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Original commit message from CVS: 2004-01-14 Benjamin Otte <in7y118@public.uni-hamburg.de> * ext/aalib/gstaasink.c: (gst_aasink_chain): * ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event): * ext/esd/esdsink.c: (gst_esdsink_chain): * ext/libcaca/gstcacasink.c: (gst_cacasink_chain): * ext/mas/massink.c: (gst_massink_chain): * ext/sdl/sdlvideosink.c: (gst_sdlvideosink_chain): * gst/matroska/matroska-demux.c: (gst_matroska_demux_parse_index), (gst_matroska_demux_parse_metadata): * gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_loop), (gst_mpeg_parse_release_locks): * gst/tcp/gsttcpsink.c: (gst_tcpsink_chain): * gst/udp/gstudpsink.c: (gst_udpsink_chain): * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_get): * sys/oss/gstosssink.c: (gst_osssink_init), (gst_osssink_chain), (gst_osssink_change_state): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_chain): * sys/ximage/ximagesink.c: (gst_ximagesink_chain): * sys/xvideo/xvideosink.c: (gst_xvideosink_chain), (gst_xvideosink_release_locks): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_chain): use element time. * ext/alsa/gstalsaclock.c: (gst_alsa_clock_start), (gst_alsa_clock_stop): * gst-libs/gst/audio/audioclock.c: (gst_audio_clock_set_active), (gst_audio_clock_get_internal_time): simplify for use with new clocking code. * testsuite/alsa/Makefile.am: * testsuite/alsa/sinesrc.c: (sinesrc_init), (sinesrc_force_caps): fix testsuite for new caps system
553 lines
16 KiB
C
553 lines
16 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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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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/*
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* Portions derived from maswavplay.c (distributed under the X11
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* license):
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*
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* Copyright (c) 2001-2003 Shiman Associates Inc. All Rights Reserved.
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* Copyright (c) 2000, 2001 by Shiman Associates Inc. and Sun
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* Microsystems, Inc. All Rights Reserved.
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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 "massink.h"
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#define BUFFER_SIZE 640
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/* Signals and args */
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enum {
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/* FILL ME */
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LAST_SIGNAL
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};
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enum {
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ARG_0,
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ARG_MUTE,
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ARG_DEPTH,
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ARG_CHANNELS,
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ARG_RATE,
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ARG_HOST,
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};
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static GstStaticPadTemplate sink_factory =
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GST_STATIC_PAD_TEMPLATE (
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"sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int")
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);
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static void gst_massink_base_init (gpointer g_class);
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static void gst_massink_class_init (GstMassinkClass *klass);
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static void gst_massink_init (GstMassink *massink);
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static void gst_massink_set_clock (GstElement *element, GstClock *clock);
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static gboolean gst_massink_open_audio (GstMassink *sink);
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//static void gst_massink_close_audio (GstMassink *sink);
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static GstElementStateReturn gst_massink_change_state (GstElement *element);
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static gboolean gst_massink_sync_parms (GstMassink *massink);
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static GstPadLinkReturn gst_massink_sinkconnect (GstPad *pad, const GstCaps *caps);
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static void gst_massink_chain (GstPad *pad, GstData *_data);
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static void gst_massink_set_property (GObject *object, guint prop_id,
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const GValue *value, GParamSpec *pspec);
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static void gst_massink_get_property (GObject *object, guint prop_id,
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GValue *value, GParamSpec *pspec);
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#define GST_TYPE_MASSINK_DEPTHS (gst_massink_depths_get_type())
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static GType
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gst_massink_depths_get_type (void)
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{
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static GType massink_depths_type = 0;
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static GEnumValue massink_depths[] = {
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{8, "8", "8 Bits"},
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{16, "16", "16 Bits"},
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{0, NULL, NULL},
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};
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if (!massink_depths_type) {
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massink_depths_type = g_enum_register_static("GstMassinkDepths", massink_depths);
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}
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return massink_depths_type;
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}
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static GstElementClass *parent_class = NULL;
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/*static guint gst_massink_signals[LAST_SIGNAL] = { 0 }; */
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GType
