mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-25 01:30:38 +00:00
Code Issues 1 Projects Releases Wiki Activity

libgstplay has a new home. it still needs to be packaged though

Browse source 
Original commit message from CVS:
libgstplay has a new home. it still needs to be packaged though
This commit is contained in:
Steve Baker 2002-10-27 20:59:41 +00:00
parent 246c51cc7e
commit ad565493fc
5 changed files with 1833 additions and 2 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

4
gst-libs/gst/Makefile.am
View file

@ -4,6 +4,6 @@ else
GCONF_DIR=
endif
SUBDIRS = audio idct resample riff floatcast $(GCONF_DIR) video
SUBDIRS = audio idct resample riff floatcast $(GCONF_DIR) video play
DIST_SUBDIRS = audio idct resample riff floatcast gconf video
DIST_SUBDIRS = audio idct resample riff floatcast gconf video play

13
gst-libs/gst/play/Makefile.am Normal file
View file

@ -0,0 +1,13 @@
librarydir = $(libdir)
library_LTLIBRARIES = libgstplay.la
libgstplay_la_SOURCES = play.c
libgstplayincludedir = $(includedir)/@PACKAGE@-@VERSION@/gst/play
libgstplayinclude_HEADERS = play.h
libgstplay_la_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_CFLAGS)
libgstplay_la_LIBADD = $(GST_LIBS) $(GST_PLUGINS_LIBS)
noinst_HEADERS = playpipelines.c

890
gst-libs/gst/play/play.old.c Normal file
View file

@ -0,0 +1,890 @@
/* GStreamer
* Copyright (C) 1999,2000,2001,2002 Erik Walthinsen <omega@cse.ogi.edu>
* 2000,2001,2002 Wim Taymans <wtay@chello.be>
* 2002 Steve Baker <steve@stevebaker.org>
*
* play.c: GstPlay object code
*
* 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.
*/
#include "play.h"
enum {
STREAM_END,
INFORMATION,
STATE_CHANGE,
STREAM_LENGTH,
TIME_TICK,
HAVE_XID,
HAVE_VIDEO_SIZE,
LAST_SIGNAL,
};
/* this struct is used to decouple signals coming out of threaded pipelines */
typedef struct _GstPlaySignal GstPlaySignal;
struct _GstPlaySignal
{
gint signal_id;
union {
struct {
gint width;
gint height;
} video_size;
struct {
gint xid;
} video_xid;
struct {
GstElementState old_state;
GstElementState new_state;
} state;
struct {
GstElement* element;
GParamSpec* param;
} info;
} signal_data;
};
enum
{
ARG_0,
ARG_LOCATION,
ARG_VOLUME,
ARG_MUTE,
/* FILL ME */
};
static guint gst_play_signals [LAST_SIGNAL] = { 0 };
static void gst_play_init (GstPlay *play);
static void gst_play_class_init (GstPlayClass *klass);
static void gst_play_dispose (GObject *object);
static void gst_play_default_timeout_add (guint interval, GSourceFunc function, gpointer data);
static void gst_play_default_idle_add (GSourceFunc function, gpointer data);
static void gst_play_set_property (GObject *object, guint prop_id,
const GValue *value, GParamSpec *pspec);
static void gst_play_get_property (GObject *object, guint prop_id,
GValue *value, GParamSpec *pspec);
static void callback_pipeline_state_change (GstElement *element, GstElementState old,
GstElementState state, GstPlay *play);
static void callback_pipeline_deep_notify (GstElement *element, GstElement *orig,
GParamSpec *param, GstPlay *play);
static void callback_audio_sink_eos (GstElement *element, GstPlay *play);
static void callback_video_have_xid (GstElement *element, gint xid, GstPlay *play);
static void callback_video_have_size (GstElement *element, gint width, gint height, GstPlay *play);
static void callback_bin_pre_iterate (GstBin *bin, GMutex *mutex);
static void callback_bin_post_iterate (GstBin *bin, GMutex *mutex);
static gboolean gst_play_idle_signal (GstPlay *play);
static gboolean gst_play_idle_callback (GstPlay *play);
static gboolean gst_play_get_length_callback (GstPlay *play);
static gboolean gst_play_tick_callback (GstPlay *play);
GQuark
gst_play_error_quark (void)
{
static GQuark quark = 0;
if (quark == 0) {
quark = g_quark_from_static_string ("gst-play-error-quark");
}
return quark;
}
/* GError creation when plugin is missing */
/* If we want to make error messages less generic and have more errors
* than only plug-ins, move the message creation to the switch */
static void
gst_play_error_plugin (GstPlayError type, GError **error)
{
gchar *name;
if (error == NULL) return;
switch (type)
{
case GST_PLAY_ERROR_THREAD:
name = g_strdup ("thread");
break;
case GST_PLAY_ERROR_QUEUE:
name = g_strdup ("queue");
break;
case GST_PLAY_ERROR_FAKESINK:
name = g_strdup ("fakesink");
break;
case GST_PLAY_ERROR_VOLUME:
name = g_strdup ("volume");
break;
case GST_PLAY_ERROR_COLORSPACE:
name = g_strdup ("colorspace");
break;
case GST_PLAY_ERROR_GNOMEVFSSRC:
name = g_strdup ("gnomevfssrc");
break;
default:
name = g_strdup ("unknown");
break;
}
*error = g_error_new (GST_PLAY_ERROR, type,
"The %s plug-in could not be found. "
"This plug-in is essential for gst-player. "
"Please install it and verify that it works "
"by running 'gst-inspect %s'",
name, name);
g_free (name);
return;
}
/* split static pipeline functions to a seperate file */
#include "playpipelines.c"
static GstElementClass * parent_class = NULL;
GType
gst_play_get_type (void)
{
static GType play_type = 0;
if (!play_type)
{
static const GTypeInfo play_info = {
sizeof (GstPlayClass),
(GBaseInitFunc) NULL,
(GBaseFinalizeFunc) NULL,
(GClassInitFunc) gst_play_class_init,
NULL, NULL, sizeof (GstPlay),
0, (GInstanceInitFunc) gst_play_init,
NULL
};
play_type = g_type_register_static (G_TYPE_OBJECT, "GstPlay", &play_info, 0);
}
return play_type;
}
static void
gst_play_class_init (GstPlayClass *klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
parent_class = g_type_class_ref(GST_TYPE_OBJECT);
klass->information = NULL;
klass->state_changed = NULL;
klass->stream_end = NULL;
gobject_class->dispose = gst_play_dispose;
gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_play_set_property);
gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_play_get_property);
g_object_class_install_property (gobject_class, ARG_LOCATION,
g_param_spec_string ("location", "location of file",
"location of the file to play",
NULL, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_VOLUME,
g_param_spec_float ("volume", "Playing volume",
"Playing volume",
0, 1.0, 0, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_MUTE,
g_param_spec_boolean ("mute", "Volume muted", "Playing volume muted",
