configurable at some point,
 * since people seem to get all whiny if they don't have a buffer
 * size slider to play with.
 *
 * the volume element is used for crossfading.
 */
static RBXFadeStream *
create_stream (RBPlayerGstXFade *player, const char *uri, gpointer stream_data, GDestroyNotify stream_data_destroy)
{
	RBXFadeStream *stream;
	GstCaps *caps;
	GArray *stream_filters = NULL;
	GstElement *tail;
	gint i;

	rb_debug ("creating new stream for %s (stream data %p)", uri, stream_data);
	stream = g_object_new (RB_TYPE_XFADE_STREAM, NULL, NULL);
	stream->player = g_object_ref (player);
	stream->stream_data = stream_data;
	stream->stream_data_destroy = stream_data_destroy;
	stream->uri = g_strdup (uri);
	stream->state = WAITING;

	stream->use_buffering = FALSE;
	for (i = 0; i < G_N_ELEMENTS (stream_schemes); i++) {
		if (gst_uri_has_protocol (uri, stream_schemes[i])) {
			stream->use_buffering = TRUE;
			break;
		}
	}

	/* kill the floating reference */
	gst_object_ref_sink (stream);
	gst_element_set_locked_state (GST_ELEMENT (stream), TRUE);
	stream->decoder = gst_element_factory_make ("uridecodebin", NULL);
	if (stream->decoder == NULL) {
		rb_debug ("unable to create uridecodebin");
		g_object_unref (stream);
		return NULL;
	}
	gst_object_ref (stream->decoder);
	g_object_set (stream->decoder, "uri", uri, NULL);

	/* connect uridecodebin to audioconvert when it creates its output pad */
	g_signal_connect_object (stream->decoder,
				 "source-setup",
				 G_CALLBACK (stream_source_setup_cb),
				 stream,
				 0);
	g_signal_connect_object (stream->decoder,
				 "pad-added",
				 G_CALLBACK (stream_pad_added_cb),
				 stream,
				 0);
	g_signal_connect_object (stream->decoder,
				 "pad-removed",
				 G_CALLBACK (stream_pad_removed_cb),
				 stream,
				 0);

	stream->identity = gst_element_factory_make ("identity", NULL);
	if (stream->identity == NULL) {
		rb_debug ("unable to create identity");
		g_object_unref (stream);
		return NULL;
	}

	stream->audioconvert = gst_element_factory_make ("audioconvert", NULL);
	if (stream->audioconvert == NULL) {
		rb_debug ("unable to create audio converter");
		g_object_unref (stream);
		return NULL;
	}
	gst_object_ref (stream->audioconvert);

	stream->audioresample = gst_element_factory_make ("audioresample", NULL);
	if (stream->audioresample == NULL) {
		rb_debug ("unable to create audio resample");
		g_object_unref (stream);
		return NULL;
	}
	gst_object_ref (stream->audioresample);

	stream->capsfilter = gst_element_factory_make ("capsfilter", NULL);
	if (stream->capsfilter == NULL) {
		rb_debug ("unable to create capsfilter");
		g_object_unref (stream);
		return NULL;
	}
	gst_object_ref (stream->capsfilter);

	caps = gst_caps_new_simple ("audio/x-raw",
				    "format",   G_TYPE_STRING, "S16LE",			/* appropriate? */
				    "channels", G_TYPE_INT, 2,
				    "layout",	G_TYPE_STRING, "interleaved",
				    "rate",	G_TYPE_INT, 44100,
				    NULL);
	g_object_set (stream->capsfilter, "caps", caps, NULL);
	gst_caps_unref (caps);

	stream->volume = gst_element_factory_make ("volume", NULL);
	if (stream->volume == NULL) {
		rb_debug ("unable to create volume control");
		g_object_unref (stream);
		return NULL;
	}
	gst_object_ref (stream->volume);

	g_signal_connect_object (stream->volume,
				 "notify::volume",
				 G_CALLBACK (volume_changed_cb),
				 player, 0);

	stream->fader = GST_TIMED_VALUE_CONTROL_SOURCE (gst_interpolation_control_source_new ());
	gst_timed_value_control_source_set (GST_TIMED_VALUE_CONTROL_SOURCE (stream->fader), 0, 0.1);
	g_object_set (stream->fader, "mode", GST_INTERPOLATION_MODE_LINEAR, NULL);

	gst_object_add_control_binding (GST_OBJECT (stream->volume),
					gst_direct_control_binding_new (GST_OBJECT (stream->volume), "volume", GST_CONTROL_SOURCE (stream->fader)));

	stream->preroll = gst_element_factory_make ("queue", NULL);
	if (stream->preroll == NULL) {
		rb_debug ("unable to create preroll queue");
		g_object_unref (stream);
		return NULL;
	}
	/* decode at least a second during prerolling, to hopefully avoid underruns.
	 * we clear this when prerolling is finished.  bump the max buffer count up
	 * a bit (from 200) as with some formats it often takes more buffers to
	 * make up a whole second.  don't really want to remove it altogether, though.
	 */
	g_object_set (stream->preroll,
		      "min-threshold-time", GST_SECOND,
		      "max-size-buffers", 1000,
		      NULL);

	gst_bin_add_many (GST_BIN (stream),
			  stream->decoder,
			  stream->identity,
			  stream->audioconvert,
			  stream->audioresample,
			  stream->capsfilter,
			  stream->preroll,
			  stream->volume,
			  NULL);
	gst_element_link_many (stream->audioconvert,
			       stream->audioresample,
			       stream->capsfilter,
			       stream->preroll,
			       stream->volume,
			       NULL);

	if (rb_debug_matches ("check-imperfect", __FILE__)) {

		if (rb_debug_matches ("check-imperfect-timestamp", __FILE__)) {
			g_object_set (stream->identity, "check-imperfect-timestamp", TRUE, NULL);
		}
		if (rb_debug_matches ("check-imperfect-offset", __FILE__)) {
			g_object_set (stream->identity, "check-imperfect-offset", TRUE, NULL);
		}
	}
	stream->src_pad = gst_element_get_static_pad (stream->volume, "src");

	/* link in any per-stream filters after the identity element, with an
	 * audioconvert before each.
	 */
	tail = stream->identity;
	g_signal_emit (player, signals[GET_STREAM_FILTERS], 0, uri, &stream_filters);
	if (stream_filters != NULL) {
		int i;
		for (i = 0; i < stream_filters->len; i++) {
			GValue *v = &g_array_index (stream_filters, GValue, i);
			GstElement *filter;
			GstElement *audioconvert;

			audioconvert = gst_element_factory_make ("audioconvert", NULL);
			filter = GST_ELEMENT (g_value_get_object (v));

			gst_bin_add_many (GST_BIN (stream), audioconvert, filter, NULL);
			gst_element_link_many (tail, audioconvert, filter, NULL);
			tail = filter;
		}

		g_array_unref (stream_filters);
	}
	gst_element_link (tail, stream->audioconvert);

	/* ghost the stream src pad up to the bin */
	stream->ghost_pad = gst_ghost_pad_new ("src", stream->src_pad);
	gst_element_add_pad (GST_ELEMENT (stream), stream->ghost_pad);

	/* watch for EOS events using a pad probe */
	gst_pad_add_probe (stream->src_pad,
			   GST_PAD_PROBE_TYPE_EVENT_BOTH | GST_PAD_PROBE_TYPE_EVENT_FLUSH,
			   (GstPadProbeCallback) stream_src_event_cb,
			   stream,
			   NULL);

