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gstreamer/gst-libs/gst/play/play.c
Thomas Vander Stichele 9d0344af8c don't use stupid colorspace, do use hermes, make macro, mark for translation 
Original commit message from CVS:
don't use stupid colorspace, do use hermes, make macro, mark for translation
2004-07-30 13:41:55 +00:00

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/* GStreamer
* Copyright (C) 2003 Julien Moutte <julien@moutte.net>
*
* 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 <string.h>
#include "play.h"
#include <gst/gst-i18n-plugin.h>
#define TICK_INTERVAL_MSEC 200
GST_DEBUG_CATEGORY_STATIC (play_debug);
#define GST_CAT_DEFAULT play_debug
enum
{
TIME_TICK,
STREAM_LENGTH,
HAVE_VIDEO_SIZE,
LAST_SIGNAL
};
struct _GstPlayPrivate
{
char *location;
GHashTable *elements;
gint64 time_nanos;
gint64 length_nanos;
gint get_length_attempt;
gint tick_unblock_remaining; /* how many msecs left
to unblock due to seeking */
guint tick_id;
guint length_id;
gulong handoff_hid;
/* error/debug handling */
GError *error;
gchar *debug;
};
static guint gst_play_signals[LAST_SIGNAL] = { 0 };
static GstPipelineClass *parent_class = NULL;
/* ======================================================= */
/* */
/* Private Methods */
/* */
/* ======================================================= */
static GstCaps *gst_play_video_fixate (GstPad * pad, const GstCaps * caps,
gpointer user_data);
static GstCaps *gst_play_audio_fixate (GstPad * pad, const GstCaps * caps,
gpointer user_data);
static 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;
}
/* General GError creation */
static void
gst_play_error_create (GError ** error, const gchar * message)
{
/* check if caller wanted an error reported */
if (error == NULL)
return;
*error = g_error_new (GST_PLAY_ERROR, 0, message);
return;
}
/* GError creation when plugin is missing */
/* FIXME: what if multiple elements could have been used and they're all
* missing ? varargs ? */
static void
gst_play_error_plugin (const gchar * element, GError ** error)
{
gchar *message;
message = g_strdup_printf (_("The %s element could not be found. "
"This element is essential for playback. "
"Please install the right plug-in and verify "
"that it works by running 'gst-inspect %s'"), element, element);
gst_play_error_create (error, message);
g_free (message);
return;
}
#define GST_PLAY_MAKE_OR_ERROR(el, factory, name, error) \
G_STMT_START { \
el = gst_element_factory_make (factory, name); \
if (!GST_IS_ELEMENT (el)) { \
gst_play_error_plugin (factory, error); \
return FALSE; \
} \
} G_STMT_END
/* Create a colorspace element from the list of acceptable ones;
* set error and fail if none found. */
#define GST_PLAY_MAKE_CS_OR_ERROR(el, name, error) \
G_STMT_START { \
el = gst_element_factory_make ("ffmpegcolorspace", name); \
if (!GST_IS_ELEMENT (el)) \
el = gst_element_factory_make ("ffcolorspace", name); \
if (!GST_IS_ELEMENT (el)) \
el = gst_element_factory_make ("hermescolorspace", name); \
if (!GST_IS_ELEMENT (el)) { \
gst_play_error_create (error, \
_("No usable colorspace element could be found.\n" \
"Please install one and restart.")); \
return FALSE; \
} \
} G_STMT_END
#define GST_PLAY_ERROR_RETURN(error, message) \
G_STMT_START { \
gst_play_error_create (error, message); \
return FALSE; \
} G_STMT_END
#define GST_PLAY_HASH_LOOKUP(element, key, retval_if_fail) \
G_STMT_START { \
(element) = g_hash_table_lookup (play->priv->elements, (key));\
if (!element) \
return (retval_if_fail); \
} G_STMT_END
/* setup parts of the pipeline
* only put decoding part in the thread
* create all others and keep them around
*/
static gboolean
gst_play_pipeline_setup (GstPlay * play, GError ** error)
{
/* Threads */
GstElement *work_thread, *audio_thread, *video_thread;
/* output bin */
GstElement *output_bin;
/* Main Thread elements */
GstElement *source, *autoplugger;
/* output bin elements */
GstElement *audioconvert, *volume, *tee, *identity;
GstElement *identity_cs;
/* Visualization bin */
GstElement *vis_bin, *vis_queue, *vis_element, *vis_cs;
/* Video Thread elements */
GstElement *video_queue, *video_switch, *video_cs, *video_balance;
GstElement *balance_cs, *video_scaler, *video_sink;
/* Audio Thread elements */
GstElement *audio_queue, *audio_sink;
/* Some useful pads */
GstPad *tee_pad1, *tee_pad2;
g_return_val_if_fail (play != NULL, FALSE);
g_return_val_if_fail (GST_IS_PLAY (play), FALSE);
GST_DEBUG_OBJECT (play, "setting up pipeline");
/* Creating main thread and its elements */
{
GST_PLAY_MAKE_OR_ERROR (work_thread, "thread", "work_thread", error);
g_hash_table_insert (play->priv->elements, "work_thread", work_thread);
gst_bin_add (GST_BIN (play), work_thread);
/* Placeholder for datasrc */
GST_PLAY_MAKE_OR_ERROR (source, "fakesrc", "source", error);
g_hash_table_insert (play->priv->elements, "source", source);
/* Autoplugger */
GST_PLAY_MAKE_OR_ERROR (autoplugger, "spider", "autoplugger", error);
g_hash_table_insert (play->priv->elements, "autoplugger", autoplugger);
/* adding and linking */
gst_bin_add_many (GST_BIN (work_thread), source, autoplugger, NULL);