FALSE, G_PARAM_READWRITE));
gst_play_signals [INFORMATION] =
g_signal_new ("information",
G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_FIRST,
G_STRUCT_OFFSET (GstPlayClass, information),
NULL, NULL,
gst_marshal_VOID__OBJECT_PARAM,
G_TYPE_NONE, 2,
G_TYPE_OBJECT, G_TYPE_PARAM);
gst_play_signals [STATE_CHANGE] =
g_signal_new ("state_change",
G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_FIRST,
G_STRUCT_OFFSET (GstPlayClass, state_changed),
NULL, NULL,
gst_marshal_VOID__INT_INT,
G_TYPE_NONE, 2,
G_TYPE_INT, G_TYPE_INT);
gst_play_signals [STREAM_END] =
g_signal_new ("stream_end",
G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_FIRST,
G_STRUCT_OFFSET (GstPlayClass, stream_end),
NULL, NULL,
gst_marshal_VOID__VOID,
G_TYPE_NONE, 0);
gst_play_signals [TIME_TICK] =
g_signal_new ("time_tick",
G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_FIRST,
G_STRUCT_OFFSET (GstPlayClass, time_tick),
NULL, NULL,
gst_marshal_VOID__INT64,
G_TYPE_NONE, 1,
G_TYPE_INT64);
gst_play_signals [STREAM_LENGTH] =
g_signal_new ("stream_length",
G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_FIRST,
G_STRUCT_OFFSET (GstPlayClass, stream_length),
NULL, NULL,
gst_marshal_VOID__INT64,
G_TYPE_NONE, 1,
G_TYPE_INT64);
gst_play_signals [HAVE_XID] =
g_signal_new ("have_xid",
G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_FIRST,
G_STRUCT_OFFSET (GstPlayClass, have_xid),
NULL, NULL,
gst_marshal_VOID__INT,
G_TYPE_NONE, 1,
G_TYPE_INT);
gst_play_signals [HAVE_VIDEO_SIZE] =
g_signal_new ("have_video_size",
G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_FIRST,
G_STRUCT_OFFSET (GstPlayClass, have_video_size),
NULL, NULL,
gst_marshal_VOID__INT_INT,
G_TYPE_NONE, 2,
G_TYPE_INT, G_TYPE_INT);
gst_control_init(NULL,NULL);
}
static void
gst_play_init (GstPlay *play)
{
play->pipeline = NULL;
play->source = NULL;
play->autoplugger = NULL;
play->audio_sink = NULL;
play->audio_sink_element = NULL;
play->video_sink = NULL;
play->video_sink_element = NULL;
play->volume = NULL;
play->other_elements = g_hash_table_new(g_str_hash, g_str_equal);
play->audio_bin_mutex = g_mutex_new();
play->video_bin_mutex = g_mutex_new();
gst_play_set_idle_timeout_funcs(play, gst_play_default_timeout_add, gst_play_default_idle_add);
}
GstPlay *
gst_play_new (GstPlayPipeType pipe_type, GError **error)
{
GstPlay *play;
play = g_object_new (GST_TYPE_PLAY, NULL);
/* FIXME: looks like only VIDEO ever gets used ! */
switch (pipe_type){
case GST_PLAY_PIPE_VIDEO:
play->setup_pipeline = gst_play_video_setup;
play->teardown_pipeline = NULL;
play->set_autoplugger = gst_play_video_set_auto;
play->set_video_sink = gst_play_video_set_video;
play->set_audio_sink = gst_play_video_set_audio;
break;
case GST_PLAY_PIPE_VIDEO_THREADSAFE:
play->setup_pipeline = gst_play_videots_setup;
play->teardown_pipeline = NULL;
play->set_autoplugger = gst_play_videots_set_auto;
play->set_video_sink = gst_play_videots_set_video;
play->set_audio_sink = gst_play_videots_set_audio;
break;
case GST_PLAY_PIPE_AUDIO_THREADED:
play->setup_pipeline = gst_play_audiot_setup;
play->teardown_pipeline = NULL;
play->set_autoplugger = gst_play_audiot_set_auto;
play->set_video_sink = NULL;
play->set_audio_sink = gst_play_audiot_set_audio;
break;
case GST_PLAY_PIPE_AUDIO_HYPER_THREADED:
play->setup_pipeline = gst_play_audioht_setup;
play->teardown_pipeline = NULL;
play->set_autoplugger = gst_play_audioht_set_auto;
play->set_video_sink = NULL;
play->set_audio_sink = gst_play_audioht_set_audio;
break;
default:
g_warning("unknown pipeline type: %d\n", pipe_type);
}
/* init pipeline */
if ((play->setup_pipeline) &&
(! play->setup_pipeline (play, error)))
{
g_object_unref (play);
return NULL;
}
if (play->pipeline){
/* connect to pipeline events */
g_signal_connect (G_OBJECT (play->pipeline), "deep_notify", G_CALLBACK (callback_pipeline_deep_notify), play);
g_signal_connect (G_OBJECT (play->pipeline), "state_change", G_CALLBACK (callback_pipeline_state_change), play);
}
if (play->volume){
play->vol_dpman = gst_dpman_get_manager(play->volume);
play->vol_dparam = gst_dpsmooth_new(G_TYPE_FLOAT);
g_object_set(G_OBJECT(play->vol_dparam), "update_period", 2000000LL, NULL);
g_object_set(G_OBJECT(play->vol_dparam), "slope_delta_float", 0.1F, NULL);
g_object_set(G_OBJECT(play->vol_dparam), "slope_time", 10000000LL, NULL);
if (!gst_dpman_attach_dparam (play->vol_dpman, "volume", play->vol_dparam)){
g_warning("could not attach dparam to volume element\n");
}
gst_dpman_set_mode(play->vol_dpman, "asynchronous");
gst_play_set_volume(play, 0.9);
}
play->signal_queue = g_async_queue_new();
return play;
}
static void
gst_play_dispose (GObject *object)
{
GstPlay *play = GST_PLAY (object);
G_OBJECT_CLASS (parent_class)->dispose (object);
g_mutex_free(play->audio_bin_mutex);
g_mutex_free(play->video_bin_mutex);
}
static void
callback_pipeline_deep_notify (GstElement *element, GstElement *orig, GParamSpec *param, GstPlay* play)
{
GstPlaySignal *signal;
signal = g_new0(GstPlaySignal, 1);
signal->signal_id = INFORMATION;
signal->signal_data.info.element = orig;
signal->signal_data.info.param = param;
g_async_queue_push(play->signal_queue, signal);
play->idle_add_func ((GSourceFunc) gst_play_idle_signal, play);
}
static void
callback_pipeline_state_change (GstElement *element, GstElementState old, GstElementState state, GstPlay* play)
{
GstPlaySignal *signal;
g_return_if_fail (GST_IS_ELEMENT (element));
g_return_if_fail (GST_IS_PLAY (play));
g_return_if_fail (element == play->pipeline);
/*g_print ("got state change %s to %s\n", gst_element_state_get_name (old), gst_element_state_get_name (state));*/
/* do additional stuff depending on state */
if (GST_IS_PIPELINE (play->pipeline)){
switch (state) {
case GST_STATE_PLAYING:
play->idle_add_func ((GSourceFunc) gst_play_idle_callback, play);
play->timeout_add_func (200, (GSourceFunc) gst_play_tick_callback, play);
if (play->length_nanos == 0LL){
/* try to get the length up to 16 times */
play->get_length_attempt = 16;
play->timeout_add_func (200, (GSourceFunc) gst_play_get_length_callback, play);
}
break;
default:
break;
}
}
signal = g_new0(GstPlaySignal, 1);
signal->signal_id = STATE_CHANGE;
signal->signal_data.state.old_state = old;