	/* use the pipeline bus even when not inside the pipeline (?) */
	gst_element_set_bus (GST_ELEMENT (stream), gst_element_get_bus (player->priv->pipeline));

	return stream;
}

/* starts playback for a stream.
 * - links to adder and unblocks
 * - if play_type is CROSSFADE:
 *   - starts the fade in of the new stream
 *   - starts the fade out of the old stream
 *   - sets the stream to PLAYING state
 * - if play_type is WAIT_EOS:
 *   - if something is playing, set the stream to wait-eos state
 *   - otherwise, starts it
 * - if play_type is REPLACE:
 *   - stops any existing stream
 *   - starts the new stream
 */
static gboolean
actually_start_stream (RBXFadeStream *stream, GError **error)
{
	RBPlayerGstXFade *player = stream->player;
	gboolean ret = TRUE;
	gboolean need_reap = FALSE;
	gboolean playing;
	GList *l;
	GList *to_fade;

	rb_debug ("going to start playback for stream %s (play type %d, crossfade %" G_GINT64_FORMAT ") -> FADING_IN | PLAYING", stream->uri, stream->play_type, stream->crossfade);
	switch (stream->play_type) {
	case RB_PLAYER_PLAY_CROSSFADE:

		to_fade = NULL;
		g_rec_mutex_lock (&player->priv->stream_list_lock);
		for (l = player->priv->streams; l != NULL; l = l->next) {
			RBXFadeStream *pstream = (RBXFadeStream *)l->data;

			if (pstream == stream)
				continue;

			switch (pstream->state) {
			case FADING_IN:
			case PLAYING:
				rb_debug ("stream %s is playing; crossfading -> FADING_OUT", pstream->uri);
				to_fade = g_list_prepend (to_fade, g_object_ref (pstream));
				break;

			case PAUSED:
			case WAITING_EOS:
			case SEEKING:
			case SEEKING_PAUSED:
			case PREROLLING:
			case PREROLL_PLAY:
				rb_debug ("stream %s is paused; replacing it", pstream->uri);
				pstream->state = PENDING_REMOVE;
				/* fall through */
			case PENDING_REMOVE:
				need_reap = TRUE;
				break;

			default:
				break;
			}
		}

		g_rec_mutex_unlock (&player->priv->stream_list_lock);

		for (l = to_fade; l != NULL; l = l->next) {
			RBXFadeStream *pstream = (RBXFadeStream *)l->data;
			double fade_out_start = 1.0f;
			gint64 fade_out_time = stream->crossfade;

			switch (pstream->state) {
			case FADING_IN:
				/* fade out from where the fade in got up to */
				g_object_get (pstream->volume, "volume", &fade_out_start, NULL);
				fade_out_time = (gint64)(((double) stream->crossfade) * fade_out_start);
				/* fall through */

			case PLAYING:
				start_stream_fade (pstream, fade_out_start, 0.0f, fade_out_time);
				pstream->state = FADING_OUT;

				start_stream_fade (stream, 0.0f, 1.0f, stream->crossfade);
				break;

			default:
				/* shouldn't happen, but ignore it if it does */
				break;
			}

			g_object_unref (pstream);
		}
		g_list_free (to_fade);

		if (stream->fading == FALSE) {
			rb_debug ("stream isn't fading; setting volume to 1.0");
			gst_timed_value_control_source_set (GST_TIMED_VALUE_CONTROL_SOURCE (stream->fader), 0, 0.1);
			gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (stream->volume), TRUE);
		}

		ret = link_and_unblock_stream (stream, error);
		break;

	case RB_PLAYER_PLAY_AFTER_EOS:

		g_rec_mutex_lock (&player->priv->stream_list_lock);

		playing = FALSE;
		for (l = player->priv->streams; l != NULL; l = l->next) {
			RBXFadeStream *pstream = (RBXFadeStream *)l->data;
			if (pstream == stream)
				continue;

			switch (pstream->state) {
			case PLAYING:
			case FADING_IN:
			case FADING_OUT:
				rb_debug ("stream %s is already playing", pstream->uri);
				playing = TRUE;
				break;
			case PAUSED:
				rb_debug ("stream %s is paused; replacing it", pstream->uri);
				pstream->state = PENDING_REMOVE;
			case PENDING_REMOVE:
				need_reap = TRUE;
				break;
			default:
				break;
			}
		}

		g_rec_mutex_unlock (&player->priv->stream_list_lock);
	
		if (playing) {
			/* wait for current stream's EOS */
			rb_debug ("existing playing stream found; waiting for its EOS -> WAITING_EOS");
			stream->state = WAITING_EOS;
			stream->starting_eos = FALSE;
		} else {
			rb_debug ("no playing stream found, so starting immediately");
			ret = link_and_unblock_stream (stream, error);
		}
		break;

	case RB_PLAYER_PLAY_REPLACE:
		/* replace any existing playing stream */
		g_rec_mutex_lock (&player->priv->stream_list_lock);

		for (l = player->priv->streams; l != NULL; l = l->next) {
			RBXFadeStream *pstream = (RBXFadeStream *)l->data;
			if (pstream == stream)
				continue;

			switch (pstream->state) {
			case PLAYING:
			case PAUSED:
			case FADING_IN:
			case PENDING_REMOVE:
				/* kill this one */
				rb_debug ("stopping stream %s (replaced by new stream)", pstream->uri);
				need_reap = TRUE;
				pstream->state = PENDING_REMOVE;
				break;

			default:
				/* let it go */
				break;
			}
		}

		g_rec_mutex_unlock (&player->priv->stream_list_lock);

		ret = link_and_unblock_stream (stream, error);
		break;

	default:
		g_assert_not_reached ();
	}

	if (need_reap) {
		schedule_stream_reap (player);
	}

	return ret;
}

/* called on a streaming thread when the stream src pad is blocked
 * (that is, when prerolling is complete).  in some situations we
 * start playback immediately, otherwise we wait for something else
 * to happen.
 */
static GstPadProbeReturn
stream_src_blocked_cb (GstPad *pad, GstPadProbeInfo *info, RBXFadeStream *stream)
{
	GError *error = NULL;
	gboolean start_stream = FALSE;
	GstElement *src;
	GstQuery *query;

	g_mutex_lock (&stream->lock);
	if (stream->src_blocked) {
		/*rb_debug ("stream %s already blocked", stream->uri);*/
		g_mutex_unlock (&stream->lock);
		return GST_PAD_PROBE_OK;
	}
	stream->src_blocked = TRUE;

	g_object_set (stream->preroll,
		      "min-threshold-time", G_GINT64_CONSTANT (0),
		      "max-size-buffers", 200,		/* back to normal value */
		      NULL);

	g_object_get (stream->decoder, "source", &src, NULL);
	query = gst_query_new_scheduling ();
	if (gst_element_query (src, query)) {
		GstSchedulingFlags flags;
		gst_query_parse_scheduling (query, &flags, NULL, NULL, NULL);

		/* this matches how uridecodebin decides whether to do buffering */
		stream->use_buffering = (flags & GST_SCHEDULING_FLAG_BANDWIDTH_LIMITED);
	}
	gst_query_unref (query);
	g_object_unref (src);

	if (stream->use_buffering) {
		rb_debug ("stream %s requires buffering", stream->uri);
		switch (stream->state) {
		case PREROLL_PLAY:
			post_stream_playing_message (stream, TRUE);
			break;
		default:
			break;
		}
		g_mutex_unlock (&stream->lock);
		return GST_PAD_PROBE_OK;
	}

