mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-09 05:38:58 +00:00
Various live555 based products are using the wrong "mode" string or seem to assume case-insensitive matching, which is wrong. Examples for this are the Yuan SC6C0N1 mini and the Kiloview E2. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/727>
815 lines
27 KiB
C
815 lines
27 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 <stdlib.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpmp4gdepay.h"
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#include "gstrtputils.h"
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GST_DEBUG_CATEGORY_STATIC (rtpmp4gdepay_debug);
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#define GST_CAT_DEFAULT (rtpmp4gdepay_debug)
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static GstStaticPadTemplate gst_rtp_mp4g_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,"
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"systemstream=(boolean)false;"
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"audio/mpeg," "mpegversion=(int) 4, " "stream-format=(string)raw")
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);
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static GstStaticPadTemplate gst_rtp_mp4g_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\", \"audio\", \"application\" }, "
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"clock-rate = (int) [1, MAX ], "
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"encoding-name = (string) \"MPEG4-GENERIC\", "
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/* required string params */
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/* "streamtype = (string) { \"4\", \"5\" }, " Not set by Wowza 4 = video, 5 = audio */
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/* "profile-level-id = (string) [1,MAX], " */
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/* "config = (string) [1,MAX]" */
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"mode = (string) { \"generic\", \"CELP-cbr\", \"CELP-vbr\", \"AAC-lbr\", \"AAC-hbr\", \"aac-hbr\" } "
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/* Optional general parameters */
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/* "objecttype = (string) [1,MAX], " */
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/* "constantsize = (string) [1,MAX], " *//* constant size of each AU */
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/* "constantduration = (string) [1,MAX], " *//* constant duration of each AU */
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/* "maxdisplacement = (string) [1,MAX], " */
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/* "de-interleavebuffersize = (string) [1,MAX], " */
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/* Optional configuration parameters */
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/* "sizelength = (string) [1, 32], " */
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/* "indexlength = (string) [1, 32], " */
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/* "indexdeltalength = (string) [1, 32], " */
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/* "ctsdeltalength = (string) [1, 32], " */
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/* "dtsdeltalength = (string) [1, 32], " */
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/* "randomaccessindication = (string) {0, 1}, " */
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/* "streamstateindication = (string) [0, 32], " */
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/* "auxiliarydatasizelength = (string) [0, 32]" */ )
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);
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/* simple bitstream parser */
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typedef struct
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{
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const guint8 *data;
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const guint8 *end;
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gint head; /* bitpos in the cache of next bit */
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guint64 cache; /* cached bytes */
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} GstBsParse;
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static void
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gst_bs_parse_init (GstBsParse * bs, const guint8 * data, guint size)
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{
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bs->data = data;
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bs->end = data + size;
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bs->head = 0;
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bs->cache = 0xffffffff;
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}
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static guint32
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gst_bs_parse_read (GstBsParse * bs, guint n)
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{
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guint32 res = 0;
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gint shift;
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if (n == 0)
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return res;
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/* fill up the cache if we need to */
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while (bs->head < n) {
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if (bs->data >= bs->end) {
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/* we're at the end, can't produce more than head number of bits */
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n = bs->head;
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break;
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}
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/* shift bytes in cache, moving the head bits of the cache left */
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bs->cache = (bs->cache << 8) | *bs->data++;
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bs->head += 8;
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}
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/* bring the required bits down and truncate */
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if ((shift = bs->head - n) > 0)
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res = bs->cache >> shift;
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else
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res = bs->cache;
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/* mask out required bits */
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if (n < 32)
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res &= (1 << n) - 1;
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bs->head = shift;
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return res;