signal->signal_data.state.new_state = state;
g_async_queue_push(play->signal_queue, signal);
play->idle_add_func ((GSourceFunc) gst_play_idle_signal, play);
}
static gboolean
gst_play_idle_signal (GstPlay *play)
{
GstPlaySignal *signal;
gint queue_length;
signal = g_async_queue_try_pop(play->signal_queue);
if (signal == NULL){
return FALSE;
}
switch (signal->signal_id){
case HAVE_XID:
g_signal_emit (G_OBJECT (play), gst_play_signals[HAVE_XID], 0,
signal->signal_data.video_xid.xid);
break;
case HAVE_VIDEO_SIZE:
g_signal_emit (G_OBJECT (play), gst_play_signals[HAVE_VIDEO_SIZE], 0,
signal->signal_data.video_size.width, signal->signal_data.video_size.height);
break;
case STATE_CHANGE:
g_signal_emit (G_OBJECT (play), gst_play_signals[STATE_CHANGE], 0,
signal->signal_data.state.old_state, signal->signal_data.state.new_state);
break;
case INFORMATION:
g_signal_emit (G_OBJECT (play), gst_play_signals[INFORMATION], 0,
signal->signal_data.info.element, signal->signal_data.info.param);
break;
default:
break;
}
g_free(signal);
queue_length = g_async_queue_length (play->signal_queue);
return (queue_length > 0);
}
static gboolean
gst_play_idle_eos (GstPlay* play)
{
g_signal_emit (G_OBJECT (play), gst_play_signals [STREAM_END], 0);
return FALSE;
}
static void
callback_audio_sink_eos (GstElement *element, GstPlay *play)
{
play->idle_add_func ((GSourceFunc) gst_play_idle_eos, play);
}
static void
callback_video_have_xid (GstElement *element, gint xid, GstPlay *play)
{
GstPlaySignal *signal;
signal = g_new0(GstPlaySignal, 1);
signal->signal_id = HAVE_XID;
signal->signal_data.video_xid.xid = xid;
g_async_queue_push(play->signal_queue, signal);
play->idle_add_func ((GSourceFunc) gst_play_idle_signal, play);
/*g_print("have xid %d\n", xid);*/
}
static void
callback_video_have_size (GstElement *element, gint width, gint height, GstPlay *play)
{
GstPlaySignal *signal;
signal = g_new0(GstPlaySignal, 1);
signal->signal_id = HAVE_VIDEO_SIZE;
signal->signal_data.video_size.width = width;
signal->signal_data.video_size.height = height;
g_async_queue_push(play->signal_queue, signal);
play->idle_add_func ((GSourceFunc) gst_play_idle_signal, play);
/*g_print("have size %d x %d\n", width, height);*/
}
static void
callback_bin_pre_iterate (GstBin *bin, GMutex *mutex)
{
g_mutex_lock(mutex);
}
static void
callback_bin_post_iterate (GstBin *bin, GMutex *mutex)
{
g_mutex_unlock(mutex);
}
static gboolean
gst_play_get_length_callback (GstPlay *play)
{
gint64 value;
GstFormat format = GST_FORMAT_TIME;
gboolean query_worked = FALSE;
g_print("trying to get length\n");
if (play->audio_sink_element != NULL){
g_mutex_lock(play->audio_bin_mutex);
query_worked = gst_element_query (play->audio_sink_element, GST_PAD_QUERY_TOTAL, &format, &value);
g_mutex_unlock(play->audio_bin_mutex);
}
else if (play->video_sink_element != NULL){
g_mutex_lock(play->video_bin_mutex);
query_worked = gst_element_query (play->video_sink_element, GST_PAD_QUERY_TOTAL, &format, &value);
g_mutex_unlock(play->video_bin_mutex);
}
if (query_worked){
g_print("got length %lld\n", value);
g_signal_emit (G_OBJECT (play), gst_play_signals [STREAM_LENGTH], 0, value);
play->length_nanos = value;
return FALSE;
}
else {
if (play->get_length_attempt-- < 1){
/* we've tried enough times, give up */
return FALSE;
}
}
return (gst_element_get_state(play->pipeline) == GST_STATE_PLAYING);
}
static gboolean
gst_play_tick_callback (GstPlay *play)
{
gint secs;
play->clock = gst_bin_get_clock (GST_BIN (play->pipeline));
play->time_nanos = gst_clock_get_time(play->clock);
secs = (gint) (play->time_nanos / GST_SECOND);
if (secs != play->time_seconds){
play->time_seconds = secs;
g_signal_emit (G_OBJECT (play), gst_play_signals [TIME_TICK], 0, play->time_nanos);
}
return (gst_element_get_state(play->pipeline) == GST_STATE_PLAYING);
}
static gboolean
gst_play_idle_callback (GstPlay *play)
{
return gst_bin_iterate (GST_BIN (play->pipeline));
}
static void
gst_play_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
{
GstPlay *play = GST_PLAY (object);
g_return_if_fail (GST_IS_PLAY (play));
switch (prop_id) {
case ARG_LOCATION:
gst_play_set_location(play, g_value_get_string (value));
break;
case ARG_VOLUME:
gst_play_set_volume(play, g_value_get_float (value));
break;
case ARG_MUTE:
gst_play_set_mute(play, g_value_get_boolean (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_play_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
{
GstPlay *play = GST_PLAY (object);
g_return_if_fail (GST_IS_PLAY (play));
switch (prop_id) {
case ARG_LOCATION:
g_value_set_string (value, gst_play_get_location(play));
break;
case ARG_VOLUME:
g_value_set_float(value, gst_play_get_volume(play));
break;
case ARG_MUTE:
g_value_set_boolean (value, gst_play_get_mute(play));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
void
gst_play_seek_to_time (GstPlay *play, gint64 time_nanos)
{
GstEvent *s_event;
gboolean audio_seek_worked = FALSE;
gboolean video_seek_worked = FALSE;
g_return_if_fail (GST_IS_PLAY (play));
if (time_nanos < 0LL){
play->seek_time = 0LL;
}
else if (time_nanos < 0LL){
play->seek_time = play->length_nanos;
}
else {
play->seek_time = time_nanos;
}
/*g_print("doing seek to %lld\n", play->seek_time);*/
gst_element_set_state(play->pipeline, GST_STATE_PAUSED);
s_event = gst_event_new_seek (GST_FORMAT_TIME |
GST_SEEK_METHOD_SET |
GST_SEEK_FLAG_FLUSH, play->seek_time);
if (play->audio_sink_element != NULL){
gst_event_ref (s_event);
audio_seek_worked = gst_element_send_event (play->audio_sink_element, s_event);
}
if (play->video_sink_element != NULL){
gst_event_ref (s_event);
video_seek_worked = gst_element_send_event (play->video_sink_element, s_event);
}
gst_event_unref (s_event);
if (audio_seek_worked || video_seek_worked){
play->time_nanos = gst_clock_get_time(play->clock);
g_signal_emit (G_OBJECT (play), gst_play_signals [TIME_TICK], 0, play->time_nanos);
}
gst_element_set_state(play->pipeline, GST_STATE_PLAYING);
}
void
gst_play_need_new_video_window(GstPlay *play)
{
g_return_if_fail (GST_IS_PLAY (play));
if (GST_IS_ELEMENT(play->video_sink_element)){
g_object_set(G_OBJECT(play->video_sink_element), "need_new_window", TRUE, NULL);
}
}
static gboolean
