mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-27 04:01:08 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/subprojects/gst-plugins-good/gst/rtp/gstrtpmp4gdepay.c

811 lines
27 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) <2005> Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpelements.h"
#include "gstrtpmp4gdepay.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpmp4gdepay_debug);
#define GST_CAT_DEFAULT (rtpmp4gdepay_debug)
static GstStaticPadTemplate gst_rtp_mp4g_depay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/mpeg,"
"mpegversion=(int) 4,"
"systemstream=(boolean)false;"
"audio/mpeg," "mpegversion=(int) 4, " "stream-format=(string)raw")
);
static GstStaticPadTemplate gst_rtp_mp4g_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) { \"video\", \"audio\", \"application\" }, "
"clock-rate = (int) [1, MAX ], "
"encoding-name = (string) \"MPEG4-GENERIC\", "
/* required string params */
/* "streamtype = (string) { \"4\", \"5\" }, " Not set by Wowza 4 = video, 5 = audio */
/* "profile-level-id = (string) [1,MAX], " */
/* "config = (string) [1,MAX]" */
"mode = (string) { \"generic\", \"CELP-cbr\", \"CELP-vbr\", \"AAC-lbr\", \"AAC-hbr\", \"aac-hbr\" } "
/* Optional general parameters */
/* "objecttype = (string) [1,MAX], " */
/* "constantsize = (string) [1,MAX], " *//* constant size of each AU */
/* "constantduration = (string) [1,MAX], " *//* constant duration of each AU */
/* "maxdisplacement = (string) [1,MAX], " */
/* "de-interleavebuffersize = (string) [1,MAX], " */
/* Optional configuration parameters */
/* "sizelength = (string) [1, 32], " */
/* "indexlength = (string) [1, 32], " */
/* "indexdeltalength = (string) [1, 32], " */
/* "ctsdeltalength = (string) [1, 32], " */
/* "dtsdeltalength = (string) [1, 32], " */
/* "randomaccessindication = (string) {0, 1}, " */
/* "streamstateindication = (string) [0, 32], " */
/* "auxiliarydatasizelength = (string) [0, 32]" */ )
);
/* simple bitstream parser */
typedef struct
{
const guint8 *data;
const guint8 *end;
gint head; /* bitpos in the cache of next bit */
guint64 cache; /* cached bytes */
} GstBsParse;
static void
gst_bs_parse_init (GstBsParse * bs, const guint8 * data, guint size)
{
bs->data = data;
bs->end = data + size;
bs->head = 0;
bs->cache = 0xffffffff;
}
static guint32
gst_bs_parse_read (GstBsParse * bs, guint n)
{
guint32 res = 0;
gint shift;
if (n == 0)
return res;
/* fill up the cache if we need to */
while (bs->head < n) {
if (bs->data >= bs->end) {
/* we're at the end, can't produce more than head number of bits */
n = bs->head;
break;
}
/* shift bytes in cache, moving the head bits of the cache left */
bs->cache = (bs->cache << 8) | *bs->data++;
bs->head += 8;
}
/* bring the required bits down and truncate */
if ((shift = bs->head - n) > 0)
res = bs->cache >> shift;
else
res = bs->cache;
/* mask out required bits */
if (n < 32)
res &= (1 << n) - 1;
bs->head = shift;
return res;
}
#define gst_rtp_mp4g_depay_parent_class parent_class
G_DEFINE_TYPE (GstRtpMP4GDepay, gst_rtp_mp4g_depay,
GST_TYPE_RTP_BASE_DEPAYLOAD);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpmp4gdepay, "rtpmp4gdepay",
GST_RANK_SECONDARY, GST_TYPE_RTP_MP4G_DEPAY, rtp_element_init (plugin));
static void gst_rtp_mp4g_depay_finalize (GObject * object);
static gboolean gst_rtp_mp4g_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
static GstBuffer *gst_rtp_mp4g_depay_process (GstRTPBaseDepayload * depayload,
GstRTPBuffer * rtp);
static gboolean gst_rtp_mp4g_depay_handle_event (GstRTPBaseDepayload * filter,
GstEvent * event);
static GstStateChangeReturn gst_rtp_mp4g_depay_change_state (GstElement *
element, GstStateChange transition);
static void
gst_rtp_mp4g_depay_class_init (GstRtpMP4GDepayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