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gst_massink_get_type (void)
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{
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static GType massink_type = 0;
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if (!massink_type) {
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static const GTypeInfo massink_info = {
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sizeof(GstMassinkClass),
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gst_massink_base_init,
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NULL,
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(GClassInitFunc)gst_massink_class_init,
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NULL,
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NULL,
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sizeof(GstMassink),
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0,
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(GInstanceInitFunc)gst_massink_init,
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};
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massink_type = g_type_register_static(GST_TYPE_ELEMENT, "GstMassink", &massink_info, 0);
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}
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return massink_type;
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}
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static void
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gst_massink_base_init (gpointer g_class)
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{
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static GstElementDetails massink_details = GST_ELEMENT_DETAILS (
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"Esound audio sink",
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"Sink/Audio",
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"Plays audio to a MAS server",
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"Zeeshan Ali <zak147@yahoo.com>"
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);
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_factory));
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gst_element_class_set_details (element_class, &massink_details);
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}
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static void
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gst_massink_class_init (GstMassinkClass *klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass*)klass;
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gstelement_class = (GstElementClass*)klass;
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parent_class = g_type_class_ref(GST_TYPE_ELEMENT);
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g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_MUTE,
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g_param_spec_boolean("mute","mute","mute",
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TRUE,G_PARAM_READWRITE)); /* CHECKME */
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g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_DEPTH,
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g_param_spec_enum("depth","depth","depth",
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GST_TYPE_MASSINK_DEPTHS,16,G_PARAM_READWRITE)); /* CHECKME! */
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g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_RATE,
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g_param_spec_int("frequency","frequency","frequency",
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G_MININT,G_MAXINT,0,G_PARAM_READWRITE)); /* CHECKME */
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g_object_class_install_property(G_OBJECT_CLASS(klass), ARG_HOST,
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g_param_spec_string("host","host","host",
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NULL, G_PARAM_READWRITE)); /* CHECKME */
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gobject_class->set_property = gst_massink_set_property;
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gobject_class->get_property = gst_massink_get_property;
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gstelement_class->change_state = gst_massink_change_state;
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gstelement_class->set_clock = gst_massink_set_clock;
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}
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static void
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gst_massink_set_clock (GstElement *element, GstClock *clock)
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{
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GstMassink *massink;
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massink = GST_MASSINK (element);
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massink->clock = clock;
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}
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static void
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gst_massink_init(GstMassink *massink)
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{
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massink->sinkpad = gst_pad_new_from_template (
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gst_static_pad_template_get (&sink_factory), "sink");
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gst_element_add_pad(GST_ELEMENT(massink), massink->sinkpad);
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gst_pad_set_chain_function(massink->sinkpad, GST_DEBUG_FUNCPTR(gst_massink_chain));
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gst_pad_set_link_function(massink->sinkpad, gst_massink_sinkconnect);
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massink->mute = FALSE;
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massink->format = 16;
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massink->depth = 16;
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massink->channels = 2;
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massink->frequency = 44100;
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massink->host = NULL;
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}
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static gboolean
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gst_massink_sync_parms (GstMassink *massink)
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{
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g_return_val_if_fail (massink != NULL, FALSE);
