mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-09 10:59:39 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/playback/gstplaybasebin.c
Sebastian Dröge 49deb0c05d Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u... 
Original commit message from CVS:
* configure.ac:
* ext/alsa/gstalsamixerelement.c:
(gst_alsa_mixer_element_class_init):
* ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
* ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
* ext/cdparanoia/gstcdparanoiasrc.c:
(gst_cd_paranoia_src_class_init):
* ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
* ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
* ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
* ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
* ext/pango/gsttextrender.c: (gst_text_render_class_init):
* ext/theora/theoradec.c: (gst_theora_dec_class_init):
* ext/theora/theoraenc.c: (gst_theora_enc_class_init):
* ext/theora/theoraparse.c: (gst_theora_parse_class_init):
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_class_init):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_class_init):
* gst-libs/gst/interfaces/mixertrack.c:
(gst_mixer_track_class_init):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init):
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init):
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init):
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init):
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(preroll_unlinked):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
* gst/playback/gstplaysink.c: (gst_play_sink_class_init):
* gst/playback/gstqueue2.c: (gst_queue_class_init):
* gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
* gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
(gst_stream_selector_class_init):
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
* gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_video_test_src_class_init):
* gst/volume/gstvolume.c: (gst_volume_class_init):
* sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
* sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
* sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
static strings (i.e. all). This gives us less memory usage,
fewer allocations and thus less memory defragmentation. Depend
on core CVS for this. Fixes bug #523806.
2008-03-22 15:00:53 +00:00

2760 lines
84 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst-i18n-plugin.h>
#include <string.h>
#include "gstplaybasebin.h"
#include "gststreamselector.h"
#include "gstplay-marshal.h"
#include <gst/pbutils/pbutils.h>
GST_DEBUG_CATEGORY_STATIC (gst_play_base_bin_debug);
#define GST_CAT_DEFAULT gst_play_base_bin_debug
#define DEFAULT_QUEUE_SIZE (3 * GST_SECOND)
#define DEFAULT_QUEUE_MIN_THRESHOLD ((DEFAULT_QUEUE_SIZE * 30) / 100)
#define DEFAULT_QUEUE_THRESHOLD ((DEFAULT_QUEUE_SIZE * 95) / 100)
#define DEFAULT_CONNECTION_SPEED 0
#define GROUP_LOCK(pbb) g_mutex_lock (pbb->group_lock)
#define GROUP_UNLOCK(pbb) g_mutex_unlock (pbb->group_lock)
#define GROUP_WAIT(pbb) g_cond_wait (pbb->group_cond, pbb->group_lock)
#define GROUP_SIGNAL(pbb) g_cond_signal (pbb->group_cond)
/* props */
enum
{
ARG_0,
ARG_URI,
ARG_SUBURI,
ARG_QUEUE_SIZE,
ARG_QUEUE_THRESHOLD,
ARG_QUEUE_MIN_THRESHOLD,
ARG_NSTREAMS,
ARG_STREAMINFO,
ARG_STREAMINFO_VALUES,
ARG_SOURCE,
ARG_VIDEO,
ARG_AUDIO,
ARG_TEXT,
ARG_SUBTITLE_ENCODING,
ARG_CONNECTION_SPEED
};
static void gst_play_base_bin_class_init (GstPlayBaseBinClass * klass);
static void gst_play_base_bin_init (GstPlayBaseBin * play_base_bin);
static void gst_play_base_bin_dispose (GObject * object);
static void gst_play_base_bin_finalize (GObject * object);
static void gst_play_base_bin_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * spec);
static void gst_play_base_bin_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * spec);
static void gst_play_base_bin_handle_message_func (GstBin * bin,
GstMessage * msg);
static GstStateChangeReturn gst_play_base_bin_change_state (GstElement *
element, GstStateChange transition);
static const GList *gst_play_base_bin_get_streaminfo (GstPlayBaseBin * bin);
static GValueArray *gst_play_base_bin_get_streaminfo_value_array (GstPlayBaseBin
* play_base_bin);
static void preroll_remove_overrun (GstElement * element,
GstPlayBaseBin * play_base_bin);
static void queue_remove_probe (GstElement * queue, GstPlayBaseBin
* play_base_bin);
static GstElement *make_decoder (GstPlayBaseBin * play_base_bin);
static gboolean has_all_raw_caps (GstPad * pad, gboolean * all_raw);
static gboolean prepare_output (GstPlayBaseBin * play_base_bin);
static void set_active_source (GstPlayBaseBin * play_base_bin,
GstStreamType type, gint source_num);
static gboolean probe_triggered (GstPad * pad, GstEvent * event,
gpointer user_data);
static void setup_substreams (GstPlayBaseBin * play_base_bin);
static GstPipelineClass *parent_class;
/*
* GObject playbasebin wrappers.
*/
GType
gst_play_base_bin_get_type (void)
{
static GType gst_play_base_bin_type = 0;
if (!gst_play_base_bin_type) {
static const GTypeInfo gst_play_base_bin_info = {
sizeof (GstPlayBaseBinClass),
NULL,
NULL,
(GClassInitFunc) gst_play_base_bin_class_init,
NULL,
NULL,
sizeof (GstPlayBaseBin),
0,
(GInstanceInitFunc) gst_play_base_bin_init,
NULL
};
gst_play_base_bin_type = g_type_register_static (GST_TYPE_PIPELINE,
"GstPlayBaseBin", &gst_play_base_bin_info, 0);
}
return gst_play_base_bin_type;
}
static void
gst_play_base_bin_class_init (GstPlayBaseBinClass * klass)
{
GObjectClass *gobject_klass;
GstElementClass *gstelement_klass;
GstBinClass *gstbin_klass;
gobject_klass = (GObjectClass *) klass;
gstelement_klass = (GstElementClass *) klass;
gstbin_klass = (GstBinClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_klass->set_property = gst_play_base_bin_set_property;
gobject_klass->get_property = gst_play_base_bin_get_property;
g_object_class_install_property (gobject_klass, ARG_URI,
g_param_spec_string ("uri", "URI", "URI of the media to play",
NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_klass, ARG_SUBURI,
g_param_spec_string ("suburi", ".sub-URI", "Optional URI of a subtitle",
NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_klass, ARG_QUEUE_SIZE,
g_param_spec_uint64 ("queue-size", "Queue size",
"Size of internal queues in nanoseconds", 0, G_MAXINT64,
DEFAULT_QUEUE_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_klass, ARG_QUEUE_THRESHOLD,
g_param_spec_uint64 ("queue-threshold", "Queue threshold",
"Buffering threshold of internal queues in nanoseconds", 0,
G_MAXINT64, DEFAULT_QUEUE_THRESHOLD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_klass, ARG_QUEUE_MIN_THRESHOLD,
g_param_spec_uint64 ("queue-min-threshold", "Queue min threshold",
"Buffering low threshold of internal queues in nanoseconds", 0,
G_MAXINT64, DEFAULT_QUEUE_MIN_THRESHOLD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_klass, ARG_NSTREAMS,
g_param_spec_int ("nstreams", "NStreams", "number of streams",
0, G_MAXINT, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_klass, ARG_STREAMINFO,
g_param_spec_pointer ("stream-info", "Stream info", "List of streaminfo",
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_klass, ARG_STREAMINFO_VALUES,
g_param_spec_value_array ("stream-info-value-array",
"StreamInfo GValueArray", "value array of streaminfo",
g_param_spec_object ("streaminfo", "StreamInfo", "Streaminfo object",
GST_TYPE_STREAM_INFO, G_PARAM_READABLE),
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_klass, ARG_SOURCE,
g_param_spec_object ("source", "Source", "Source element",
GST_TYPE_ELEMENT, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_klass, ARG_VIDEO,
g_param_spec_int ("current-video", "Current video",
"Currently playing video stream (-1 = none)",
-1, G_MAXINT, -1, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_klass, ARG_AUDIO,
g_param_spec_int ("current-audio", "Current audio",
"Currently playing audio stream (-1 = none)",
-1, G_MAXINT, -1, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_klass, ARG_TEXT,
g_param_spec_int ("current-text", "Current text",
"Currently playing text stream (-1 = none)",
-1, G_MAXINT, -1, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_klass, ARG_SUBTITLE_ENCODING,
g_param_spec_string ("subtitle-encoding", "subtitle encoding",
"Encoding to assume if input subtitles are not in UTF-8 encoding. "
"If not set, the GST_SUBTITLE_ENCODING environment variable will "
"be checked for an encoding to use. If that is not set either, "
"ISO-8859-15 will be assumed.", NULL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstPlayBin::connection-speed
*
* Network connection speed in kbps (0 = unknown)
*
* Since: 0.10.10 at gstplaybin.c, 0.10.15 moved to gstplaybasebin
**/
g_object_class_install_property (gobject_klass, ARG_CONNECTION_SPEED,
g_param_spec_uint ("connection-speed", "Connection Speed",
"Network connection speed in kbps (0 = unknown)",
0, G_MAXUINT, DEFAULT_CONNECTION_SPEED,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
GST_DEBUG_CATEGORY_INIT (gst_play_base_bin_debug, "playbasebin", 0,
"playbasebin");
gobject_klass->dispose = GST_DEBUG_FUNCPTR (gst_play_base_bin_dispose);
gobject_klass->finalize = GST_DEBUG_FUNCPTR (gst_play_base_bin_finalize);
gstbin_klass->handle_message =
GST_DEBUG_FUNCPTR (gst_play_base_bin_handle_message_func);
gstelement_klass->change_state =
GST_DEBUG_FUNCPTR (gst_play_base_bin_change_state);
}
static void
gst_play_base_bin_init (GstPlayBaseBin * play_base_bin)
{
play_base_bin->uri = NULL;
play_base_bin->suburi = NULL;
play_base_bin->need_rebuild = TRUE;
play_base_bin->is_stream = FALSE;
play_base_bin->source = NULL;
play_base_bin->decoders = NULL;
play_base_bin->subtitle = NULL;
play_base_bin->subencoding = NULL;
play_base_bin->subtitle_elements = NULL;
play_base_bin->sub_lock = g_mutex_new ();
play_base_bin->group_lock = g_mutex_new ();
play_base_bin->group_cond = g_cond_new ();
play_base_bin->building_group = NULL;
play_base_bin->queued_groups = NULL;
play_base_bin->queue_size = DEFAULT_QUEUE_SIZE;
play_base_bin->queue_threshold = DEFAULT_QUEUE_THRESHOLD;
play_base_bin->queue_min_threshold = DEFAULT_QUEUE_MIN_THRESHOLD;
play_base_bin->connection_speed = DEFAULT_CONNECTION_SPEED;
}
static void
gst_play_base_bin_dispose (GObject * object)
{
GstPlayBaseBin *play_base_bin;
play_base_bin = GST_PLAY_BASE_BIN (object);
g_free (play_base_bin->uri);
play_base_bin->uri = NULL;
g_free (play_base_bin->suburi);
play_base_bin->suburi = NULL;
g_free (play_base_bin->subencoding);
play_base_bin->subencoding = NULL;
if (play_base_bin->subtitle_elements) {
g_slist_free (play_base_bin->subtitle_elements);
play_base_bin->subtitle_elements = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_play_base_bin_finalize (GObject * object)
{
GstPlayBaseBin *play_base_bin = GST_PLAY_BASE_BIN (object);
g_mutex_free (play_base_bin->group_lock);
g_cond_free (play_base_bin->group_cond);
g_mutex_free (play_base_bin->sub_lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
/*
* playbasebingroup stuff.