if (!gst_element_link (source, autoplugger))
GST_PLAY_ERROR_RETURN (error, "Could not link source and autoplugger");
}
/* output bin */
{
GST_PLAY_MAKE_OR_ERROR (output_bin, "bin", "output_bin", error);
g_hash_table_insert (play->priv->elements, "output_bin", output_bin);
/* Make sure we convert audio to the needed format */
GST_PLAY_MAKE_OR_ERROR (audioconvert, "audioconvert", "audioconvert",
error);
g_hash_table_insert (play->priv->elements, "audioconvert", audioconvert);
/* Duplicate audio signal to audio sink and visualization thread */
GST_PLAY_MAKE_OR_ERROR (tee, "tee", "tee", error);
tee_pad1 = gst_element_get_request_pad (tee, "src%d");
tee_pad2 = gst_element_get_request_pad (tee, "src%d");
g_hash_table_insert (play->priv->elements, "tee_pad1", tee_pad1);
g_hash_table_insert (play->priv->elements, "tee_pad2", tee_pad2);
g_hash_table_insert (play->priv->elements, "tee", tee);
gst_bin_add_many (GST_BIN (output_bin), audioconvert, tee, NULL);
if (!gst_element_link (audioconvert, tee))
GST_PLAY_ERROR_RETURN (error, "Could not link audio thread elements");
/* identity ! colorspace ! switch */
GST_PLAY_MAKE_OR_ERROR (identity, "identity", "identity", error);
g_hash_table_insert (play->priv->elements, "identity", identity);
GST_PLAY_MAKE_CS_OR_ERROR (identity_cs, "identity_cs", error);
g_hash_table_insert (play->priv->elements, "identity_cs", identity_cs);
gst_bin_add_many (GST_BIN (output_bin), identity, identity_cs, NULL);
if (!gst_element_link (identity, identity_cs))
GST_PLAY_ERROR_RETURN (error, "Could not link work thread elements");
/* we ref the output bin so we can put it in and out the work_thread
* whenever we want */
gst_object_ref (GST_OBJECT (output_bin));
GST_DEBUG_OBJECT (play, "adding output bin to work thread in setup");
gst_bin_add (GST_BIN (work_thread), output_bin);
}
/* Visualization bin (note: it s not added to the pipeline yet) */
{
vis_bin = gst_bin_new ("vis_bin");
if (!GST_IS_ELEMENT (vis_bin)) {
gst_play_error_plugin ("bin", error);
return FALSE;
}
g_hash_table_insert (play->priv->elements, "vis_bin", vis_bin);
/* Buffer queue for video data */
GST_PLAY_MAKE_OR_ERROR (vis_queue, "queue", "vis_queue", error);
g_hash_table_insert (play->priv->elements, "vis_queue", vis_queue);
/* Visualization element placeholder */
GST_PLAY_MAKE_OR_ERROR (vis_element, "identity", "vis_element", error);
g_hash_table_insert (play->priv->elements, "vis_element", vis_element);
/* Colorspace conversion */
GST_PLAY_MAKE_CS_OR_ERROR (vis_cs, "vis_cs", error);
g_hash_table_insert (play->priv->elements, "vis_cs", vis_cs);
gst_bin_add_many (GST_BIN (vis_bin), vis_queue, vis_element, vis_cs, NULL);
if (!gst_element_link_many (vis_queue, vis_element, vis_cs, NULL))
GST_PLAY_ERROR_RETURN (error,
"Could not link visualisation thread elements");
gst_element_add_ghost_pad (vis_bin, gst_element_get_pad (vis_cs, "src"),
"src");
}
/* Creating our video output bin */
{
GST_PLAY_MAKE_OR_ERROR (video_thread, "thread", "video_thread", error);
g_hash_table_insert (play->priv->elements, "video_thread", video_thread);
gst_bin_add (GST_BIN (output_bin), video_thread);
/* Buffer queue for video data */
GST_PLAY_MAKE_OR_ERROR (video_queue, "queue", "video_queue", error);
g_hash_table_insert (play->priv->elements, "video_queue", video_queue);
GST_PLAY_MAKE_OR_ERROR (video_switch, "switch", "video_switch", error);
g_hash_table_insert (play->priv->elements, "video_switch", video_switch);
/* Colorspace conversion */
GST_PLAY_MAKE_CS_OR_ERROR (video_cs, "video_cs", error);
g_hash_table_insert (play->priv->elements, "video_cs", video_cs);
/* Software colorbalance */
GST_PLAY_MAKE_OR_ERROR (video_balance, "videobalance", "video_balance",
error);
g_hash_table_insert (play->priv->elements, "video_balance", video_balance);
/* Colorspace conversion */
GST_PLAY_MAKE_CS_OR_ERROR (balance_cs, "balance_cs", error);
g_hash_table_insert (play->priv->elements, "balance_cs", balance_cs);
/* Software scaling of video stream */
GST_PLAY_MAKE_OR_ERROR (video_scaler, "videoscale", "video_scaler", error);
g_hash_table_insert (play->priv->elements, "video_scaler", video_scaler);
g_signal_connect (gst_element_get_pad (video_scaler, "src"), "fixate",
G_CALLBACK (gst_play_video_fixate), play);
/* Placeholder for future video sink bin */
GST_PLAY_MAKE_OR_ERROR (video_sink, "fakesink", "video_sink", error);
g_hash_table_insert (play->priv->elements, "video_sink", video_sink);
gst_bin_add_many (GST_BIN (video_thread), video_queue, video_switch,
video_cs, video_balance, balance_cs, video_scaler, video_sink, NULL);
/* break down linking so we can figure out what might be failing */
if (!gst_element_link (video_queue, video_switch))
GST_PLAY_ERROR_RETURN (error,
"Could not link video output thread (queue and switch)");
if (!gst_element_link (video_switch, video_cs))
GST_PLAY_ERROR_RETURN (error,
"Could not link video output thread (switch and cs)");
if (!gst_element_link (video_cs, video_balance))
GST_PLAY_ERROR_RETURN (error,
"Could not link video output thread (cs and balance)");
if (!gst_element_link (video_balance, balance_cs))
GST_PLAY_ERROR_RETURN (error,
"Could not link video output thread (balance and balance_cs)");