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}
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#define gst_rtp_mp4g_depay_parent_class parent_class
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G_DEFINE_TYPE (GstRtpMP4GDepay, gst_rtp_mp4g_depay,
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GST_TYPE_RTP_BASE_DEPAYLOAD);
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static void gst_rtp_mp4g_depay_finalize (GObject * object);
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static gboolean gst_rtp_mp4g_depay_setcaps (GstRTPBaseDepayload * depayload,
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GstCaps * caps);
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static GstBuffer *gst_rtp_mp4g_depay_process (GstRTPBaseDepayload * depayload,
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GstRTPBuffer * rtp);
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static gboolean gst_rtp_mp4g_depay_handle_event (GstRTPBaseDepayload * filter,
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GstEvent * event);
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static GstStateChangeReturn gst_rtp_mp4g_depay_change_state (GstElement *
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element, GstStateChange transition);
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static void
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gst_rtp_mp4g_depay_class_init (GstRtpMP4GDepayClass * 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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gobject_class->finalize = gst_rtp_mp4g_depay_finalize;
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gstelement_class->change_state = gst_rtp_mp4g_depay_change_state;
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gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_mp4g_depay_process;
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gstrtpbasedepayload_class->set_caps = gst_rtp_mp4g_depay_setcaps;
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gstrtpbasedepayload_class->handle_event = gst_rtp_mp4g_depay_handle_event;
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_mp4g_depay_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_rtp_mp4g_depay_sink_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP MPEG4 ES depayloader", "Codec/Depayloader/Network/RTP",
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"Extracts MPEG4 elementary streams from RTP packets (RFC 3640)",
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"Wim Taymans <wim.taymans@gmail.com>");
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GST_DEBUG_CATEGORY_INIT (rtpmp4gdepay_debug, "rtpmp4gdepay", 0,
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"MP4-generic RTP Depayloader");
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}
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static void
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gst_rtp_mp4g_depay_init (GstRtpMP4GDepay * rtpmp4gdepay)
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{
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rtpmp4gdepay->adapter = gst_adapter_new ();
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rtpmp4gdepay->packets = g_queue_new ();
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}
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static void
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gst_rtp_mp4g_depay_finalize (GObject * object)
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{
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GstRtpMP4GDepay *rtpmp4gdepay;
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rtpmp4gdepay = GST_RTP_MP4G_DEPAY (object);
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g_object_unref (rtpmp4gdepay->adapter);
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rtpmp4gdepay->adapter = NULL;
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g_queue_free (rtpmp4gdepay->packets);
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rtpmp4gdepay->packets = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gint
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gst_rtp_mp4g_depay_parse_int (GstStructure * structure, const gchar * field,
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gint def)
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{
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const gchar *str;
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gint res;
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if ((str = gst_structure_get_string (structure, field)))
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return atoi (str);
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if (gst_structure_get_int (structure, field, &res))
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return res;
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return def;
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}
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static gboolean
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gst_rtp_mp4g_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
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{
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GstStructure *structure;
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GstRtpMP4GDepay *rtpmp4gdepay;
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GstCaps *srccaps = NULL;
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const gchar *str;
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gint clock_rate;
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gint someint;
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gboolean res;
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rtpmp4gdepay = GST_RTP_MP4G_DEPAY (depayload);
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structure = gst_caps_get_structure (caps, 0);
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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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depayload->clock_rate = clock_rate;
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rtpmp4gdepay->check_adts = FALSE;
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if ((str = gst_structure_get_string (structure, "media"))) {
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if (strcmp (str, "audio") == 0) {
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srccaps = gst_caps_new_simple ("audio/mpeg",
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"mpegversion", G_TYPE_INT, 4, "stream-format", G_TYPE_STRING, "raw",
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NULL);
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rtpmp4gdepay->check_adts = TRUE;
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rtpmp4gdepay->warn_adts = TRUE;
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} else if (strcmp (str, "video") == 0) {