gst_play_default_idle (GstPlayIdleData *idle_data)
{
if(idle_data->func(idle_data->data)){
/* call this function again in the future */
return TRUE;
}
/* this function should no longer be called */
g_free(idle_data);
return FALSE;
}
static void
gst_play_default_timeout_add (guint interval, GSourceFunc function, gpointer data)
{
GstPlayIdleData *idle_data = g_new0(GstPlayIdleData, 1);
idle_data->func = function;
idle_data->data = data;
g_timeout_add (interval, (GSourceFunc)gst_play_default_idle, idle_data);
}
static void
gst_play_default_idle_add (GSourceFunc function, gpointer data)
{
GstPlayIdleData *idle_data = g_new0(GstPlayIdleData, 1);
idle_data->func = function;
idle_data->data = data;
g_idle_add ((GSourceFunc)gst_play_default_idle, idle_data);
}
void
gst_play_set_idle_timeout_funcs (GstPlay *play, GstPlayTimeoutAdd timeout_add_func, GstPlayIdleAdd idle_add_func)
{
g_return_if_fail (GST_IS_PLAY (play));
play->timeout_add_func = timeout_add_func;
play->idle_add_func = idle_add_func;
}
GstElement*
gst_play_get_sink_element (GstPlay *play, GstElement *element){
GstPad *pad = NULL;
GList *elements = NULL;
const GList *pads = NULL;
gboolean has_src;
g_return_val_if_fail (GST_IS_PLAY (play), NULL);
g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
if (!GST_IS_BIN(element)){
/* since its not a bin, we'll presume this
* element is a sink element */
return element;
}
elements = (GList *) gst_bin_get_list (GST_BIN(element));
/* traverse all elements looking for a src pad */
while (elements && pad == NULL) {
element = GST_ELEMENT (elements->data);
pads = gst_element_get_pad_list (element);
has_src = FALSE;
while (pads) {
/* check for src pad */
if (GST_PAD_DIRECTION (GST_PAD (pads->data)) == GST_PAD_SRC) {
has_src = TRUE;
break;
}
pads = g_list_next (pads);
}
if (!has_src){
return element;
}
elements = g_list_next (elements);
}
/* we didn't find a sink element */
return NULL;
}
gboolean
gst_play_set_video_sink (GstPlay *play, GstElement *video_sink)
{
g_return_val_if_fail (GST_IS_PLAY (play), FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (video_sink), FALSE);
if (gst_play_get_state (play) != GST_STATE_READY){
gst_play_set_state (play, GST_STATE_READY);
}
if (play->set_video_sink){
return play->set_video_sink(play, video_sink);
}
/* if there is no set_video_sink func, fail quietly */
return FALSE;
}
gboolean
gst_play_set_audio_sink (GstPlay *play, GstElement *audio_sink)
{
g_return_val_if_fail (GST_IS_PLAY (play), FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (audio_sink), FALSE);
if (gst_play_get_state (play) != GST_STATE_READY){
gst_play_set_state (play, GST_STATE_READY);
}
if (play->set_audio_sink){
return play->set_audio_sink(play, audio_sink);
}
/* if there is no set_audio_sink func, fail quietly */
return FALSE;
}
GstElementStateReturn
gst_play_set_state (GstPlay *play, GstElementState state)
{
g_return_val_if_fail (GST_IS_PLAY (play), GST_STATE_FAILURE);
g_return_val_if_fail (GST_IS_ELEMENT(play->pipeline), GST_STATE_FAILURE);
/*g_print("setting state to %d\n", state);*/
return gst_element_set_state(play->pipeline, state);
}
GstElementState
gst_play_get_state (GstPlay *play)
{
g_return_val_if_fail (GST_IS_PLAY (play), GST_STATE_FAILURE);
g_return_val_if_fail (play->pipeline, GST_STATE_FAILURE);
return gst_element_get_state(play->pipeline);
}
gboolean
gst_play_set_location (GstPlay *play, const gchar *location)
{
GstElementState current_state;
g_return_val_if_fail (GST_IS_PLAY (play), FALSE);
g_return_val_if_fail (location != NULL, FALSE);
current_state = gst_play_get_state (play);
if (current_state != GST_STATE_READY){
gst_play_set_state (play, GST_STATE_READY);
}
if (play->set_autoplugger){
if (! play->set_autoplugger(play, gst_element_factory_make ("spider", "autoplugger"))){
g_warning ("couldn't replace autoplugger\n");
return FALSE;
}
}
/* FIXME check for valid location (somehow) */
g_object_set (G_OBJECT (play->source), "location", location, NULL);
/* reset time/length values */
play->time_seconds = 0;
play->length_nanos = 0LL;
play->time_nanos = 0LL;
g_signal_emit (G_OBJECT (play), gst_play_signals [STREAM_LENGTH], 0, 0LL);
g_signal_emit (G_OBJECT (play), gst_play_signals [TIME_TICK], 0, 0LL);
play->need_stream_length = TRUE;
return TRUE;
}
gchar*
gst_play_get_location (GstPlay *play)
{
gchar* location;
g_return_val_if_fail (GST_IS_PLAY (play), NULL);
g_return_val_if_fail (GST_IS_ELEMENT(play->source), NULL);
g_object_get (G_OBJECT (play->source), "location", &location, NULL);
return location;
}
void
gst_play_set_volume (GstPlay *play, gfloat volume)
{
g_return_if_fail (GST_IS_PLAY (play));
g_object_set(G_OBJECT(play->vol_dparam), "value_float", volume, NULL);
}
gfloat
gst_play_get_volume (GstPlay *play)
{
gfloat volume;
g_return_val_if_fail (GST_IS_PLAY (play), 0);
g_object_get(G_OBJECT(play->vol_dparam), "value_float", &volume, NULL);
return volume;
}
void
gst_play_set_mute (GstPlay *play, gboolean mute)
{
g_return_if_fail (GST_IS_PLAY (play));
g_object_set (G_OBJECT (play->volume), "mute", mute, NULL);
}
gboolean
gst_play_get_mute (GstPlay *play)
{
gboolean mute;
g_return_val_if_fail (GST_IS_PLAY (play), 0);
g_object_get (G_OBJECT (play->volume), "mute", &mute, NULL);
return mute;
}
/* modelines */
/* vim:set ts=8:sw=8:noet */

176
gst-libs/gst/play/play.old.h Normal file
View file

@ -0,0 +1,176 @@
/* GStreamer
* Copyright (C) 1999,2000,2001,2002 Erik Walthinsen <omega@cse.ogi.edu>
* 2000,2001,2002 Wim Taymans <wtay@chello.be>
* 2002 Steve Baker <steve@stevebaker.org>
*
* play.h: GstPlay object code
*
*
* 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.
*/
#ifndef __GSTPLAY_H__
#define __GSTPLAY_H__
#include <gst/gst.h>
#include <gst/control/control.h>
/*
* GstPlay is a simple class for audio and video playback.
* It's job is to get the media (supplied by a URI) played.
* More specific it should get the media from source to the output elements.
* How that is done should not be relevant for developers using this class.
* A user using this class should not have to know very much about how
* GStreamer works, other than that it plays back media.
* Additionally it supplies signals to get information about the current
* playing state.