*/
static GstPlayBaseGroup *
group_create (GstPlayBaseBin * play_base_bin)
{
GstPlayBaseGroup *group;
group = g_new0 (GstPlayBaseGroup, 1);
group->bin = play_base_bin;
group->streaminfo_value_array = g_value_array_new (0);
GST_DEBUG_OBJECT (play_base_bin, "created new group %p", group);
return group;
}
/*
* Gets the currently playing group.
*
* Callers must have group-lock held when calling this.
*/
static GstPlayBaseGroup *
get_active_group (GstPlayBaseBin * play_base_bin)
{
GstPlayBaseGroup *group = NULL;
if (play_base_bin->queued_groups)
group = play_base_bin->queued_groups->data;
return group;
}
/*
* get the group used for discovering the different streams.
* This function creates a group is there is none.
*
* Callers must have group-lock held when calling this.
*/
static GstPlayBaseGroup *
get_building_group (GstPlayBaseBin * play_base_bin)
{
GstPlayBaseGroup *group;
group = play_base_bin->building_group;
if (group == NULL) {
group = group_create (play_base_bin);
play_base_bin->building_group = group;
}
return group;
}
/*
* Callers must have lock held when calling this!
*/
static void
group_destroy (GstPlayBaseGroup * group)
{
GstPlayBaseBin *play_base_bin = group->bin;
gint n;
GST_LOG ("removing group %p", group);
/* remove the preroll queues */
for (n = 0; n < NUM_TYPES; n++) {
GstElement *element = group->type[n].preroll;
GstElement *fakesrc;
GstElement *sel;
const GList *item;
if (!element)
continue;
sel = group->type[n].selector;
/* remove any fakesrc elements for this preroll element */
for (item = sel->pads; item != NULL; item = item->next) {
GstPad *pad = GST_PAD (item->data);
guint sig_id;
if (GST_PAD_DIRECTION (pad) != GST_PAD_SINK)
continue;
sig_id =
GPOINTER_TO_INT (g_object_get_data (G_OBJECT (pad), "unlinked_id"));
if (sig_id != 0) {
GST_LOG ("removing unlink signal %s:%s", GST_DEBUG_PAD_NAME (pad));
g_signal_handler_disconnect (G_OBJECT (pad), sig_id);
g_object_set_data (G_OBJECT (pad), "unlinked_id", GINT_TO_POINTER (0));
}
fakesrc = (GstElement *) g_object_get_data (G_OBJECT (pad), "fakesrc");
if (fakesrc != NULL) {
GST_LOG ("removing fakesrc from %s:%s",
GST_PAD_NAME (pad), GST_ELEMENT_NAME (GST_PAD_PARENT (pad)));
gst_element_set_state (fakesrc, GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (play_base_bin), fakesrc);
}
}
/* if the group is currently being played, we have to remove the element
* from the thread */
gst_element_set_state (element, GST_STATE_NULL);
gst_element_set_state (group->type[n].selector, GST_STATE_NULL);
GST_LOG ("removing preroll element %s", GST_ELEMENT_NAME (element));
gst_bin_remove (group->type[n].bin, element);
gst_bin_remove (group->type[n].bin, group->type[n].selector);
group->type[n].preroll = NULL;
group->type[n].selector = NULL;
group->type[n].bin = NULL;
}
/* free the streaminfo too */
g_list_foreach (group->streaminfo, (GFunc) g_object_unref, NULL);
g_list_free (group->streaminfo);
g_value_array_free (group->streaminfo_value_array);
g_free (group);
}
/*
* is called when the current building group is completely finished
* and ready for playback
*
* This function grabs lock, so take care when calling.
*/
static void
group_commit (GstPlayBaseBin * play_base_bin, gboolean fatal, gboolean subtitle)
{
GstPlayBaseGroup *group;
gboolean had_active_group;
gboolean res;
GROUP_LOCK (play_base_bin);
group = play_base_bin->building_group;
had_active_group = (get_active_group (play_base_bin) != NULL);
GST_DEBUG_OBJECT (play_base_bin, "commit group %p, had active %d",
group, had_active_group);
/* if an element signalled a no-more-pads after we stopped due
* to preroll, the group is NULL. This is not an error */
if (group == NULL) {
if (!fatal) {
GROUP_UNLOCK (play_base_bin);
return;
} else {
GST_DEBUG_OBJECT (play_base_bin, "Group loading failed, bailing out");
}
} else {
if (!subtitle) {
gint n;
GST_DEBUG_OBJECT (play_base_bin, "group %p done", group);
play_base_bin->queued_groups =
g_list_append (play_base_bin->queued_groups, group);
play_base_bin->building_group = NULL;
/* remove signals. We don't want anymore signals from the preroll
* elements at this stage. */
for (n = 0; n < NUM_TYPES; n++) {
GstElement *element = group->type[n].preroll;
if (!element)
continue;
preroll_remove_overrun (element, play_base_bin);
/* if overrun is removed, probe alse has to be removed */
queue_remove_probe (element, play_base_bin);
}
} else {
/* this is a special subtitle bin, we don't commit the group but
* mark the subtitles as detected before we signal. */
GST_DEBUG_OBJECT (play_base_bin, "marking subtitle bin as complete");
play_base_bin->subtitle_done = TRUE;
}
}
GST_DEBUG_OBJECT (play_base_bin, "signal group done");
GROUP_SIGNAL (play_base_bin);
GST_DEBUG_OBJECT (play_base_bin, "signaled group done");
if (!subtitle && !had_active_group) {
if (!prepare_output (play_base_bin)) {
GROUP_UNLOCK (play_base_bin);
return;
}
setup_substreams (play_base_bin);
GST_DEBUG_OBJECT (play_base_bin, "Emitting signal");
res = GST_PLAY_BASE_BIN_GET_CLASS (play_base_bin)->
setup_output_pads (play_base_bin, group);
GST_DEBUG_OBJECT (play_base_bin, "done");
GROUP_UNLOCK (play_base_bin);
g_object_notify (G_OBJECT (play_base_bin), "stream-info");
} else {
GROUP_UNLOCK (play_base_bin);
}
}
/*
* check if there are streams in the group that are not muted
*
* Callers must have group-lock held when calling this.
*/
static gboolean
group_is_muted (GstPlayBaseGroup * group)
{
gint n;
for (n = 0; n < NUM_TYPES; n++) {
if (group->type[n].preroll && !group->type[n].done)
return FALSE;
}
return TRUE;
}
/*
* Buffer/cache checking.
*/
static inline void
fill_buffer (GstPlayBaseBin * play_base_bin, gint percent)
{
GST_DEBUG_OBJECT (play_base_bin, "buffering %d", percent);
gst_element_post_message (GST_ELEMENT_CAST (play_base_bin),
gst_message_new_buffering (GST_OBJECT_CAST (play_base_bin), percent));
}
static gboolean
check_queue_event (GstPad * pad, GstEvent * event, gpointer user_data)
{
GstElement *queue = GST_ELEMENT_CAST (user_data);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
GST_DEBUG ("EOS event, mark EOS");
g_object_set_data (G_OBJECT (queue), "eos", "1");
break;
case GST_EVENT_FLUSH_STOP:
GST_DEBUG ("FLUSH_STOP event, remove EOS");
g_object_set_data (G_OBJECT (queue), "eos", NULL);
break;
default:
GST_DEBUG ("uninteresting event %s", GST_EVENT_TYPE_NAME (event));
break;
}
return TRUE;
}
static gboolean
check_queue (GstPad * pad, GstBuffer * data, gpointer user_data)
{
GstElement *queue = GST_ELEMENT_CAST (user_data);
GstPlayBaseBin *play_base_bin = g_object_get_data (G_OBJECT (queue), "pbb");
guint64 level = 0;
GST_DEBUG_OBJECT (queue, "check queue triggered");
g_object_get (G_OBJECT (queue), "current-level-time", &level, NULL);
GST_DEBUG_OBJECT (play_base_bin, "Queue size: %" GST_TIME_FORMAT,
GST_TIME_ARGS (level));
if (play_base_bin->queue_threshold > 0) {
level = level * 99 / play_base_bin->queue_threshold;
if (level > 99)
level = 99;
} else
level = 99;
fill_buffer (play_base_bin, level);
/* continue! */
return TRUE;
}
/* If a queue overruns and we are buffer in streaming mode (we have a min-time)
* we can potentially create a deadlock when:
*
* 1) the max-bytes is hit and
* 2) the min-time is not hit.
*
* We recover from this situation in this callback by
* setting the max-bytes to unlimited if we see that there is
* a current-time-level (which means some sort of timestamping is
* done).
*/
static void
queue_deadlock_check (GstElement * queue, GstPlayBaseBin * play_base_bin)
{
guint64 time, min_time;
guint bytes;
GST_DEBUG_OBJECT (play_base_bin, "overrun signal received from queue %s",
GST_ELEMENT_NAME (queue));
/* figure out where we are */
g_object_get (G_OBJECT (queue), "current-level-time", &time,
"current-level-bytes", &bytes, "min-threshold-time", &min_time, NULL);
GST_DEBUG_OBJECT (play_base_bin, "streaming mode, queue %s current %"
GST_TIME_FORMAT ", min %" GST_TIME_FORMAT
", bytes %d", GST_ELEMENT_NAME (queue),
GST_TIME_ARGS (time), GST_TIME_ARGS (min_time), bytes);
/* if the bytes in the queue represent time, we disable bytes
* overrun checking to not cause deadlocks.
