2006-02-09 14:20:14 +00:00
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/* GStreamer
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* Copyright (C) <2005> Wim Taymans <wim@fluendo.com>
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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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/* Element-Checklist-Version: 5 */
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2006-06-20 14:57:09 +00:00
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/**
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* SECTION:element-rtpdec
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*
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* <refsect2>
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* <para>
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* A simple RTP session manager used internally by rtspsrc.
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* </para>
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* </refsect2>
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*
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* Last reviewed on 2006-06-20 (0.10.4)
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*/
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2006-02-09 14:20:14 +00:00
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#include "gstrtpdec.h"
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2006-06-22 19:31:04 +00:00
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GST_DEBUG_CATEGORY_STATIC (rtpdec_debug);
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2006-02-09 14:20:14 +00:00
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#define GST_CAT_DEFAULT (rtpdec_debug)
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/* elementfactory information */
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Define GstElementDetails as const and also static (when defined as global)
Original commit message from CVS:
* ext/aalib/gstaasink.c:
* ext/annodex/gstcmmldec.c:
* ext/annodex/gstcmmlenc.c:
* ext/cairo/gsttextoverlay.c:
* ext/cairo/gsttimeoverlay.c:
* ext/cdio/gstcdiocddasrc.c:
* ext/dv/gstdvdec.c:
* ext/dv/gstdvdemux.c:
* ext/esd/esdmon.c:
* ext/esd/esdsink.c:
* ext/flac/gstflacenc.c:
* ext/flac/gstflactag.c:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init):
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init):
* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init):
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init):
* ext/gdk_pixbuf/pixbufscale.c:
* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
* ext/jpeg/gstjpegdec.c:
* ext/jpeg/gstjpegenc.c:
* ext/jpeg/gstsmokedec.c:
* ext/jpeg/gstsmokeenc.c:
* ext/libcaca/gstcacasink.c:
* ext/libmng/gstmngdec.c:
* ext/libmng/gstmngenc.c:
* ext/libpng/gstpngdec.c:
* ext/libpng/gstpngenc.c:
* ext/mikmod/gstmikmod.c:
* ext/raw1394/gstdv1394src.c:
* ext/shout2/gstshout2.c: (gst_shout2send_init):
* ext/shout2/gstshout2.h:
* ext/speex/gstspeexdec.c:
* ext/speex/gstspeexenc.c:
* gst/alpha/gstalpha.c:
* gst/alpha/gstalphacolor.c:
* gst/apetag/gstapedemux.c:
* gst/auparse/gstauparse.c:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_base_init):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_base_init):
* gst/avi/gstavidemux.c: (gst_avi_demux_base_init):
* gst/avi/gstavimux.c: (gst_avimux_base_init):
* gst/cutter/gstcutter.c:
* gst/debug/breakmydata.c:
* gst/debug/efence.c:
* gst/debug/gstnavigationtest.c:
* gst/debug/gstnavseek.c:
* gst/debug/negotiation.c:
* gst/debug/progressreport.c:
* gst/debug/testplugin.c:
* gst/effectv/gstaging.c:
* gst/effectv/gstdice.c:
* gst/effectv/gstedge.c:
* gst/effectv/gstquark.c:
* gst/effectv/gstrev.c:
* gst/effectv/gstshagadelic.c:
* gst/effectv/gstvertigo.c:
* gst/effectv/gstwarp.c:
* gst/flx/gstflxdec.c:
* gst/goom/gstgoom.c:
* gst/icydemux/gsticydemux.c:
* gst/id3demux/gstid3demux.c:
* gst/interleave/deinterleave.c:
* gst/interleave/interleave.c:
* gst/law/alaw-decode.c: (gst_alawdec_base_init):
* gst/law/alaw-encode.c: (gst_alawenc_base_init):
* gst/law/mulaw-decode.c: (gst_mulawdec_base_init):
* gst/law/mulaw-encode.c: (gst_mulawenc_base_init):
* gst/level/gstlevel.c:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init):
* gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init):
* gst/median/gstmedian.c:
* gst/monoscope/gstmonoscope.c:
* gst/multipart/multipartdemux.c:
* gst/multipart/multipartmux.c:
* gst/oldcore/gstaggregator.c:
* gst/oldcore/gstfdsink.c:
* gst/oldcore/gstmd5sink.c:
* gst/oldcore/gstmultifilesrc.c:
* gst/oldcore/gstpipefilter.c:
* gst/oldcore/gstshaper.c:
* gst/oldcore/gststatistics.c:
* gst/rtp/gstasteriskh263.c:
* gst/rtp/gstrtpL16depay.c:
* gst/rtp/gstrtpL16pay.c:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpamrpay.c:
* gst/rtp/gstrtpdepay.c:
* gst/rtp/gstrtpgsmpay.c:
* gst/rtp/gstrtph263pay.c:
* gst/rtp/gstrtph263pdepay.c:
* gst/rtp/gstrtph263ppay.c:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4vdepay.c:
* gst/rtp/gstrtpmp4vpay.c:
* gst/rtp/gstrtpmpadepay.c:
* gst/rtp/gstrtpmpapay.c:
* gst/rtp/gstrtppcmadepay.c:
* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmudepay.c:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtpspeexdepay.c:
* gst/rtp/gstrtpspeexpay.c:
* gst/rtsp/gstrtpdec.c:
* gst/rtsp/gstrtspsrc.c:
* gst/smpte/gstsmpte.c:
* gst/udp/gstdynudpsink.c:
* gst/udp/gstmultiudpsink.c:
* gst/udp/gstudpsink.c:
* gst/udp/gstudpsrc.c:
* gst/videobox/gstvideobox.c:
* gst/videofilter/gstgamma.c: (gst_gamma_base_init):
* gst/videofilter/gstvideobalance.c:
* gst/videofilter/gstvideoflip.c:
* gst/videofilter/gstvideotemplate.c:
(gst_videotemplate_base_init):
* gst/videomixer/videomixer.c:
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
(gst_wavparse_class_init), (gst_wavparse_dispose),
(gst_wavparse_reset), (gst_wavparse_init),
(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
(gst_wavparse_peek_chunk), (gst_wavparse_stream_headers),
(gst_wavparse_parse_stream_init), (gst_wavparse_send_event),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
(gst_wavparse_chain), (gst_wavparse_srcpad_event),
(gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull),
(gst_wavparse_change_state):
* gst/wavparse/gstwavparse.h:
* sys/oss/gstossmixerelement.c:
* sys/oss/gstosssink.c:
* sys/oss/gstosssrc.c:
* sys/osxaudio/gstosxaudioelement.c:
* sys/osxaudio/gstosxaudiosink.c:
* sys/osxaudio/gstosxaudiosrc.c:
* sys/sunaudio/gstsunaudiomixer.c:
* sys/sunaudio/gstsunaudiosink.c:
Define GstElementDetails as const and also static (when defined as
global)
2006-04-25 21:39:46 +00:00
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static const GstElementDetails rtpdec_details =
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GST_ELEMENT_DETAILS ("RTP Decoder",
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2006-02-09 14:20:14 +00:00
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"Codec/Parser/Network",
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"Accepts raw RTP and RTCP packets and sends them forward",
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"Wim Taymans <wim@fluendo.com>");
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/* GstRTPDec signals and args */
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enum
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{
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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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{
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ARG_0,
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ARG_SKIP
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/* FILL ME */
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};
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static GstStaticPadTemplate gst_rtpdec_src_rtp_template =
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GST_STATIC_PAD_TEMPLATE ("srcrtp",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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static GstStaticPadTemplate gst_rtpdec_src_rtcp_template =
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GST_STATIC_PAD_TEMPLATE ("srcrtcp",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static GstStaticPadTemplate gst_rtpdec_sink_rtp_template =
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GST_STATIC_PAD_TEMPLATE ("sinkrtp",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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static GstStaticPadTemplate gst_rtpdec_sink_rtcp_template =
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GST_STATIC_PAD_TEMPLATE ("sinkrtcp",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static void gst_rtpdec_class_init (gpointer g_class);
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static void gst_rtpdec_init (GstRTPDec * rtpdec);
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static GstCaps *gst_rtpdec_getcaps (GstPad * pad);
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static GstFlowReturn gst_rtpdec_chain_rtp (GstPad * pad, GstBuffer * buffer);
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static GstFlowReturn gst_rtpdec_chain_rtcp (GstPad * pad, GstBuffer * buffer);
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static void gst_rtpdec_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_rtpdec_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static GstStateChangeReturn gst_rtpdec_change_state (GstElement * element,
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GstStateChange transition);
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static GstElementClass *parent_class = NULL;
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/*static guint gst_rtpdec_signals[LAST_SIGNAL] = { 0 };*/
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GType
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gst_rtpdec_get_type (void)
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{
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static GType rtpdec_type = 0;
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if (!rtpdec_type) {
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static const GTypeInfo rtpdec_info = {
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sizeof (GstRTPDecClass), NULL,
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NULL,
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(GClassInitFunc) gst_rtpdec_class_init,
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NULL,
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NULL,
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sizeof (GstRTPDec),
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0,
