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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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9569929252
Original commit message from CVS: new method. various debugging
1475 lines
46 KiB
C
1475 lines
46 KiB
C
/* GStreamer
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* Copyright (C) 2003 Julien Moutte <julien@moutte.net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include "play.h"
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#define TICK_INTERVAL_MSEC 200
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GST_DEBUG_CATEGORY_STATIC (play_debug);
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#define GST_CAT_DEFAULT play_debug
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enum
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{
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TIME_TICK,
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STREAM_LENGTH,
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HAVE_VIDEO_SIZE,
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LAST_SIGNAL
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};
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struct _GstPlayPrivate
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{
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char *location;
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GHashTable *elements;
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gint64 time_nanos;
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gint64 length_nanos;
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gint get_length_attempt;
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gint tick_unblock_remaining; /* how many msecs left
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to unblock due to seeking */
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guint tick_id;
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guint length_id;
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gulong handoff_hid;
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/* error/debug handling */
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GError *error;
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gchar *debug;
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};
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static guint gst_play_signals[LAST_SIGNAL] = { 0 };
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static GstPipelineClass *parent_class = NULL;
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/* ======================================================= */
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/* */
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/* Private Methods */
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/* */
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/* ======================================================= */
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static GstCaps *gst_play_video_fixate (GstPad * pad, const GstCaps * caps,
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gpointer user_data);
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static GstCaps *gst_play_audio_fixate (GstPad * pad, const GstCaps * caps,
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gpointer user_data);
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static GQuark
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gst_play_error_quark (void)
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{
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static GQuark quark = 0;
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if (quark == 0)
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quark = g_quark_from_static_string ("gst-play-error-quark");
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return quark;
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}
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/* General GError creation */
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static void
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gst_play_error_create (GError ** error, const gchar * message)
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{
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/* check if caller wanted an error reported */
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if (error == NULL)
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return;
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*error = g_error_new (GST_PLAY_ERROR, 0, message);
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return;
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}
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/* GError creation when plugin is missing */
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/* FIXME: what if multiple elements could have been used and they're all
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* missing ? varargs ? */
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static void
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gst_play_error_plugin (const gchar * element, GError ** error)
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{
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gchar *message;
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message = g_strdup_printf ("The %s element could not be found. "
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"This element is essential for playback. "
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"Please install the right plug-in and verify "
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"that it works by running 'gst-inspect %s'", element, element);
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gst_play_error_create (error, message);
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g_free (message);
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return;
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}
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#define GST_PLAY_MAKE_OR_ERROR(el, factory, name, error) \
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G_STMT_START { \
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el = gst_element_factory_make (factory, name); \
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if (!GST_IS_ELEMENT (el)) \
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{ \
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gst_play_error_plugin (factory, error); \
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return FALSE; \
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} \
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} G_STMT_END
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#define GST_PLAY_ERROR_RETURN(error, message) \
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G_STMT_START { \
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gst_play_error_create (error, message); \
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return FALSE; \
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} G_STMT_END
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#define GST_PLAY_HASH_LOOKUP(element, key, retval_if_fail) \
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G_STMT_START { \
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(element) = g_hash_table_lookup (play->priv->elements, (key));\
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if (!element) \
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return (retval_if_fail); \
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} G_STMT_END
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/* setup parts of the pipeline
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* only put decoding part in the thread
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* create all others and keep them around
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*/
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static gboolean
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gst_play_pipeline_setup (GstPlay * play, GError ** error)