*/
if (bytes && time != 0 && time < min_time) {
GST_DEBUG_OBJECT (play_base_bin,
"possible deadlock found, removing byte limit");
/* queue knows about time but is filled with bytes that do
* not represent min-threshold time, disable bytes checking so
* the queue can grow some more. */
g_object_set (G_OBJECT (queue), "max-size-bytes", 0, NULL);
/* bytes limit is removed, we cannot deadlock anymore */
g_signal_handlers_disconnect_by_func (queue,
(gpointer) queue_deadlock_check, play_base_bin);
} else {
GST_DEBUG_OBJECT (play_base_bin, "no deadlock");
}
}
static void
queue_remove_probe (GstElement * queue, GstPlayBaseBin * play_base_bin)
{
gpointer data;
GstPad *sinkpad;
data = g_object_get_data (G_OBJECT (queue), "probe");
sinkpad = gst_element_get_pad (queue, "sink");
if (data) {
GST_DEBUG_OBJECT (play_base_bin,
"Removing buffer probe from pad %s:%s (%p)",
GST_DEBUG_PAD_NAME (sinkpad), sinkpad);
g_object_set_data (G_OBJECT (queue), "probe", NULL);
gst_pad_remove_buffer_probe (sinkpad, GPOINTER_TO_INT (data));
} else {
GST_DEBUG_OBJECT (play_base_bin,
"No buffer probe to remove from %s:%s (%p)",
GST_DEBUG_PAD_NAME (sinkpad), sinkpad);
}
gst_object_unref (sinkpad);
}
/* Used for time-based buffering in streaming mode and is called when a queue
* emits the running signal. This means that the high watermark threshold is
* reached and the buffering is completed. */
static void
queue_threshold_reached (GstElement * queue, GstPlayBaseBin * play_base_bin)
{
GstPlayBaseGroup *group;
gpointer data;
gint n;
GST_DEBUG_OBJECT (play_base_bin, "running signal received from queue %s",
GST_ELEMENT_NAME (queue));
/* we disconnect the signal so that we don't get called for every buffer. */
g_signal_handlers_disconnect_by_func (queue,
(gpointer) queue_threshold_reached, play_base_bin);
data = g_object_get_data (G_OBJECT (queue), "eos");
if (data) {
GST_DEBUG_OBJECT (play_base_bin, "disable min threshold time, we are EOS");
g_object_set (queue, "min-threshold-time", (guint64) 0, NULL);
} else {
/* now place the limits at the low threshold. When we hit this limit, the
* underrun signal will be called. The underrun signal is always connected. */
GST_DEBUG_OBJECT (play_base_bin,
"setting min threshold time to %" G_GUINT64_FORMAT,
play_base_bin->queue_min_threshold);
g_object_set (queue, "min-threshold-time",
play_base_bin->queue_min_threshold, NULL);
}
GROUP_LOCK (play_base_bin);
group = get_active_group (play_base_bin);
if (!group) {
GROUP_UNLOCK (play_base_bin);
return;
}
/* we remove the probe now because we don't need it anymore to give progress
* about the buffering. */
for (n = 0; n < NUM_TYPES; n++) {
GstElement *element = group->type[n].preroll;
if (!element)
continue;
queue_remove_probe (element, play_base_bin);
}
GROUP_UNLOCK (play_base_bin);
/* we post a 100% buffering message to notify the app that buffering is
* completed and playback can start/continue */
if (play_base_bin->is_stream)
fill_buffer (play_base_bin, 100);
}
/* this signal will be fired when one of the queues with raw
* data is filled. This means that the group building stage is over
* and playback of the new queued group should start. This is a rather unusual
* situation because normally the group is commited when the "no_more_pads"
* signal is fired.
*/
static void
queue_overrun (GstElement * queue, GstPlayBaseBin * play_base_bin)
{
GST_DEBUG_OBJECT (play_base_bin, "queue %s overrun",
GST_ELEMENT_NAME (queue));
preroll_remove_overrun (queue, play_base_bin);
group_commit (play_base_bin, FALSE,
GST_OBJECT_PARENT (GST_OBJECT_CAST (queue)) ==
GST_OBJECT_CAST (play_base_bin->subtitle));
/* notify end of buffering */
queue_threshold_reached (queue, play_base_bin);
}
/* this signal is only added when in streaming mode to catch underruns
*/
static void
queue_out_of_data (GstElement * queue, GstPlayBaseBin * play_base_bin)
{
GST_DEBUG_OBJECT (play_base_bin, "underrun signal received from queue %s",
GST_ELEMENT_NAME (queue));
/* On underrun, we want to temoprarily pause playback, set a "min-size"
* threshold and wait for the running signal and then play again.
*
* This signal could never be called because the queue max-size limits are set
* too low. We take care of this possible deadlock in the the overrun signal
* handler. */
g_signal_connect (G_OBJECT (queue), "pushing",
G_CALLBACK (queue_threshold_reached), play_base_bin);
GST_DEBUG_OBJECT (play_base_bin,
"setting min threshold time to %" G_GUINT64_FORMAT,
(guint64) play_base_bin->queue_threshold);
g_object_set (queue, "min-threshold-time",
(guint64) play_base_bin->queue_threshold, NULL);
/* re-connect probe, this will fire feedback about the percentage that we
* buffered and is posted in the BUFFERING message. */
if (!g_object_get_data (G_OBJECT (queue), "probe")) {
GstPad *sinkpad;
guint id;
sinkpad = gst_element_get_pad (queue, "sink");
id = gst_pad_add_buffer_probe (sinkpad, G_CALLBACK (check_queue), queue);
g_object_set_data (G_OBJECT (queue), "probe", GINT_TO_POINTER (id));
GST_DEBUG_OBJECT (play_base_bin,
"Re-attaching buffering probe to pad %s:%s %p",
GST_DEBUG_PAD_NAME (sinkpad), sinkpad);
gst_object_unref (sinkpad);
fill_buffer (play_base_bin, 0);
}
}
/*
* generate a preroll element which is simply a queue. While there
* are still dynamic elements in the pipeline, we wait for one
* of the queues to fill. The assumption is that all the dynamic
* streams will be detected by that time.
*
* Callers must have the group-lock held when calling this.
*/
static void
gen_preroll_element (GstPlayBaseBin * play_base_bin,
GstPlayBaseGroup * group, GstStreamType type, GstPad * pad,
GstStreamInfo * info)
{
GstElement *selector, *preroll;
gchar *name, *padname;
const gchar *prename;
guint overrun_sig;
GstPad *preroll_pad;
GstBin *target;
GstState state;
if (type == GST_STREAM_TYPE_VIDEO)
prename = "video";
else if (type == GST_STREAM_TYPE_TEXT)
prename = "text";
else if (type == GST_STREAM_TYPE_AUDIO)
prename = "audio";
else
g_return_if_reached ();
/* create stream selector */
selector = g_object_new (GST_TYPE_STREAM_SELECTOR, NULL);
padname = gst_pad_get_name (pad);
name = g_strdup_printf ("selector_%s_%s", prename, padname);
gst_object_set_name (GST_OBJECT_CAST (selector), name);
g_free (name);
/* create preroll queue */
name = g_strdup_printf ("preroll_%s_%s", prename, padname);
preroll = gst_element_factory_make ("queue", name);
g_free (name);
g_free (padname);
/* for buffering of raw data we ideally want to buffer a
* very small amount of buffers since the memory used by
* this raw data can be enormously huge.
*
* We use an upper limit of typically a few seconds here but
* cap in case no timestamps are set on the raw data (bad!).
*
* FIXME: we abuse this buffer to do network buffering since
* we can then easily do time-based buffering. The better
* solution would be to add a specific network queue right
* after the source that measures the datarate and scales this
* queue of encoded data instead.
*/
g_object_set (G_OBJECT (preroll),
"max-size-buffers", 0, "max-size-bytes",
((type == GST_STREAM_TYPE_VIDEO) ? 25 : 2) * 1024 * 1024,
"max-size-time", play_base_bin->queue_size, NULL);
/* the overrun signal is always attached and serves two purposes:
*
* 1) when we are building a group and the overrun is called, we commit the
* group. The reason being that if we fill the entire queue without a
* normal group commit (with _no_more_pads()) we can assume the
* audio/video is completely wacked or the element just does not know when
* it is ready with all the pads (mpeg).
* 2) When we are doing network buffering, we keep track of low/high
* watermarks in the queue. It is possible that we set the high watermark
* higher than the max-size limits to trigger an overrun. In this case we
* will never get a running signal but we can use the overrun signal to
* detect this deadlock and correct it.
*/
overrun_sig = g_signal_connect (G_OBJECT (preroll), "overrun",
G_CALLBACK (queue_overrun), play_base_bin);
/* keep a ref to the signal id so that we can disconnect the signal callback
* when we are done with the preroll */
g_object_set_data (G_OBJECT (preroll), "overrun_signal_id",
GINT_TO_POINTER (overrun_sig));
if (play_base_bin->is_stream &&
((type == GST_STREAM_TYPE_VIDEO &&
group->type[GST_STREAM_TYPE_AUDIO - 1].npads == 0) ||
(type == GST_STREAM_TYPE_AUDIO &&
group->type[GST_STREAM_TYPE_VIDEO - 1].npads == 0))) {
GstPad *sinkpad;
guint id;
/* catch deadlocks when we are network buffering in time but the max-limit
* in bytes is hit. */
g_signal_connect (G_OBJECT (preroll), "overrun",
G_CALLBACK (queue_deadlock_check), play_base_bin);
/* attach pointer to playbasebin */
g_object_set_data (G_OBJECT (preroll), "pbb", play_base_bin);
/* give updates on queue size */
sinkpad = gst_element_get_pad (preroll, "sink");
id = gst_pad_add_buffer_probe (sinkpad, G_CALLBACK (check_queue), preroll);
GST_DEBUG_OBJECT (play_base_bin, "Attaching probe to pad %s:%s (%p)",
GST_DEBUG_PAD_NAME (sinkpad), sinkpad);
gst_object_unref (sinkpad);
g_object_set_data (G_OBJECT (preroll), "probe", GINT_TO_POINTER (id));
/* catch eos and flush events so that we can ignore underruns */
id = gst_pad_add_event_probe (sinkpad, G_CALLBACK (check_queue_event),
preroll);
g_object_set_data (G_OBJECT (preroll), "eos_probe", GINT_TO_POINTER (id));
/* When we connect this queue, it will start running and immediatly
* fire an underrun. */
g_signal_connect (G_OBJECT (preroll), "underrun",
G_CALLBACK (queue_out_of_data), play_base_bin);
/* configure threshold and callbacks */
queue_out_of_data (preroll, play_base_bin);
}
/* listen for EOS so we can switch groups when one ended. */
preroll_pad = gst_element_get_pad (preroll, "src");
gst_pad_add_event_probe (preroll_pad, G_CALLBACK (probe_triggered), info);
gst_object_unref (preroll_pad);
/* add to group list */
group->type[type - 1].selector = selector;
group->type[type - 1].preroll = preroll;
/* figure out where the preroll element should go */
if (type == GST_STREAM_TYPE_TEXT && play_base_bin->subtitle)
target = GST_BIN_CAST (play_base_bin->subtitle);
else
target = GST_BIN_CAST (play_base_bin);
group->type[type - 1].bin = target;
gst_bin_add (target, selector);
gst_bin_add (target, preroll);
gst_element_link (selector, preroll);
/* figure out target state and set */
state = (GST_STATE (play_base_bin) == GST_STATE_PLAYING ?
GST_STATE_PLAYING : GST_STATE_PAUSED);
gst_element_set_state (selector, state);
gst_element_set_state (preroll, state);
}
static void
preroll_remove_overrun (GstElement * element, GstPlayBaseBin * play_base_bin)
{
guint overrun_sig;
GObject *obj = G_OBJECT (element);
overrun_sig = GPOINTER_TO_INT (g_object_get_data (obj, "overrun_signal_id"));
if (overrun_sig) {
GST_LOG_OBJECT (play_base_bin, "removing preroll signal %s",
GST_ELEMENT_NAME (element));
g_signal_handler_disconnect (obj, overrun_sig);
/* We have disconnected this signal, remove the signal_id from the object
* data */
g_object_set_data (obj, "overrun_signal_id", NULL);
}
}
static void
remove_groups (GstPlayBaseBin * play_base_bin)
{
GROUP_LOCK (play_base_bin);
/* first destroy the group we were building if any */
if (play_base_bin->building_group) {
group_destroy (play_base_bin->building_group);
play_base_bin->building_group = NULL;
}
/* remove the queued groups */
g_list_foreach (play_base_bin->queued_groups, (GFunc) group_destroy, NULL);
g_list_free (play_base_bin->queued_groups);
play_base_bin->queued_groups = NULL;
/* clear subs */
if (play_base_bin->subtitle) {
gst_element_set_state (play_base_bin->subtitle, GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (play_base_bin), play_base_bin->subtitle);
play_base_bin->subtitle = NULL;
}
GROUP_UNLOCK (play_base_bin);
}
/*
* Add/remove a single stream to current building group.