if (!gst_element_link (balance_cs, video_scaler))
GST_PLAY_ERROR_RETURN (error,
"Could not link video output thread (balance_cs and scaler)");
if (!gst_element_link (video_scaler, video_sink))
GST_PLAY_ERROR_RETURN (error,
"Could not link video output thread (balance_cs and scaler)");
gst_element_add_ghost_pad (video_thread, gst_element_get_pad (video_queue,
"sink"), "sink");
if (!gst_element_link (identity_cs, video_thread))
GST_PLAY_ERROR_RETURN (error,
"Could not link video output thread elements");
}
/* Creating our audio output bin
{ queue ! fakesink } */
{
GST_PLAY_MAKE_OR_ERROR (audio_thread, "thread", "audio_thread", error);
g_hash_table_insert (play->priv->elements, "audio_thread", audio_thread);
gst_bin_add (GST_BIN (output_bin), audio_thread);
/* Buffer queue for our audio thread */
GST_PLAY_MAKE_OR_ERROR (audio_queue, "queue", "audio_queue", error);
g_hash_table_insert (play->priv->elements, "audio_queue", audio_queue);
/* Volume control */
GST_PLAY_MAKE_OR_ERROR (volume, "volume", "volume", error);
g_hash_table_insert (play->priv->elements, "volume", volume);
g_signal_connect (gst_element_get_pad (volume, "src"), "fixate",
G_CALLBACK (gst_play_audio_fixate), play);
/* Placeholder for future audio sink bin */
GST_PLAY_MAKE_OR_ERROR (audio_sink, "fakesink", "audio_sink", error);
g_hash_table_insert (play->priv->elements, "audio_sink", audio_sink);
gst_bin_add_many (GST_BIN (audio_thread), audio_queue, volume, audio_sink,
NULL);
if (!gst_element_link_many (audio_queue, volume, audio_sink, NULL))
GST_PLAY_ERROR_RETURN (error,
"Could not link audio output thread elements");
gst_element_add_ghost_pad (audio_thread, gst_element_get_pad (audio_queue,
"sink"), "sink");
gst_pad_link (tee_pad2, gst_element_get_pad (audio_queue, "sink"));
}
GST_DEBUG_OBJECT (play, "setting up pipeline succeeded.");
return TRUE;
}
static void
gst_play_have_video_size (GstElement * element, gint width,
gint height, GstPlay * play)
{
g_return_if_fail (play != NULL);
g_return_if_fail (GST_IS_PLAY (play));
g_signal_emit (G_OBJECT (play), gst_play_signals[HAVE_VIDEO_SIZE],
0, width, height);
}
static gboolean
gst_play_tick_callback (GstPlay * play)
{
GstFormat format = GST_FORMAT_TIME;
gboolean q = FALSE;
GstElement *audio_sink_element = NULL;
g_return_val_if_fail (play != NULL, FALSE);
/* just return without updating the UI when we are in the middle of seeking */
if (play->priv->tick_unblock_remaining > 0) {
play->priv->tick_unblock_remaining -= TICK_INTERVAL_MSEC;
return TRUE;
}
if (!GST_IS_PLAY (play)) {
play->priv->tick_id = 0;
return FALSE;
}
audio_sink_element = g_hash_table_lookup (play->priv->elements,
"audio_sink_element");
if (!GST_IS_ELEMENT (audio_sink_element)) {
play->priv->tick_id = 0;
return FALSE;
}
q = gst_element_query (audio_sink_element, GST_QUERY_POSITION, &format,
&(play->priv->time_nanos));
if (q)
g_signal_emit (G_OBJECT (play), gst_play_signals[TIME_TICK],
0, play->priv->time_nanos);
if (GST_STATE (GST_ELEMENT (play)) == GST_STATE_PLAYING)
return TRUE;
else {
play->priv->tick_id = 0;
return FALSE;
}
}
static gboolean
gst_play_get_length_callback (GstPlay * play)
{
GstElement *audio_sink_element, *video_sink_element;
GstFormat format = GST_FORMAT_TIME;
gint64 value;
gboolean q = FALSE;
g_return_val_if_fail (play != NULL, FALSE);
g_return_val_if_fail (GST_IS_PLAY (play), FALSE);
GST_DEBUG_OBJECT (play, "trying to get length");
/* We try to get length from all real sink elements */
audio_sink_element = g_hash_table_lookup (play->priv->elements,
"audio_sink_element");
video_sink_element = g_hash_table_lookup (play->priv->elements,
"video_sink_element");
if (!GST_IS_ELEMENT (audio_sink_element) &&
!GST_IS_ELEMENT (video_sink_element)) {
play->priv->length_id = 0;
return FALSE;
}
/* Audio first and then Video */
if (GST_IS_ELEMENT (audio_sink_element)) {
GST_DEBUG_OBJECT (play, "querying for length on audio sink");
q = gst_element_query (audio_sink_element, GST_QUERY_TOTAL, &format,
&value);
} else
GST_DEBUG_OBJECT (play, "no audio sink element");
if (!q) {
GST_DEBUG_OBJECT (play, "no query result from audio sink");
if (GST_IS_ELEMENT (video_sink_element)) {
GST_DEBUG_OBJECT (play, "querying for length on video sink");
q = gst_element_query (video_sink_element, GST_QUERY_TOTAL, &format,
&value);
}
}
if (q) {
play->priv->length_nanos = value;
GST_DEBUG_OBJECT (play, "got length, %" GST_TIME_FORMAT,
GST_TIME_ARGS ((GstClockTime) value));
g_signal_emit (G_OBJECT (play), gst_play_signals[STREAM_LENGTH],
0, play->priv->length_nanos);
play->priv->length_id = 0;
return FALSE;
}
play->priv->get_length_attempt++;
GST_DEBUG_OBJECT (play, "no length yet, was attempt %d",
play->priv->get_length_attempt);
/* We try 16 times */
if (play->priv->get_length_attempt > 15) {
play->priv->length_id = 0;
return FALSE;
} else
return TRUE;
}
static GstCaps *
gst_play_video_fixate (GstPad * pad, const GstCaps * caps, gpointer user_data)
{
GstStructure *structure;
GstCaps *newcaps;
GST_DEBUG ("video fixate %p %" GST_PTR_FORMAT, pad, caps);
if (gst_caps_get_size (caps) > 1)
return NULL;
newcaps = gst_caps_copy (caps);
