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e7f919986a
Original commit message from CVS: * gst/rtp/gstasteriskh263.c: * gst/rtp/gstrtpL16depay.c: * gst/rtp/gstrtpac3depay.c: * gst/rtp/gstrtpamrpay.c: * gst/rtp/gstrtpdepay.c: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtph263depay.c: * gst/rtp/gstrtph263pdepay.c: * gst/rtp/gstrtph263ppay.c: * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtpmp2tdepay.c: * gst/rtp/gstrtpmp4adepay.c: * gst/rtp/gstrtpmp4gdepay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vdepay.c: * gst/rtp/gstrtpmpadepay.c: * gst/rtp/gstrtpmpvdepay.c: * gst/rtp/gstrtpsv3vdepay.c: * gst/rtp/gstrtptheoradepay.c: * gst/rtp/gstrtptheorapay.c: * gst/rtp/gstrtpvorbisdepay.c: * gst/rtp/gstrtpvorbispay.c: Remove copy/paste unused code (property setters and getter) found by the coverage suite (yay, saves ~20k on disk).
249 lines
7 KiB
C
249 lines
7 KiB
C
/* 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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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <gst/rtp/gstrtpbuffer.h>
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#include <string.h>
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#include "gstrtpmp4vdepay.h"
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GST_DEBUG_CATEGORY_STATIC (rtpmp4vdepay_debug);
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#define GST_CAT_DEFAULT (rtpmp4vdepay_debug)
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/* elementfactory information */
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static const GstElementDetails gst_rtp_mp4vdepay_details =
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GST_ELEMENT_DETAILS ("RTP packet depayloader",
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"Codec/Depayloader/Network",
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"Extracts MPEG4 video from RTP packets (RFC 3016)",
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"Wim Taymans <wim@fluendo.com>");
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static GstStaticPadTemplate gst_rtp_mp4v_depay_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("video/mpeg,"
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"mpegversion=(int) 4," "systemstream=(boolean)false")
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);
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static GstStaticPadTemplate gst_rtp_mp4v_depay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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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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"media = (string) \"video\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) [1, MAX ], " "encoding-name = (string) \"MP4V-ES\""
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/* All optional parameters
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*
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* "profile-level-id=[1,MAX]"
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* "config="
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*/
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)
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);
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GST_BOILERPLATE (GstRtpMP4VDepay, gst_rtp_mp4v_depay, GstBaseRTPDepayload,
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GST_TYPE_BASE_RTP_DEPAYLOAD);
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static void gst_rtp_mp4v_depay_finalize (GObject * object);
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static gboolean gst_rtp_mp4v_depay_setcaps (GstBaseRTPDepayload * depayload,
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GstCaps * caps);
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static GstBuffer *gst_rtp_mp4v_depay_process (GstBaseRTPDepayload * depayload,
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GstBuffer * buf);
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static GstStateChangeReturn gst_rtp_mp4v_depay_change_state (GstElement *
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element, GstStateChange transition);
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static void
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gst_rtp_mp4v_depay_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_mp4v_depay_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_mp4v_depay_sink_template));
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gst_element_class_set_details (element_class, &gst_rtp_mp4vdepay_details);
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}
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static void
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gst_rtp_mp4v_depay_class_init (GstRtpMP4VDepayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
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gobject_class->finalize = gst_rtp_mp4v_depay_finalize;
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gstelement_class->change_state = gst_rtp_mp4v_depay_change_state;
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gstbasertpdepayload_class->process = gst_rtp_mp4v_depay_process;
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gstbasertpdepayload_class->set_caps = gst_rtp_mp4v_depay_setcaps;
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GST_DEBUG_CATEGORY_INIT (rtpmp4vdepay_debug, "rtpmp4vdepay", 0,
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"MPEG4 video RTP Depayloader");
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}
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static void
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gst_rtp_mp4v_depay_init (GstRtpMP4VDepay * rtpmp4vdepay,
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GstRtpMP4VDepayClass * klass)
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{
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rtpmp4vdepay->adapter = gst_adapter_new ();
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}
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static void
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gst_rtp_mp4v_depay_finalize (GObject * object)
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{
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GstRtpMP4VDepay *rtpmp4vdepay;
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rtpmp4vdepay = GST_RTP_MP4V_DEPAY (object);
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g_object_unref (rtpmp4vdepay->adapter);
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rtpmp4vdepay->adapter = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_rtp_mp4v_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
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{