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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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}
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}
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if (srccaps == NULL)
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goto unknown_media;
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/* these values are optional and have a default value of 0 (no header) */
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rtpmp4gdepay->sizelength =
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gst_rtp_mp4g_depay_parse_int (structure, "sizelength", 0);
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rtpmp4gdepay->indexlength =
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gst_rtp_mp4g_depay_parse_int (structure, "indexlength", 0);
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rtpmp4gdepay->indexdeltalength =
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gst_rtp_mp4g_depay_parse_int (structure, "indexdeltalength", 0);
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rtpmp4gdepay->ctsdeltalength =
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gst_rtp_mp4g_depay_parse_int (structure, "ctsdeltalength", 0);
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rtpmp4gdepay->dtsdeltalength =
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gst_rtp_mp4g_depay_parse_int (structure, "dtsdeltalength", 0);
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someint =
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gst_rtp_mp4g_depay_parse_int (structure, "randomaccessindication", 0);
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rtpmp4gdepay->randomaccessindication = someint > 0 ? 1 : 0;
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rtpmp4gdepay->streamstateindication =
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gst_rtp_mp4g_depay_parse_int (structure, "streamstateindication", 0);
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rtpmp4gdepay->auxiliarydatasizelength =
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gst_rtp_mp4g_depay_parse_int (structure, "auxiliarydatasizelength", 0);
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rtpmp4gdepay->constantSize =
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gst_rtp_mp4g_depay_parse_int (structure, "constantsize", 0);
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rtpmp4gdepay->constantDuration =
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gst_rtp_mp4g_depay_parse_int (structure, "constantduration", 0);
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rtpmp4gdepay->maxDisplacement =
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gst_rtp_mp4g_depay_parse_int (structure, "maxdisplacement", 0);
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/* get config string */
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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_caps_set_simple (srccaps,
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"codec_data", GST_TYPE_BUFFER, buffer, NULL);
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g_value_unset (&v);
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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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res = gst_pad_set_caps (depayload->srcpad, srccaps);
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gst_caps_unref (srccaps);
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return res;
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/* ERRORS */
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unknown_media:
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{
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GST_DEBUG_OBJECT (rtpmp4gdepay, "Unknown media type");
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return FALSE;
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}
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}
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static void
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gst_rtp_mp4g_depay_clear_queue (GstRtpMP4GDepay * rtpmp4gdepay)
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{
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GstBuffer *outbuf;
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while ((outbuf = g_queue_pop_head (rtpmp4gdepay->packets)))
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gst_buffer_unref (outbuf);
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}
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static void
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gst_rtp_mp4g_depay_reset (GstRtpMP4GDepay * rtpmp4gdepay)
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{
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gst_adapter_clear (rtpmp4gdepay->adapter);
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rtpmp4gdepay->max_AU_index = -1;
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rtpmp4gdepay->next_AU_index = -1;
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rtpmp4gdepay->prev_AU_index = -1;
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rtpmp4gdepay->prev_rtptime = -1;
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rtpmp4gdepay->last_AU_index = -1;
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gst_rtp_mp4g_depay_clear_queue (rtpmp4gdepay);
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}
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static void
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gst_rtp_mp4g_depay_push_outbuf (GstRtpMP4GDepay * rtpmp4gdepay,
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GstBuffer * outbuf, guint AU_index)
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{
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gboolean discont = FALSE;
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if (AU_index != rtpmp4gdepay->next_AU_index) {
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GST_DEBUG_OBJECT (rtpmp4gdepay, "discont, expected AU_index %u",
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rtpmp4gdepay->next_AU_index);
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GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
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discont = TRUE;
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}
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GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing %sAU_index %u",
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discont ? "" : "expected ", AU_index);
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gst_rtp_drop_meta (GST_ELEMENT_CAST (rtpmp4gdepay), outbuf, 0);
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gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpmp4gdepay), outbuf);
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rtpmp4gdepay->next_AU_index = AU_index + 1;
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}