structure = gst_caps_get_structure (newcaps, 0);
if (gst_structure_has_field (structure, "width") &&
gst_caps_structure_fixate_field_nearest_int (structure, "width", 320)) {
return newcaps;
}
if (gst_structure_has_field (structure, "height") &&
gst_caps_structure_fixate_field_nearest_int (structure, "height", 240)) {
return newcaps;
}
if (gst_structure_has_field (structure, "framerate") &&
gst_caps_structure_fixate_field_nearest_double (structure, "framerate",
30.0)) {
return newcaps;
}
/* failed to fixate */
gst_caps_free (newcaps);
return NULL;
}
static GstCaps *
gst_play_audio_fixate (GstPad * pad, const GstCaps * caps, gpointer user_data)
{
GstCaps *newcaps;
GstStructure *structure;
GST_DEBUG ("audio fixate %p %" GST_PTR_FORMAT, pad, caps);
newcaps =
gst_caps_new_full (gst_structure_copy (gst_caps_get_structure (caps, 0)),
NULL);
structure = gst_caps_get_structure (newcaps, 0);
if (gst_structure_has_field (structure, "rate") &&
gst_caps_structure_fixate_field_nearest_int (structure, "rate", 44100)) {
return newcaps;
}
if (gst_structure_has_field (structure, "depth") &&
gst_caps_structure_fixate_field_nearest_int (structure, "depth", 16)) {
return newcaps;
}
if (gst_structure_has_field (structure, "width") &&
gst_caps_structure_fixate_field_nearest_int (structure, "width", 16)) {
return newcaps;
}
if (gst_structure_has_field (structure, "channels") &&
gst_caps_structure_fixate_field_nearest_int (structure, "channels", 2)) {
return newcaps;
}
gst_caps_free (newcaps);
return NULL;
}
/* this is a signal handler because we want this called AFTER the state
* change has passed. FIXME: core should rename signal to state-changed
* to make this clear. */
static void
gst_play_state_change (GstElement * element, GstElementState old,
GstElementState state)
{
GstPlay *play;
g_return_if_fail (element != NULL);
g_return_if_fail (GST_IS_PLAY (element));
play = GST_PLAY (element);
if (state == GST_STATE_PLAYING) {
if (play->priv->tick_id) {
g_source_remove (play->priv->tick_id);
play->priv->tick_id = 0;
}
play->priv->tick_id = g_timeout_add (TICK_INTERVAL_MSEC,
(GSourceFunc) gst_play_tick_callback, play);
play->priv->get_length_attempt = 0;
if (play->priv->length_id) {
g_source_remove (play->priv->length_id);
play->priv->length_id = 0;
}
play->priv->length_id = g_timeout_add (TICK_INTERVAL_MSEC,
(GSourceFunc) gst_play_get_length_callback, play);
} else {
if (play->priv->tick_id) {
g_source_remove (play->priv->tick_id);
play->priv->tick_id = 0;
}
if (play->priv->length_id) {
g_source_remove (play->priv->length_id);
play->priv->length_id = 0;
}
}
if (GST_ELEMENT_CLASS (parent_class)->state_change)
GST_ELEMENT_CLASS (parent_class)->state_change (element, old, state);
}
static void
gst_play_identity_handoff (GstElement * identity, GstBuffer * buf,
GstPlay * play)
{
g_signal_handler_disconnect (G_OBJECT (identity), play->priv->handoff_hid);
play->priv->handoff_hid = 0;
gst_play_connect_visualization (play, FALSE);
}
/* =========================================== */
/* */
/* Init & Dispose & Class init */
/* */
/* =========================================== */
static void
gst_play_dispose (GObject * object)
{
GstPlay *play;
GstElement *output_bin;
g_return_if_fail (object != NULL);
g_return_if_fail (GST_IS_PLAY (object));
play = GST_PLAY (object);
if (play->priv->length_id) {
g_source_remove (play->priv->length_id);
play->priv->length_id = 0;
}
if (play->priv->tick_id) {
g_source_remove (play->priv->tick_id);
play->priv->tick_id = 0;
}
if (play->priv->location) {
g_free (play->priv->location);
play->priv->location = NULL;
}
/* since we reffed our output bin to keep it around, unref it here */
output_bin = g_hash_table_lookup (play->priv->elements, "output_bin");
if (output_bin)
gst_object_unref (GST_OBJECT (output_bin));
if (play->priv->elements) {
g_hash_table_destroy (play->priv->elements);
play->priv->elements = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_play_init (GstPlay * play)
{
play->priv = g_new0 (GstPlayPrivate, 1);
play->priv->location = NULL;
play->priv->length_nanos = 0;
play->priv->time_nanos = 0;
play->priv->elements = g_hash_table_new (g_str_hash, g_str_equal);
play->priv->error = NULL;
play->priv->debug = NULL;
if (!gst_play_pipeline_setup (play, &play->priv->error)) {
g_warning ("libgstplay: failed initializing pipeline, error: %s",
play->priv->error->message);
}
}
static void
gst_play_class_init (GstPlayClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
gobject_class->dispose = gst_play_dispose;
element_class->state_change = gst_play_state_change;
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_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_DEBUG_CATEGORY_INIT (play_debug, "GST_PLAY", 0, "GStreamer Play library");
GST_DEBUG ("Play class initialized");
}
/* ======================================================= */
/* */
/* Public Methods */
/* */
/* ======================================================= */
/**
* gst_play_set_location:
* @play: a #GstPlay.
* @location: a const #char* indicating location to play
*
* Set location of @play to @location.
*
* Returns: TRUE if location was set successfully.