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{
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/* Threads */
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GstElement *work_thread, *audio_thread, *video_thread;
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/* output bin */
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GstElement *output_bin;
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/* Main Thread elements */
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GstElement *source, *autoplugger;
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/* output bin elements */
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GstElement *audioconvert, *volume, *tee, *identity;
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GstElement *identity_cs;
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/* Visualization bin */
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GstElement *vis_bin, *vis_queue, *vis_element, *vis_cs;
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/* Video Thread elements */
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GstElement *video_queue, *video_switch, *video_cs, *video_balance;
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GstElement *balance_cs, *video_scaler, *video_sink;
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/* Audio Thread elements */
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GstElement *audio_queue, *audio_sink;
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/* Some useful pads */
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GstPad *tee_pad1, *tee_pad2;
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g_return_val_if_fail (play != NULL, FALSE);
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g_return_val_if_fail (GST_IS_PLAY (play), FALSE);
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GST_DEBUG_OBJECT (play, "setting up pipeline");
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/* Creating main thread and its elements */
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{
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GST_PLAY_MAKE_OR_ERROR (work_thread, "thread", "work_thread", error);
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g_hash_table_insert (play->priv->elements, "work_thread", work_thread);
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gst_bin_add (GST_BIN (play), work_thread);
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/* Placeholder for datasrc */
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GST_PLAY_MAKE_OR_ERROR (source, "fakesrc", "source", error);
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g_hash_table_insert (play->priv->elements, "source", source);
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/* Autoplugger */
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GST_PLAY_MAKE_OR_ERROR (autoplugger, "spider", "autoplugger", error);
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g_hash_table_insert (play->priv->elements, "autoplugger", autoplugger);
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/* adding and linking */
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gst_bin_add_many (GST_BIN (work_thread), source, autoplugger, NULL);
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if (!gst_element_link (source, autoplugger))
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GST_PLAY_ERROR_RETURN (error, "Could not link source and autoplugger");
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}
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/* output bin */
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{
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GST_PLAY_MAKE_OR_ERROR (output_bin, "bin", "output_bin", error);
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g_hash_table_insert (play->priv->elements, "output_bin", output_bin);
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/* Make sure we convert audio to the needed format */
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GST_PLAY_MAKE_OR_ERROR (audioconvert, "audioconvert", "audioconvert",
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error);
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g_hash_table_insert (play->priv->elements, "audioconvert", audioconvert);
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/* Duplicate audio signal to audio sink and visualization thread */
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GST_PLAY_MAKE_OR_ERROR (tee, "tee", "tee", error);
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tee_pad1 = gst_element_get_request_pad (tee, "src%d");
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tee_pad2 = gst_element_get_request_pad (tee, "src%d");
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g_hash_table_insert (play->priv->elements, "tee_pad1", tee_pad1);
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g_hash_table_insert (play->priv->elements, "tee_pad2", tee_pad2);
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g_hash_table_insert (play->priv->elements, "tee", tee);
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gst_bin_add_many (GST_BIN (output_bin), audioconvert, tee, NULL);
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if (!gst_element_link (audioconvert, tee))
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GST_PLAY_ERROR_RETURN (error, "Could not link audio thread elements");
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/* identity ! colorspace ! switch */
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GST_PLAY_MAKE_OR_ERROR (identity, "identity", "identity", error);
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g_hash_table_insert (play->priv->elements, "identity", identity);
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identity_cs = gst_element_factory_make ("ffcolorspace", "identity_cs");
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if (!GST_IS_ELEMENT (identity_cs)) {
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identity_cs =
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gst_element_factory_make ("ffmpegcolorspace", "identity_cs");
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if (!GST_IS_ELEMENT (identity_cs)) {
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identity_cs = gst_element_factory_make ("colorspace", "identity_cs");
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if (!GST_IS_ELEMENT (identity_cs)) {
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gst_play_error_plugin ("colorspace", error);
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return FALSE;
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}
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}
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}
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g_hash_table_insert (play->priv->elements, "identity_cs", identity_cs);
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gst_bin_add_many (GST_BIN (output_bin), identity, identity_cs, NULL);
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if (!gst_element_link_many (autoplugger, identity, identity_cs, NULL))
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GST_PLAY_ERROR_RETURN (error, "Could not link work thread elements");
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/* we ref the output bin so we can put it in and out the work_thread
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* whenever we want */
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gst_object_ref (GST_OBJECT (output_bin));
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GST_DEBUG_OBJECT (play, "adding output bin to work thread in setup");
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gst_bin_add (GST_BIN (work_thread), output_bin);
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}
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/* Visualization bin (note: it s not added to the pipeline yet) */
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{
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vis_bin = gst_bin_new ("vis_bin");