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g_return_val_if_fail (GST_IS_MASSINK (massink), FALSE);
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//gst_massink_close_audio (massink);
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//return gst_massink_open_audio (massink);
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return 1;
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}
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static GstPadLinkReturn
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gst_massink_sinkconnect (GstPad *pad, const GstCaps *caps)
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{
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GstMassink *massink;
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massink = GST_MASSINK (gst_pad_get_parent (pad));
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if (gst_massink_sync_parms (massink))
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return GST_PAD_LINK_OK;
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return GST_PAD_LINK_REFUSED;
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}
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static void
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gst_massink_chain (GstPad *pad, GstData *_data)
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{
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GstBuffer *buf = GST_BUFFER (_data);
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gint32 err;
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g_return_if_fail(pad != NULL);
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g_return_if_fail(GST_IS_PAD(pad));
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g_return_if_fail(buf != NULL);
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GstMassink *massink = GST_MASSINK (gst_pad_get_parent (pad));
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if (massink->clock && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
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GST_DEBUG ("massink: clock wait: %llu\n", GST_BUFFER_TIMESTAMP (buf));
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gst_element_wait (GST_ELEMENT (massink), GST_BUFFER_TIMESTAMP (buf));
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}
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if (GST_BUFFER_DATA (buf) != NULL) {
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if (!massink->mute) {
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GST_DEBUG ("massink: data=%p size=%d", GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
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if (GST_BUFFER_SIZE (buf) > BUFFER_SIZE) {
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gst_buffer_unref (buf);
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return;
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}
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massink->data->length = GST_BUFFER_SIZE (buf);
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memcpy (massink->data->segment, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf));
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err = mas_send (massink->audio_channel, massink->data);
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if (err < 0) {
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g_print ("error sending data to MAS server\n");
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gst_buffer_unref (buf);
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return;
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}
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/* FIXME: Please correct the Timestamping if its wrong */
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massink->data->header.media_timestamp += massink->data->length / 4;
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massink->data->header.sequence++;
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}
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}
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gst_buffer_unref (buf);
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}
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static void
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gst_massink_set_property (GObject *object, guint prop_id, const GValue *value, GParamSpec *pspec)
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{
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GstMassink *massink;
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/* it's not null if we got it, but it might not be ours */
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g_return_if_fail(GST_IS_MASSINK(object));
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massink = GST_MASSINK(object);
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switch (prop_id) {
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case ARG_MUTE:
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massink->mute = g_value_get_boolean (value);
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break;
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case ARG_DEPTH:
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massink->depth = g_value_get_enum (value);
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gst_massink_sync_parms (massink);
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break;
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case ARG_CHANNELS:
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massink->channels = g_value_get_enum (value);
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gst_massink_sync_parms (massink);
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break;
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case ARG_RATE:
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massink->frequency = g_value_get_int (value);
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gst_massink_sync_parms (massink);
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break;
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case ARG_HOST:
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if (massink->host != NULL) g_free(massink->host);
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if (g_value_get_string (value) == NULL)
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massink->host = NULL;
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else
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massink->host = g_strdup (g_value_get_string (value));
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break;
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default:
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break;
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}