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GstStructure *structure;
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GstRtpMP4VDepay *rtpmp4vdepay;
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GstCaps *srccaps;
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const gchar *str;
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gint clock_rate = 90000; /* default */
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rtpmp4vdepay = GST_RTP_MP4V_DEPAY (depayload);
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structure = gst_caps_get_structure (caps, 0);
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if (gst_structure_has_field (structure, "clock-rate"))
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gst_structure_get_int (structure, "clock-rate", &clock_rate);
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depayload->clock_rate = clock_rate;
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srccaps = gst_caps_new_simple ("video/mpeg",
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"mpegversion", G_TYPE_INT, 4,
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"systemstream", G_TYPE_BOOLEAN, FALSE, NULL);
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if ((str = gst_structure_get_string (structure, "config"))) {
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GValue v = { 0 };
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g_value_init (&v, GST_TYPE_BUFFER);
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if (gst_value_deserialize (&v, str)) {
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GstBuffer *buffer;
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buffer = gst_value_get_buffer (&v);
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gst_buffer_ref (buffer);
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g_value_unset (&v);
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gst_caps_set_simple (srccaps,
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"codec_data", GST_TYPE_BUFFER, buffer, NULL);
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} else {
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g_warning ("cannot convert config to buffer");
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}
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}
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gst_pad_set_caps (depayload->srcpad, srccaps);
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gst_caps_unref (srccaps);
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return TRUE;
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}
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static GstBuffer *
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gst_rtp_mp4v_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
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{
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GstRtpMP4VDepay *rtpmp4vdepay;
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GstBuffer *outbuf;
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rtpmp4vdepay = GST_RTP_MP4V_DEPAY (depayload);
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if (!gst_rtp_buffer_validate (buf))
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goto bad_packet;
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/* flush remaining data on discont */
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if (GST_BUFFER_IS_DISCONT (buf))
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gst_adapter_clear (rtpmp4vdepay->adapter);
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outbuf = gst_rtp_buffer_get_payload_buffer (buf);
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gst_adapter_push (rtpmp4vdepay->adapter, outbuf);
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/* if this was the last packet of the VOP, create and push a buffer */
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if (gst_rtp_buffer_get_marker (buf)) {
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guint avail;
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avail = gst_adapter_available (rtpmp4vdepay->adapter);
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outbuf = gst_adapter_take_buffer (rtpmp4vdepay->adapter, avail);
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gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
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GST_DEBUG ("gst_rtp_mp4v_depay_chain: pushing buffer of size %d",
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GST_BUFFER_SIZE (outbuf));
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return outbuf;
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}
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return NULL;
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bad_packet:
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{
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GST_ELEMENT_WARNING (rtpmp4vdepay, STREAM, DECODE,
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("Packet did not validate"), (NULL));
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return NULL;
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}
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}
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static GstStateChangeReturn
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gst_rtp_mp4v_depay_change_state (GstElement * element,
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GstStateChange transition)
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{
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GstRtpMP4VDepay *rtpmp4vdepay;
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GstStateChangeReturn ret;
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rtpmp4vdepay = GST_RTP_MP4V_DEPAY (element);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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gst_adapter_clear (rtpmp4vdepay->adapter);
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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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default:
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break;
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}
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return ret;
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}
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gboolean
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gst_rtp_mp4v_depay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpmp4vdepay",
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GST_RANK_MARGINAL, GST_TYPE_RTP_MP4V_DEPAY);
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}
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