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if (!GST_IS_ELEMENT (vis_bin)) {
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gst_play_error_plugin ("bin", error);
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return FALSE;
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}
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g_hash_table_insert (play->priv->elements, "vis_bin", vis_bin);
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/* Buffer queue for video data */
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GST_PLAY_MAKE_OR_ERROR (vis_queue, "queue", "vis_queue", error);
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g_hash_table_insert (play->priv->elements, "vis_queue", vis_queue);
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/* Visualization element placeholder */
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GST_PLAY_MAKE_OR_ERROR (vis_element, "identity", "vis_element", error);
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g_hash_table_insert (play->priv->elements, "vis_element", vis_element);
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/* Colorspace conversion */
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vis_cs = gst_element_factory_make ("ffcolorspace", "vis_cs");
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if (!GST_IS_ELEMENT (vis_cs)) {
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vis_cs = gst_element_factory_make ("ffmpegcolorspace", "vis_cs");
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if (!GST_IS_ELEMENT (vis_cs)) {
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vis_cs = gst_element_factory_make ("colorspace", "vis_cs");
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if (!GST_IS_ELEMENT (vis_cs)) {
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gst_play_error_plugin ("colorspace", error);
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return FALSE;
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}
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}
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}
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g_hash_table_insert (play->priv->elements, "vis_cs", vis_cs);
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gst_bin_add_many (GST_BIN (vis_bin), vis_queue, vis_element, vis_cs, NULL);
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if (!gst_element_link_many (vis_queue, vis_element, vis_cs, NULL))
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GST_PLAY_ERROR_RETURN (error,
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"Could not link visualisation thread elements");
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gst_element_add_ghost_pad (vis_bin, gst_element_get_pad (vis_cs, "src"),
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"src");
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}
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/* Creating our video output bin */
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{
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GST_PLAY_MAKE_OR_ERROR (video_thread, "thread", "video_thread", error);
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g_hash_table_insert (play->priv->elements, "video_thread", video_thread);
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gst_bin_add (GST_BIN (output_bin), video_thread);
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/* Buffer queue for video data */
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GST_PLAY_MAKE_OR_ERROR (video_queue, "queue", "video_queue", error);
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g_hash_table_insert (play->priv->elements, "video_queue", video_queue);
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GST_PLAY_MAKE_OR_ERROR (video_switch, "switch", "video_switch", error);
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g_hash_table_insert (play->priv->elements, "video_switch", video_switch);
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/* Colorspace conversion */
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video_cs = gst_element_factory_make ("ffcolorspace", "video_cs");
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if (!GST_IS_ELEMENT (video_cs)) {
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video_cs = gst_element_factory_make ("ffmpegcolorspace", "video_cs");
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if (!GST_IS_ELEMENT (video_cs)) {
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video_cs = gst_element_factory_make ("colorspace", "video_cs");
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if (!GST_IS_ELEMENT (video_cs)) {
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gst_play_error_plugin ("colorspace", error);
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return FALSE;
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}
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}
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}
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g_hash_table_insert (play->priv->elements, "video_cs", video_cs);
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/* Software colorbalance */
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GST_PLAY_MAKE_OR_ERROR (video_balance, "videobalance", "video_balance",
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error);
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g_hash_table_insert (play->priv->elements, "video_balance", video_balance);
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/* Colorspace conversion */
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balance_cs = gst_element_factory_make ("ffcolorspace", "balance_cs");
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if (!GST_IS_ELEMENT (balance_cs)) {
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balance_cs = gst_element_factory_make ("ffmpegcolorspace", "balance_cs");
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if (!GST_IS_ELEMENT (balance_cs)) {
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balance_cs = gst_element_factory_make ("colorspace", "balance_cs");
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if (!GST_IS_ELEMENT (balance_cs)) {
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gst_play_error_plugin ("colorspace", error);
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return FALSE;
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}
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}
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}
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g_hash_table_insert (play->priv->elements, "balance_cs", balance_cs);
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/* Software scaling of video stream */
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GST_PLAY_MAKE_OR_ERROR (video_scaler, "videoscale", "video_scaler", error);
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g_hash_table_insert (play->priv->elements, "video_scaler", video_scaler);
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g_signal_connect (gst_element_get_pad (video_scaler, "src"), "fixate",
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G_CALLBACK (gst_play_video_fixate), play);
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/* Placeholder for future video sink bin */
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GST_PLAY_MAKE_OR_ERROR (video_sink, "fakesink", "video_sink", error);
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g_hash_table_insert (play->priv->elements, "video_sink", video_sink);
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gst_bin_add_many (GST_BIN (video_thread), video_queue, video_switch,
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video_cs, video_balance, balance_cs, video_scaler, video_sink, NULL);
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/* break down linking so we can figure out what might be failing */
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if (!gst_element_link (video_queue, video_switch))
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GST_PLAY_ERROR_RETURN (error,
