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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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4397c8ffbf
Remove the pt restrictions for all the depayloaders that have an encoding-name. We can use this to autoplug decoders. Remove the encoding-name for all the payloaders with a fixed payload type. We now either have an encoding-name or a pt in the sinkpad caps of a depayloader. See https://bugzilla.gnome.org/show_bug.cgi?id=639292
323 lines
9.4 KiB
C
323 lines
9.4 KiB
C
/* GStreamer
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* Copyright (C) <2005> Wim Taymans <wim.taymans@gmail.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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, 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 <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpsv3vdepay.h"
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GST_DEBUG_CATEGORY (rtpsv3vdepay_debug);
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#define GST_CAT_DEFAULT rtpsv3vdepay_debug
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static GstStaticPadTemplate gst_rtp_sv3v_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/x-svq, " "svqversion = (int) 3")
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);
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static GstStaticPadTemplate gst_rtp_sv3v_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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"clock-rate = (int) 90000, "
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"encoding-name = (string) { \"X-SV3V-ES\", \"X-SORENSON-VIDEO\" , \"X-SORENSONVIDEO\" , \"X-SorensonVideo\" }")
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);
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#define gst_rtp_sv3v_depay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpSV3VDepay, gst_rtp_sv3v_depay,
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GST_TYPE_RTP_BASE_DEPAYLOAD);
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static void gst_rtp_sv3v_depay_finalize (GObject * object);
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static GstStateChangeReturn gst_rtp_sv3v_depay_change_state (GstElement *
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element, GstStateChange transition);
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static GstBuffer *gst_rtp_sv3v_depay_process (GstRTPBaseDepayload * depayload,
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GstBuffer * buf);
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gboolean gst_rtp_sv3v_depay_setcaps (GstRTPBaseDepayload * filter,
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GstCaps * caps);
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static void
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gst_rtp_sv3v_depay_class_init (GstRtpSV3VDepayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
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gstrtpbasedepayload_class->process = gst_rtp_sv3v_depay_process;
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gstrtpbasedepayload_class->set_caps = gst_rtp_sv3v_depay_setcaps;
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gobject_class->finalize = gst_rtp_sv3v_depay_finalize;
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gstelement_class->change_state = gst_rtp_sv3v_depay_change_state;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_sv3v_depay_src_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_sv3v_depay_sink_template));
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP SVQ3 depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts SVQ3 video from RTP packets (no RFC)",
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"Wim Taymans <wim.taymans@gmail.com>");
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}
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static void
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gst_rtp_sv3v_depay_init (GstRtpSV3VDepay * rtpsv3vdepay)
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{
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rtpsv3vdepay->adapter = gst_adapter_new ();
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}
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static void
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gst_rtp_sv3v_depay_finalize (GObject * object)
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{
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GstRtpSV3VDepay *rtpsv3vdepay;
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rtpsv3vdepay = GST_RTP_SV3V_DEPAY (object);
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g_object_unref (rtpsv3vdepay->adapter);
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rtpsv3vdepay->adapter = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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/* only on the sink */
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gboolean
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gst_rtp_sv3v_depay_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps)
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{
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GstStructure *structure = gst_caps_get_structure (caps, 0);
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gint clock_rate;
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if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
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clock_rate = 90000; /* default */
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filter->clock_rate = clock_rate;
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/* will set caps later */
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return TRUE;
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}
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static GstBuffer *
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gst_rtp_sv3v_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
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{
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GstRtpSV3VDepay *rtpsv3vdepay;
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static struct
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{
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guint width, height;
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} resolutions[7] = {
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{
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160, 128}, {
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128, 96}, {
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176, 144}, {
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352, 288}, {
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704, 576}, {
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240, 180}, {
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320, 240}
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};
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gint payload_len;
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guint8 *payload;
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gboolean M;
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gboolean C, S, E;
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GstBuffer *outbuf = NULL;
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guint16 seq;
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GstRTPBuffer rtp = { NULL };
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rtpsv3vdepay = GST_RTP_SV3V_DEPAY (depayload);
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gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
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/* flush on sequence number gaps */
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seq = gst_rtp_buffer_get_seq (&rtp);
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GST_DEBUG ("timestamp %" GST_TIME_FORMAT ", sequence number:%d",
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), seq);
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if (seq != rtpsv3vdepay->nextseq) {
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GST_DEBUG ("Sequence discontinuity, clearing adapter");
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gst_adapter_clear (rtpsv3vdepay->adapter);
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}
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rtpsv3vdepay->nextseq = seq + 1;
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payload_len = gst_rtp_buffer_get_payload_len (&rtp);
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if (payload_len < 3)
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goto bad_packet;
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payload = gst_rtp_buffer_get_payload (&rtp);
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M = gst_rtp_buffer_get_marker (&rtp);
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/* This is all a guess:
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* 1 1 1 1 1 1