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}
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static void
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gst_massink_get_property (GObject *object, guint prop_id, GValue *value, GParamSpec *pspec)
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{
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GstMassink *massink;
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/* it's not null if we got it, but it might not be ours */
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g_return_if_fail(GST_IS_MASSINK(object));
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massink = GST_MASSINK(object);
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switch (prop_id) {
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case ARG_MUTE:
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g_value_set_boolean (value, massink->mute);
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break;
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case ARG_DEPTH:
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g_value_set_enum (value, massink->depth);
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break;
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case ARG_CHANNELS:
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g_value_set_enum (value, massink->channels);
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break;
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case ARG_RATE:
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g_value_set_int (value, massink->frequency);
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break;
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case ARG_HOST:
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g_value_set_string (value, massink->host);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static gboolean
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plugin_init (GstPlugin *plugin)
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{
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if (!gst_element_register (plugin, "massink", GST_RANK_NONE,
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GST_TYPE_MASSINK)){
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return FALSE;
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}
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return TRUE;
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}
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GST_PLUGIN_DEFINE (
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GST_VERSION_MAJOR,
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GST_VERSION_MINOR,
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"massink",
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"uses MAS for audio output",
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plugin_init,
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VERSION,
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"LGPL",
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GST_PACKAGE,
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GST_ORIGIN
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);
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static gboolean
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gst_massink_open_audio (GstMassink *massink)
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{
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gint32 err;
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char *ratestring = g_malloc (16);
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char *bps = g_malloc (8);
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struct mas_data_characteristic* dc;
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g_print ("Connecting to MAS.\n");
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masc_log_verbosity (MAS_VERBLVL_DEBUG);
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err = mas_init();
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if (err < 0) {
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GST_DEBUG ("connection with local MAS server failed.");
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exit (1);
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}
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GST_DEBUG ("Establishing audio output channel.");
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mas_make_data_channel ("Gstreamer", &massink->audio_channel, &massink->audio_source, &massink->audio_sink);
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mas_asm_get_port_by_name (0, "default_mix_sink", &massink->mix_sink);
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GST_DEBUG ("Instantiating endian device.");
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err = mas_asm_instantiate_device ("endian", 0, 0, &massink->endian);
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if (err < 0) {
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GST_DEBUG ("Failed to instantiate endian converter device");
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exit(1);
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}
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mas_asm_get_port_by_name (massink->endian, "endian_sink", &massink->endian_sink);
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mas_asm_get_port_by_name (massink->endian, "endian_source", &massink->endian_source);
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sprintf (bps, "%u", massink->depth);
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sprintf (ratestring, "%u", massink->frequency);
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GST_DEBUG ("Connecting net -> endian.");
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masc_make_dc (&dc, 6);
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/* wav weirdness: 8 bit data is unsigned, >8 data is signed. */
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masc_append_dc_key_value (dc, "format", (massink->depth==8) ? "ulinear":"linear");
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masc_append_dc_key_value (dc, "resolution", bps);
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masc_append_dc_key_value (dc, "sampling rate", ratestring);
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masc_append_dc_key_value (dc, "channels", "2");
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masc_append_dc_key_value (dc, "endian", "little");
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err = mas_asm_connect_source_sink (massink->audio_source, massink->endian_sink, dc);
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if ( err < 0 ) {