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"Could not link video output thread (queue and switch)");
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if (!gst_element_link (video_switch, video_cs))
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GST_PLAY_ERROR_RETURN (error,
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"Could not link video output thread (switch and cs)");
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if (!gst_element_link (video_cs, video_balance))
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GST_PLAY_ERROR_RETURN (error,
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"Could not link video output thread (cs and balance)");
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if (!gst_element_link (video_balance, balance_cs))
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GST_PLAY_ERROR_RETURN (error,
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"Could not link video output thread (balance and balance_cs)");
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if (!gst_element_link (balance_cs, video_scaler))
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GST_PLAY_ERROR_RETURN (error,
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"Could not link video output thread (balance_cs and scaler)");
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if (!gst_element_link (video_scaler, video_sink))
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GST_PLAY_ERROR_RETURN (error,
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"Could not link video output thread (balance_cs and scaler)");
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gst_element_add_ghost_pad (video_thread, gst_element_get_pad (video_queue,
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"sink"), "sink");
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if (!gst_element_link (identity_cs, video_thread))
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GST_PLAY_ERROR_RETURN (error,
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"Could not link video output thread elements");
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}
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/* Creating our audio output bin
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{ queue ! fakesink } */
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{
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GST_PLAY_MAKE_OR_ERROR (audio_thread, "thread", "audio_thread", error);
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g_hash_table_insert (play->priv->elements, "audio_thread", audio_thread);
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gst_bin_add (GST_BIN (output_bin), audio_thread);
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/* Buffer queue for our audio thread */
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GST_PLAY_MAKE_OR_ERROR (audio_queue, "queue", "audio_queue", error);
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g_hash_table_insert (play->priv->elements, "audio_queue", audio_queue);
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/* Volume control */
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GST_PLAY_MAKE_OR_ERROR (volume, "volume", "volume", error);
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g_hash_table_insert (play->priv->elements, "volume", volume);
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g_signal_connect (gst_element_get_pad (volume, "src"), "fixate",
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G_CALLBACK (gst_play_audio_fixate), play);
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/* Placeholder for future audio sink bin */
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GST_PLAY_MAKE_OR_ERROR (audio_sink, "fakesink", "audio_sink", error);
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g_hash_table_insert (play->priv->elements, "audio_sink", audio_sink);
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gst_bin_add_many (GST_BIN (audio_thread), audio_queue, volume, audio_sink,
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NULL);
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if (!gst_element_link_many (audio_queue, volume, audio_sink, NULL))
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GST_PLAY_ERROR_RETURN (error,
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"Could not link audio output thread elements");
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gst_element_add_ghost_pad (audio_thread, gst_element_get_pad (audio_queue,
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"sink"), "sink");
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gst_pad_link (tee_pad2, gst_element_get_pad (audio_queue, "sink"));
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}
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GST_DEBUG_OBJECT (play, "setting up pipeline succeeded.");
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return TRUE;
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}
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static void
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gst_play_have_video_size (GstElement * element, gint width,
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gint height, GstPlay * play)
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{
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g_return_if_fail (play != NULL);
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g_return_if_fail (GST_IS_PLAY (play));
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g_signal_emit (G_OBJECT (play), gst_play_signals[HAVE_VIDEO_SIZE],
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0, width, height);
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}
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static gboolean
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gst_play_tick_callback (GstPlay * play)
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{
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GstFormat format = GST_FORMAT_TIME;
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gboolean q = FALSE;
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GstElement *audio_sink_element = NULL;
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g_return_val_if_fail (play != NULL, FALSE);
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/* just return without updating the UI when we are in the middle of seeking */
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if (play->priv->tick_unblock_remaining > 0) {
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play->priv->tick_unblock_remaining -= TICK_INTERVAL_MSEC;
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return TRUE;
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}
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if (!GST_IS_PLAY (play)) {
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play->priv->tick_id = 0;
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return FALSE;
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}
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audio_sink_element = g_hash_table_lookup (play->priv->elements,
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"audio_sink_element");
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if (!GST_IS_ELEMENT (audio_sink_element)) {
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play->priv->tick_id = 0;
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return FALSE;
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}
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q = gst_element_query (audio_sink_element, GST_QUERY_POSITION, &format,
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&(play->priv->time_nanos));
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if (q)
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g_signal_emit (G_OBJECT (play), gst_play_signals[TIME_TICK],
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0, play->priv->time_nanos);
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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;
|
|
}
|