*
* Must be called with group-lock held.
*/
static void
add_stream (GstPlayBaseGroup * group, GstStreamInfo * info)
{
GValue v = { 0, };
GST_DEBUG ("add stream to group %p", group);
/* keep ref to the group */
g_object_set_data (G_OBJECT (info), "group", group);
g_value_init (&v, G_TYPE_OBJECT);
g_value_set_object (&v, info);
g_value_array_append (group->streaminfo_value_array, &v);
g_value_unset (&v);
group->streaminfo = g_list_append (group->streaminfo, info);
if (info->type > 0 && info->type <= NUM_TYPES) {
group->type[info->type - 1].npads++;
}
}
static gboolean
string_arr_has_str (const gchar * values[], const gchar * value)
{
if (values && value) {
while (*values != NULL) {
if (strcmp (value, *values) == 0)
return TRUE;
++values;
}
}
return FALSE;
}
/* mime types we are not handling on purpose right now, don't post a
* missing-plugin message for these */
static const gchar *blacklisted_mimes[] = {
"video/x-dvd-subpicture", NULL
};
#define IS_BLACKLISTED_MIME(type) (string_arr_has_str(blacklisted_mimes,type))
static void
gst_play_base_bin_handle_message_func (GstBin * bin, GstMessage * msg)
{
if (gst_is_missing_plugin_message (msg)) {
gchar *detail;
guint i;
detail = gst_missing_plugin_message_get_installer_detail (msg);
for (i = 0; detail != NULL && blacklisted_mimes[i] != NULL; ++i) {
if (strstr (detail, "|decoder-") && strstr (detail, blacklisted_mimes[i])) {
GST_LOG_OBJECT (bin, "suppressing message %" GST_PTR_FORMAT, msg);
gst_message_unref (msg);
g_free (detail);
return;
}
}
g_free (detail);
}
GST_BIN_CLASS (parent_class)->handle_message (bin, msg);
}
/*
* signal fired when an unknown stream is found. We create a new
* UNKNOWN streaminfo object.
*/
static void
unknown_type (GstElement * element, GstPad * pad, GstCaps * caps,
GstPlayBaseBin * play_base_bin)
{
const gchar *type_name;
GstStreamInfo *info;
GstPlayBaseGroup *group;
type_name = gst_structure_get_name (gst_caps_get_structure (caps, 0));
if (type_name && !IS_BLACKLISTED_MIME (type_name)) {
gchar *capsstr;
capsstr = gst_caps_to_string (caps);
GST_DEBUG_OBJECT (play_base_bin, "don't know how to handle %s", capsstr);
/* FIXME, g_message() ? */
g_message ("don't know how to handle %s", capsstr);
g_free (capsstr);
} else {
/* don't spew stuff to the terminal or send message if it's blacklisted */
GST_DEBUG_OBJECT (play_base_bin, "media type %s not handled on purpose, "
"not posting a missing-plugin message on the bus", type_name);
}
GROUP_LOCK (play_base_bin);
group = get_building_group (play_base_bin);
/* add the stream to the list */
info = gst_stream_info_new (GST_OBJECT_CAST (pad), GST_STREAM_TYPE_UNKNOWN,
NULL, caps);
info->origin = GST_OBJECT_CAST (pad);
add_stream (group, info);
GROUP_UNLOCK (play_base_bin);
}
/* add a streaminfo that indicates that the stream is handled by the
* given element. This usually means that a stream without actual data is
* produced but one that is sunken by an element. Examples of this are:
* cdaudio, a hardware decoder/sink, dvd meta bins etc...
*/
static void
add_element_stream (GstElement * element, GstPlayBaseBin * play_base_bin)
{
GstStreamInfo *info;
GstPlayBaseGroup *group;
GROUP_LOCK (play_base_bin);
group = get_building_group (play_base_bin);
/* add the stream to the list */
info =
gst_stream_info_new (GST_OBJECT_CAST (element), GST_STREAM_TYPE_ELEMENT,
NULL, NULL);
info->origin = GST_OBJECT_CAST (element);
add_stream (group, info);
GROUP_UNLOCK (play_base_bin);
}
/* when the decoder element signals that no more pads will be generated, we
* can commit the current group.
*/
static void
no_more_pads_full (GstElement * element, gboolean subs,
GstPlayBaseBin * play_base_bin)
{
/* setup phase */
GST_DEBUG_OBJECT (element, "no more pads, %d pending",
play_base_bin->pending);
/* nothing pending, we can exit */
if (play_base_bin->pending == 0)
return;
/* the object has no pending no_more_pads */
if (!g_object_get_data (G_OBJECT (element), "pending"))
return;
g_object_set_data (G_OBJECT (element), "pending", NULL);
play_base_bin->pending--;
GST_DEBUG_OBJECT (element, "remove pending, now %d pending",
play_base_bin->pending);
if (play_base_bin->pending == 0) {
/* we can commit this group for playback now */
group_commit (play_base_bin, play_base_bin->is_stream, subs);
}
}
static void
no_more_pads (GstElement * element, GstPlayBaseBin * play_base_bin)
{
no_more_pads_full (element, FALSE, play_base_bin);
}
static void
sub_no_more_pads (GstElement * element, GstPlayBaseBin * play_base_bin)
{
no_more_pads_full (element, TRUE, play_base_bin);
}
static void
source_no_more_pads (GstElement * element, GstPlayBaseBin * bin)
{
GST_DEBUG_OBJECT (bin, "No more pads in source element %s.",
GST_ELEMENT_NAME (element));
g_signal_handler_disconnect (G_OBJECT (element), bin->src_np_sig_id);
bin->src_np_sig_id = 0;
g_signal_handler_disconnect (G_OBJECT (element), bin->src_nmp_sig_id);
bin->src_nmp_sig_id = 0;
no_more_pads_full (element, FALSE, bin);
}
static gboolean
probe_triggered (GstPad * pad, GstEvent * event, gpointer user_data)
{
GstPlayBaseGroup *group;
GstPlayBaseBin *play_base_bin;
GstStreamInfo *info;
gboolean res;
GstEventType type;
type = GST_EVENT_TYPE (event);
GST_LOG ("probe triggered, (%d) %s", type, gst_event_type_get_name (type));
/* we only care about EOS */
if (type != GST_EVENT_EOS)
return TRUE;
info = GST_STREAM_INFO (user_data);
group = (GstPlayBaseGroup *) g_object_get_data (G_OBJECT (info), "group");
play_base_bin = group->bin;
if (type == GST_EVENT_EOS) {
gint num_groups = 0;
gboolean have_left;
GST_DEBUG_OBJECT (play_base_bin, "probe got EOS in group %p", group);
GROUP_LOCK (play_base_bin);
/* mute this stream */
g_object_set (G_OBJECT (info), "mute", TRUE, NULL);
if (info->type > 0 && info->type <= NUM_TYPES)
group->type[info->type - 1].done = TRUE;
/* see if we have some more groups left to play */
num_groups = g_list_length (play_base_bin->queued_groups);
if (play_base_bin->building_group)
num_groups++;
have_left = (num_groups > 1);
/* see if the complete group is muted */
if (!group_is_muted (group)) {
/* group is not completely muted, we remove the EOS event
* and continue, eventually the other streams will be EOSed and
* we can switch out this group. */
GST_DEBUG ("group %p not completely muted", group);
GROUP_UNLOCK (play_base_bin);
/* remove the EOS if we have something left */
return !have_left;
}
if (have_left) {
/* ok, get rid of the current group then */
//group_destroy (group);
/* removing the current group brings the next group
* active */
play_base_bin->queued_groups =
g_list_remove (play_base_bin->queued_groups, group);
/* and wait for the next one to be ready */
while (!play_base_bin->queued_groups) {
GROUP_WAIT (play_base_bin);
}
group = play_base_bin->queued_groups->data;
/* now activate the next one */
setup_substreams (play_base_bin);
GST_DEBUG ("switching to next group %p - emitting signal", group);
/* and signal the new group */
res = GST_PLAY_BASE_BIN_GET_CLASS (play_base_bin)->
setup_output_pads (play_base_bin, group);
GROUP_UNLOCK (play_base_bin);
g_object_notify (G_OBJECT (play_base_bin), "stream-info");
/* get rid of the EOS event */
return FALSE;
} else {
GROUP_UNLOCK (play_base_bin);
GST_LOG ("Last group done, EOS");
}
}
return TRUE;
}
/* This function will be called when the sinkpad of the preroll element
* is unlinked, we have to connect something to the sinkpad or else the
* state change will fail..