	/* update stream state */
	switch (stream->state) {
	case PREROLLING:
		rb_debug ("stream %s is prerolled, not starting yet -> WAITING", stream->uri);
		stream->state = WAITING;
		break;
	case PREROLL_PLAY:
		rb_debug ("stream %s is prerolled, need to start it", stream->uri);
		start_stream = TRUE;
		break;
	default:
		rb_debug ("didn't expect to get preroll completion callback in this state (%d)", stream->state);
		break;
	}
	
	g_mutex_unlock (&stream->lock);
	
	if (start_stream == TRUE) {	
		/* not sure this is actually an acceptable thing to do on a streaming thread.. */
		if (actually_start_stream (stream, &error) == FALSE) {
			emit_stream_error (stream, error);
		}
	}

	return GST_PAD_PROBE_OK;
}

/*
 * starts prerolling for a stream.
 * since the stream isn't linked to anything yet, we
 * block the src pad.  when the pad block callback
 * is called, prerolling is complete and the stream
 * can be linked and played immediately if required.
 *
 * must be called *without* the stream list lock?
 */
static void
preroll_stream (RBPlayerGstXFade *player, RBXFadeStream *stream)
{
	GstStateChangeReturn state;
	GstMessage *message;
	GList *messages;
	GstBus *bus;

	stream->block_probe_id =
		gst_pad_add_probe (stream->src_pad,
				   GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM,
				   (GstPadProbeCallback) stream_src_blocked_cb,
				   stream,
				   NULL);

	stream->emitted_playing = FALSE;
	stream->state = PREROLLING;
	state = gst_element_set_state (GST_ELEMENT (stream), GST_STATE_PAUSED);
	switch (state) {
	case GST_STATE_CHANGE_FAILURE:
		rb_debug ("preroll for stream %s failed (state change failed)", stream->uri);

		/* process messages in an idle handler in case we got a redirect */
		bus = gst_element_get_bus (GST_ELEMENT (player->priv->pipeline));
		messages = NULL;
		message = gst_bus_pop (bus);
		while (message != NULL) {
			messages = g_list_prepend (messages, message);
			message = gst_bus_pop (bus);
		}
		g_object_unref (bus);

		g_rec_mutex_lock (&player->priv->sink_lock);
		player->priv->idle_messages = g_list_concat (player->priv->idle_messages, g_list_reverse (messages));
		if (player->priv->bus_idle_id == 0)
			player->priv->bus_idle_id = g_idle_add ((GSourceFunc) bus_idle_cb, player);
		g_rec_mutex_unlock (&player->priv->sink_lock);
		break;

	case GST_STATE_CHANGE_NO_PREROLL:
		rb_debug ("no preroll for stream %s, setting to PLAYING instead?", stream->uri);
		gst_element_set_state (GST_ELEMENT (stream), GST_STATE_PLAYING);
		break;
	case GST_STATE_CHANGE_SUCCESS:
	case GST_STATE_CHANGE_ASYNC:
		/* uridecodebin returns SUCCESS from state changes when streaming, so we can't
		 * use that to figure out what to do next.  instead, we wait for pads to be added
		 * and for our pad block callbacks to be called.
		 */
		break;
	default:
		g_assert_not_reached();
	}
}

/*
 * returns the RBXFadeStream, playback position, and duration of the current
 * playing stream.
 */
static gboolean
get_times_and_stream (RBPlayerGstXFade *player, RBXFadeStream **pstream, gint64 *pos, gint64 *duration)
{
	gboolean got_time = FALSE;
	gboolean buffering = FALSE;
	RBXFadeStream *stream;

	if (player->priv->pipeline == NULL)
		return FALSE;

	g_rec_mutex_lock (&player->priv->stream_list_lock);
	
	/* first look for a network stream that is buffering during preroll */
	stream = find_stream_by_state (player, PREROLLING | PREROLL_PLAY);
	if (stream != NULL) {
		if (stream->emitted_fake_playing == FALSE) {
			g_object_unref (stream);
			stream = NULL;
		} else {
			rb_debug ("found buffering stream %s as current", stream->uri);
			buffering = TRUE;
		}
	}

	/* otherwise, the stream that is playing */
	if (stream == NULL) {
		stream = find_stream_by_state (player, FADING_IN | PLAYING | FADING_OUT_PAUSED | PAUSED | PENDING_REMOVE | REUSING);
	}
	g_rec_mutex_unlock (&player->priv->stream_list_lock);

	if (stream != NULL) {
		if (pstream != NULL) {
			*pstream = stream;
		}

		if (pos != NULL) {
			if (buffering) {
				*pos = 0;
			} else if (stream->state == PAUSED) {
				*pos = -1;

				gst_element_query_position (stream->volume, GST_FORMAT_TIME, pos);
			} else {
				/* for playing streams, we subtract the current output position
				 * (a running counter generated by the adder) from the position
				 * at which we started playback.
				 */
				*pos = -1;
				gst_element_query_position (player->priv->pipeline, GST_FORMAT_TIME, pos);
				if (*pos != -1) {
					*pos -= stream->base_time;
				} else {
					rb_debug ("position query failed");
				}
			}
		}

		if (duration != NULL) {
			*duration = -1;
			/* queries are supposed to go to sinks, but the closest thing we
			 * have in the stream bin is the volume element, which is the last
			 * linked element.
			 */
			gst_element_query_duration (stream->volume, GST_FORMAT_TIME, duration);
		}
		got_time = TRUE;
		if (pstream == NULL) {
			g_object_unref (stream);
		}
	} else {
		rb_debug ("not playing");
	}

	return got_time;
}

static gboolean
tick_timeout (RBPlayerGstXFade *player)
{
	gint64 pos = -1;
	gint64 duration = -1;
	RBXFadeStream *stream = NULL;

	if (get_times_and_stream (player, &stream, &pos, &duration)) {
		_rb_player_emit_tick (RB_PLAYER (player), stream->stream_data, pos, duration);
		g_object_unref (stream);
	}

	return TRUE;
}

static gboolean
emit_volume_changed_idle (RBPlayerGstXFade *player)
{
	double vol;

	if (GST_IS_STREAM_VOLUME (player->priv->volume_handler)) {
		vol = gst_stream_volume_get_volume (GST_STREAM_VOLUME (player->priv->volume_handler),
						    GST_STREAM_VOLUME_FORMAT_CUBIC);
	} else {
		vol = player->priv->cur_volume;
	}

	_rb_player_emit_volume_changed (RB_PLAYER (player), vol);
	return FALSE;
}

static void
stream_volume_changed (GObject *element, GParamSpec *pspec, RBPlayerGstXFade *player)
{
	double v;

	g_object_get (element, "volume", &v, NULL);
	player->priv->cur_volume = v;

	g_idle_add ((GSourceFunc) emit_volume_changed_idle, player);
}

/*
 * output sink + adder pipeline:
 *
 * outputcaps = audio/x-raw,channels=2,rate=44100,format=S16LE
 * outputbin = outputcaps ! volume ! filterbin ! audioconvert ! audioresample ! tee ! queue ! audiosink
 * silencebin = appsrc ! outputcaps
 *
 * pipeline = silencebin ! adder ! outputbin
 *
 * the tee in output bin has branches attached to it using the
 * RBPlayerGstTee interface.  filterbin contains elements inserted
 * using the RBPlayerGstFilter interface.
 *
 * silencebin is there for two reasons:
 * - lets us start the sink without having any streams playing
 * - probably helps keep things from falling over between streams
 */

static void
add_bus_watch (RBPlayerGstXFade *player)
{
	GstBus *bus;

	bus = gst_element_get_bus (GST_ELEMENT (player->priv->pipeline));
	player->priv->bus_watch_id = gst_bus_add_watch (bus, (GstBusFunc) rb_player_gst_xfade_bus_cb, player);
	gst_object_unref (bus);
}

static gboolean
start_sink_locked (RBPlayerGstXFade *player, GList **messages, GError **error)
{
	GstStateChangeReturn sr;
	gboolean waiting;
	GError *generic_error = NULL;
	GstBus *bus;

	g_set_error (&generic_error,
		     RB_PLAYER_ERROR,
		     RB_PLAYER_ERROR_INTERNAL,		/* ? */
		     _("Failed to open output device"));

	rb_debug ("starting sink");

	/* first, start the output bin.
	 * this won't preroll until we start the silence bin.
	 */
	sr = gst_element_set_state (player->priv->outputbin, GST_STATE_PAUSED);
	if (sr == GST_STATE_CHANGE_FAILURE) {
		rb_debug ("output bin state change failed");
		g_propagate_error (error, generic_error);
		return FALSE;
	}