*/
gboolean
gst_play_set_location (GstPlay * play, const char *location)
{
GstElement *work_thread, *source, *autoplugger;
GstElement *audioconvert, *identity;
g_return_val_if_fail (play != NULL, FALSE);
g_return_val_if_fail (GST_IS_PLAY (play), FALSE);
if (play->priv->location)
g_free (play->priv->location);
play->priv->location = g_strdup (location);
if (GST_STATE (GST_ELEMENT (play)) != GST_STATE_READY) {
GstElementStateReturn ret;
ret = gst_element_set_state (GST_ELEMENT (play), GST_STATE_READY);
if (ret == GST_STATE_FAILURE) {
GST_ERROR_OBJECT (play, "failed setting to READY");
return FALSE;
}
}
GST_PLAY_HASH_LOOKUP (work_thread, "work_thread", FALSE);
GST_PLAY_HASH_LOOKUP (source, "source", FALSE);
GST_PLAY_HASH_LOOKUP (autoplugger, "autoplugger", FALSE);
GST_PLAY_HASH_LOOKUP (audioconvert, "audioconvert", FALSE);
GST_PLAY_HASH_LOOKUP (identity, "identity", FALSE);
/* Spider can autoplug only once. We remove the actual one and put a new
autoplugger */
gst_element_unlink (source, autoplugger);
gst_element_unlink (autoplugger, identity);
gst_element_unlink (autoplugger, audioconvert);
gst_bin_remove (GST_BIN (work_thread), autoplugger);
autoplugger = gst_element_factory_make ("spider", "autoplugger");
if (!GST_IS_ELEMENT (autoplugger))
return FALSE;
gst_bin_add (GST_BIN (work_thread), autoplugger);
gst_element_link (source, autoplugger);
gst_element_link (autoplugger, audioconvert);
gst_element_link (autoplugger, identity);
g_hash_table_replace (play->priv->elements, "autoplugger", autoplugger);
/* FIXME: Why don't we have an interface to do that kind of stuff ? */
g_object_set (G_OBJECT (source), "location", play->priv->location, NULL);
play->priv->length_nanos = 0LL;
play->priv->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);
return TRUE;
}
/**
* gst_play_get_location:
* @play: a #GstPlay.
*
* Get current location of @play.
*
* Returns: a const #char* pointer to current location.
*/
char *
gst_play_get_location (GstPlay * play)
{
g_return_val_if_fail (play != NULL, NULL);
g_return_val_if_fail (GST_IS_PLAY (play), NULL);
return g_strdup (play->priv->location);
}
/**
* gst_play_seek_to_time:
* @play: a #GstPlay.
* @time_nanos: a #gint64 indicating a time position.
*
* Performs a seek on @play until @time_nanos.
*/
/* FIXME: use GstClockTime for 0.9 */
gboolean
gst_play_seek_to_time (GstPlay * play, gint64 time_nanos)
{
GstElement *audio_seek_element, *video_seek_element, *audio_sink_element;
GstClockTime seek_to;
g_return_val_if_fail (play != NULL, FALSE);
g_return_val_if_fail (GST_IS_PLAY (play), FALSE);
g_return_val_if_fail (time_nanos >= 0L, FALSE);
seek_to = (GstClockTime) time_nanos;
GST_DEBUG_OBJECT (play, "seeking to time %" GST_TIME_FORMAT,
GST_TIME_ARGS (seek_to));
audio_seek_element = g_hash_table_lookup (play->priv->elements,
"audioconvert");
audio_sink_element = g_hash_table_lookup (play->priv->elements,
"audio_sink_element");
video_seek_element = g_hash_table_lookup (play->priv->elements, "identity");
if (GST_IS_ELEMENT (audio_seek_element) &&
GST_IS_ELEMENT (video_seek_element) &&
GST_IS_ELEMENT (audio_sink_element)) {
gboolean s = FALSE;
/* HACK: block tick signal from idler for 500 msec */
/* GStreamer can't currently report when seeking is finished,
so we just chose a .5 sec default block time */
play->priv->tick_unblock_remaining = 500;
s = gst_element_seek (video_seek_element, GST_FORMAT_TIME |
GST_SEEK_METHOD_SET | GST_SEEK_FLAG_FLUSH, time_nanos);
if (!s) {
s = gst_element_seek (audio_seek_element, GST_FORMAT_TIME |
GST_SEEK_METHOD_SET | GST_SEEK_FLAG_FLUSH, time_nanos);
}
if (s) {
GstFormat format = GST_FORMAT_TIME;
gboolean q = FALSE;
q = gst_element_query (audio_sink_element, GST_QUERY_POSITION, &format,
&(play->priv->time_nanos));
if (q)
g_signal_emit (G_OBJECT (play), gst_play_signals[TIME_TICK],
0, play->priv->time_nanos);
}
}
return TRUE;
}
/**
* gst_play_set_data_src:
* @play: a #GstPlay.
* @data_src: a #GstElement.
*
* Set @data_src as the source element of @play.
*
* Returns: TRUE if call succeeded.
*/
gboolean
gst_play_set_data_src (GstPlay * play, GstElement * data_src)
{
GstElement *work_thread, *old_data_src, *autoplugger;
g_return_val_if_fail (play != NULL, FALSE);
g_return_val_if_fail (GST_IS_PLAY (play), FALSE);
GST_DEBUG_OBJECT (play, "setting new data src element %s",
GST_ELEMENT_NAME (data_src));
/* We bring back the pipeline to READY */
if (GST_STATE (GST_ELEMENT (play)) != GST_STATE_READY) {
GstElementStateReturn ret;
ret = gst_element_set_state (GST_ELEMENT (play), GST_STATE_READY);
if (ret == GST_STATE_FAILURE) {
GST_ERROR_OBJECT (play, "failed setting to READY");
return FALSE;
}
}
/* Getting needed objects */
GST_PLAY_HASH_LOOKUP (work_thread, "work_thread", FALSE);
GST_PLAY_HASH_LOOKUP (old_data_src, "source", FALSE);
GST_PLAY_HASH_LOOKUP (autoplugger, "autoplugger", FALSE);
/* Unlinking old source from autoplugger, removing it from pipeline, adding
the new one and connecting it to autoplugger FIXME: we should put a new
autoplugger here as spider can autoplugg only once */
gst_element_unlink (old_data_src, autoplugger);
gst_bin_remove (GST_BIN (work_thread), old_data_src);
gst_bin_add (GST_BIN (work_thread), data_src);
if (!gst_element_link (data_src, autoplugger)) {
GST_ERROR_OBJECT (play, "could not link source to autoplugger");
return FALSE;
}
g_hash_table_replace (play->priv->elements, "source", data_src);
return TRUE;
}
/**
* gst_play_set_video_sink:
* @play: a #GstPlay.