*/
typedef enum {
GST_PLAY_OK,
GST_PLAY_UNKNOWN_MEDIA,
GST_PLAY_CANNOT_PLAY,
GST_PLAY_ERROR,
} GstPlayReturn;
typedef enum {
GST_PLAY_PIPE_AUDIO,
GST_PLAY_PIPE_AUDIO_THREADED,
GST_PLAY_PIPE_AUDIO_HYPER_THREADED,
GST_PLAY_PIPE_VIDEO_THREADSAFE,
GST_PLAY_PIPE_VIDEO,
} GstPlayPipeType;
typedef enum {
GST_PLAY_ERROR_FAKESINK,
GST_PLAY_ERROR_THREAD,
GST_PLAY_ERROR_QUEUE,
GST_PLAY_ERROR_GNOMEVFSSRC,
GST_PLAY_ERROR_VOLUME,
GST_PLAY_ERROR_COLORSPACE,
GST_PLAY_ERROR_LAST,
} GstPlayError;
#define GST_PLAY_ERROR gst_play_error_quark ()
#define GST_TYPE_PLAY (gst_play_get_type())
#define GST_PLAY(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_PLAY, GstPlay))
#define GST_PLAY_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_PLAY, GstPlayClass))
#define GST_IS_PLAY(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_PLAY))
#define GST_IS_PLAY_CLASS(obj) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_PLAY))
#define GST_PLAY_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GTK_TYPE_PLAY, GstPlayClass))
typedef struct _GstPlay GstPlay;
typedef struct _GstPlayClass GstPlayClass;
typedef struct _GstPlayIdleData GstPlayIdleData;
typedef void (*GstPlayTimeoutAdd) (guint interval, GSourceFunc function, gpointer data);
typedef void (*GstPlayIdleAdd) (GSourceFunc function, gpointer data);
struct _GstPlay
{
GObject parent;
gboolean (*setup_pipeline) (GstPlay *play, GError **error);
void (*teardown_pipeline) (GstPlay *play);
gboolean (*set_autoplugger) (GstPlay *play, GstElement *autoplugger);
gboolean (*set_video_sink) (GstPlay *play, GstElement *videosink);
gboolean (*set_audio_sink) (GstPlay *play, GstElement *audiosink);
/* core elements */
GstElement *pipeline;
GstElement *volume;
GstElement *source;
GstElement *autoplugger;
GstElement *video_sink;
GstElement *video_sink_element;
GstElement *audio_sink;
GstElement *audio_sink_element;
GstDParamManager *vol_dpman;
GstDParam *vol_dparam;
GHashTable *other_elements;
GstClock *clock;
GMutex *audio_bin_mutex;
GMutex *video_bin_mutex;
gboolean need_stream_length;
gboolean need_seek;
gint time_seconds;
gint get_length_attempt;
gint64 seek_time;
gint64 time_nanos;
gint64 length_nanos;
GAsyncQueue *signal_queue;
GstPlayTimeoutAdd timeout_add_func;
GstPlayIdleAdd idle_add_func;
};
struct _GstPlayClass
{
GObjectClass parent_class;
/* signals */
void (*information) (GstPlay* play, GstElement* element, GParamSpec *param);
void (*state_changed) (GstPlay* play, GstElementState old_state, GstElementState new_state);
void (*stream_end) (GstPlay* play);
void (*time_tick) (GstPlay* play, gint64 time_nanos);
void (*stream_length) (GstPlay* play, gint64 length_nanos);
void (*have_xid) (GstPlay* play, gint xid);
void (*have_video_size) (GstPlay* play, gint width, gint height);
};
struct _GstPlayIdleData
{
GSourceFunc func;
gpointer data;
};
GType gst_play_get_type (void);
GstPlay* gst_play_new (GstPlayPipeType pipe_type, GError **error);
void gst_play_seek_to_time (GstPlay *play, gint64 time_nanos);
GstElement* gst_play_get_sink_element (GstPlay *play, GstElement *element);
gboolean gst_play_set_video_sink (GstPlay *play, GstElement *element);
gboolean gst_play_set_audio_sink (GstPlay *play, GstElement *element);
void gst_play_need_new_video_window (GstPlay *play);
GstElementStateReturn gst_play_set_state (GstPlay *play, GstElementState state);
GstElementState gst_play_get_state (GstPlay *play);
gboolean gst_play_set_location (GstPlay *play, const gchar *location);
gchar* gst_play_get_location (GstPlay *play);
void gst_play_set_volume (GstPlay *play, gfloat volume);
gfloat gst_play_get_volume (GstPlay *play);
void gst_play_set_mute (GstPlay *play, gboolean mute);
gboolean gst_play_get_mute (GstPlay *play);
void gst_play_set_idle_timeout_funcs (GstPlay *play, GstPlayTimeoutAdd timeout_add_func, GstPlayIdleAdd idle_add_func);
#endif /* __GSTPLAY_H__ */
/* modelines */
/* vim:set ts=8:sw=8:noet */

752
gst-libs/gst/play/playpipelines.c Normal file
View file

@ -0,0 +1,752 @@
/* GStreamer
* Copyright (C) 1999,2000,2001,2002 Erik Walthinsen <omega@cse.ogi.edu>
* 2000,2001,2002 Wim Taymans <wtay@chello.be>
* 2002 Steve Baker <steve@stevebaker.org>
*
* playpipelines.c: Set up pipelines for playback
*
* 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.
*/
/*
* GST_PLAY_PIPE_AUDIO_THREADED
* { gnomevfssrc ! spider ! volume ! osssink }
*/
static gboolean
gst_play_audiot_setup (GstPlay *play, GError **error)
{
g_return_val_if_fail (GST_IS_PLAY(play), FALSE);
/* creating gst_thread */
play->pipeline = gst_thread_new ("main_pipeline");
g_return_val_if_fail (GST_IS_THREAD (play->pipeline), FALSE);
/* create source element */
play->source = gst_element_factory_make ("gnomevfssrc", "source");
if (!play->source)
{
gst_play_error_plugin (GST_PLAY_ERROR_GNOMEVFSSRC, error);
return FALSE;
}
/* Adding element to bin */
gst_bin_add (GST_BIN (play->pipeline), play->source);
/* create audio elements */
play->volume = gst_element_factory_make ("volume", "volume");
if (!play->volume)
{
gst_play_error_plugin (GST_PLAY_ERROR_VOLUME, error);
return FALSE;
}
/* create audiosink.