	/* then the adder */
	sr = gst_element_set_state (player->priv->adder, GST_STATE_PAUSED);
	if (sr == GST_STATE_CHANGE_FAILURE) {
		rb_debug ("adder state change failed");
		g_propagate_error (error, generic_error);
		return FALSE;
	}

	/* then the silence bin */
	sr = gst_element_set_state (player->priv->silencebin, GST_STATE_PAUSED);
	if (sr == GST_STATE_CHANGE_FAILURE) {
		rb_debug ("silence bin state change failed");
		g_propagate_error (error, generic_error);
		return FALSE;
	}

	/* give the silence bin some data so it can preroll */
	silencesrc_push (player);

	/* now wait for everything to finish */
	waiting = TRUE;
	bus = gst_element_get_bus (GST_ELEMENT (player->priv->pipeline));
	while (waiting) {
		GstMessage *message;
		GstState oldstate;
		GstState newstate;
		GstState pending;

		/*
		 * when a second message is posted immediately after the first,
		 * this loop finishes processing the first at roughly the time the
		 * second is posted, so occasionally we'll hit this bug:
		 * https://bugzilla.gnome.org/show_bug.cgi?id=750397
		 * sleeping for 10us makes the read loop slow enough to avoid the
		 * race with state-changed and async-done messages, but hopefully
		 * not so much slower that we run into it in other conditions.
		 */
		g_usleep (10);

		message = gst_bus_timed_pop (bus, GST_SECOND * 5);
		if (message == NULL) {
			rb_debug ("sink is taking too long to start..");
			g_propagate_error (error, generic_error);
			gst_object_unref (bus);
			return FALSE;
		}

		switch (GST_MESSAGE_TYPE (message)) {
		case GST_MESSAGE_ERROR:
			{
				char *debug;
				GError *gst_error = NULL;
				RBXFadeStream *stream;

				/* we only want to process errors from the sink here.
				 * errors from streams should go to the normal message handler.
				 */
				stream = find_stream_for_message (player, message);
				if (stream != NULL) {
					rb_debug ("got an error from a stream; passing it to the bus handler");
					*messages = g_list_append (*messages, gst_message_ref (message));
					g_object_unref (stream);
				} else {
					gst_message_parse_error (message, &gst_error, &debug);
					rb_debug ("got error message: %s (%s)", gst_error->message, debug);
					gst_message_unref (message);
					g_free (debug);

					if (error != NULL && *error == NULL) {
						/* Translators: the parameter here is an error message */
						g_set_error (error,
							     RB_PLAYER_ERROR,
							     RB_PLAYER_ERROR_INTERNAL,		/* ? */
							     _("Failed to open output device: %s"),
							     gst_error->message);
					}
					g_error_free (gst_error);
					g_error_free (generic_error);

					gst_element_set_state (player->priv->outputbin, GST_STATE_NULL);
					gst_element_set_state (player->priv->adder, GST_STATE_NULL);
					gst_element_set_state (player->priv->silencebin, GST_STATE_NULL);
					gst_object_unref (bus);
					return FALSE;
				}
			}
			break;

		case GST_MESSAGE_STATE_CHANGED:
			{
				gst_message_parse_state_changed (message, &oldstate, &newstate, &pending);
				if (newstate == GST_STATE_PAUSED && pending == GST_STATE_VOID_PENDING) {
					if (GST_MESSAGE_SRC (message) == GST_OBJECT (player->priv->outputbin)) {
						rb_debug ("outputbin is now PAUSED");
						waiting = FALSE;
					} else if (GST_MESSAGE_SRC (message) == GST_OBJECT (player->priv->adder)) {
						rb_debug ("adder is now PAUSED");
					} else if (GST_MESSAGE_SRC (message) == GST_OBJECT (player->priv->silencebin)) {
						rb_debug ("silencebin is now PAUSED");
					}
				}
			}
			break;

		default:
			/* save the message to pass to the bus callback once we've dropped
			 * the sink lock.
			 */
			*messages = g_list_append (*messages, gst_message_ref (message));
			break;
		}

		gst_message_unref (message);
	}
	gst_object_unref (bus);

	/* if the sink provides a 'volume' property, use that to control output volume */
	player->priv->volume_handler = rb_player_gst_find_element_with_property (player->priv->sink, "volume");
	if (player->priv->volume_handler == NULL) {
		rb_debug ("sink doesn't provide volume control, using volume element");
		player->priv->volume_handler = g_object_ref (player->priv->volume);
	} else if (player->priv->volume_applied == 0) {
		/* ignore the initial volume setting, allowing the
		 * sink to restore its own volume.
		 */
		player->priv->volume_applied = 1;
	}

	/* if there has been a volume change that we haven't applied, apply it now */
	if (player->priv->volume_applied < player->priv->volume_changed) {
		g_object_set (player->priv->volume_handler, "volume", player->priv->cur_volume, NULL);
		player->priv->volume_applied = player->priv->volume_changed;
	}

	g_signal_connect_object (player->priv->volume_handler,
				 "notify::volume",
				 G_CALLBACK (stream_volume_changed),
				 player, 0);


	sr = gst_element_set_state (player->priv->silencebin, GST_STATE_PLAYING);
	if (sr == GST_STATE_CHANGE_FAILURE) {
		rb_debug ("silence bin state change failed");
		g_propagate_error (error, generic_error);
		return FALSE;
	}

	sr = gst_element_set_state (player->priv->adder, GST_STATE_PLAYING);
	if (sr == GST_STATE_CHANGE_FAILURE) {
		rb_debug ("adder state change failed");
		g_propagate_error (error, generic_error);
		return FALSE;
	}

	sr = gst_element_set_state (player->priv->outputbin, GST_STATE_PLAYING);
	if (sr == GST_STATE_CHANGE_FAILURE) {
		rb_debug ("output bin state change failed");
		g_propagate_error (error, generic_error);
		return FALSE;
	}

	rb_debug ("sink playing");
	player->priv->sink_state = SINK_PLAYING;

	/* set the pipeline to PLAYING so it selects a clock */
	gst_element_set_state (player->priv->pipeline, GST_STATE_PLAYING);

	/* now that the sink is running, start polling for playing position.
	 * might want to replace this with a complicated set of pad probes
	 * to avoid polling, but duration queries on the sink are better
	 * as they account for internal buffering etc.  maybe there's a way
	 * to account for that in a pad probe callback on the sink's sink pad?
	 */
	if (player->priv->tick_timeout_id == 0) {
		gint ms_period = 1000 / RB_PLAYER_GST_XFADE_TICK_HZ;
		player->priv->tick_timeout_id =
			g_timeout_add (ms_period,
				      (GSourceFunc) tick_timeout,
				      player);
	}
	return TRUE;
}

static gboolean
start_sink (RBPlayerGstXFade *player, GError **error)
{
	GList *messages = NULL;
	gboolean ret;

	g_rec_mutex_lock (&player->priv->sink_lock);
	switch (player->priv->sink_state) {
	case SINK_NULL:
		g_assert_not_reached ();
		break;

	case SINK_STOPPED:
		/* prevent messages from being processed by the main thread while we're starting the sink */
		g_source_remove (player->priv->bus_watch_id);
		ret = start_sink_locked (player, &messages, error);

		player->priv->idle_messages = g_list_concat (player->priv->idle_messages, messages);
		if (player->priv->bus_idle_id == 0)
			player->priv->bus_idle_id = g_idle_add ((GSourceFunc) bus_idle_cb, player);
		add_bus_watch (player);
		break;

	case SINK_PLAYING:
		ret = TRUE;
		break;

	default:
		g_assert_not_reached ();
	}
	g_rec_mutex_unlock (&player->priv->sink_lock);

	return ret;
}

static gboolean
stop_sink (RBPlayerGstXFade *player)
{
#if 0
	GstStateChangeReturn sr;

	switch (player->priv->sink_state) {
	case SINK_PLAYING:
		rb_debug ("stopping sink");

		if (player->priv->tick_timeout_id != 0) {
			g_source_remove (player->priv->tick_timeout_id);
			player->priv->tick_timeout_id = 0;
		}

		sr = gst_element_set_state (player->priv->outputbin, GST_STATE_READY);
		if (sr == GST_STATE_CHANGE_FAILURE) {
			rb_debug ("couldn't stop output bin");
			return FALSE;
		}

		sr = gst_element_set_state (player->priv->adder, GST_STATE_READY);
		if (sr == GST_STATE_CHANGE_FAILURE) {
			rb_debug ("couldn't stop adder");
			return FALSE;
		}

		sr = gst_element_set_state (player->priv->silencebin, GST_STATE_READY);
		if (sr == GST_STATE_CHANGE_FAILURE) {
			rb_debug ("couldn't stop silence bin");
			return FALSE;
		}