* @video_sink: a #GstElement.
*
* Set @video_sink as the video sink element of @play.
*
* Returns: TRUE if call succeeded.
*/
gboolean
gst_play_set_video_sink (GstPlay * play, GstElement * video_sink)
{
GstElement *video_thread, *old_video_sink, *video_scaler, *video_sink_element;
GstElementStateReturn ret;
g_return_val_if_fail (play != NULL, FALSE);
g_return_val_if_fail (GST_IS_PLAY (play), FALSE);
g_return_val_if_fail (video_sink != NULL, FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (video_sink), FALSE);
/* We bring back the pipeline to READY */
if (GST_STATE (GST_ELEMENT (play)) != GST_STATE_READY) {
GstElementStateReturn ret;
ret = gst_element_set_state (GST_ELEMENT (play), GST_STATE_READY);
if (ret == GST_STATE_FAILURE) {
GST_ERROR_OBJECT (play, "failed setting to READY");
return FALSE;
}
}
/* Getting needed objects */
GST_PLAY_HASH_LOOKUP (video_thread, "video_thread", FALSE);
GST_PLAY_HASH_LOOKUP (old_video_sink, "video_sink", FALSE);
GST_PLAY_HASH_LOOKUP (video_scaler, "video_scaler", FALSE);
/* Unlinking old video sink from video scaler, removing it from pipeline,
adding the new one and linking it */
gst_element_unlink (video_scaler, old_video_sink);
gst_bin_remove (GST_BIN (video_thread), old_video_sink);
gst_bin_add (GST_BIN (video_thread), video_sink);
if (!gst_element_link (video_scaler, video_sink)) {
GST_ERROR_OBJECT (play, "could not link video_scaler to video_sink");
return FALSE;
}
g_hash_table_replace (play->priv->elements, "video_sink", video_sink);
video_sink_element = gst_play_get_sink_element (play, video_sink,
GST_PLAY_SINK_TYPE_VIDEO);
if (GST_IS_ELEMENT (video_sink_element)) {
g_hash_table_replace (play->priv->elements, "video_sink_element",
video_sink_element);
if (GST_IS_X_OVERLAY (video_sink_element)) {
g_signal_connect (G_OBJECT (video_sink_element),
"desired_size_changed", G_CALLBACK (gst_play_have_video_size), play);
}
}
ret = gst_element_set_state (video_sink, GST_STATE (GST_ELEMENT (play)));
if (ret == GST_STATE_FAILURE) {
GST_ERROR_OBJECT (play, "failed setting to READY");
return FALSE;
}
return TRUE;
}
/**
* gst_play_set_audio_sink:
* @play: a #GstPlay.
* @audio_sink: a #GstElement.
*
* Set @audio_sink as the audio sink element of @play.
*
* Returns: TRUE if call succeeded.
*/
gboolean
gst_play_set_audio_sink (GstPlay * play, GstElement * audio_sink)
{
GstElement *old_audio_sink, *audio_thread, *volume, *audio_sink_element;
GstElementStateReturn ret;
g_return_val_if_fail (play != NULL, FALSE);
g_return_val_if_fail (GST_IS_PLAY (play), FALSE);
g_return_val_if_fail (audio_sink != NULL, FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (audio_sink), FALSE);
/* We bring back the pipeline to READY */
if (GST_STATE (GST_ELEMENT (play)) != GST_STATE_READY) {
GstElementStateReturn ret;
ret = gst_element_set_state (GST_ELEMENT (play), GST_STATE_READY);
if (ret == GST_STATE_FAILURE) {
GST_ERROR_OBJECT (play, "failed setting to READY");
return FALSE;
}
}
/* Getting needed objects */
GST_PLAY_HASH_LOOKUP (audio_thread, "audio_thread", FALSE);
GST_PLAY_HASH_LOOKUP (volume, "volume", FALSE);
GST_PLAY_HASH_LOOKUP (old_audio_sink, "audio_sink", FALSE);
/* Unlinking old audiosink, removing it from pipeline, putting the new one
and linking it */
gst_element_unlink (volume, old_audio_sink);
gst_bin_remove (GST_BIN (audio_thread), old_audio_sink);
gst_bin_add (GST_BIN (audio_thread), audio_sink);
if (!gst_element_link (volume, audio_sink)) {
GST_ERROR_OBJECT (play, "could not link volume to audio_sink");
return FALSE;
}
g_hash_table_replace (play->priv->elements, "audio_sink", audio_sink);
audio_sink_element = gst_play_get_sink_element (play, audio_sink,
GST_PLAY_SINK_TYPE_AUDIO);
if (GST_IS_ELEMENT (audio_sink_element)) {
g_hash_table_replace (play->priv->elements, "audio_sink_element",
audio_sink_element);
}
ret = gst_element_set_state (audio_sink, GST_STATE (GST_ELEMENT (play)));
if (ret == GST_STATE_FAILURE) {
GST_ERROR_OBJECT (play, "failed setting to READY");
return FALSE;
}
return TRUE;
}
/**
* gst_play_set_visualization:
* @play: a #GstPlay.
* @element: a #GstElement.
*
* Set @video_sink as the video sink element of @play.
*
* Returns: TRUE if call succeeded.