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static void
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gst_rtp_mp4g_depay_flush_queue (GstRtpMP4GDepay * rtpmp4gdepay)
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{
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GstBuffer *outbuf;
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guint AU_index;
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while ((outbuf = g_queue_pop_head (rtpmp4gdepay->packets))) {
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AU_index = GST_BUFFER_OFFSET (outbuf);
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GST_DEBUG_OBJECT (rtpmp4gdepay, "next available AU_index %u", AU_index);
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gst_rtp_mp4g_depay_push_outbuf (rtpmp4gdepay, outbuf, AU_index);
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}
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}
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static void
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gst_rtp_mp4g_depay_queue (GstRtpMP4GDepay * rtpmp4gdepay, GstBuffer * outbuf)
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{
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guint AU_index = GST_BUFFER_OFFSET (outbuf);
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if (rtpmp4gdepay->next_AU_index == -1) {
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GST_DEBUG_OBJECT (rtpmp4gdepay, "Init AU counter %u", AU_index);
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rtpmp4gdepay->next_AU_index = AU_index;
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}
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if (rtpmp4gdepay->next_AU_index == AU_index) {
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GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing expected AU_index %u", AU_index);
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/* we received the expected packet, push it and flush as much as we can from
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* the queue */
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gst_rtp_mp4g_depay_push_outbuf (rtpmp4gdepay, outbuf, AU_index);
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while ((outbuf = g_queue_peek_head (rtpmp4gdepay->packets))) {
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AU_index = GST_BUFFER_OFFSET (outbuf);
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GST_DEBUG_OBJECT (rtpmp4gdepay, "next available AU_index %u", AU_index);
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if (rtpmp4gdepay->next_AU_index == AU_index) {
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outbuf = g_queue_pop_head (rtpmp4gdepay->packets);
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gst_rtp_mp4g_depay_push_outbuf (rtpmp4gdepay, outbuf, AU_index);
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} else {
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GST_DEBUG_OBJECT (rtpmp4gdepay, "waiting for next AU_index %u",
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rtpmp4gdepay->next_AU_index);
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break;
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}
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}
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} else {
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GList *list;
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GST_DEBUG_OBJECT (rtpmp4gdepay, "queueing AU_index %u", AU_index);
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/* loop the list to skip strictly smaller AU_index buffers */
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for (list = rtpmp4gdepay->packets->head; list; list = g_list_next (list)) {
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guint idx;
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gint gap;
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idx = GST_BUFFER_OFFSET (GST_BUFFER_CAST (list->data));
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/* compare the new seqnum to the one in the buffer */
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gap = (gint) (idx - AU_index);
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GST_DEBUG_OBJECT (rtpmp4gdepay, "compare with AU_index %u, gap %d", idx,
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gap);
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/* AU_index <= idx, we can stop looking */
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if (G_LIKELY (gap > 0))
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break;
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}
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if (G_LIKELY (list))
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g_queue_insert_before (rtpmp4gdepay->packets, list, outbuf);
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else
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g_queue_push_tail (rtpmp4gdepay->packets, outbuf);
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}
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}
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|
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static GstBuffer *
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gst_rtp_mp4g_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
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{
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GstRtpMP4GDepay *rtpmp4gdepay;
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GstBuffer *outbuf = NULL;
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GstClockTime timestamp;
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rtpmp4gdepay = GST_RTP_MP4G_DEPAY (depayload);
|
|
|
|
/* flush remaining data on discont */
|
|
if (GST_BUFFER_IS_DISCONT (rtp->buffer)) {
|
|
GST_DEBUG_OBJECT (rtpmp4gdepay, "received DISCONT");
|
|
gst_adapter_clear (rtpmp4gdepay->adapter);
|
|
}
|
|
|
|
timestamp = GST_BUFFER_PTS (rtp->buffer);
|
|
|
|
{
|
|
gint payload_len, payload_AU;
|
|
guint8 *payload;
|
|
guint32 rtptime;
|
|
guint AU_headers_len;
|
|
guint AU_size, AU_index, AU_index_delta, payload_AU_size;
|
|
gboolean M;
|
|
|
|
payload_len = gst_rtp_buffer_get_payload_len (rtp);
|
|
payload = gst_rtp_buffer_get_payload (rtp);
|
|
|
|
GST_DEBUG_OBJECT (rtpmp4gdepay, "received payload of %d", payload_len);
|
|
|
|
rtptime = gst_rtp_buffer_get_timestamp (rtp);
|
|
M = gst_rtp_buffer_get_marker (rtp);
|
|
|
|
if (rtpmp4gdepay->sizelength > 0) {
|
|
gint num_AU_headers, AU_headers_bytes, i;
|
|
GstBsParse bs;
|
|
|
|
if (payload_len < 2)
|
|
goto short_payload;
|
|
|
|
/* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+
|
|
* |AU-headers-length|AU-header|AU-header| |AU-header|padding|
|
|
* | | (1) | (2) | | (n) * | bits |
|
|
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+
|
|
*
|
|
* The length is 2 bytes and contains the length of the following
|
|
* AU-headers in bits.