		/* try stopping the sink, but don't worry if we can't */
		sr = gst_element_set_state (player->priv->sink, GST_STATE_NULL);
		if (sr == GST_STATE_CHANGE_FAILURE) {
			rb_debug ("couldn't set audio sink to NULL state");
		}

		if (player->priv->volume_handler) {
			g_object_unref (player->priv->volume_handler);
			player->priv->volume_handler = NULL;
		}

		/* set the pipeline to READY so we can make it select a clock when we
		 * start the sink again.  everything inside the pipeline has its state
		 * locked, so this doesn't affect anything else.
		 */
		gst_element_set_state (player->priv->pipeline, GST_STATE_READY);

		player->priv->sink_state = SINK_STOPPED;
		break;

	case SINK_STOPPED:
	case SINK_NULL:
		break;
	}

	return TRUE;
#endif
	return TRUE;
}

static void
silencesrc_free_buffer(gpointer d)
{
}

static gboolean
silencesrc_push (RBPlayerGstXFade *player)
{
	GstBuffer *buffer;
	GstFlowReturn ret;

	buffer = gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY,
					      player->priv->silence_buffer,
					      sizeof(player->priv->silence_buffer),
					      0,
					      sizeof(player->priv->silence_buffer),
					      silencesrc_free_buffer,
					      NULL);
	g_signal_emit_by_name (player->priv->silencesrc, "push-buffer", buffer, &ret);
	gst_buffer_unref (buffer);

	return (ret == GST_FLOW_OK);
}

static void
silencesrc_need_data_cb (GstElement *appsrc, guint size, RBPlayerGstXFade *player)
{
	silencesrc_push (player);
}

static gboolean
create_sink (RBPlayerGstXFade *player, GError **error)
{
	const char *try_sinks[] = { "gsettingsaudiosink", "gconfaudiosink", "autoaudiosink" };
	GstElement *audioconvert;
	GstElement *audioresample;
	GstElement *capsfilter;
	GstElement *queue;
	GstCaps *caps;
	GstPad *filterpad;
	GstPad *outputghostpad;
	GstPad *ghostpad;
	GstPad *reqpad;
	GstPad *addersrcpad;
	GstPadLinkReturn plr;
	GList *l;
	int i;

	if (player->priv->sink_state != SINK_NULL)
		return TRUE;

	/* set filter caps.
	 * 44100Hz is about the most reasonable thing to use;
	 * we have audioconvert+audioresample afterwards in
	 * case the output device doesn't actually support
	 * that rate.
	 */
	caps = gst_caps_new_simple ("audio/x-raw",
				    "format" ,  G_TYPE_STRING, "S16LE",
				    "channels", G_TYPE_INT, 2,
				    "rate",	G_TYPE_INT, 44100,
				    "layout",	G_TYPE_STRING, "interleaved",
				    NULL);

	player->priv->pipeline = gst_pipeline_new ("rbplayer");
	add_bus_watch (player);
	g_object_notify (G_OBJECT (player), "bus");

	player->priv->outputbin = gst_bin_new ("outputbin");
	gst_element_set_locked_state (player->priv->outputbin, TRUE);
	player->priv->adder = gst_element_factory_make ("adder", "outputadder");
	player->priv->capsfilter = gst_element_factory_make ("capsfilter", "outputcapsfilter");
	audioconvert = gst_element_factory_make ("audioconvert", "outputconvert");
	audioresample = gst_element_factory_make ("audioresample", "outputresample");
	player->priv->tee = gst_element_factory_make ("tee", "outputtee");
	queue = gst_element_factory_make ("queue", NULL);
	player->priv->volume = gst_element_factory_make ("volume", "outputvolume");
	player->priv->filterbin = rb_gst_create_filter_bin ();
	capsfilter = gst_element_factory_make ("capsfilter", NULL);
	if (player->priv->pipeline == NULL ||
	    player->priv->adder == NULL ||
	    player->priv->capsfilter == NULL ||
	    audioconvert == NULL ||
	    audioresample == NULL ||
	    player->priv->tee == NULL ||
	    queue == NULL ||
	    player->priv->volume == NULL ||
	    player->priv->filterbin == NULL ||
	    capsfilter == NULL) {
		/* we could include the element name in the error message,
		 * but these are all fundamental elements that are always
		 * available.
		 */
		g_set_error (error,
			     RB_PLAYER_ERROR,
			     RB_PLAYER_ERROR_GENERAL,
			     _("Failed to create GStreamer element; check your installation"));
		return FALSE;
	}

	for (i = 0; i < G_N_ELEMENTS (try_sinks); i++) {
		player->priv->sink = rb_player_gst_try_audio_sink (try_sinks[i], NULL);
		if (player->priv->sink != NULL) {
			break;
		}
	}
	if (player->priv->sink == NULL) {
		g_set_error (error,
			     RB_PLAYER_ERROR,
			     RB_PLAYER_ERROR_GENERAL,
			     _("Failed to create audio output element; check your installation"));
		return FALSE;
	}

	g_object_set (player->priv->capsfilter, "caps", caps, NULL);
	g_object_set (capsfilter, "caps", caps, NULL);

	g_object_set (queue, "max-size-buffers", 10, NULL);

	gst_bin_add_many (GST_BIN (player->priv->outputbin),
			  player->priv->capsfilter,
			  player->priv->volume,
			  player->priv->filterbin,
			  audioconvert,
			  audioresample,
			  capsfilter,
			  player->priv->tee,
			  queue,
			  player->priv->sink,
			  NULL);
	if (gst_element_link_many (player->priv->capsfilter,
			       player->priv->volume,
			       player->priv->filterbin,
			       audioconvert,
			       audioresample,
			       capsfilter,
			       player->priv->tee,
			       queue,
			       player->priv->sink,
			       NULL) == FALSE) {
		g_warning ("minus one");
		g_set_error (error,
			     RB_PLAYER_ERROR,
			     RB_PLAYER_ERROR_GENERAL,
			     _("Failed to link GStreamer pipeline; check your installation"));
		return FALSE;
	}

	filterpad = gst_element_get_static_pad (player->priv->capsfilter, "sink");
	outputghostpad = gst_ghost_pad_new ("sink", filterpad);
	gst_element_add_pad (player->priv->outputbin, outputghostpad);
	gst_object_unref (filterpad);

	/* create silence bin */
	player->priv->silencebin = gst_bin_new ("silencebin");