*/
static void
preroll_unlinked (GstPad * pad, GstPad * peerpad,
GstPlayBaseBin * play_base_bin)
{
GstElement *fakesrc;
guint sig_id;
GstPad *srcpad;
/* make a fakesrc that will just emit one EOS */
fakesrc = gst_element_factory_make ("fakesrc", NULL);
g_object_set (G_OBJECT (fakesrc), "num-buffers", 0, NULL);
GST_DEBUG ("patching unlinked pad %s:%s", GST_DEBUG_PAD_NAME (pad));
srcpad = gst_element_get_pad (fakesrc, "src");
gst_bin_add (GST_BIN_CAST (play_base_bin), fakesrc);
gst_pad_link (srcpad, pad);
gst_object_unref (srcpad);
/* keep track of these patch elements */
g_object_set_data (G_OBJECT (pad), "fakesrc", fakesrc);
/* now unlink the unlinked signal so that it is not called again when
* we destroy the queue */
sig_id = GPOINTER_TO_INT (g_object_get_data (G_OBJECT (pad), "unlinked_id"));
if (sig_id != 0) {
g_signal_handler_disconnect (G_OBJECT (pad), sig_id);
g_object_set_data (G_OBJECT (pad), "unlinked_id", GINT_TO_POINTER (0));
}
}
/* Mute stream on first data - for header-is-in-stream-stuff
* (vorbis, ogmtext). */
static gboolean
mute_stream (GstPad * pad, GstBuffer * buf, gpointer data)
{
GstStreamInfo *info = GST_STREAM_INFO (data);
guint id;
GST_DEBUG ("mute stream triggered");
g_object_set (G_OBJECT (info), "mute", TRUE, NULL);
id = GPOINTER_TO_INT (g_object_get_data (G_OBJECT (info), "mute_probe"));
g_object_set_data (G_OBJECT (info), "mute_probe", NULL);
if (id > 0)
gst_pad_remove_buffer_probe (GST_PAD_CAST (info->object), id);
/* no data */
return FALSE;
}
/* Eat data. */
static gboolean
silence_stream (GstPad * pad, GstMiniObject * data, gpointer user_data)
{
GST_DEBUG ("silence stream triggered");
/* no data */
return FALSE;
}
/* Called by the signal handlers when a decodebin (main or subtitle) has
* found a new raw pad. We create a preroll element if needed and the
* appropriate streaminfo. Commits the group if there will be no more pads
* from decodebin */
static void
new_decoded_pad_full (GstElement * element, GstPad * pad, gboolean last,
GstPlayBaseBin * play_base_bin, gboolean is_subs)
{
GstStructure *structure;
const gchar *mimetype;
GstCaps *caps;
GstStreamInfo *info;
GstStreamType type = GST_STREAM_TYPE_UNKNOWN;
GstPad *sinkpad;
GstPlayBaseGroup *group;
guint sig;
GstObject *parent;
gboolean first_pad;
GST_DEBUG ("play base: new decoded pad. Last: %d", last);
/* first see if this pad has interesting caps */
caps = gst_pad_get_caps (pad);
if (caps == NULL || gst_caps_is_empty (caps) || gst_caps_is_any (caps))
goto no_type;
/* get the mime type */
structure = gst_caps_get_structure (caps, 0);
mimetype = gst_structure_get_name (structure);
GROUP_LOCK (play_base_bin);
group = get_building_group (play_base_bin);
group->nstreams++;
parent = gst_object_get_parent (GST_OBJECT_CAST (element));
if (g_str_has_prefix (mimetype, "audio/") &&
parent != GST_OBJECT_CAST (play_base_bin->subtitle)) {
type = GST_STREAM_TYPE_AUDIO;
} else if (g_str_has_prefix (mimetype, "video/") &&
parent != GST_OBJECT_CAST (play_base_bin->subtitle)) {
type = GST_STREAM_TYPE_VIDEO;
} else if (g_str_has_prefix (mimetype, "text/")) {
type = GST_STREAM_TYPE_TEXT;
}
gst_object_unref (parent);
info = gst_stream_info_new (GST_OBJECT_CAST (pad), type, NULL, caps);
gst_caps_unref (caps);
if (type == GST_STREAM_TYPE_UNKNOWN) {
/* Unknown streams get added to the group, but the data
* just gets ignored */
add_stream (group, info);
GROUP_UNLOCK (play_base_bin);
/* signal the no more pads after adding the stream */
if (last)
no_more_pads_full (element, is_subs, play_base_bin);
return;
}
/* first pad of each type gets a selector + preroll queue */
first_pad = (group->type[type - 1].npads == 0);
if (first_pad) {
GST_DEBUG ("play base: pad needs new preroll");
gen_preroll_element (play_base_bin, group, type, pad, info);
}
/* add to stream selector */
sinkpad =
gst_element_get_request_pad (group->type[type - 1].selector, "sink%d");
/* make sure we catch unlink signals */
sig = g_signal_connect (G_OBJECT (sinkpad), "unlinked",
G_CALLBACK (preroll_unlinked), play_base_bin);
/* keep a ref to the signal id so that we can disconnect the signal callback */
g_object_set_data (G_OBJECT (sinkpad), "unlinked_id", GINT_TO_POINTER (sig));
/* Store a pointer to the stream selector pad for this stream */
g_object_set_data (G_OBJECT (pad), "pb_sel_pad", sinkpad);
gst_pad_link (pad, sinkpad);
gst_object_unref (sinkpad);
/* select 1st for now - we'll select a preferred one after preroll */
if (!first_pad) {
guint id;
GST_DEBUG ("Adding silence_stream data probe on type %d (npads %d)", type,
group->type[type - 1].npads);
id = gst_pad_add_data_probe (GST_PAD_CAST (pad),
G_CALLBACK (silence_stream), info);
g_object_set_data (G_OBJECT (pad), "eat_probe", GINT_TO_POINTER (id));
}
/* add the stream to the list */
add_stream (group, info);
GROUP_UNLOCK (play_base_bin);
/* signal the no more pads after adding the stream */
if (last)
no_more_pads_full (element, is_subs, play_base_bin);
return;
/* ERRORS */
no_type:
{
g_warning ("no type on pad %s:%s", GST_DEBUG_PAD_NAME (pad));
if (caps)
gst_caps_unref (caps);
return;
}
}
static void
new_decoded_pad (GstElement * element, GstPad * pad, gboolean last,
GstPlayBaseBin * play_base_bin)
{
new_decoded_pad_full (element, pad, last, play_base_bin, FALSE);
}
static void
subs_new_decoded_pad (GstElement * element, GstPad * pad, gboolean last,
GstPlayBaseBin * play_base_bin)
{
new_decoded_pad_full (element, pad, last, play_base_bin, TRUE);
}
static void
set_encoding_element (GstElement * element, gchar * encoding)
{
GST_DEBUG_OBJECT (element, "setting encoding to %s", GST_STR_NULL (encoding));
g_object_set (G_OBJECT (element), "subtitle-encoding", encoding, NULL);
}
static void
decodebin_element_added_cb (GstBin * decodebin, GstElement * element,
gpointer data)
{
GstPlayBaseBin *play_base_bin = GST_PLAY_BASE_BIN (data);
gchar *encoding;
if (!g_object_class_find_property (G_OBJECT_GET_CLASS (element),
"subtitle-encoding")) {
return;
}
g_mutex_lock (play_base_bin->sub_lock);
play_base_bin->subtitle_elements =
g_slist_append (play_base_bin->subtitle_elements, element);
encoding = g_strdup (play_base_bin->subencoding);
g_mutex_unlock (play_base_bin->sub_lock);
set_encoding_element (element, encoding);
g_free (encoding);
}
static void
decodebin_element_removed_cb (GstBin * decodebin, GstElement * element,
gpointer data)
{
GstPlayBaseBin *play_base_bin = GST_PLAY_BASE_BIN (data);
g_mutex_lock (play_base_bin->sub_lock);
play_base_bin->subtitle_elements =
g_slist_remove (play_base_bin->subtitle_elements, element);
g_mutex_unlock (play_base_bin->sub_lock);
}
/*
* Generate source ! subparse bins.
*/
static GstElement *
setup_subtitle (GstPlayBaseBin * play_base_bin, gchar * sub_uri)
{
GstElement *source, *subdecodebin, *subbin;
if (!gst_uri_is_valid (sub_uri))
goto invalid_uri;
source = gst_element_make_from_uri (GST_URI_SRC, sub_uri, NULL);
if (!source)
goto unknown_uri;
if (g_getenv ("USE_DECODEBIN2"))
subdecodebin = gst_element_factory_make ("decodebin2", "subtitle-decoder");
else
subdecodebin = gst_element_factory_make ("decodebin", "subtitle-decoder");
g_signal_connect (subdecodebin, "element-added",
G_CALLBACK (decodebin_element_added_cb), play_base_bin);
g_signal_connect (subdecodebin, "element-removed",
G_CALLBACK (decodebin_element_removed_cb), play_base_bin);
subbin = gst_bin_new ("subtitle-bin");
gst_bin_add_many (GST_BIN_CAST (subbin), source, subdecodebin, NULL);
gst_element_link (source, subdecodebin);
/* return the subtitle GstElement object */
return subbin;
/* WARNINGS */
invalid_uri:
{
GST_ELEMENT_WARNING (play_base_bin, RESOURCE, NOT_FOUND,
(_("Invalid subtitle URI \"%s\", subtitles disabled."), sub_uri),
(NULL));
return NULL;
}
unknown_uri:
{
gchar *prot = gst_uri_get_protocol (sub_uri);
if (prot) {
gchar *desc;
gst_element_post_message (GST_ELEMENT (play_base_bin),
gst_missing_uri_source_message_new (GST_ELEMENT (play_base_bin),
prot));
desc = gst_pb_utils_get_source_description (prot);
GST_ELEMENT_ERROR (play_base_bin, CORE, MISSING_PLUGIN,
(_("A %s plugin is required to play this stream, but not installed."),
desc), ("No URI handler to handle sub_uri: %s", sub_uri));
g_free (desc);
g_free (prot);
} else
goto invalid_uri;
return NULL;
}
}
/* helper function to lookup stuff in lists */
static gboolean
array_has_value (const gchar * values[], const gchar * value)
{
gint i;
for (i = 0; values[i]; i++) {
if (g_str_has_prefix (value, values[i]))
return TRUE;
}
return FALSE;
}
/* list of URIs that we consider to be streams and that need buffering.
* We have no mechanism yet to figure this out with a query. */
static const gchar *stream_uris[] = { "http://", "mms://", "mmsh://",
"mmsu://", "mmst://", "myth://", NULL
};
/* blacklisted URIs, we know they will always fail. */
static const gchar *blacklisted_uris[] = { NULL };
/* mime types that we don't consider to be media types */
static const gchar *no_media_mimes[] = {
"application/x-executable", "application/x-bzip", "application/x-gzip",
"application/zip", "application/x-compress", NULL
};
/* mime types we consider raw media */
static const gchar *raw_mimes[] = {
"audio/x-raw", "video/x-raw", NULL
};
#define IS_STREAM_URI(uri) (array_has_value (stream_uris, uri))
#define IS_BLACKLISTED_URI(uri) (array_has_value (blacklisted_uris, uri))
#define IS_NO_MEDIA_MIME(mime) (array_has_value (no_media_mimes, mime))
#define IS_RAW_MIME(mime) (array_has_value (raw_mimes, mime))
/*
* Generate and configure a source element.
*/
static GstElement *
gen_source_element (GstPlayBaseBin * play_base_bin, GstElement ** subbin)
{
GstElement *source;
if (!play_base_bin->uri)
goto no_uri;
if (!gst_uri_is_valid (play_base_bin->uri))
goto invalid_uri;
if (IS_BLACKLISTED_URI (play_base_bin->uri))
goto uri_blacklisted;
if (play_base_bin->suburi) {
GST_LOG_OBJECT (play_base_bin, "Creating decoder for subtitles URI %s",
play_base_bin->suburi);
/* subtitle specified */
*subbin = setup_subtitle (play_base_bin, play_base_bin->suburi);
} else {
/* no subtitle specified */
*subbin = NULL;
}
source = gst_element_make_from_uri (GST_URI_SRC, play_base_bin->uri,
"source");
if (!source)
goto no_source;
play_base_bin->is_stream = IS_STREAM_URI (play_base_bin->uri);
/* make HTTP sources send extra headers so we get icecast
* metadata in case the stream is an icecast stream */
if (!strncmp (play_base_bin->uri, "http://", 7) &&
g_object_class_find_property (G_OBJECT_GET_CLASS (source),
"iradio-mode")) {
g_object_set (source, "iradio-mode", TRUE, NULL);
}
if (g_object_class_find_property (G_OBJECT_GET_CLASS (source),
"connection-speed")) {
GST_DEBUG_OBJECT (play_base_bin,
"setting connection-speed=%d to source element",
play_base_bin->connection_speed / 1000);
g_object_set (source, "connection-speed",
play_base_bin->connection_speed / 1000, NULL);
}
return source;
/* ERRORS */
no_uri:
{
GST_ELEMENT_ERROR (play_base_bin, RESOURCE, NOT_FOUND,
(_("No URI specified to play from.")), (NULL));
return NULL;
}
invalid_uri:
{
GST_ELEMENT_ERROR (play_base_bin, RESOURCE, NOT_FOUND,
(_("Invalid URI \"%s\"."), play_base_bin->uri), (NULL));
return NULL;
}
uri_blacklisted:
{
GST_ELEMENT_ERROR (play_base_bin, RESOURCE, FAILED,
(_("RTSP streams cannot be played yet.")), (NULL));
return NULL;
}
no_source:
{
gchar *prot = gst_uri_get_protocol (play_base_bin->uri);
/* whoops, could not create the source element, dig a little deeper to
* figure out what might be wrong. */
if (prot) {
gchar *desc;
gst_element_post_message (GST_ELEMENT (play_base_bin),
gst_missing_uri_source_message_new (GST_ELEMENT (play_base_bin),
prot));
desc = gst_pb_utils_get_source_description (prot);
GST_ELEMENT_ERROR (play_base_bin, CORE, MISSING_PLUGIN,
(_("A %s plugin is required to play this stream, but not installed."),
desc), ("No URI handler for %s", prot));
g_free (desc);
g_free (prot);
} else
goto invalid_uri;
return NULL;
}
}
/* is called when a dynamic source element created a new pad. */
static void
source_new_pad (GstElement * element, GstPad * pad, GstPlayBaseBin * bin)
{
GstElement *decoder;
gboolean is_raw;
GST_DEBUG_OBJECT (bin, "Found new pad %s.%s in source element %s",
GST_DEBUG_PAD_NAME (pad), GST_ELEMENT_NAME (element));
/* if this is a pad with all raw caps, we can expose it */
if (has_all_raw_caps (pad, &is_raw) && is_raw) {
/* it's all raw, create output pads. */
new_decoded_pad_full (element, pad, FALSE, bin, FALSE);
return;
}
/* not raw, create decoder */
decoder = make_decoder (bin);
if (!decoder)
goto no_decodebin;
/* and link to decoder */
if (!gst_element_link (bin->source, decoder))
goto could_not_link;
gst_element_set_state (decoder, GST_STATE_PAUSED);
return;
/* ERRORS */
no_decodebin:
{
/* error was posted */
return;
}
could_not_link:
{
GST_ELEMENT_ERROR (bin, CORE, NEGOTIATION,
(NULL), ("Can't link source to decoder element"));
return;
}
}
/*
* Setup the substreams (is called right after group_commit () when
* loading a new group, or after switching groups).