|
|
*/
|
|
AU_headers_len = (payload[0] << 8) | payload[1];
|
|
AU_headers_bytes = (AU_headers_len + 7) / 8;
|
|
num_AU_headers = AU_headers_len / 16;
|
|
|
|
GST_DEBUG_OBJECT (rtpmp4gdepay, "AU headers len %d, bytes %d, num %d",
|
|
AU_headers_len, AU_headers_bytes, num_AU_headers);
|
|
|
|
/* skip header */
|
|
payload += 2;
|
|
payload_len -= 2;
|
|
|
|
if (payload_len < AU_headers_bytes)
|
|
goto short_payload;
|
|
|
|
/* skip special headers, point to first payload AU */
|
|
payload_AU = 2 + AU_headers_bytes;
|
|
payload_AU_size = payload_len - AU_headers_bytes;
|
|
|
|
if (G_UNLIKELY (rtpmp4gdepay->auxiliarydatasizelength)) {
|
|
gint aux_size;
|
|
|
|
/* point the bitstream parser to the first auxiliary data bit */
|
|
gst_bs_parse_init (&bs, payload + AU_headers_bytes,
|
|
payload_len - AU_headers_bytes);
|
|
aux_size =
|
|
gst_bs_parse_read (&bs, rtpmp4gdepay->auxiliarydatasizelength);
|
|
/* convert to bytes */
|
|
aux_size = (aux_size + 7) / 8;
|
|
/* AU data then follows auxiliary data */
|
|
if (payload_AU_size < aux_size)
|
|
goto short_payload;
|
|
payload_AU += aux_size;
|
|
payload_AU_size -= aux_size;
|
|
}
|
|
|
|
/* point the bitstream parser to the first AU header bit */
|
|
gst_bs_parse_init (&bs, payload, payload_len);
|
|
AU_index = AU_index_delta = 0;
|
|
|
|
for (i = 0; i < num_AU_headers && payload_AU_size > 0; i++) {
|
|
/* parse AU header
|
|
* +---------------------------------------+
|
|
* | AU-size |
|
|
* +---------------------------------------+
|
|
* | AU-Index / AU-Index-delta |
|
|
* +---------------------------------------+
|
|
* | CTS-flag |
|
|
* +---------------------------------------+
|
|
* | CTS-delta |
|
|
* +---------------------------------------+
|
|
* | DTS-flag |
|
|
* +---------------------------------------+
|
|
* | DTS-delta |
|
|
* +---------------------------------------+
|
|
* | RAP-flag |
|
|
* +---------------------------------------+
|
|
* | Stream-state |
|
|
* +---------------------------------------+
|
|
*/
|
|
AU_size = gst_bs_parse_read (&bs, rtpmp4gdepay->sizelength);
|
|
|
|
/* calculate the AU_index, which is only on the first AU of the packet
|
|
* and the AU_index_delta on the other AUs. This will be used to
|
|
* reconstruct the AU ordering when interleaving. */
|
|
if (i == 0) {
|
|
AU_index = gst_bs_parse_read (&bs, rtpmp4gdepay->indexlength);
|
|
|
|
GST_DEBUG_OBJECT (rtpmp4gdepay, "AU index %u", AU_index);
|
|
|
|
if (AU_index == 0 && rtpmp4gdepay->prev_AU_index == 0) {