FIXME : Should use gconf to choose the right one */
play->audio_sink = gst_element_factory_make ("osssink", "play_audio");
if (!play->audio_sink)
g_warning ("You need the osssink element to use this program.");
g_object_set (
G_OBJECT (play->audio_sink),
"fragment", 0x00180008, NULL);
g_signal_connect (
G_OBJECT (play->audio_sink), "eos",
G_CALLBACK (callback_audio_sink_eos), play);
gst_bin_add_many (
GST_BIN (play->pipeline), play->volume,
play->audio_sink, NULL);
gst_element_connect (play->volume, play->audio_sink);
gst_bin_set_pre_iterate_function(
GST_BIN (play->pipeline),
(GstBinPrePostIterateFunction) callback_bin_pre_iterate,
play->audio_bin_mutex);
gst_bin_set_post_iterate_function(
GST_BIN (play->pipeline),
(GstBinPrePostIterateFunction) callback_bin_post_iterate,
play->audio_bin_mutex);
return TRUE;
}
static gboolean
gst_play_audiot_set_audio (GstPlay *play, GstElement *audio_sink)
{
g_return_val_if_fail (GST_IS_PLAY(play), FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (audio_sink), FALSE);
if (play->audio_sink)
{
gst_element_disconnect (play->volume, play->audio_sink);
gst_bin_remove (GST_BIN (play->pipeline), play->audio_sink);
}
play->audio_sink = audio_sink;
gst_bin_add (GST_BIN (play->pipeline), play->audio_sink);
gst_element_connect (play->volume, play->audio_sink);
return TRUE;
}
static gboolean
gst_play_audiot_set_auto (GstPlay *play, GstElement *autoplugger)
{
g_return_val_if_fail (GST_IS_PLAY(play), FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (autoplugger), FALSE);
if (play->autoplugger){
/* we need to remove the existing autoplugger before creating a new one */
gst_element_disconnect (play->autoplugger, play->volume);
gst_element_disconnect (play->autoplugger, play->source);
gst_bin_remove (GST_BIN (play->pipeline), play->autoplugger);
}
play->autoplugger = autoplugger;
g_return_val_if_fail (play->autoplugger != NULL, FALSE);
gst_bin_add (GST_BIN (play->pipeline), play->autoplugger);
gst_element_connect (play->source, play->autoplugger);
gst_element_connect (play->autoplugger, play->volume);
return TRUE;
}
/*
* GST_PLAY_PIPE_AUDIO_HYPER_THREADED
* { gnomevfssrc ! spider ! { queue ! volume ! osssink } }
*/
static gboolean
gst_play_audioht_setup (GstPlay *play, GError **error)
{
GstElement *audio_thread, *audio_queue;
g_return_val_if_fail (GST_IS_PLAY(play), FALSE);
/*
play->pipeline = gst_thread_new ("main_pipeline");
g_return_val_if_fail (GST_IS_THREAD (play->pipeline), FALSE);
*/
/* creating pipeline */
play->pipeline = gst_pipeline_new ("main_pipeline");
g_return_val_if_fail (GST_IS_PIPELINE (play->pipeline), FALSE);
/* create source element */
play->source = gst_element_factory_make ("gnomevfssrc", "source");
if (!play->source)
{
gst_play_error_plugin (GST_PLAY_ERROR_GNOMEVFSSRC, error);
return FALSE;
}
/* Adding element to bin */
gst_bin_add (GST_BIN (play->pipeline), play->source);
/* create audio thread */
audio_thread = gst_thread_new ("audio_thread");
g_return_val_if_fail (GST_IS_THREAD (audio_thread), FALSE);
g_hash_table_insert(play->other_elements, "audio_thread", audio_thread);
/* create audio queue */
audio_queue = gst_element_factory_make ("queue", "audio_queue");
if (!audio_queue)
{
gst_play_error_plugin (GST_PLAY_ERROR_QUEUE, error);
return FALSE;
}
g_hash_table_insert(play->other_elements, "audio_queue", audio_queue);
/* create source element */
play->volume = gst_element_factory_make ("volume", "volume");
if (!play->volume)
{
gst_play_error_plugin (GST_PLAY_ERROR_VOLUME, error);
return FALSE;
}
/* create audiosink.
FIXME : Should use gconf to choose the right one */
play->audio_sink = gst_element_factory_make ("osssink", "play_audio");
if (!play->audio_sink)
g_warning ("You need the osssink element to use this program.\n");
g_object_set (G_OBJECT (play->audio_sink), "fragment", 0x00180008, NULL);
g_signal_connect (G_OBJECT (play->audio_sink), "eos",
G_CALLBACK (callback_audio_sink_eos), play);
gst_bin_add_many (
GST_BIN (audio_thread), audio_queue, play->volume,
play->audio_sink, NULL);
gst_element_connect_many (audio_queue, play->volume, play->audio_sink);
gst_element_add_ghost_pad (
audio_thread, gst_element_get_pad (audio_queue, "sink"),
"sink");
gst_bin_add (GST_BIN (play->pipeline), audio_thread);
gst_bin_set_pre_iterate_function(
GST_BIN (audio_thread),
(GstBinPrePostIterateFunction) callback_bin_pre_iterate,
play->audio_bin_mutex);
gst_bin_set_post_iterate_function(
GST_BIN (audio_thread),
(GstBinPrePostIterateFunction) callback_bin_post_iterate,
play->audio_bin_mutex);
return TRUE;
}
static gboolean
gst_play_audioht_set_audio (GstPlay *play, GstElement *audio_sink)
{
GstElement *audio_thread;
g_return_val_if_fail (GST_IS_PLAY(play), FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (audio_sink), FALSE);
audio_thread = g_hash_table_lookup(play->other_elements, "audio_thread");
if (play->audio_sink)
{
gst_element_disconnect (play->volume, play->audio_sink);
gst_bin_remove (GST_BIN (audio_thread), play->audio_sink);
}
play->audio_sink = audio_sink;
gst_bin_add (GST_BIN (audio_thread), play->audio_sink);
gst_element_connect (play->volume, play->audio_sink);
return TRUE;
}
static gboolean
gst_play_audioht_set_auto (GstPlay *play, GstElement *autoplugger)
{
GstElement *audio_thread;
g_return_val_if_fail (GST_IS_PLAY(play), FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (autoplugger), FALSE);
audio_thread = g_hash_table_lookup(play->other_elements, "audio_thread");
if (play->autoplugger){
/* we need to remove the existing autoplugger before creating a new one */
gst_element_disconnect (play->autoplugger, audio_thread);
gst_element_disconnect (play->autoplugger, play->source);
gst_bin_remove (GST_BIN (play->pipeline), play->autoplugger);
}
play->autoplugger = autoplugger;
g_return_val_if_fail (play->autoplugger != NULL, FALSE);
gst_bin_add (GST_BIN (play->pipeline), play->autoplugger);
gst_element_connect (play->source, play->autoplugger);
gst_element_connect (play->autoplugger, audio_thread);
return TRUE;
}
/*
* GST_PLAY_PIPE_VIDEO
* { gnomevfssrc ! spider ! { queue ! volume ! osssink }
* spider0.src2 ! { queue ! colorspace ! (videosink) } }
*/
static gboolean
gst_play_video_setup (GstPlay *play, GError **error)
{
GstElement *audio_bin, *audio_queue;
GstElement *video_queue, *video_bin;
GstElement *work_thread, *colorspace;
g_return_val_if_fail (GST_IS_PLAY(play), FALSE);
/* creating pipeline */
play->pipeline = gst_pipeline_new ("main_pipeline");
g_return_val_if_fail (GST_IS_PIPELINE (play->pipeline), FALSE);
/* creating work thread */
work_thread = gst_thread_new ("work_thread");
g_return_val_if_fail (GST_IS_THREAD (work_thread), FALSE);
g_hash_table_insert(play->other_elements, "work_thread", work_thread);
gst_bin_add (GST_BIN (play->pipeline), work_thread);
/* create source element */
play->source = gst_element_factory_make ("gnomevfssrc", "source");
if (!play->source)
{
gst_play_error_plugin (GST_PLAY_ERROR_GNOMEVFSSRC, error);
return FALSE;
}
gst_bin_add (GST_BIN (work_thread), play->source);
/* creating volume element */
play->volume = gst_element_factory_make ("volume", "volume");
if (!play->volume)
{
gst_play_error_plugin (GST_PLAY_ERROR_VOLUME, error);
return FALSE;
}
/* creating audio_sink element */
play->audio_sink = gst_element_factory_make ("fakesink", "fake_audio");
if (!play->audio_sink)
{
gst_play_error_plugin (GST_PLAY_ERROR_FAKESINK, error);