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(GInstanceInitFunc) gst_rtpdec_init,
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};
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rtpdec_type =
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g_type_register_static (GST_TYPE_ELEMENT, "GstRTPDec", &rtpdec_info, 0);
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}
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return rtpdec_type;
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}
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static void
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gst_rtpdec_class_init (gpointer g_class)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstRTPDecClass *klass;
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klass = (GstRTPDecClass *) g_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtpdec_src_rtp_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtpdec_src_rtcp_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtpdec_sink_rtp_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtpdec_sink_rtcp_template));
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gst_element_class_set_details (gstelement_class, &rtpdec_details);
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gobject_class->set_property = gst_rtpdec_set_property;
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gobject_class->get_property = gst_rtpdec_get_property;
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2006-06-20 14:57:09 +00:00
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/* FIXME, this is unused and probably copied from somewhere */
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SKIP,
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g_param_spec_int ("skip", "Skip", "skip (unused)", G_MININT, G_MAXINT, 0,
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G_PARAM_READWRITE));
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2006-02-09 14:20:14 +00:00
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2006-04-08 21:21:45 +00:00
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parent_class = g_type_class_peek_parent (klass);
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2006-02-09 14:20:14 +00:00
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gstelement_class->change_state = gst_rtpdec_change_state;
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GST_DEBUG_CATEGORY_INIT (rtpdec_debug, "rtpdec", 0, "RTP decoder");
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}
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static void
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gst_rtpdec_init (GstRTPDec * rtpdec)
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{
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/* the input rtp pad */
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rtpdec->sink_rtp =
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2006-03-15 16:17:12 +00:00
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gst_pad_new_from_static_template (&gst_rtpdec_sink_rtp_template,
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"sinkrtp");
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2006-02-09 14:20:14 +00:00
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gst_pad_set_getcaps_function (rtpdec->sink_rtp, gst_rtpdec_getcaps);
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gst_pad_set_chain_function (rtpdec->sink_rtp, gst_rtpdec_chain_rtp);
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gst/rtsp/gstrtpdec.c: Add pads after setting them up.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps):
Add pads after setting them up.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_finalize),
(gst_rtspsrc_free_stream), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_combine_flows), (gst_rtspsrc_loop),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Fix interleaved mode.
- Protect streaming with lock.
- Combine flows
- set caps on outgoing buffers.
- strip trailing \0 from data packets.
- Configure RTP/RTCP in stream.
Use DEBUG_OBJECT more.
2006-08-16 09:48:26 +00:00
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gst_element_add_pad (GST_ELEMENT (rtpdec), rtpdec->sink_rtp);
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2006-02-09 14:20:14 +00:00
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/* the input rtcp pad */
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rtpdec->sink_rtcp =
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2006-03-15 16:17:12 +00:00
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gst_pad_new_from_static_template (&gst_rtpdec_sink_rtcp_template,
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"sinkrtcp");
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2006-02-09 14:20:14 +00:00
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gst_pad_set_chain_function (rtpdec->sink_rtcp, gst_rtpdec_chain_rtcp);
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gst/rtsp/gstrtpdec.c: Add pads after setting them up.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps):
Add pads after setting them up.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_finalize),
(gst_rtspsrc_free_stream), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_combine_flows), (gst_rtspsrc_loop),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Fix interleaved mode.