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GST_DEBUG ("Failed to connect net audio output to endian");
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return -1;
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}
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/* The next device is 'if needed' only. After the following if()
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statement, open_source will contain the current unconnected
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source in the path (will be either endian_source or
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squant_source in this case)
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*/
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massink->open_source = massink->endian_source;
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if (massink->depth != 16) {
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GST_DEBUG ("Sample resolution is not 16 bit/sample, instantiating squant device.");
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err = mas_asm_instantiate_device ("squant", 0, 0, &massink->squant);
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if (err < 0) {
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GST_DEBUG ("Failed to instantiate squant device");
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return -1;
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}
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mas_asm_get_port_by_name (massink->squant, "squant_sink", &massink->squant_sink);
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mas_asm_get_port_by_name (massink->squant, "squant_source", &massink->squant_source);
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GST_DEBUG ("Connecting endian -> squant.");
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masc_make_dc (&dc, 6);
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masc_append_dc_key_value (dc,"format",(massink->depth==8) ? "ulinear":"linear");
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masc_append_dc_key_value (dc, "resolution", bps);
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masc_append_dc_key_value (dc, "sampling rate", ratestring);
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masc_append_dc_key_value (dc, "channels", "2");
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masc_append_dc_key_value (dc, "endian", "host");
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err = mas_asm_connect_source_sink (massink->endian_source, massink->squant_sink, dc);
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if (err < 0) {
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GST_DEBUG ("Failed to connect endian output to squant");
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return -1;
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}
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/* sneaky: the squant device is optional -> pretend it isn't there */
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massink->open_source = massink->squant_source;
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}
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|
|
/* Another 'if necessary' device, as above */
|
|
if (massink->frequency != 44100) {
|
|
GST_DEBUG ("Sample rate is not 44100, instantiating srate device.");
|
|
err = mas_asm_instantiate_device ("srate", 0, 0, &massink->srate);
|
|
|
|
if (err < 0) {
|
|
GST_DEBUG ("Failed to instantiate srate device");
|
|
return -1;
|
|
}
|
|
|
|
mas_asm_get_port_by_name (massink->srate, "sink", &massink->srate_sink);
|
|
mas_asm_get_port_by_name (massink->srate, "source", &massink->srate_source);
|
|
|
|
GST_DEBUG ("Connecting to srate.");
|
|
masc_make_dc (&dc, 6);
|
|
masc_append_dc_key_value (dc, "format", "linear");
|
|
masc_append_dc_key_value (dc, "resolution", "16");
|
|
masc_append_dc_key_value (dc, "sampling rate", ratestring);
|
|
masc_append_dc_key_value (dc, "channels", "2");
|
|
masc_append_dc_key_value (dc, "endian", "host");
|
|
|
|
err = mas_asm_connect_source_sink (massink->open_source, massink->srate_sink, dc);
|
|
|
|
if ( err < 0 ) {
|
|
GST_DEBUG ("Failed to connect to srate");
|
|
return -1;
|
|
}
|
|
|
|
|
|
massink->open_source = massink->srate_source;
|
|
}
|
|
|
|
GST_DEBUG ("Connecting to mix.");
|
|
masc_make_dc(&dc, 6);
|
|
masc_append_dc_key_value (dc, "format", "linear");
|
|
masc_append_dc_key_value (dc, "resolution", "16");
|
|
masc_append_dc_key_value (dc, "sampling rate", "44100");
|
|
masc_append_dc_key_value (dc, "channels", "2");
|
|
masc_append_dc_key_value (dc, "endian", "host");
|
|
|
|
err = mas_asm_connect_source_sink (massink->open_source, massink->mix_sink, dc);
|
|
|
|
if ( err < 0 ) {
|
|
GST_DEBUG ("Failed to connect to mixer");
|
|
return -1;
|
|
}
|
|
|
|
GST_FLAG_SET (massink, GST_MASSINK_OPEN);
|
|
|
|
masc_make_mas_data (&massink->data, BUFFER_SIZE);
|
|
|
|
massink->data->header.type = 10;
|
|
|
|
massink->data->header.media_timestamp = 0;
|
|
massink->data->header.sequence = 0;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/*static void
|
|
gst_massink_close_audio (GstMassink *massink)
|
|
{
|
|
mas_free_device(massink->endian);
|
|
mas_free_device(massink->srate);
|
|
mas_free_device(massink->squant);
|
|
|
|
mas_free_port(massink->mix_sink);
|
|
mas_free_port(massink->srate_source);
|
|
mas_free_port(massink->srate_sink);
|
|
mas_free_port(massink->audio_source);
|
|
mas_free_port(massink->audio_sink);
|
|
mas_free_port(massink->endian_source);
|
|
mas_free_port(massink->endian_sink);
|
|
mas_free_port(massink->squant_source);
|
|
mas_free_port(massink->squant_sink);
|
|
mas_free_port(massink->open_source);
|
|
|
|
mas_free_channel (massink->audio_channel);
|
|
masc_destroy_mas_data (massink->data);
|
|
|
|
g_free (ratestring);
|
|
g_free (bps);
|
|
|
|
GST_FLAG_UNSET (massink, GST_MASSINK_OPEN);
|
|
|
|
GST_DEBUG ("massink: closed sound channel");
|
|
}*/
|
|
|
|
static GstElementStateReturn
|
|
gst_massink_change_state (GstElement *element)
|
|
{
|
|
g_return_val_if_fail (GST_IS_MASSINK (element), FALSE);
|
|
|
|
/* if going down into NULL state, close the fd if it's open */
|
|
if (GST_STATE_PENDING (element) == GST_STATE_NULL) {
|
|
//if (GST_FLAG_IS_SET (element, GST_MASSINK_OPEN))
|
|
//gst_massink_close_audio (GST_MASSINK (element));
|
|
/* otherwise (READY or higher) we need to open the fd */
|
|
} else {
|
|
if (!GST_FLAG_IS_SET (element, GST_MASSINK_OPEN)) {
|
|
if (!gst_massink_open_audio (GST_MASSINK (element)))
|
|
return GST_STATE_FAILURE;
|
|
}
|
|
}
|
|
|
|
if (GST_ELEMENT_CLASS (parent_class)->change_state)
|
|
return GST_ELEMENT_CLASS (parent_class)->change_state (element);
|
|
return GST_STATE_SUCCESS;
|
|
}
|
|
|