|
|
gint diff;
|
|
gint cd;
|
|
|
|
/* if we see two consecutive packets with AU_index of 0, we can
|
|
* assume we have constantDuration packets. Since we don't have
|
|
* the index we must use the AU duration to calculate the
|
|
* index. Get the diff between the timestamps first, this can be
|
|
* positive or negative. */
|
|
if (rtpmp4gdepay->prev_rtptime <= rtptime)
|
|
diff = rtptime - rtpmp4gdepay->prev_rtptime;
|
|
else
|
|
diff = -(rtpmp4gdepay->prev_rtptime - rtptime);
|
|
|
|
/* if no constantDuration was given, make one */
|
|
if (rtpmp4gdepay->constantDuration != 0) {
|
|
cd = rtpmp4gdepay->constantDuration;
|
|
GST_DEBUG_OBJECT (depayload, "using constantDuration %d", cd);
|
|
} else if (rtpmp4gdepay->prev_AU_num > 0) {
|
|
/* use number of packets and of previous frame */
|
|
cd = diff / rtpmp4gdepay->prev_AU_num;
|
|
GST_DEBUG_OBJECT (depayload, "guessing constantDuration %d", cd);
|
|
if (!GST_BUFFER_IS_DISCONT (rtp->buffer)) {
|
|
/* rfc3640 - 3.2.3.2
|
|
* if we see two consecutive packets with AU_index of 0 and
|
|
* there has been no discontinuity, we must conclude that this
|
|
* value of constantDuration is correct from now on. */
|
|
GST_DEBUG_OBJECT (depayload,
|
|
"constantDuration of %d detected", cd);
|
|
rtpmp4gdepay->constantDuration = cd;
|
|
}
|
|
} else {
|
|
/* assume this frame has the same number of packets as the
|
|
* previous one */
|
|
cd = diff / num_AU_headers;
|
|
GST_DEBUG_OBJECT (depayload, "guessing constantDuration %d", cd);
|
|
}
|
|
|
|
if (cd > 0) {
|
|
/* get the number of packets by dividing with the duration */
|
|
diff /= cd;
|
|
} else {
|
|
diff = 0;
|
|
}
|
|
|
|
rtpmp4gdepay->last_AU_index += diff;
|
|
rtpmp4gdepay->prev_AU_index = AU_index;
|
|
|
|
AU_index = rtpmp4gdepay->last_AU_index;
|
|
|
|
GST_DEBUG_OBJECT (rtpmp4gdepay, "diff %d, AU index %u", diff,
|
|
AU_index);
|
|
} else {
|
|
rtpmp4gdepay->prev_AU_index = AU_index;
|
|
rtpmp4gdepay->last_AU_index = AU_index;
|
|
}
|
|
|
|
/* keep track of the highest AU_index */
|
|
if (rtpmp4gdepay->max_AU_index != -1
|
|
&& rtpmp4gdepay->max_AU_index <= AU_index) {
|
|
GST_DEBUG_OBJECT (rtpmp4gdepay, "new interleave group, flushing");
|
|
/* a new interleave group started, flush */
|
|
gst_rtp_mp4g_depay_flush_queue (rtpmp4gdepay);
|
|
}
|
|
if (G_UNLIKELY (!rtpmp4gdepay->maxDisplacement &&
|
|
rtpmp4gdepay->max_AU_index != -1
|
|
&& rtpmp4gdepay->max_AU_index >= AU_index)) {
|
|
GstBuffer *outbuf;
|
|
|
|