*/
gboolean
gst_play_set_visualization (GstPlay * play, GstElement * vis_element)
{
GstElement *vis_bin, *vis_queue, *old_vis_element, *vis_cs;
gboolean was_playing = FALSE;
g_return_val_if_fail (play != NULL, FALSE);
g_return_val_if_fail (GST_IS_PLAY (play), FALSE);
g_return_val_if_fail (vis_element != NULL, FALSE);
g_return_val_if_fail (GST_IS_ELEMENT (vis_element), FALSE);
/* Getting needed objects */
GST_PLAY_HASH_LOOKUP (vis_bin, "vis_bin", FALSE);
GST_PLAY_HASH_LOOKUP (vis_queue, "vis_queue", FALSE);
GST_PLAY_HASH_LOOKUP (old_vis_element, "vis_element", FALSE);
GST_PLAY_HASH_LOOKUP (vis_cs, "vis_cs", FALSE);
/* We bring back the pipeline to PAUSED */
if (GST_STATE (GST_ELEMENT (play)) == GST_STATE_PLAYING) {
GstElementStateReturn ret;
ret = gst_element_set_state (GST_ELEMENT (play), GST_STATE_PAUSED);
if (ret == GST_STATE_FAILURE) {
GST_ERROR_OBJECT (play, "failed setting to READY");
return FALSE;
}
was_playing = TRUE;
}
gst_element_unlink_many (vis_queue, old_vis_element, vis_cs, NULL);
gst_bin_remove (GST_BIN (vis_bin), old_vis_element);
gst_bin_add (GST_BIN (vis_bin), vis_element);
if (!gst_element_link_many (vis_queue, vis_element, vis_cs, NULL)) {
GST_ERROR_OBJECT (play, "could not link vis bin elements");
return FALSE;
}
g_hash_table_replace (play->priv->elements, "vis_element", vis_element);
if (was_playing) {
GstElementStateReturn ret;
ret = gst_element_set_state (GST_ELEMENT (play), GST_STATE_PLAYING);
if (ret == GST_STATE_FAILURE) {
GST_ERROR_OBJECT (play, "failed setting to READY");
return FALSE;
}
}
return TRUE;
}
/**
* gst_play_connect_visualization:
* @play: a #GstPlay.
* @connect: a #gboolean indicating wether or not
* visualization should be connected.
*
* Connect or disconnect visualization bin in @play.
*
* Returns: TRUE if call succeeded.
*/
gboolean
gst_play_connect_visualization (GstPlay * play, gboolean connect)
{
GstElement *video_thread, *vis_queue, *vis_bin, *video_switch, *identity;
GstPad *tee_pad1, *vis_queue_pad, *vis_bin_pad, *switch_pad;
gboolean was_playing = FALSE;
g_return_val_if_fail (play != NULL, FALSE);
g_return_val_if_fail (GST_IS_PLAY (play), FALSE);
/* Until i fix the switch */
return TRUE;
/* Getting needed objects */
GST_PLAY_HASH_LOOKUP (video_thread, "video_thread", FALSE);
GST_PLAY_HASH_LOOKUP (vis_bin, "vis_bin", FALSE);
GST_PLAY_HASH_LOOKUP (vis_queue, "vis_queue", FALSE);
GST_PLAY_HASH_LOOKUP (video_switch, "video_switch", FALSE);
GST_PLAY_HASH_LOOKUP (identity, "identity", FALSE);
GST_PLAY_HASH_LOOKUP (tee_pad1, "tee_pad1", FALSE);
vis_queue_pad = gst_element_get_pad (vis_queue, "sink");
/* Check if the vis element is in the pipeline. That means visualization is
connected already */
if (gst_element_get_managing_bin (vis_bin)) {
/* If we are supposed to connect then nothing to do we return */
if (connect) {
return TRUE;
}
/* Disconnecting visualization */
/* We bring back the pipeline to PAUSED */
if (GST_STATE (GST_ELEMENT (play)) == GST_STATE_PLAYING) {
GstElementStateReturn ret;
ret = gst_element_set_state (GST_ELEMENT (play), GST_STATE_PAUSED);
if (ret == GST_STATE_FAILURE) {
GST_ERROR_OBJECT (play, "failed setting to READY");
return FALSE;
}
was_playing = TRUE;
}
/* Unlinking, removing */
gst_pad_unlink (tee_pad1, vis_queue_pad);
vis_bin_pad = gst_element_get_pad (vis_bin, "src");
switch_pad = gst_pad_get_peer (vis_bin_pad);
gst_pad_unlink (vis_bin_pad, switch_pad);
gst_element_release_request_pad (video_switch, switch_pad);
gst_object_ref (GST_OBJECT (vis_bin));
gst_bin_remove (GST_BIN (video_thread), vis_bin);
} else {
/* If we are supposed to disconnect then nothing to do we return */
if (!connect) {
return TRUE;
}
/* Connecting visualization */
/* We bring back the pipeline to PAUSED */
if (GST_STATE (GST_ELEMENT (play)) == GST_STATE_PLAYING) {
GstElementStateReturn ret;
ret = gst_element_set_state (GST_ELEMENT (play), GST_STATE_PAUSED);
if (ret == GST_STATE_FAILURE) {
GST_ERROR_OBJECT (play, "failed setting to READY");
return FALSE;
}
was_playing = TRUE;
}
/* Adding, linking */
play->priv->handoff_hid = g_signal_connect (G_OBJECT (identity),
"handoff", G_CALLBACK (gst_play_identity_handoff), play);
gst_bin_add (GST_BIN (video_thread), vis_bin);
gst_pad_link (tee_pad1, vis_queue_pad);
if (!gst_element_link (vis_bin, video_switch)) {
GST_ERROR_OBJECT (play, "could not link vis bin to video switch");
return FALSE;
}
}
if (was_playing) {
GstElementStateReturn ret;
ret = gst_element_set_state (GST_ELEMENT (play), GST_STATE_PLAYING);
if (ret == GST_STATE_FAILURE) {
GST_ERROR_OBJECT (play, "failed setting to READY");
return FALSE;
}
}
return TRUE;
}
/**
* gst_play_get_framerate:
* @play: a #GstPlay.
*
* Get the video framerate from @play.
*
* Returns: a #gdouble indicating video framerate in frame per second.