return FALSE;
}
play->audio_sink_element = NULL;
/* creating audio_queue element */
audio_queue = gst_element_factory_make ("queue", "audio_queue");
if (!audio_queue)
{
gst_play_error_plugin (GST_PLAY_ERROR_QUEUE, error);
return FALSE;
}
g_hash_table_insert (play->other_elements, "audio_queue", audio_queue);
/* creating audio thread */
audio_bin = gst_thread_new ("audio_bin");
if (!audio_bin)
{
gst_play_error_plugin (GST_PLAY_ERROR_THREAD, error);
return FALSE;
}
g_hash_table_insert (play->other_elements, "audio_bin", audio_bin);
/* setting up iterate functions */
gst_bin_set_pre_iterate_function (
GST_BIN (audio_bin),
(GstBinPrePostIterateFunction) callback_bin_pre_iterate,
play->audio_bin_mutex);
gst_bin_set_post_iterate_function (
GST_BIN (audio_bin),
(GstBinPrePostIterateFunction) callback_bin_post_iterate,
play->audio_bin_mutex);
/* adding all that stuff to bin */
gst_bin_add_many (
GST_BIN (audio_bin), audio_queue, play->volume,
play->audio_sink, NULL);
gst_element_connect_many (audio_queue, play->volume,
play->audio_sink, NULL);
gst_element_add_ghost_pad (
audio_bin,
gst_element_get_pad (audio_queue, "sink"),
"sink");
gst_bin_add (GST_BIN (work_thread), audio_bin);
/* create video elements */
play->video_sink = gst_element_factory_make ("fakesink", "fake_show");
if (!play->video_sink)
{
gst_play_error_plugin (GST_PLAY_ERROR_FAKESINK, error);
return FALSE;
}
play->video_sink_element = NULL;
video_queue = gst_element_factory_make ("queue", "video_queue");
if (!video_queue)
{
gst_play_error_plugin (GST_PLAY_ERROR_QUEUE, error);
return FALSE;
}
g_hash_table_insert (play->other_elements, "video_queue", video_queue);
colorspace = gst_element_factory_make ("colorspace", "colorspace");
if (!colorspace)
{
gst_play_error_plugin (GST_PLAY_ERROR_COLORSPACE, error);
return FALSE;
}
g_hash_table_insert (play->other_elements, "colorspace", colorspace);
video_bin = gst_thread_new ("video_bin");
if (!video_bin)
{
gst_play_error_plugin (GST_PLAY_ERROR_THREAD, error);
return FALSE;
}
g_hash_table_insert (play->other_elements, "video_bin", video_bin);
/* adding all that stuff to bin */
gst_bin_add_many (GST_BIN (video_bin), video_queue, colorspace,
play->video_sink, NULL);
gst_element_connect_many (video_queue, colorspace,
play->video_sink, NULL);
/* setting up iterate functions */
gst_bin_set_pre_iterate_function (
GST_BIN (video_bin),
(GstBinPrePostIterateFunction) callback_bin_pre_iterate,
play->video_bin_mutex);
gst_bin_set_post_iterate_function (
GST_BIN (video_bin),
(GstBinPrePostIterateFunction) callback_bin_post_iterate,
play->video_bin_mutex);
gst_element_add_ghost_pad (
video_bin, gst_element_get_pad (video_queue, "sink"),
"sink");
gst_bin_add (GST_BIN (work_thread), video_bin);
return TRUE;
}
static gboolean
gst_play_video_set_auto (GstPlay *play, GstElement *autoplugger){
GstElement *audio_bin, *video_bin, *work_thread;
g_return_val_if_fail (GST_IS_PLAY(play), FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (autoplugger), FALSE);
audio_bin = g_hash_table_lookup(play->other_elements, "audio_bin");
video_bin = g_hash_table_lookup(play->other_elements, "video_bin");
work_thread = g_hash_table_lookup(play->other_elements, "work_thread");
if (play->autoplugger){
/* we need to remove the existing autoplugger before creating a new one */
gst_element_disconnect (play->autoplugger, audio_bin);
gst_element_disconnect (play->autoplugger, play->source);
gst_element_disconnect (play->autoplugger, video_bin);
gst_bin_remove (GST_BIN (work_thread), play->autoplugger);
}
play->autoplugger = autoplugger;
g_return_val_if_fail (play->autoplugger != NULL, FALSE);
gst_bin_add (GST_BIN (work_thread), play->autoplugger);
gst_element_connect (play->source, play->autoplugger);
gst_element_connect (play->autoplugger, audio_bin);
gst_element_connect (play->autoplugger, video_bin);
return TRUE;
}
static gboolean
gst_play_video_set_video (GstPlay *play, GstElement *video_sink)
{
GstElement *video_mate, *video_bin;
g_return_val_if_fail (GST_IS_PLAY(play), FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (video_sink), FALSE);
video_bin = g_hash_table_lookup(play->other_elements, "video_bin");
video_mate = g_hash_table_lookup(play->other_elements, "colorspace");
if (play->video_sink){
gst_element_disconnect (video_mate, play->video_sink);
gst_bin_remove (GST_BIN (video_bin), play->video_sink);
}
play->video_sink = video_sink;
gst_bin_add (GST_BIN (video_bin), play->video_sink);
gst_element_connect (video_mate, play->video_sink);
play->video_sink_element = gst_play_get_sink_element (play, video_sink);
if (play->video_sink_element != NULL){
g_signal_connect (G_OBJECT (play->video_sink_element), "have_xid",
G_CALLBACK (callback_video_have_xid), play);
g_signal_connect (G_OBJECT (play->video_sink_element), "have_size",
G_CALLBACK (callback_video_have_size), play);
g_object_set(G_OBJECT(play->video_sink_element), "need_new_window", TRUE, "toplevel", FALSE, NULL);
}
return TRUE;
}
static gboolean
gst_play_video_set_audio (GstPlay *play, GstElement *audio_sink)
{
GstElement *audio_bin;
g_return_val_if_fail (GST_IS_PLAY(play), FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (audio_sink), FALSE);
audio_bin = g_hash_table_lookup(play->other_elements, "audio_bin");
if (play->audio_sink)
{
gst_element_disconnect (play->volume, play->audio_sink);
gst_bin_remove (GST_BIN (audio_bin), play->audio_sink);
}
play->audio_sink = audio_sink;
gst_bin_add (GST_BIN (audio_bin), play->audio_sink);
gst_element_connect (play->volume, play->audio_sink);
play->audio_sink_element = gst_play_get_sink_element (play, audio_sink);
if (play->audio_sink_element != NULL){
g_signal_connect (G_OBJECT (play->audio_sink), "eos",
G_CALLBACK (callback_audio_sink_eos), play);
}
return TRUE;
}
/*
* GST_PLAY_PIPE_VIDEO_THREADSAFE
* { gnomevfssrc ! spider ! { queue ! volume ! osssink } }
* spider0.src2 ! queue ! videosink
* (note that the xvideosink is not contained by a thread)
*/
static gboolean
gst_play_videots_setup (GstPlay *play, GError **error)
{
GstElement *audio_bin, *audio_queue, *video_queue, *auto_identity, *work_thread;
g_return_val_if_fail (GST_IS_PLAY(play), FALSE);
/* creating pipeline */
play->pipeline = gst_pipeline_new ("main_pipeline");
g_return_val_if_fail (GST_IS_PIPELINE (play->pipeline), FALSE);
/* creating work thread */
work_thread = gst_thread_new ("work_thread");
g_return_val_if_fail (GST_IS_THREAD (work_thread), FALSE);
g_hash_table_insert(play->other_elements, "work_thread", work_thread);
gst_bin_add (GST_BIN (play->pipeline), work_thread);
/* create source element */
play->source = gst_element_factory_make ("gnomevfssrc", "source");
if (!play->source)
{
gst_play_error_plugin (GST_PLAY_ERROR_GNOMEVFSSRC, error);
return FALSE;
}
gst_bin_add (GST_BIN (work_thread), play->source);
auto_identity = gst_element_factory_make ("identity", "auto_identity");
g_return_val_if_fail (auto_identity != NULL, FALSE);
g_hash_table_insert(play->other_elements, "auto_identity", auto_identity);
gst_bin_add (GST_BIN (work_thread), auto_identity);
gst_element_add_ghost_pad (work_thread,
gst_element_get_pad (auto_identity, "src"),
"src");
/* create volume elements */
play->volume = gst_element_factory_make ("volume", "volume");
if (!play->volume)
{
gst_play_error_plugin (GST_PLAY_ERROR_VOLUME, error);
return FALSE;
}
/* create audiosink.