/* some broken non-interleaved streams have AU-index jumping around
|
|
* all over the place, apparently assuming receiver disregards */
|
|
GST_DEBUG_OBJECT (rtpmp4gdepay, "non-interleaved broken AU indices;"
|
|
" forcing continuous flush");
|
|
/* reset AU to avoid repeated DISCONT in such case */
|
|
outbuf = g_queue_peek_head (rtpmp4gdepay->packets);
|
|
if (G_LIKELY (outbuf)) {
|
|
rtpmp4gdepay->next_AU_index = GST_BUFFER_OFFSET (outbuf);
|
|
gst_rtp_mp4g_depay_flush_queue (rtpmp4gdepay);
|
|
}
|
|
/* rebase next_AU_index to current rtp's first AU_index */
|
|
rtpmp4gdepay->next_AU_index = AU_index;
|
|
}
|
|
rtpmp4gdepay->prev_rtptime = rtptime;
|
|
rtpmp4gdepay->prev_AU_num = num_AU_headers;
|
|
} else {
|
|
AU_index_delta =
|
|
gst_bs_parse_read (&bs, rtpmp4gdepay->indexdeltalength);
|
|
AU_index += AU_index_delta + 1;
|
|
}
|
|
/* keep track of highest AU_index */
|
|
if (rtpmp4gdepay->max_AU_index == -1
|
|
|| AU_index > rtpmp4gdepay->max_AU_index)
|
|
rtpmp4gdepay->max_AU_index = AU_index;
|
|
|
|
/* the presentation time offset, a 2s-complement value, we need this to
|
|
* calculate the timestamp on the output packet. */
|
|
if (rtpmp4gdepay->ctsdeltalength > 0) {
|
|
if (gst_bs_parse_read (&bs, 1))
|
|
gst_bs_parse_read (&bs, rtpmp4gdepay->ctsdeltalength);
|
|
}
|
|
/* the decoding time offset, a 2s-complement value */
|
|
if (rtpmp4gdepay->dtsdeltalength > 0) {
|
|
if (gst_bs_parse_read (&bs, 1))
|
|
gst_bs_parse_read (&bs, rtpmp4gdepay->dtsdeltalength);
|
|
}
|
|
/* RAP-flag to indicate that the AU contains a keyframe */
|
|
if (rtpmp4gdepay->randomaccessindication)
|
|
gst_bs_parse_read (&bs, 1);
|
|
/* stream-state */
|
|
if (rtpmp4gdepay->streamstateindication > 0)
|
|
gst_bs_parse_read (&bs, rtpmp4gdepay->streamstateindication);
|
|
|
|
GST_DEBUG_OBJECT (rtpmp4gdepay, "size %d, index %d, delta %d", AU_size,
|
|
AU_index, AU_index_delta);
|
|
|
|
/* fragmented pakets have the AU_size set to the size of the
|
|
* unfragmented AU. */
|
|
if (AU_size > payload_AU_size)
|
|
AU_size = payload_AU_size;
|
|
|
|
/* collect stuff in the adapter, strip header from payload and push in
|
|
* the adapter */
|
|
outbuf =
|
|
gst_rtp_buffer_get_payload_subbuffer (rtp, payload_AU, AU_size);
|
|
gst_adapter_push (rtpmp4gdepay->adapter, outbuf);
|
|
|
|
if (M) {
|
|
guint32 v = 0;
|
|
guint avail;
|
|
|
|
/* packet is complete, flush */
|
|
avail = gst_adapter_available (rtpmp4gdepay->adapter);
|
|
|
|
/* Some broken senders send ADTS headers (e.g. some Sony cameras).
