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gstreamer/gst/playback/gstplaybasebin.c
Wim Taymans 56f01bc0cb gst/playback/gstplaybasebin.c: Connect to the new queue "pushing" signal instead of the broken "running" one. 
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
(queue_out_of_data):
Connect to the new queue "pushing" signal instead of the broken
"running" one.
2007-05-10 15:28:13 +00:00

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* 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 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
};
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 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_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_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_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_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_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_object_class_install_property (gobject_klass, ARG_STREAMINFO,
g_param_spec_pointer ("stream-info", "Stream info", "List of streaminfo",
G_PARAM_READABLE));
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_object_class_install_property (gobject_klass, ARG_SOURCE,
g_param_spec_object ("source", "Source", "Source element",
GST_TYPE_ELEMENT, G_PARAM_READABLE));
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_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_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_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));
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;
}
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);
}
} 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");
}
}
/* 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));
}
/* 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)
{
gpointer data;
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);
}
/* we remove the probe now because we don't need it anymore to give progress
* about the buffering. */
data = g_object_get_data (G_OBJECT (queue), "probe");
if (data) {
GstPad *sinkpad;
sinkpad = gst_element_get_pad (queue, "sink");
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));
gst_object_unref (sinkpad);
}
/* we post a 100% buffering message to notify the app that buffering is
* completed and playback can start/continue */
fill_buffer (play_base_bin, 100);
}
/* 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 : 1) * 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--;
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;
GstPad *sinkpad;
GstPlayBaseGroup *group;
guint sig;
GstObject *parent;
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;
} else {
type = GST_STREAM_TYPE_UNKNOWN;
}
gst_object_unref (parent);
info = gst_stream_info_new (GST_OBJECT_CAST (pad), type, NULL, caps);
if (type > 0 && type <= NUM_TYPES) {
/* first pad of each type gets a selector + preroll queue */
if (group->type[type - 1].npads == 0) {
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);
/* add the stream to the list */
gst_caps_unref (caps);
info->origin = GST_OBJECT_CAST (pad);
/* select 1st for now - we'll select a preferred one after preroll */
if (type == GST_STREAM_TYPE_UNKNOWN || group->type[type - 1].npads > 0) {
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_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://", 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);
}
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;
/* 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) {
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++) {
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;
}
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++;
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);
/* 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++;
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) {
GST_DEBUG_OBJECT (play_base_bin, "Source has dynamic output pads");
/* 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++;
} 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)
{
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
/*
* 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;
}
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", GST_PAD_NAME (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);
} 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_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_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);
}
GROUP_UNLOCK (play_base_bin);
return array;
}
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