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* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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* |0|C|S|E|0|0|0|0|0|0|0|0|0|0|0|0|
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* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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*
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* C: config, packet contains config info
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* S: start, packet contains start of frame
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* E: end, packet contains end of frame
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*/
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/* this seems to indicate a packet with a config string sent before each
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* keyframe */
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C = (payload[0] & 0x40) == 0x40;
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/* redundant with the RTP marker bit */
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S = (payload[0] & 0x20) == 0x20;
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E = (payload[0] & 0x10) == 0x10;
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GST_DEBUG ("M:%d, C:%d, S:%d, E:%d", M, C, S, E);
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GST_MEMDUMP ("incoming buffer", payload, payload_len);
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if (G_UNLIKELY (C)) {
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GstCaps *caps;
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GstBuffer *codec_data;
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GstMapInfo cmap;
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guint8 res;
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GST_DEBUG ("Configuration packet");
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/* if we already have caps, we don't need to do anything. FIXME, check if
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* something changed. */
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if (G_UNLIKELY (gst_pad_has_current_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD
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(depayload)))) {
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GST_DEBUG ("Already configured, skipping config parsing");
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goto beach;
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}
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res = payload[2] >> 5;
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/* width and height, according to http://wiki.multimedia.cx/index.php?title=Sorenson_Video_1#Stream_Format_And_Header */
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if (G_LIKELY (res < 7)) {
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rtpsv3vdepay->width = resolutions[res].width;
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rtpsv3vdepay->height = resolutions[res].height;
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} else {
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/* extended width/height, they're contained in the following 24bit */
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rtpsv3vdepay->width = ((payload[2] & 0x1f) << 7) | (payload[3] >> 1);
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rtpsv3vdepay->height =
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(payload[3] & 0x1) << 11 | payload[4] << 3 | (payload[5] >> 5);
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}
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/* CodecData needs to be 'SEQH' + len (32bit) + data according to
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* ffmpeg's libavcodec/svq3.c:svq3_decode_init */
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codec_data = gst_buffer_new_and_alloc (payload_len + 6);
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gst_buffer_map (codec_data, &cmap, GST_MAP_WRITE);
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memcpy (cmap.data, "SEQH", 4);
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GST_WRITE_UINT32_LE (cmap.data + 4, payload_len - 2);
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memcpy (cmap.data + 8, payload + 2, payload_len - 2);
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GST_MEMDUMP ("codec_data", cmap.data, gst_buffer_get_size (codec_data));
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gst_buffer_unmap (codec_data, &cmap);
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caps = gst_caps_new_simple ("video/x-svq",
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"svqversion", G_TYPE_INT, 3,
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"width", G_TYPE_INT, rtpsv3vdepay->width,
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"height", G_TYPE_INT, rtpsv3vdepay->height,
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"codec_data", GST_TYPE_BUFFER, codec_data, NULL);
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gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (depayload), caps);
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gst_caps_unref (caps);
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GST_DEBUG ("Depayloader now configured");
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rtpsv3vdepay->configured = TRUE;
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goto beach;
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}
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if (G_LIKELY (rtpsv3vdepay->configured)) {
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GstBuffer *tmpbuf;
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GST_DEBUG ("Storing incoming payload");
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/* store data in adapter, stip off 2 bytes header */
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tmpbuf = gst_rtp_buffer_get_payload_subbuffer (&rtp, 2, -1);
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gst_adapter_push (rtpsv3vdepay->adapter, tmpbuf);
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if (G_UNLIKELY (M)) {
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/* frame is completed: push contents of adapter */
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guint avail;
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avail = gst_adapter_available (rtpsv3vdepay->adapter);
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GST_DEBUG ("Returning completed output buffer [%d bytes]", avail);
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outbuf = gst_adapter_take_buffer (rtpsv3vdepay->adapter, avail);
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}
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}
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beach:
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gst_rtp_buffer_unmap (&rtp);
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return outbuf;
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/* ERRORS */
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bad_packet:
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{
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GST_ELEMENT_WARNING (rtpsv3vdepay, STREAM, DECODE,
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(NULL), ("Packet was too short"));
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gst_rtp_buffer_unmap (&rtp);
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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_sv3v_depay_change_state (GstElement * element,
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GstStateChange transition)
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{
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GstRtpSV3VDepay *rtpsv3vdepay;
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GstStateChangeReturn ret;
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rtpsv3vdepay = GST_RTP_SV3V_DEPAY (element);
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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gst_adapter_clear (rtpsv3vdepay->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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case GST_STATE_CHANGE_READY_TO_NULL:
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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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}
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gboolean
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gst_rtp_sv3v_depay_plugin_init (GstPlugin * plugin)
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{
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GST_DEBUG_CATEGORY_INIT (rtpsv3vdepay_debug, "rtpsv3vdepay", 0,
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"RTP SV3V depayloader");
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return gst_element_register (plugin, "rtpsv3vdepay",
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GST_RANK_SECONDARY, GST_TYPE_RTP_SV3V_DEPAY);
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}
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