	/*
	 * audiotestsrc is the sensible thing to use here, except that with the
	 * silent waveform it produces buffers with the GAP flag set, which currently
	 * cause pulsesink to screw up.
	 *
	 * to get around this, for now we produce silence using an appsrc instead.
	 */
	player->priv->silencesrc = gst_element_factory_make ("appsrc", "silencesrc");
	g_object_set (player->priv->silencesrc, "caps", caps, "format", GST_FORMAT_TIME, NULL);

	g_signal_connect (player->priv->silencesrc, "need-data", G_CALLBACK (silencesrc_need_data_cb), player);

	audioconvert = gst_element_factory_make ("audioconvert", "silenceconvert");

	capsfilter = gst_element_factory_make ("capsfilter", "silencecapsfilter");
	g_object_set (capsfilter, "caps", caps, NULL);
	gst_caps_unref (caps);

	if (player->priv->silencesrc == NULL ||
	    audioconvert == NULL ||
	    capsfilter == NULL) {
		g_set_error (error,
			     RB_PLAYER_ERROR,
			     RB_PLAYER_ERROR_GENERAL,
			     _("Failed to create GStreamer element; check your installation"));
		return FALSE;
	}

	gst_bin_add_many (GST_BIN (player->priv->silencebin),
			  player->priv->silencesrc,
			  audioconvert,
			  capsfilter,
			  NULL);
	if (gst_element_link_many (player->priv->silencesrc,
				   audioconvert,
				   capsfilter,
				   NULL) == FALSE) {
		g_warning ("zero");
		g_set_error (error,
			     RB_PLAYER_ERROR,
			     RB_PLAYER_ERROR_GENERAL,
			     _("Failed to link GStreamer pipeline; check your installation"));
		return FALSE;
	}

	filterpad = gst_element_get_static_pad (capsfilter, "src");
	ghostpad = gst_ghost_pad_new (NULL, filterpad);
	gst_element_add_pad (player->priv->silencebin, ghostpad);
	gst_object_unref (filterpad);

	/* assemble stuff:
	 * - add everything to the pipeline
	 * - link adder to output bin
	 * - link silence bin to adder
	 */
	gst_bin_add_many (GST_BIN (player->priv->pipeline),
			  player->priv->adder,
			  player->priv->outputbin,
			  player->priv->silencebin,
			  NULL);

	addersrcpad = gst_element_get_static_pad (player->priv->adder, "src");
	plr = gst_pad_link (addersrcpad, outputghostpad);
	if (plr != GST_PAD_LINK_OK) {
		g_warning ("one");
		g_set_error (error,
			     RB_PLAYER_ERROR,
			     RB_PLAYER_ERROR_GENERAL,
			     _("Failed to link GStreamer pipeline; check your installation"));
		return FALSE;
	}

	reqpad = gst_element_get_request_pad (player->priv->adder, "sink_%u");
	if (reqpad == NULL) {
		g_warning ("two");
		g_set_error (error,
			     RB_PLAYER_ERROR,
			     RB_PLAYER_ERROR_GENERAL,
			     _("Failed to link GStreamer pipeline; check your installation"));
		return FALSE;
	}

	plr = gst_pad_link (ghostpad, reqpad);
	if (plr != GST_PAD_LINK_OK) {
		g_warning ("three");
		g_set_error (error,
			     RB_PLAYER_ERROR,
			     RB_PLAYER_ERROR_GENERAL,
			     _("Failed to link GStreamer pipeline; check your installation"));
		return FALSE;
	}

	/* add any tees and filters that were waiting for us */
	for (l = player->priv->waiting_tees; l != NULL; l = g_list_next (l)) {
		rb_player_gst_tee_add_tee (RB_PLAYER_GST_TEE (player), GST_ELEMENT (l->data));
	}
	g_list_free (player->priv->waiting_tees);
	player->priv->waiting_tees = NULL;

	for (l = player->priv->waiting_filters; l != NULL; l = g_list_next (l)) {
		rb_player_gst_filter_add_filter (RB_PLAYER_GST_FILTER (player), GST_ELEMENT (l->data));
	}
	g_list_free (player->priv->waiting_filters);
	player->priv->waiting_filters = NULL;

	player->priv->sink_state = SINK_STOPPED;
	return TRUE;
}



static gboolean
rb_player_gst_xfade_open (RBPlayer *iplayer,
			  const char *uri,
			  gpointer stream_data,
			  GDestroyNotify stream_data_destroy,
			  GError **error)
{
	RBXFadeStream *stream;
	RBPlayerGstXFade *player = RB_PLAYER_GST_XFADE (iplayer);
	gboolean reused = FALSE;
	GList *t;

	/* create sink if we don't already have one */
	if (create_sink (player, error) == FALSE)
		return FALSE;

	/* see if anyone wants us to reuse an existing stream */
	g_rec_mutex_lock (&player->priv->stream_list_lock);
	for (t = player->priv->streams; t != NULL; t = t->next) {
		RBXFadeStream *stream = (RBXFadeStream *)t->data;

		switch (stream->state) {
		case WAITING:
		case PENDING_REMOVE:
		case REUSING:
		case SEEKING:
		case SEEKING_PAUSED:
		case SEEKING_EOS:
		case PREROLLING:
		case PREROLL_PLAY:
			break;

		case PLAYING:
		case FADING_IN:
		case FADING_OUT:
		case FADING_OUT_PAUSED:
		case WAITING_EOS:
		case PAUSED:
			g_signal_emit (player,
				       signals[CAN_REUSE_STREAM], 0,
				       uri, stream->uri, GST_ELEMENT (stream),
				       &reused);
			break;
		}

		if (reused) {
			rb_debug ("reusing stream %s for new stream %s", stream->uri, uri);
			stream->state = REUSING;
			stream->new_uri = g_strdup (uri);
			stream->new_stream_data = stream_data;
			stream->new_stream_data_destroy = stream_data_destroy;

			/* move the stream to the front of the list so it'll be started when
			 * _play is called (it's probably already there, but just in case..)
			 */
			player->priv->streams = g_list_remove (player->priv->streams, stream);
			player->priv->streams = g_list_prepend (player->priv->streams, stream);
			break;
		}
	}
	g_rec_mutex_unlock (&player->priv->stream_list_lock);
	if (reused) {
		return TRUE;
	}

	/* construct new stream */
	stream = create_stream (player, uri, stream_data, stream_data_destroy);
	if (stream == NULL) {
		rb_debug ("unable to create pipeline to play %s", uri);
		g_set_error (error,
			     RB_PLAYER_ERROR,
			     RB_PLAYER_ERROR_GENERAL,
			     _("Failed to create GStreamer pipeline to play %s"),
			     uri);
		return FALSE;
	}

	g_rec_mutex_lock (&player->priv->stream_list_lock);
	player->priv->streams = g_list_prepend (player->priv->streams, stream);
	dump_stream_list (player);
	g_rec_mutex_unlock (&player->priv->stream_list_lock);

	/* start prerolling it */
	preroll_stream (player, stream);

	return TRUE;
}

static gboolean
stop_sink_later (RBPlayerGstXFade *player)
{
	g_rec_mutex_lock (&player->priv->sink_lock);
	player->priv->stop_sink_id = 0;
	if (g_atomic_int_get (&player->priv->linked_streams) == 0) {
		stop_sink (player);
	}
	g_rec_mutex_unlock (&player->priv->sink_lock);

	return FALSE;
}

static void
maybe_stop_sink (RBPlayerGstXFade *player)
{
	g_rec_mutex_lock (&player->priv->sink_lock);
	if (player->priv->stop_sink_id == 0) {
		player->priv->stop_sink_id =
			g_timeout_add (1000,
				       (GSourceFunc) stop_sink_later,
				       player);
	}
	g_rec_mutex_unlock (&player->priv->sink_lock);
}

static gboolean
rb_player_gst_xfade_close (RBPlayer *iplayer, const char *uri, GError **error)
{
	RBPlayerGstXFade *player = RB_PLAYER_GST_XFADE (iplayer);
	gboolean ret = TRUE;

	if (uri == NULL) {
		GList *list;
		GList *l;