|
|
* Try to detect those and skip them (still needs config set), but
|
|
* don't check every frame, only the first (unless we detect ADTS) */
|
|
if (rtpmp4gdepay->check_adts && avail >= 7) {
|
|
if (gst_adapter_masked_scan_uint32_peek (rtpmp4gdepay->adapter,
|
|
0xfffe0000, 0xfff00000, 0, 4, &v) == 0) {
|
|
guint adts_hdr_len = (((v >> 16) & 0x1) == 0) ? 9 : 7;
|
|
if (avail > adts_hdr_len) {
|
|
if (rtpmp4gdepay->warn_adts) {
|
|
GST_WARNING_OBJECT (rtpmp4gdepay, "Detected ADTS header of "
|
|
"%u bytes, skipping", adts_hdr_len);
|
|
rtpmp4gdepay->warn_adts = FALSE;
|
|
}
|
|
gst_adapter_flush (rtpmp4gdepay->adapter, adts_hdr_len);
|
|
avail -= adts_hdr_len;
|
|
}
|
|
} else {
|
|
rtpmp4gdepay->check_adts = FALSE;
|
|
rtpmp4gdepay->warn_adts = TRUE;
|
|
}
|
|
}
|
|
|
|
outbuf = gst_adapter_take_buffer (rtpmp4gdepay->adapter, avail);
|
|
|
|
/* copy some of the fields we calculated above on the buffer. We also
|
|
* copy the AU_index so that we can sort the packets in our queue. */
|
|
GST_BUFFER_PTS (outbuf) = timestamp;
|
|
GST_BUFFER_OFFSET (outbuf) = AU_index;
|
|
|
|
if (rtpmp4gdepay->constantDuration != 0) {
|
|
/* if we have constantDuration, calculate timestamp for next AU
|
|
* in this RTP packet. */
|
|
timestamp += (rtpmp4gdepay->constantDuration * GST_SECOND) /
|
|
depayload->clock_rate;
|
|
} else {
|
|
/* otherwise, make sure we don't use the timestamp again for other
|
|
* AUs. */
|
|
timestamp = GST_CLOCK_TIME_NONE;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (depayload,
|
|
"pushing buffer of size %" G_GSIZE_FORMAT,
|
|
gst_buffer_get_size (outbuf));
|
|
|
|
gst_rtp_mp4g_depay_queue (rtpmp4gdepay, outbuf);
|
|
|
|
}
|
|
payload_AU += AU_size;
|
|
payload_AU_size -= AU_size;
|
|
}
|
|
} else {
|
|
/* push complete buffer in adapter */
|
|
outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, 0, payload_len);
|
|
gst_adapter_push (rtpmp4gdepay->adapter, outbuf);
|
|
|
|
/* if this was the last packet of the VOP, create and push a buffer */
|
|
if (M) {
|
|
guint avail;
|
|
|
|
avail = gst_adapter_available (rtpmp4gdepay->adapter);
|
|
|
|
outbuf = gst_adapter_take_buffer (rtpmp4gdepay->adapter, avail);
|
|
|
|
GST_DEBUG ("gst_rtp_mp4g_depay_chain: pushing buffer of size %"
|
|
G_GSIZE_FORMAT, gst_buffer_get_size (outbuf));
|
|
|
|
return outbuf;
|
|
}
|
|
}
|
|
}
|
|
|
|
return NULL;
|
|
|
|
/* ERRORS */
|
|
short_payload:
|
|
{
|
|
GST_ELEMENT_WARNING (rtpmp4gdepay, STREAM, DECODE,
|
|
("Packet payload was too short."), (NULL));
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_mp4g_depay_handle_event (GstRTPBaseDepayload * filter, GstEvent * event)
|
|
{
|
|
gboolean ret;
|
|
GstRtpMP4GDepay *rtpmp4gdepay;
|
|
|
|
rtpmp4gdepay = GST_RTP_MP4G_DEPAY (filter);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_STOP:
|
|
gst_rtp_mp4g_depay_reset (rtpmp4gdepay);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret =
|
|
GST_RTP_BASE_DEPAYLOAD_CLASS (parent_class)->handle_event (filter, event);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_mp4g_depay_change_state (GstElement * element,
|
|
GstStateChange transition)
|
|
{
|
|
GstRtpMP4GDepay *rtpmp4gdepay;
|
|
GstStateChangeReturn ret;
|
|
|
|
rtpmp4gdepay = GST_RTP_MP4G_DEPAY (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
gst_rtp_mp4g_depay_reset (rtpmp4gdepay);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
gst_rtp_mp4g_depay_reset (rtpmp4gdepay);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
gboolean
|
|
gst_rtp_mp4g_depay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtpmp4gdepay",
|
|
GST_RANK_SECONDARY, GST_TYPE_RTP_MP4G_DEPAY);
|
|
}
|