*
* Should be called with group-lock held.
*/
static void
setup_substreams (GstPlayBaseBin * play_base_bin)
{
GstPlayBaseGroup *group;
gint n;
const GList *item;
GST_DEBUG_OBJECT (play_base_bin, "setting up substreams");
/* Remove the eat probes */
group = get_active_group (play_base_bin);
for (item = group->streaminfo; item; item = item->next) {
GstStreamInfo *info = item->data;
gpointer data;
data = g_object_get_data (G_OBJECT (info->object), "eat_probe");
if (data) {
gst_pad_remove_data_probe (GST_PAD_CAST (info->object),
GPOINTER_TO_INT (data));
g_object_set_data (G_OBJECT (info->object), "eat_probe", NULL);
}
/* now remove unknown pads */
if (info->type == GST_STREAM_TYPE_UNKNOWN) {
guint id;
id = GPOINTER_TO_INT (g_object_get_data (G_OBJECT (info), "mute_probe"));
if (id == 0) {
id = gst_pad_add_buffer_probe (GST_PAD_CAST (info->object),
G_CALLBACK (mute_stream), info);
g_object_set_data (G_OBJECT (info), "mute_probe", GINT_TO_POINTER (id));
}
}
}
/* now check if the requested current streams exist. If
* current >= num_streams, decrease current so at least
* we have output. Always keep it enabled. */
for (n = 0; n < NUM_TYPES; n++) {
if (play_base_bin->current[n] >= group->type[n].npads) {
GST_DEBUG_OBJECT (play_base_bin, "reset type %d to current 0", n);
play_base_bin->current[n] = 0;
}
}
/* now activate the right sources. Don't forget that during preroll,
* we set the first source to forwarding and ignored the rest. */
for (n = 0; n < NUM_TYPES; n++) {
GST_DEBUG_OBJECT (play_base_bin, "setting type %d to current %d", n,
play_base_bin->current[n]);
set_active_source (play_base_bin, n + 1, play_base_bin->current[n]);
}
}
/**
* has_all_raw_caps:
* @pad: a #GstPad
* @all_raw: pointer to hold the result
*
* check if the caps of the pad are all raw. The caps are all raw if
* all of its structures contain audio/x-raw or video/x-raw.
*
* Returns: %FALSE @pad has no caps. Else TRUE and @all_raw set t the result.
*/
static gboolean
has_all_raw_caps (GstPad * pad, gboolean * all_raw)
{
GstCaps *caps;
gint capssize;
guint i, num_raw = 0;
gboolean res = FALSE;
caps = gst_pad_get_caps (pad);
if (caps == NULL)
return FALSE;
capssize = gst_caps_get_size (caps);
/* no caps, skip and move to the next pad */
if (capssize == 0 || gst_caps_is_empty (caps) || gst_caps_is_any (caps))
goto done;
/* count the number of raw formats in the caps */
for (i = 0; i < capssize; ++i) {
GstStructure *s;
const gchar *mime_type;
s = gst_caps_get_structure (caps, i);
mime_type = gst_structure_get_name (s);
if (IS_RAW_MIME (mime_type))
++num_raw;
}
*all_raw = (num_raw == capssize);
res = TRUE;
done:
gst_caps_unref (caps);
return res;
}
/**
* analyse_source:
* @play_base_bin: a #GstPlayBaseBin
* @is_raw: are all pads raw data
* @have_out: does the source have output
* @is_dynamic: is this a dynamic source
*
* Check the source of @play_base_bin and collect information about it.
*
* @is_raw will be set to TRUE if the source only produces raw pads. When this
* function returns, all of the raw pad of the source will be added
* to @play_base_bin.
*
* @have_out: will be set to TRUE if the source has output pads.
*
* @is_dynamic: TRUE if the element will create (more) pads dynamically later
* on.
*
* Returns: FALSE if a fatal error occured while scanning.
*/
static gboolean
analyse_source (GstPlayBaseBin * play_base_bin, gboolean * is_raw,
gboolean * have_out, gboolean * is_dynamic)
{
GstIterator *pads_iter;
gboolean done = FALSE;
gboolean res = TRUE;
*have_out = FALSE;
*is_raw = FALSE;
*is_dynamic = FALSE;
pads_iter = gst_element_iterate_src_pads (play_base_bin->source);
while (!done) {
GstPad *pad = NULL;
switch (gst_iterator_next (pads_iter, (gpointer) & pad)) {
case GST_ITERATOR_ERROR:
res = FALSE;
/* FALLTROUGH */
case GST_ITERATOR_DONE:
done = TRUE;
break;
case GST_ITERATOR_RESYNC:
/* reset results and resync */
*have_out = FALSE;
*is_raw = FALSE;
*is_dynamic = FALSE;
gst_iterator_resync (pads_iter);
break;
case GST_ITERATOR_OK:
/* we now officially have an ouput pad */
*have_out = TRUE;
/* if FALSE, this pad has no caps and we continue with the next pad. */
if (!has_all_raw_caps (pad, is_raw)) {
gst_object_unref (pad);
break;
}
/* caps on source pad are all raw, we can add the pad */
if (*is_raw) {
new_decoded_pad_full (play_base_bin->source, pad, FALSE,
play_base_bin, FALSE);
}
gst_object_unref (pad);
break;
}
}
gst_iterator_free (pads_iter);
if (!*have_out) {
GstElementClass *elemclass;
GList *walk;
/* element has no output pads, check for padtemplates that list SOMETIMES
* pads. */
elemclass = GST_ELEMENT_GET_CLASS (play_base_bin->source);
walk = gst_element_class_get_pad_template_list (elemclass);
while (walk != NULL) {
GstPadTemplate *templ;
templ = (GstPadTemplate *) walk->data;
if (GST_PAD_TEMPLATE_DIRECTION (templ) == GST_PAD_SRC) {
if (GST_PAD_TEMPLATE_PRESENCE (templ) == GST_PAD_SOMETIMES) {
*is_dynamic = TRUE;
break; /* only break out if we found a sometimes src pad
continue walking through if say a request src pad is found
elements such as mpegtsparse and dvbbasebin have request
and sometimes src pads */
}
}
walk = g_list_next (walk);
}
}
return res;
}
static void
remove_decoders (GstPlayBaseBin * bin)
{
GSList *walk;
for (walk = bin->decoders; walk; walk = g_slist_next (walk)) {
GstElement *decoder = GST_ELEMENT_CAST (walk->data);
GST_DEBUG_OBJECT (bin, "removing old decoder element");
gst_element_set_state (decoder, GST_STATE_NULL);
gst_bin_remove (GST_BIN_CAST (bin), decoder);
}
g_slist_free (bin->decoders);
bin->decoders = NULL;
}
static GstElement *
make_decoder (GstPlayBaseBin * play_base_bin)
{
GstElement *decoder;
/* now create the decoder element */
if (g_getenv ("USE_DECODEBIN2"))
decoder = gst_element_factory_make ("decodebin2", NULL);
else
decoder = gst_element_factory_make ("decodebin", NULL);
if (!decoder)
goto no_decodebin;
g_signal_connect (decoder, "element-added",
G_CALLBACK (decodebin_element_added_cb), play_base_bin);
g_signal_connect (decoder, "element-removed",
G_CALLBACK (decodebin_element_removed_cb), play_base_bin);
gst_bin_add (GST_BIN_CAST (play_base_bin), decoder);
/* set up callbacks to create the links between decoded data
* and video/audio/subtitle rendering/output. */
g_signal_connect (G_OBJECT (decoder),
"new-decoded-pad", G_CALLBACK (new_decoded_pad), play_base_bin);
g_signal_connect (G_OBJECT (decoder), "no-more-pads",
G_CALLBACK (no_more_pads), play_base_bin);
g_signal_connect (G_OBJECT (decoder),
"unknown-type", G_CALLBACK (unknown_type), play_base_bin);
g_object_set_data (G_OBJECT (decoder), "pending", "1");
play_base_bin->pending++;
GST_DEBUG_OBJECT (play_base_bin, "created decodebin, %d pending",
play_base_bin->pending);
play_base_bin->decoders = g_slist_prepend (play_base_bin->decoders, decoder);
return decoder;
/* ERRORS */
no_decodebin:
{
GST_ELEMENT_ERROR (play_base_bin, CORE, MISSING_PLUGIN,
(_("Could not create \"decodebin\" element.")), (NULL));
return NULL;
}
}
static void
remove_source (GstPlayBaseBin * bin)
{
GstElement *source = bin->source;
if (source) {
GST_DEBUG_OBJECT (bin, "removing old src element");
gst_element_set_state (source, GST_STATE_NULL);
if (bin->src_np_sig_id) {
g_signal_handler_disconnect (G_OBJECT (source), bin->src_np_sig_id);
bin->src_np_sig_id = 0;
}
if (bin->src_nmp_sig_id) {
g_signal_handler_disconnect (G_OBJECT (source), bin->src_nmp_sig_id);
bin->src_nmp_sig_id = 0;
}
gst_bin_remove (GST_BIN_CAST (bin), source);
bin->source = NULL;
}
}
static GstBusSyncReply
subbin_startup_sync_msg (GstBus * bus, GstMessage * msg, gpointer user_data)
{
if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ERROR) {
GstPlayBaseBin *play_base_bin;
play_base_bin = GST_PLAY_BASE_BIN (user_data);
if (!play_base_bin->subtitle_done) {
GST_WARNING_OBJECT (play_base_bin, "error starting up subtitle bin: %"
GST_PTR_FORMAT, msg);
play_base_bin->subtitle_done = TRUE;
GST_DEBUG_OBJECT (play_base_bin, "signal group done");
GROUP_SIGNAL (play_base_bin);
GST_DEBUG_OBJECT (play_base_bin, "signaled group done");
}
}
return GST_BUS_PASS;
}
/* construct and run the source and decoder elements until we found
* all the streams or until a preroll queue has been filled.