		/* need to copy the list as unlink_and_dispose_stream modifies it */
		g_rec_mutex_lock (&player->priv->stream_list_lock);
		list = g_list_copy (player->priv->streams);
		for (l = list; l != NULL; l = l->next) {
			RBXFadeStream *stream = (RBXFadeStream *)l->data;
			g_object_ref (stream);
		}
		g_rec_mutex_unlock (&player->priv->stream_list_lock);

		for (l = list; l != NULL; l = l->next) {
			RBXFadeStream *stream = (RBXFadeStream *)l->data;
			unlink_and_dispose_stream (player, stream);
			g_object_unref (stream);
		}
		g_list_free (list);
	} else {
		/* just stop and close the stream for the specified uri */
		RBXFadeStream *stream;

		g_rec_mutex_lock (&player->priv->stream_list_lock);
		stream = find_stream_by_uri (player, uri);
		g_rec_mutex_unlock (&player->priv->stream_list_lock);

		if (stream != NULL) {
			unlink_and_dispose_stream (player, stream);
			g_object_unref (stream);
		} else {
			rb_debug ("can't find stream for %s", uri);
			/* XXX set error ?*/
			ret = FALSE;
		}
	}

	return ret;
}


static gboolean
rb_player_gst_xfade_opened (RBPlayer *iplayer)
{
	RBPlayerGstXFade *player = RB_PLAYER_GST_XFADE (iplayer);
	RBXFadeStream *stream;
	gboolean opened = FALSE;

	/* maybe replace this with just a flag somewhere? */

	g_rec_mutex_lock (&player->priv->stream_list_lock);

	stream = find_stream_by_state (player, PREROLLING | PREROLL_PLAY | WAITING_EOS | WAITING | FADING_IN | PLAYING | PAUSED);
	if (stream != NULL) {
		opened = TRUE;
		g_object_unref (stream);
	}

	g_rec_mutex_unlock (&player->priv->stream_list_lock);

	return opened;
}

static gboolean
rb_player_gst_xfade_play (RBPlayer *iplayer,
			  RBPlayerPlayType play_type,
			  gint64 crossfade,
			  GError **error)
{
	RBXFadeStream *stream;
	int stream_state;
	RBPlayerGstXFade *player = RB_PLAYER_GST_XFADE (iplayer);
	gboolean ret = TRUE;

	g_rec_mutex_lock (&player->priv->stream_list_lock);

	/* is there anything to play? */
	if (player->priv->streams == NULL) {
		g_set_error (error,
			     RB_PLAYER_ERROR,
			     RB_PLAYER_ERROR_GENERAL,
			     "Nothing to play");		/* should never happen */

		g_rec_mutex_unlock (&player->priv->stream_list_lock);
		return FALSE;
	}
	
	stream = g_list_first (player->priv->streams)->data;
	g_object_ref (stream);
	g_rec_mutex_unlock (&player->priv->stream_list_lock);

	/* make sure the sink is playing */
	if (start_sink (player, error) == FALSE) {
		g_object_unref (stream);
		return FALSE;
	}

	g_mutex_lock (&stream->lock);

	rb_debug ("playing stream %s, play type %d, crossfade %" G_GINT64_FORMAT, stream->uri, play_type, crossfade);

	/* handle transitional states while holding the lock, and handle states that
	 * require action outside it (lock precedence, mostly)
	 */
	switch (stream->state) {
	case PREROLLING:
	case PREROLL_PLAY:
		rb_debug ("stream %s is prerolling; will start playback once prerolling is complete -> PREROLL_PLAY", stream->uri);
		stream->play_type = play_type;
		stream->crossfade = crossfade;
		stream->state = PREROLL_PLAY;
		break;
	
	case SEEKING_PAUSED:
		rb_debug ("unpausing seeking stream %s", stream->uri);
		stream->state = SEEKING;
		break;

	case PENDING_REMOVE:
		rb_debug ("hmm, can't play streams in PENDING_REMOVE state..");
		break;

	default:
		break;
	}

	stream_state = stream->state;
	g_mutex_unlock (&stream->lock);

	/* is the head stream already playing? */
	switch (stream_state) {
	case FADING_IN:
	case FADING_OUT:
	case FADING_OUT_PAUSED:
	case PLAYING:
	case SEEKING:
	case SEEKING_EOS:
		rb_debug ("stream %s is already playing", stream->uri);
		_rb_player_emit_playing_stream (RB_PLAYER (player), stream->stream_data);
		break;

	case PAUSED:
		rb_debug ("unpausing stream %s", stream->uri);
		start_stream_fade (stream, 0.0f, 1.0f, PAUSE_FADE_LENGTH);
		ret = link_and_unblock_stream (stream, error);
		break;

	case WAITING_EOS:
	case WAITING:
		stream->play_type = play_type;
		stream->crossfade = crossfade;
		ret = actually_start_stream (stream, error);
		break;

	case REUSING:
		switch (play_type) {
		case RB_PLAYER_PLAY_REPLACE:
		case RB_PLAYER_PLAY_CROSSFADE:
			/* probably should split this into two states.. */
			if (stream->src_blocked) {
				rb_debug ("reusing and restarting paused stream %s", stream->uri);
				reuse_stream (stream);
				ret = link_and_unblock_stream (stream, error);
			} else {
				rb_debug ("unlinking stream %s for reuse", stream->uri);
				unlink_and_block_stream (stream);
			}
			break;
		case RB_PLAYER_PLAY_AFTER_EOS:
			rb_debug ("waiting for EOS before reusing stream %s", stream->uri);
			break;
		}
		break;

	default:
		break;
	}

	g_object_unref (stream);

	return ret;
}

static void
rb_player_gst_xfade_pause (RBPlayer *iplayer)
{
	RBPlayerGstXFade *player = RB_PLAYER_GST_XFADE (iplayer);
	GList *l;
	GList *to_fade = NULL;
	gboolean done = FALSE;
	double fade_out_start = 1.0f;
	gint64 fade_out_time = PAUSE_FADE_LENGTH;

	g_rec_mutex_lock (&player->priv->stream_list_lock);

	for (l = player->priv->streams; l != NULL; l = l->next) {
		RBXFadeStream *stream;
		stream = (RBXFadeStream *)l->data;
		switch (stream->state) {
		case WAITING:
		case WAITING_EOS:
			rb_debug ("stream %s is not yet playing, can't pause", stream->uri);
			break;

		case PREROLLING:
		case PREROLL_PLAY:
			rb_debug ("stream %s is prerolling, can't pause", stream->uri);
			break;

		case REUSING:
			rb_debug ("stream %s is being reused, can't pause", stream->uri);
			break;

		case PAUSED:
		case SEEKING_PAUSED:
		case FADING_OUT_PAUSED:
			rb_debug ("stream %s is already paused", stream->uri);
			done = TRUE;
			break;

		case FADING_IN:
		case PLAYING:
			rb_debug ("pausing stream %s -> FADING_OUT_PAUSED", stream->uri);
			to_fade = g_list_prepend (to_fade, g_object_ref (stream));
			done = TRUE;
			break;

		case SEEKING:
			rb_debug ("pausing seeking stream %s -> SEEKING_PAUSED", stream->uri);
			stream->state = SEEKING_PAUSED;
			done = TRUE;
			break;
		case SEEKING_EOS:
			rb_debug ("stream %s is seeking after EOS -> SEEKING_PAUSED", stream->uri);
			stream->state = SEEKING_PAUSED;
			done = TRUE;
			break;

		case FADING_OUT:
			rb_debug ("stream %s is fading out, can't be bothered pausing it", stream->uri);
			break;

		case PENDING_REMOVE:
			rb_debug ("stream %s is done, can't pause", stream->uri);
			break;
		}

		if (done)
			break;
	}

	g_rec_mutex_unlock (&player->priv->stream_list_lock);

	for (l = to_fade; l != NULL; l = l->next) {
		RBXFadeStream *stream = (RBXFadeStream *)l->data;

		switch (stream->state) {
		case FADING_IN:
			g_object_get (stream->volume, "volume", &fade_out_start, NULL);
			fade_out_time = (gint64)(((double) PAUSE_FADE_LENGTH) * fade_out_start);

		case PLAYING:
			stream->state = FADING_OUT_PAUSED;
			start_stream_fade (stream, fade_out_start, 0.0f, fade_out_time);

		default:
			/* shouldn't happen, but ignore it if it does */
			break;
		}

		g_object_unref (stream);
	}
	g_list_free (to_fade);
	
	if (done == FALSE)
		rb_debug ("couldn't find a stream to pause");
}

static gboolean
rb_player_gst_xfade_playing (RBPlayer *iplayer)
{
	RBPlayerGstXFade *player = RB_PLAYER_GST_XFADE (iplayer);
	gboolean playing = FALSE;
	RBXFadeStream *stream;

	if (player->priv->sink_state != SINK_PLAYING)
		return FALSE;