*/
gdouble
gst_play_get_framerate (GstPlay * play)
{
GstElement *video_element = NULL;
GstPad *video_pad = NULL;
GstCaps *video_pad_caps = NULL;
GstStructure *structure = NULL;
g_return_val_if_fail (GST_IS_PLAY (play), 0);
GST_PLAY_HASH_LOOKUP (video_element, "video_sink", 0);
video_pad = gst_element_get_pad (video_element, "sink");
if (!GST_IS_PAD (video_pad))
return 0;
video_pad_caps = (GstCaps *) gst_pad_get_negotiated_caps (video_pad);
if (!GST_IS_CAPS (video_pad_caps))
return 0;
structure = gst_caps_get_structure (video_pad_caps, 0);
if (structure) {
gdouble value;
gst_structure_get_double (structure, "framerate", &value);
return value;
}
return 0;
}
/**
* gst_play_get_sink_element:
* @play: a #GstPlay.
* @element: a #GstElement.
* @sink_type: a #GstPlaySinkType.
*
* Searches recursively for a sink #GstElement with
* type @sink_type in @element which is supposed to be a #GstBin.
*
* Returns: the sink #GstElement of @element.
*/
GstElement *
gst_play_get_sink_element (GstPlay * play,
GstElement * element, GstPlaySinkType sink_type)
{
GList *elements = NULL;
const GList *pads = NULL;
gboolean has_src, has_correct_type;
g_return_val_if_fail (GST_IS_PLAY (play), NULL);
g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
GST_DEBUG_OBJECT (play, "looking for sink element in %s",
GST_ELEMENT_NAME (element));
if (!GST_IS_BIN (element)) {
/* since its not a bin, we'll assume this
* element is a sink element */
GST_DEBUG_OBJECT (play, "not a bin, returning %s as sink element",
GST_ELEMENT_NAME (element));
return element;
}
elements = (GList *) gst_bin_get_list (GST_BIN (element));
/* traverse all elements looking for one without src pad */
while (elements) {
element = GST_ELEMENT (elements->data);
GST_DEBUG_OBJECT (play, "looking at element %s",
GST_ELEMENT_NAME (element));
/* Recursivity :) */
if (GST_IS_BIN (element)) {
element = gst_play_get_sink_element (play, element, sink_type);
if (GST_IS_ELEMENT (element))
return element;
} else {
pads = gst_element_get_pad_list (element);
has_src = FALSE;
has_correct_type = FALSE;
while (pads) {
/* check for src pad */
if (GST_PAD_DIRECTION (GST_PAD (pads->data)) == GST_PAD_SRC) {
GST_DEBUG_OBJECT (play, "element %s has a src pad",
GST_ELEMENT_NAME (element));
has_src = TRUE;
break;
} else {
/* If not a src pad checking caps */
GstPad *pad;
GstCaps *caps;
GstStructure *structure;
int i;
gboolean has_video_cap = FALSE;
gboolean has_audio_cap = FALSE;
pad = GST_PAD (pads->data);
caps = gst_pad_get_caps (pad);
/* loop over all caps members to find mime types */
for (i = 0; i < gst_caps_get_size (caps); ++i) {
structure = gst_caps_get_structure (caps, i);
GST_DEBUG_OBJECT (play,
"looking at caps %d pad %s:%s on element %s with mime %s", i,
GST_DEBUG_PAD_NAME (pad),
GST_ELEMENT_NAME (element), gst_structure_get_name (structure));
if (strcmp (gst_structure_get_name (structure),
"audio/x-raw-int") == 0) {
has_audio_cap = TRUE;
}
if (strcmp (gst_structure_get_name (structure),
"video/x-raw-yuv") == 0 ||
strcmp (gst_structure_get_name (structure),
"video/x-raw-rgb") == 0) {
has_video_cap = TRUE;
}
}
gst_caps_free (caps);
switch (sink_type) {
case GST_PLAY_SINK_TYPE_AUDIO:
if (has_audio_cap)
has_correct_type = TRUE;
break;
case GST_PLAY_SINK_TYPE_VIDEO:
if (has_video_cap)
has_correct_type = TRUE;
break;
case GST_PLAY_SINK_TYPE_ANY:
if ((has_video_cap) || (has_audio_cap))
has_correct_type = TRUE;
break;
default:
has_correct_type = FALSE;
}
}
pads = g_list_next (pads);
}
if ((!has_src) && (has_correct_type)) {
GST_DEBUG_OBJECT (play, "found %s with src pad and correct type",
GST_ELEMENT_NAME (element));
return element;
}
}
elements = g_list_next (elements);
}
/* we didn't find a sink element */
return NULL;
}
/**
* gst_play_get_all_by_interface:
* @play: a #GstPlay.
* @interface: an interface.
*
* Returns all elements that are used by @play implementing the given interface.
*
* Returns: a #GList of #GstElement implementing the interface.
*/
GList *
gst_play_get_all_by_interface (GstPlay * play, GType interface)
{
GstElement *output_bin;
GST_PLAY_HASH_LOOKUP (output_bin, "output_bin", NULL);
return gst_bin_get_all_by_interface (GST_BIN (output_bin), interface);
}
GstPlay *
gst_play_new (GError ** error)
{
GstPlay *play = g_object_new (GST_TYPE_PLAY, NULL);
if (play->priv->error) {
if (error) {
*error = play->priv->error;
play->priv->error = NULL;
} else {
g_warning ("Error creating GstPlay object.\n%s",
play->priv->error->message);
g_error_free (play->priv->error);
}
}
return play;
}
/* =========================================== */
/* */
/* Object typing & Creation */
/* */
/* =========================================== */
GType
gst_play_get_type (void)
{
static GType play_type = 0;
if (!play_type) {
static const GTypeInfo play_info = {
sizeof (GstPlayClass),
NULL,
NULL,
(GClassInitFunc) gst_play_class_init,
NULL,
NULL,
sizeof (GstPlay),
0,
(GInstanceInitFunc) gst_play_init,
NULL
};
play_type = g_type_register_static (GST_TYPE_PIPELINE, "GstPlay",
&play_info, 0);
}
return play_type;
}
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