*/
static gboolean
setup_source (GstPlayBaseBin * play_base_bin)
{
GstElement *subbin = NULL;
gboolean is_raw, have_out, is_dynamic;
if (!play_base_bin->need_rebuild)
return TRUE;
GST_DEBUG_OBJECT (play_base_bin, "setup source");
/* delete old src */
remove_source (play_base_bin);
/* create and configure an element that can handle the uri */
if (!(play_base_bin->source = gen_source_element (play_base_bin, &subbin)))
goto no_source;
/* state will be merged later - if file is not found, error will be
* handled by the application right after. */
gst_bin_add (GST_BIN_CAST (play_base_bin), play_base_bin->source);
g_object_notify (G_OBJECT (play_base_bin), "source");
/* remove the old decoders now, if any */
remove_decoders (play_base_bin);
/* remove our previous preroll queues */
remove_groups (play_base_bin);
/* clear pending dynamic elements */
play_base_bin->pending = 0;
/* do subs */
if (subbin) {
GstElement *db;
GstBus *bus;
play_base_bin->subtitle = subbin;
db = gst_bin_get_by_name (GST_BIN_CAST (subbin), "subtitle-decoder");
/* do type detection, without adding (so no preroll) */
g_signal_connect (G_OBJECT (db), "new-decoded-pad",
G_CALLBACK (subs_new_decoded_pad), play_base_bin);
g_signal_connect (G_OBJECT (db), "no-more-pads",
G_CALLBACK (sub_no_more_pads), play_base_bin);
g_signal_connect (G_OBJECT (db), "unknown-type",
G_CALLBACK (unknown_type), play_base_bin);
g_object_set_data (G_OBJECT (db), "pending", "1");
play_base_bin->pending++;
GST_DEBUG_OBJECT (play_base_bin, "we have subtitles, %d pending",
play_base_bin->pending);
if (!play_base_bin->is_stream) {
GstStateChangeReturn sret;
/* either when the queues are filled or when the decoder element
* has no more dynamic streams, the cond is unlocked. We can remove
* the signal handlers then
*/
GST_DEBUG_OBJECT (play_base_bin, "starting subtitle bin");
/* for subtitles in a separate bin we will not commit the
* current building group since we need to add the other
* audio/video streams to the group. We check if we managed
* to commit the subtitle group using an extra flag. */
play_base_bin->subtitle_done = FALSE;
/* since subbin is still a stand-alone bin, we need to add a custom bus
* to intercept error messages, so we can stop waiting and continue */
bus = gst_bus_new ();
gst_element_set_bus (subbin, bus);
gst_bus_set_sync_handler (bus, subbin_startup_sync_msg, play_base_bin);
sret = gst_element_set_state (subbin, GST_STATE_PAUSED);
if (sret != GST_STATE_CHANGE_FAILURE) {
GROUP_LOCK (play_base_bin);
GST_DEBUG ("waiting for subtitle to complete...");
while (!play_base_bin->subtitle_done)
GROUP_WAIT (play_base_bin);
GST_DEBUG ("group done !");
GROUP_UNLOCK (play_base_bin);
if (!play_base_bin->building_group ||
play_base_bin->building_group->type[GST_STREAM_TYPE_TEXT -
1].npads == 0) {
GST_DEBUG ("No subtitle found - ignoring");
gst_element_set_state (subbin, GST_STATE_NULL);
gst_object_unref (play_base_bin->subtitle);
play_base_bin->subtitle = NULL;
} else {
GST_DEBUG_OBJECT (play_base_bin, "Subtitle set-up successful");
}
} else {
GST_WARNING_OBJECT (play_base_bin, "Failed to start subtitle bin");
gst_element_set_state (subbin, GST_STATE_NULL);
gst_object_unref (play_base_bin->subtitle);
play_base_bin->subtitle = NULL;
}
gst_bus_set_sync_handler (bus, NULL, NULL);
gst_element_set_bus (subbin, NULL);
gst_object_unref (bus);
}
gst_object_unref (db);
}
/* see if the source element emits raw audio/video all by itself,
* if so, we can create streams for the pads and be done with it.
* Also check that is has source pads, if not, we assume it will
* do everything itself. */
if (!analyse_source (play_base_bin, &is_raw, &have_out, &is_dynamic))
goto invalid_source;
if (is_raw) {
GST_DEBUG_OBJECT (play_base_bin, "Source provides all raw data");
/* source provides raw data, we added the pads and we can now signal a
* no_more pads because we are done. */
group_commit (play_base_bin, play_base_bin->is_stream, FALSE);
return TRUE;
}
if (!have_out && !is_dynamic) {
GST_DEBUG_OBJECT (play_base_bin, "Source has no output pads");
/* create a stream to indicate that this uri is handled by a self
* contained element. We are now done. */
add_element_stream (play_base_bin->source, play_base_bin);
group_commit (play_base_bin, play_base_bin->is_stream, FALSE);
return TRUE;
}
if (is_dynamic) {
/* connect a handler for the new-pad signal */
play_base_bin->src_np_sig_id =
g_signal_connect (G_OBJECT (play_base_bin->source), "pad-added",
G_CALLBACK (source_new_pad), play_base_bin);
play_base_bin->src_nmp_sig_id =
g_signal_connect (G_OBJECT (play_base_bin->source), "no-more-pads",
G_CALLBACK (source_no_more_pads), play_base_bin);
g_object_set_data (G_OBJECT (play_base_bin->source), "pending", "1");
play_base_bin->pending++;
GST_DEBUG_OBJECT (play_base_bin,
"Source has dynamic output pads, %d pending", play_base_bin->pending);
} else {
GstElement *decoder;
/* no dynamic source, we can link now */
decoder = make_decoder (play_base_bin);
if (!decoder)
goto no_decodebin;
if (!gst_element_link (play_base_bin->source, decoder))
goto could_not_link;
}
if (play_base_bin->subtitle)
gst_bin_add (GST_BIN_CAST (play_base_bin), play_base_bin->subtitle);
play_base_bin->need_rebuild = FALSE;
return TRUE;
/* ERRORS */
no_source:
{
/* error message was already posted */
return FALSE;
}
invalid_source:
{
GST_ELEMENT_ERROR (play_base_bin, CORE, FAILED,
(_("Source element is invalid.")), (NULL));
return FALSE;
}
no_decodebin:
{
/* message was posted */
return FALSE;
}
could_not_link:
{
GST_ELEMENT_ERROR (play_base_bin, CORE, NEGOTIATION,
(NULL), ("Can't link source to decoder element"));
return FALSE;
}
}
static void
finish_source (GstPlayBaseBin * play_base_bin)
{
/* FIXME: no need to grab the group lock here? (tpm) */
if (get_active_group (play_base_bin) != NULL) {
if (play_base_bin->subtitle) {
/* make subs iterate from now on */
gst_bin_add (GST_BIN_CAST (play_base_bin), play_base_bin->subtitle);
}
}
}
/*
* Caller must have group-lock held.
*
* We iterate over all detected streams in the streaminfo and try to find
* impossible cases, like subtitles without video.
*/
static gboolean
prepare_output (GstPlayBaseBin * play_base_bin)
{
const GList *item;
gboolean stream_found = FALSE, no_media = FALSE;
gboolean got_video = FALSE, got_subtitle = FALSE;
GstPlayBaseGroup *group;
group = get_active_group (play_base_bin);
/* check if we found any supported stream... if not, then
* we detected stream type (or the above would've failed),
* but linking/decoding failed - plugin probably missing. */
for (item = group ? group->streaminfo : NULL; item != NULL; item = item->next) {
GstStreamInfo *info = GST_STREAM_INFO (item->data);
if (info->type == GST_STREAM_TYPE_VIDEO) {
stream_found = TRUE;
got_video = TRUE;
break;
} else if (info->type == GST_STREAM_TYPE_ELEMENT) {
stream_found = TRUE;
} else if (info->type == GST_STREAM_TYPE_AUDIO) {
stream_found = TRUE;
} else if (info->type == GST_STREAM_TYPE_TEXT ||
info->type == GST_STREAM_TYPE_SUBPICTURE) {
got_subtitle = TRUE;
} else if (!item->prev && !item->next) {
/* We're no audio/video and the only stream... We could
* be something not-media that's detected because then our
* typefind doesn't mess up with mp3 (bz2, gz, elf, ...) */
if (info->caps && !gst_caps_is_empty (info->caps)) {
const gchar *mime =
gst_structure_get_name (gst_caps_get_structure (info->caps, 0));
no_media = IS_NO_MEDIA_MIME (mime);
}
}
}
if (!stream_found) {
if (got_subtitle) {
GST_ELEMENT_ERROR (play_base_bin, STREAM, WRONG_TYPE,
(_("Only a subtitle stream was detected. "
"Either you are loading a subtitle file or some other type of "
"text file, or the media file was not recognized.")), (NULL));
} else if (!no_media) {
GST_ELEMENT_ERROR (play_base_bin, STREAM, CODEC_NOT_FOUND,
(_("You do not have a decoder installed to handle this file. "
"You might need to install the necessary plugins.")), (NULL));
} else {
GST_ELEMENT_ERROR (play_base_bin, STREAM, WRONG_TYPE,
(_("This is not a media file")), (NULL));
}
return FALSE;
} else if (got_subtitle && !got_video) {
GST_ELEMENT_ERROR (play_base_bin, STREAM, WRONG_TYPE,
(_("A subtitle stream was detected, but no video stream.")), (NULL));
return FALSE;
}
return TRUE;
}
/*
* Multi-stream management. -1 = none.
*
* Caller has group-lock held.
*/
static gint
get_active_source (GstPlayBaseBin * play_base_bin, GstStreamType type)
{
GstPlayBaseGroup *group;
GList *s;
gint num = 0;
group = get_active_group (play_base_bin);
if (!group)
return -1;
for (s = group->streaminfo; s; s = s->next) {
GstStreamInfo *info = s->data;
if (info->type == type) {
if (!info->mute && !g_object_get_data (G_OBJECT (info), "mute_probe")) {
return num;
} else {
num++;
}
}
}
return -1;
}
/* Kill pad reactivation on state change. */
#if 0
static void muted_group_change_state (GstElement * element,
gint old_state, gint new_state, gpointer data);
#endif
static void
mute_group_type (GstPlayBaseGroup * group, GstStreamType type, gboolean mute)
{
gboolean active = !mute;
GstPad *pad;
pad = gst_element_get_pad (group->type[type - 1].preroll, "src");
gst_pad_set_active (pad, active);
gst_object_unref (pad);
pad = gst_element_get_pad (group->type[type - 1].preroll, "sink");
gst_pad_set_active (pad, active);
gst_object_unref (pad);
pad = gst_element_get_pad (group->type[type - 1].selector, "src");
gst_pad_set_active (pad, active);
gst_object_unref (pad);
#if 0
if (mute) {
g_signal_connect (group->type[type - 1].preroll, "state-changed",
G_CALLBACK (muted_group_change_state), group);
} else {
g_signal_handlers_disconnect_by_func (group->type[type - 1].preroll,
G_CALLBACK (muted_group_change_state), group);
}
#endif
}
#if 0
static void
muted_group_change_state (GstElement * element,
gint old_state, gint new_state, gpointer data)
{
GstPlayBaseGroup *group = data;
GROUP_LOCK (group->bin);
if (new_state == GST_STATE_PLAYING) {
gint n;
for (n = 0; n < NUM_TYPES; n++) {
if (group->type[n].selector == element) {
mute_group_type (group, n + 1, TRUE);
}
}
}
GROUP_UNLOCK (group->bin);
}
#endif
static void
set_subtitles_visible (GstPlayBaseBin * play_base_bin, gboolean visible)
{
GstPlayBaseBinClass *klass = GST_PLAY_BASE_BIN_GET_CLASS (play_base_bin);
/* we use a vfunc for this since we don't have a reference to the
* textoverlay element, but playbin does */
if (klass != NULL && klass->set_subtitles_visible != NULL)
klass->set_subtitles_visible (play_base_bin, visible);
}
/*
* Caller has group-lock held.