	/* XXX maybe replace with just a flag? */

	g_rec_mutex_lock (&player->priv->stream_list_lock);

	stream = find_stream_by_state (player, PLAYING | FADING_IN);
	if (stream != NULL) {
		playing = TRUE;
		g_object_unref (stream);
	}
	g_rec_mutex_unlock (&player->priv->stream_list_lock);
	return playing;
}


static void
rb_player_gst_xfade_set_volume (RBPlayer *iplayer, float volume)
{
	RBPlayerGstXFade *player = RB_PLAYER_GST_XFADE (iplayer);

	player->priv->volume_changed++;
	if (player->priv->volume_handler != NULL) {
		gdouble v = (gdouble)volume;

		/* maybe use a controller here for smoother changes? */
		if (GST_IS_STREAM_VOLUME (player->priv->volume_handler)) {
			gst_stream_volume_set_volume (GST_STREAM_VOLUME (player->priv->volume_handler),
						      GST_STREAM_VOLUME_FORMAT_CUBIC, v);
		} else {
			g_object_set (player->priv->volume_handler, "volume", v, NULL);
		}
		player->priv->volume_applied = player->priv->volume_changed;
	}
	player->priv->cur_volume = volume;
}


static float
rb_player_gst_xfade_get_volume (RBPlayer *iplayer)
{
	RBPlayerGstXFade *player = RB_PLAYER_GST_XFADE (iplayer);

	if (GST_IS_STREAM_VOLUME (player->priv->volume_handler))
		return gst_stream_volume_get_volume (GST_STREAM_VOLUME (player->priv->volume_handler),
						     GST_STREAM_VOLUME_FORMAT_CUBIC);

	return player->priv->cur_volume;
}

static gboolean
rb_player_gst_xfade_seekable (RBPlayer *iplayer)
{
	RBPlayerGstXFade *player = RB_PLAYER_GST_XFADE (iplayer);
	gboolean can_seek = TRUE;
	RBXFadeStream *stream;

	/* is this supposed to query the most recently opened stream,
	 * or the current playing stream?  I really don't know.
	 */
	g_rec_mutex_lock (&player->priv->stream_list_lock);
	stream = find_stream_by_state (player, FADING_IN | PAUSED | PLAYING);
	g_rec_mutex_unlock (&player->priv->stream_list_lock);

	if (stream) {
		GstQuery *query = NULL;
		query = gst_query_new_seeking (GST_FORMAT_TIME);
		if (gst_element_query (stream->volume, query)) {
			gst_query_parse_seeking (query, NULL, &can_seek, NULL, NULL);
		} else {
			gst_query_unref (query);

			query = gst_query_new_duration (GST_FORMAT_TIME);
			can_seek = gst_element_query (stream->volume, query);
		}
		gst_query_unref (query);
		g_object_unref (stream);
	}

	return can_seek;
}

static void
rb_player_gst_xfade_set_time (RBPlayer *iplayer, gint64 time)
{
	RBPlayerGstXFade *player = RB_PLAYER_GST_XFADE (iplayer);
	RBXFadeStream *stream;

	g_rec_mutex_lock (&player->priv->stream_list_lock);
	stream = find_stream_by_state (player, FADING_IN | PLAYING | PAUSED | FADING_OUT_PAUSED | PENDING_REMOVE);
	g_rec_mutex_unlock (&player->priv->stream_list_lock);

	if (stream == NULL) {
		rb_debug ("got seek while no playing streams exist");
		return;
	}

	stream->seek_target = time;
	switch (stream->state) {
	case PAUSED:
		rb_debug ("seeking in paused stream %s; target %" 
		    G_GINT64_FORMAT, stream->uri, stream->seek_target);
		perform_seek (stream);
		break;

	case FADING_OUT_PAUSED:
		/* don't unblock and relink when the seek is done */
		stream->state = SEEKING_PAUSED;
		rb_debug ("seeking in pausing stream %s; target %"
			  G_GINT64_FORMAT, stream->uri, stream->seek_target);
		unlink_and_block_stream (stream);
		break;

	case FADING_IN:
	case PLAYING:
		stream->state = SEEKING;
		rb_debug ("seeking in playing stream %s; target %"
			  G_GINT64_FORMAT, stream->uri, stream->seek_target);
		perform_seek (stream);
		break;

	case PENDING_REMOVE:
		/* this should only happen when the stream has ended,
		 * which means we can't wait for the src pad to be blocked
		 * before we seek.  we unlink the stream when it reaches EOS,
		 * so now we just perform the seek and relink.
		 */
		rb_debug ("seeking in EOS stream %s; target %"
			  G_GINT64_FORMAT, stream->uri, stream->seek_target);
		stream->state = SEEKING_EOS;
		gst_pad_add_probe (stream->src_pad,
				   GST_PAD_PROBE_TYPE_BLOCK_DOWNSTREAM,
				   (GstPadProbeCallback) post_eos_seek_blocked_cb,
				   stream,
				   NULL);
		perform_seek (stream);
		break;
	default:
		g_assert_not_reached ();
	}

	g_object_unref (stream);
}

static gint64
rb_player_gst_xfade_get_time (RBPlayer *iplayer)
{
	gint64 pos = -1;
	RBPlayerGstXFade *player = RB_PLAYER_GST_XFADE (iplayer);

	get_times_and_stream (player, NULL, &pos, NULL);
	return pos;
}

static gboolean
need_pad_block (RBPlayerGstXFade *player)
{
	return (player->priv->sink_state == SINK_PLAYING);
}

static gboolean
rb_player_gst_xfade_add_tee (RBPlayerGstTee *iplayer, GstElement *element)
{
	RBPlayerGstXFade *player = RB_PLAYER_GST_XFADE (iplayer);
	if (player->priv->tee == NULL) {
		player->priv->waiting_tees = g_list_prepend (player->priv->waiting_tees, element);
		return TRUE;
	}

	return rb_gst_add_tee (RB_PLAYER (player), player->priv->tee, element, need_pad_block (player));
}

static gboolean
rb_player_gst_xfade_remove_tee (RBPlayerGstTee *iplayer, GstElement *element)
{
	RBPlayerGstXFade *player = RB_PLAYER_GST_XFADE (iplayer);
	if (player->priv->tee == NULL) {
		gst_object_ref_sink (element);
		player->priv->waiting_tees = g_list_remove (player->priv->waiting_tees, element);
		return TRUE;
	}

	return rb_gst_remove_tee (RB_PLAYER (player), player->priv->tee, element, need_pad_block (player));
}


static gboolean
rb_player_gst_xfade_add_filter (RBPlayerGstFilter *iplayer, GstElement *element)
{
	RBPlayerGstXFade *player = RB_PLAYER_GST_XFADE (iplayer);
	if (player->priv->filterbin == NULL) {
		player->priv->waiting_filters = g_list_prepend (player->priv->waiting_filters, element);
		return TRUE;
	}

	return rb_gst_add_filter (RB_PLAYER (player), player->priv->filterbin, element, need_pad_block (player));
}


static gboolean
rb_player_gst_xfade_remove_filter (RBPlayerGstFilter *iplayer, GstElement *element)
{
	RBPlayerGstXFade *player = RB_PLAYER_GST_XFADE (iplayer);
	if (player->priv->filterbin == NULL) {
		gst_object_ref_sink (element);
		player->priv->waiting_filters = g_list_remove (player->priv->waiting_filters, element);
		return TRUE;
	}

	return rb_gst_remove_filter (RB_PLAYER (player), player->priv->filterbin, element, need_pad_block (player));
}