FIXME : Should use gconf to choose the right one */
play->audio_sink = gst_element_factory_make ("osssink", "play_audio");
if (!play->audio_sink)
g_warning ("You need the osssink element to use this program.\n");
g_object_set (G_OBJECT (play->audio_sink), "fragment", 0x00180008, NULL);
g_signal_connect (
G_OBJECT (play->audio_sink), "eos",
G_CALLBACK (callback_audio_sink_eos), play);
audio_queue = gst_element_factory_make ("queue", "audio_queue");
if (!audio_queue)
{
gst_play_error_plugin (GST_PLAY_ERROR_QUEUE, error);
return FALSE;
}
g_hash_table_insert(play->other_elements, "audio_queue", audio_queue);
audio_bin = gst_thread_new ("audio_bin");
if (!audio_bin)
{
gst_play_error_plugin (GST_PLAY_ERROR_THREAD, error);
return FALSE;
}
g_hash_table_insert(play->other_elements, "audio_bin", audio_bin);
gst_bin_set_pre_iterate_function(
GST_BIN (audio_bin),
(GstBinPrePostIterateFunction) callback_bin_pre_iterate,
play->audio_bin_mutex);
gst_bin_set_post_iterate_function(
GST_BIN (audio_bin),
(GstBinPrePostIterateFunction) callback_bin_post_iterate,
play->audio_bin_mutex);
gst_bin_add_many (
GST_BIN (audio_bin), audio_queue,
play->volume, play->audio_sink, NULL);
gst_element_connect_many (
audio_queue, play->volume,
play->audio_sink, NULL);
gst_element_add_ghost_pad (
audio_bin,
gst_element_get_pad (audio_queue, "sink"),
"sink");
gst_bin_add (GST_BIN (work_thread), audio_bin);
/* create video elements */
play->video_sink = gst_element_factory_make ("xvideosink", "show");
g_object_set (G_OBJECT (play->video_sink), "toplevel", FALSE, NULL);
g_signal_connect (
G_OBJECT (play->video_sink), "have_xid",
G_CALLBACK (callback_video_have_xid), play);
g_signal_connect (
G_OBJECT (play->video_sink), "have_size",
G_CALLBACK (callback_video_have_size), play);
g_return_val_if_fail (play->video_sink != NULL, FALSE);
video_queue = gst_element_factory_make ("queue", "video_queue");
g_return_val_if_fail (video_queue != NULL, FALSE);
g_hash_table_insert(play->other_elements, "video_queue", video_queue);
g_object_set (G_OBJECT (video_queue), "block_timeout", 1000, NULL);
gst_bin_add_many (
GST_BIN (play->pipeline), video_queue,
play->video_sink, NULL);
gst_element_connect (video_queue, play->video_sink);
gst_bin_set_pre_iterate_function(
GST_BIN (play->pipeline),
(GstBinPrePostIterateFunction) callback_bin_pre_iterate,
play->video_bin_mutex);
gst_bin_set_post_iterate_function(
GST_BIN (play->pipeline),
(GstBinPrePostIterateFunction) callback_bin_post_iterate,
play->video_bin_mutex);
gst_element_connect (work_thread, video_queue);
return TRUE;
}
static gboolean
gst_play_videots_set_auto (GstPlay *play, GstElement *autoplugger){
GstElement *audio_bin, *auto_identity, *work_thread;
g_return_val_if_fail (GST_IS_PLAY(play), FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (autoplugger), FALSE);
audio_bin = g_hash_table_lookup(play->other_elements, "audio_bin");
auto_identity = g_hash_table_lookup(play->other_elements, "auto_identity");
work_thread = g_hash_table_lookup(play->other_elements, "work_thread");
if (play->autoplugger){
/* we need to remove the existing autoplugger before creating a new one */
gst_element_disconnect (play->autoplugger, audio_bin);
gst_element_disconnect (play->autoplugger, play->source);
gst_element_disconnect (play->autoplugger, auto_identity);
gst_bin_remove (GST_BIN (work_thread), play->autoplugger);
}
play->autoplugger = autoplugger;
g_return_val_if_fail (play->autoplugger != NULL, FALSE);
gst_bin_add (GST_BIN (work_thread), play->autoplugger);
gst_element_connect (play->source, play->autoplugger);
gst_element_connect (play->autoplugger, audio_bin);
gst_element_connect (play->autoplugger, auto_identity);
return TRUE;
}
static gboolean
gst_play_videots_set_video (GstPlay *play, GstElement *video_sink)
{
GstElement *video_mate;
g_return_val_if_fail (GST_IS_PLAY(play), FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (video_sink), FALSE);
video_mate = g_hash_table_lookup(play->other_elements, "video_queue");
if (play->video_sink){
gst_element_disconnect (video_mate, play->video_sink);
gst_bin_remove (GST_BIN (play->pipeline), play->video_sink);
}
play->video_sink = video_sink;
gst_bin_add (GST_BIN (play->pipeline), play->video_sink);
gst_element_connect (video_mate, play->video_sink);
return TRUE;
}
static gboolean
gst_play_videots_set_audio (GstPlay *play, GstElement *audio_sink)
{
GstElement *audio_bin;
g_return_val_if_fail (GST_IS_PLAY(play), FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (audio_sink), FALSE);
audio_bin = g_hash_table_lookup(play->other_elements, "audio_bin");
if (play->audio_sink)
{
gst_element_disconnect (play->volume, play->audio_sink);
gst_bin_remove (GST_BIN (audio_bin), play->audio_sink);
}
play->audio_sink = audio_sink;
gst_bin_add (GST_BIN (audio_bin), play->audio_sink);
gst_element_connect (play->volume, play->audio_sink);
return TRUE;
}
/* modelines */
/* vim:set ts=8:sw=8:noet */