*/
static void
set_active_source (GstPlayBaseBin * play_base_bin,
GstStreamType type, gint source_num)
{
GstPlayBaseGroup *group;
GList *s;
gint num = 0;
gboolean have_active = FALSE;
GstElement *sel;
GST_LOG ("Changing active source of type %d to %d", type, source_num);
play_base_bin->current[type - 1] = source_num;
group = get_active_group (play_base_bin);
if (!group || !group->type[type - 1].preroll) {
GST_LOG ("No active group, or group for type %d has no preroll", type);
return;
}
/* HACK: instead of unlinking the subtitle input (= lots of hassle,
* especially if subtitles come from an external source), just tell
* textoverlay not to render them */
if (type == GST_STREAM_TYPE_TEXT) {
gboolean visible = (source_num != -1);
set_subtitles_visible (play_base_bin, visible);
if (!visible)
return;
}
sel = group->type[type - 1].selector;
for (s = group->streaminfo; s; s = s->next) {
GstStreamInfo *info = s->data;
if (info->type == type) {
if (num == source_num) {
GstPad *sel_pad;
GST_LOG ("Unmuting (if already muted) source %d of type %d", source_num,
type);
g_object_set (info, "mute", FALSE, NULL);
/* Tell the stream selector which pad to accept */
sel_pad = GST_PAD_CAST (g_object_get_data (G_OBJECT (info->object),
"pb_sel_pad"));
if (sel && sel_pad != NULL) {
g_object_set (G_OBJECT (sel), "active-pad", sel_pad, NULL);
}
have_active = TRUE;
} else {
guint id;
GST_LOG_OBJECT (info->object, "Muting source %d of type %d", num, type);
id = gst_pad_add_buffer_probe (GST_PAD_CAST (info->object),
G_CALLBACK (mute_stream), info);
g_object_set_data (G_OBJECT (info), "mute_probe", GINT_TO_POINTER (id));
}
num++;
}
}
if (!have_active) {
GST_LOG ("Muting group type: %d", type);
g_object_set (sel, "active-pad", NULL, NULL);
} else {
GST_LOG ("Unmuting group type: %d", type);
}
mute_group_type (group, type, !have_active);
}
static void
gst_play_base_bin_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstPlayBaseBin *play_base_bin;
g_return_if_fail (GST_IS_PLAY_BASE_BIN (object));
play_base_bin = GST_PLAY_BASE_BIN (object);
switch (prop_id) {
case ARG_URI:
{
const gchar *uri = g_value_get_string (value);
if (uri == NULL) {
g_warning ("cannot set NULL uri");
return;
}
/* if we have no previous uri, or the new uri is different from the
* old one, replug */
if (play_base_bin->uri == NULL || strcmp (play_base_bin->uri, uri) != 0) {
g_free (play_base_bin->uri);
play_base_bin->uri = g_strdup (uri);
GST_DEBUG ("setting new uri to %s", uri);
play_base_bin->need_rebuild = TRUE;
}
break;
}
case ARG_SUBURI:{
const gchar *suburi = g_value_get_string (value);
if ((!suburi && !play_base_bin->suburi) ||
(suburi && play_base_bin->suburi &&
!strcmp (play_base_bin->suburi, suburi)))
return;
g_free (play_base_bin->suburi);
play_base_bin->suburi = g_strdup (suburi);
GST_DEBUG ("setting new .sub uri to %s", suburi);
play_base_bin->need_rebuild = TRUE;
break;
}
case ARG_QUEUE_SIZE:
play_base_bin->queue_size = g_value_get_uint64 (value);
break;
case ARG_QUEUE_THRESHOLD:
play_base_bin->queue_threshold = g_value_get_uint64 (value);
break;
case ARG_QUEUE_MIN_THRESHOLD:
play_base_bin->queue_min_threshold = g_value_get_uint64 (value);
break;
case ARG_CONNECTION_SPEED:
play_base_bin->connection_speed = g_value_get_uint (value) * 1000;
break;
case ARG_VIDEO:
GROUP_LOCK (play_base_bin);
set_active_source (play_base_bin,
GST_STREAM_TYPE_VIDEO, g_value_get_int (value));
GROUP_UNLOCK (play_base_bin);
break;
case ARG_AUDIO:
GROUP_LOCK (play_base_bin);
set_active_source (play_base_bin,
GST_STREAM_TYPE_AUDIO, g_value_get_int (value));
GROUP_UNLOCK (play_base_bin);
break;
case ARG_TEXT:
GROUP_LOCK (play_base_bin);
set_active_source (play_base_bin,
GST_STREAM_TYPE_TEXT, g_value_get_int (value));
GROUP_UNLOCK (play_base_bin);
break;
case ARG_SUBTITLE_ENCODING:
{
const gchar *encoding;
GSList *list;
encoding = g_value_get_string (value);
if (encoding && play_base_bin->subencoding &&
!strcmp (play_base_bin->subencoding, encoding)) {
return;
}
if (encoding == NULL && play_base_bin->subencoding == NULL)
return;
g_mutex_lock (play_base_bin->sub_lock);
g_free (play_base_bin->subencoding);
play_base_bin->subencoding = g_strdup (encoding);
list = g_slist_copy (play_base_bin->subtitle_elements);
g_slist_foreach (list, (GFunc) gst_object_ref, NULL);
g_mutex_unlock (play_base_bin->sub_lock);
/* we can't hold a lock when calling g_object_set() on a child, since
* the notify event will trigger GstObject to send a deep-notify event
* which will try to take the lock ... */
g_slist_foreach (list, (GFunc) set_encoding_element, (gpointer) encoding);
g_slist_foreach (list, (GFunc) gst_object_unref, NULL);
g_slist_free (list);
break;
}
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_play_base_bin_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstPlayBaseBin *play_base_bin;
g_return_if_fail (GST_IS_PLAY_BASE_BIN (object));
play_base_bin = GST_PLAY_BASE_BIN (object);
switch (prop_id) {
case ARG_URI:
g_value_set_string (value, play_base_bin->uri);
break;
case ARG_SUBURI:
g_value_set_string (value, play_base_bin->suburi);
break;
case ARG_NSTREAMS:
{
GstPlayBaseGroup *group;
GROUP_LOCK (play_base_bin);
group = get_active_group (play_base_bin);
if (group) {
g_value_set_int (value, group->nstreams);
} else {
g_value_set_int (value, 0);
}
GROUP_UNLOCK (play_base_bin);
break;
}
case ARG_QUEUE_SIZE:
g_value_set_uint64 (value, play_base_bin->queue_size);
break;
case ARG_QUEUE_THRESHOLD:
g_value_set_uint64 (value, play_base_bin->queue_threshold);
break;
case ARG_QUEUE_MIN_THRESHOLD:
g_value_set_uint64 (value, play_base_bin->queue_min_threshold);
break;
case ARG_CONNECTION_SPEED:
g_value_set_uint (value, play_base_bin->connection_speed / 1000);
break;
case ARG_STREAMINFO:
/* FIXME: hold some kind of lock here, use iterator */
g_value_set_pointer (value,
(gpointer) gst_play_base_bin_get_streaminfo (play_base_bin));
break;
case ARG_STREAMINFO_VALUES:{
GValueArray *copy;
copy = gst_play_base_bin_get_streaminfo_value_array (play_base_bin);
g_value_take_boxed (value, copy);
break;
}
case ARG_SOURCE:
g_value_set_object (value, play_base_bin->source);
break;
case ARG_VIDEO:
GROUP_LOCK (play_base_bin);
g_value_set_int (value, get_active_source (play_base_bin,
GST_STREAM_TYPE_VIDEO));
GROUP_UNLOCK (play_base_bin);
break;
case ARG_AUDIO:
GROUP_LOCK (play_base_bin);
g_value_set_int (value, get_active_source (play_base_bin,
GST_STREAM_TYPE_AUDIO));
GROUP_UNLOCK (play_base_bin);
break;
case ARG_TEXT:
GROUP_LOCK (play_base_bin);
g_value_set_int (value, get_active_source (play_base_bin,
GST_STREAM_TYPE_TEXT));
GROUP_UNLOCK (play_base_bin);
break;
case ARG_SUBTITLE_ENCODING:
GST_OBJECT_LOCK (play_base_bin);
g_value_set_string (value, play_base_bin->subencoding);
GST_OBJECT_UNLOCK (play_base_bin);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_play_base_bin_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret;
GstPlayBaseBin *play_base_bin;
play_base_bin = GST_PLAY_BASE_BIN (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
if (!setup_source (play_base_bin))
goto source_failed;
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
if (ret == GST_STATE_CHANGE_FAILURE)
goto cleanup_groups;
finish_source (play_base_bin);
break;
/* clean-up in both cases, READY=>NULL clean-up is if there was an error */
case GST_STATE_CHANGE_PAUSED_TO_READY:
case GST_STATE_CHANGE_READY_TO_NULL:
play_base_bin->need_rebuild = TRUE;
remove_decoders (play_base_bin);
remove_groups (play_base_bin);
remove_source (play_base_bin);
break;
default:
break;
}
return ret;
/* ERRORS */
source_failed:
{
play_base_bin->need_rebuild = TRUE;
return GST_STATE_CHANGE_FAILURE;
}
cleanup_groups:
{
/* clean up leftover groups */
remove_groups (play_base_bin);
play_base_bin->need_rebuild = TRUE;
return GST_STATE_CHANGE_FAILURE;
}
}
static const GList *
gst_play_base_bin_get_streaminfo (GstPlayBaseBin * play_base_bin)
{
GstPlayBaseGroup *group = get_active_group (play_base_bin);
GList *info = NULL;
if (group) {
info = group->streaminfo;
}
return info;
}
static GValueArray *
gst_play_base_bin_get_streaminfo_value_array (GstPlayBaseBin * play_base_bin)
{
GstPlayBaseGroup *group;
GValueArray *array = NULL;
GROUP_LOCK (play_base_bin);
group = get_active_group (play_base_bin);
if (group) {
array = g_value_array_copy (group->streaminfo_value_array);
} else {
array = g_value_array_new (0);
}
GROUP_UNLOCK (play_base_bin);
return array;
}
Reference in a new issue View git blame Copy permalink