- Protect streaming with lock.
- Combine flows
- set caps on outgoing buffers.
- strip trailing \0 from data packets.
- Configure RTP/RTCP in stream.
Use DEBUG_OBJECT more.
2006-08-16 09:48:26 +00:00
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gst_element_add_pad (GST_ELEMENT (rtpdec), rtpdec->sink_rtcp);
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2006-02-09 14:20:14 +00:00
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/* the output rtp pad */
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rtpdec->src_rtp =
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2006-03-15 16:17:12 +00:00
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gst_pad_new_from_static_template (&gst_rtpdec_src_rtp_template, "srcrtp");
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2006-02-09 14:20:14 +00:00
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gst_pad_set_getcaps_function (rtpdec->src_rtp, gst_rtpdec_getcaps);
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gst_element_add_pad (GST_ELEMENT (rtpdec), rtpdec->src_rtp);
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/* the output rtcp pad */
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rtpdec->src_rtcp =
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2006-03-15 16:17:12 +00:00
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gst_pad_new_from_static_template (&gst_rtpdec_src_rtcp_template,
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"srcrtcp");
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2006-02-09 14:20:14 +00:00
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gst_element_add_pad (GST_ELEMENT (rtpdec), rtpdec->src_rtcp);
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}
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static GstCaps *
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gst_rtpdec_getcaps (GstPad * pad)
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{
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GstRTPDec *src;
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GstPad *other;
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GstCaps *caps;
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src = GST_RTPDEC (GST_PAD_PARENT (pad));
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gst/rtsp/gstrtpdec.c: Add pads after setting them up.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps):
Add pads after setting them up.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_finalize),
(gst_rtspsrc_free_stream), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_combine_flows), (gst_rtspsrc_loop),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Fix interleaved mode.
- Protect streaming with lock.
- Combine flows
- set caps on outgoing buffers.
- strip trailing \0 from data packets.
- Configure RTP/RTCP in stream.
Use DEBUG_OBJECT more.
2006-08-16 09:48:26 +00:00
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other = (pad == src->src_rtp ? src->sink_rtp : src->src_rtp);
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2006-02-09 14:20:14 +00:00
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caps = gst_pad_peer_get_caps (other);
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if (caps == NULL)
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caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
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return caps;
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}
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static GstFlowReturn
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gst_rtpdec_chain_rtp (GstPad * pad, GstBuffer * buffer)
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{
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GstRTPDec *src;
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src = GST_RTPDEC (GST_PAD_PARENT (pad));
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GST_DEBUG ("got rtp packet");
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return gst_pad_push (src->src_rtp, buffer);
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}
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static GstFlowReturn
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gst_rtpdec_chain_rtcp (GstPad * pad, GstBuffer * buffer)
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{
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GST_DEBUG ("got rtcp packet");
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gst_buffer_unref (buffer);
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return GST_FLOW_OK;
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}
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static void
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gst_rtpdec_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstRTPDec *src;
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src = GST_RTPDEC (object);
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switch (prop_id) {
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case ARG_SKIP:
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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_rtpdec_get_property (GObject * object, guint prop_id, GValue * value,
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GParamSpec * pspec)
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{
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GstRTPDec *src;
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src = GST_RTPDEC (object);
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switch (prop_id) {
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case ARG_SKIP:
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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 GstStateChangeReturn
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gst_rtpdec_change_state (GstElement * element, GstStateChange transition)
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{
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GstStateChangeReturn ret;
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GstRTPDec *rtpdec;
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rtpdec = GST_RTPDEC (element);
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switch (transition) {
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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break;
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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switch (transition) {
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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break;
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default:
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break;
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}
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return ret;
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}
|