gobject_class->finalize = gst_rtp_mp4g_depay_finalize;
gstelement_class->change_state = gst_rtp_mp4g_depay_change_state;
gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_mp4g_depay_process;
gstrtpbasedepayload_class->set_caps = gst_rtp_mp4g_depay_setcaps;
gstrtpbasedepayload_class->handle_event = gst_rtp_mp4g_depay_handle_event;
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_mp4g_depay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_mp4g_depay_sink_template);
gst_element_class_set_static_metadata (gstelement_class,
"RTP MPEG4 ES depayloader", "Codec/Depayloader/Network/RTP",
"Extracts MPEG4 elementary streams from RTP packets (RFC 3640)",
"Wim Taymans <wim.taymans@gmail.com>");
GST_DEBUG_CATEGORY_INIT (rtpmp4gdepay_debug, "rtpmp4gdepay", 0,
"MP4-generic RTP Depayloader");
}
static void
gst_rtp_mp4g_depay_init (GstRtpMP4GDepay * rtpmp4gdepay)
{
rtpmp4gdepay->adapter = gst_adapter_new ();
rtpmp4gdepay->packets = g_queue_new ();
}
static void
gst_rtp_mp4g_depay_finalize (GObject * object)
{
GstRtpMP4GDepay *rtpmp4gdepay;
rtpmp4gdepay = GST_RTP_MP4G_DEPAY (object);
g_object_unref (rtpmp4gdepay->adapter);
rtpmp4gdepay->adapter = NULL;
g_queue_free (rtpmp4gdepay->packets);
rtpmp4gdepay->packets = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gint
gst_rtp_mp4g_depay_parse_int (GstStructure * structure, const gchar * field,
gint def)
{
const gchar *str;
gint res;
if ((str = gst_structure_get_string (structure, field)))
return atoi (str);
if (gst_structure_get_int (structure, field, &res))
return res;
return def;
}
static gboolean
gst_rtp_mp4g_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpMP4GDepay *rtpmp4gdepay;
GstCaps *srccaps = NULL;
const gchar *str;
gint clock_rate;
gint someint;
gboolean res;
rtpmp4gdepay = GST_RTP_MP4G_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
clock_rate = 90000; /* default */
depayload->clock_rate = clock_rate;
rtpmp4gdepay->check_adts = FALSE;
if ((str = gst_structure_get_string (structure, "media"))) {
if (strcmp (str, "audio") == 0) {
srccaps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, 4, "stream-format", G_TYPE_STRING, "raw",
NULL);
rtpmp4gdepay->check_adts = TRUE;
rtpmp4gdepay->warn_adts = TRUE;
} else if (strcmp (str, "video") == 0) {
srccaps = gst_caps_new_simple ("video/mpeg",
"mpegversion", G_TYPE_INT, 4,
"systemstream", G_TYPE_BOOLEAN, FALSE, NULL);
}
}
if (srccaps == NULL)
goto unknown_media;
/* these values are optional and have a default value of 0 (no header) */
rtpmp4gdepay->sizelength =
gst_rtp_mp4g_depay_parse_int (structure, "sizelength", 0);
rtpmp4gdepay->indexlength =
gst_rtp_mp4g_depay_parse_int (structure, "indexlength", 0);
rtpmp4gdepay->indexdeltalength =
gst_rtp_mp4g_depay_parse_int (structure, "indexdeltalength", 0);
rtpmp4gdepay->ctsdeltalength =
gst_rtp_mp4g_depay_parse_int (structure, "ctsdeltalength", 0);
rtpmp4gdepay->dtsdeltalength =
gst_rtp_mp4g_depay_parse_int (structure, "dtsdeltalength", 0);
someint =
gst_rtp_mp4g_depay_parse_int (structure, "randomaccessindication", 0);
rtpmp4gdepay->randomaccessindication = someint > 0 ? 1 : 0;
rtpmp4gdepay->streamstateindication =
gst_rtp_mp4g_depay_parse_int (structure, "streamstateindication", 0);
rtpmp4gdepay->auxiliarydatasizelength =
gst_rtp_mp4g_depay_parse_int (structure, "auxiliarydatasizelength", 0);
rtpmp4gdepay->constantSize =
gst_rtp_mp4g_depay_parse_int (structure, "constantsize", 0);
rtpmp4gdepay->constantDuration =
gst_rtp_mp4g_depay_parse_int (structure, "constantduration", 0);
rtpmp4gdepay->maxDisplacement =
gst_rtp_mp4g_depay_parse_int (structure, "maxdisplacement", 0);
/* get config string */
if ((str = gst_structure_get_string (structure, "config"))) {
GValue v = { 0 };
g_value_init (&v, GST_TYPE_BUFFER);
if (gst_value_deserialize (&v, str)) {
GstBuffer *buffer;
buffer = gst_value_get_buffer (&v);
gst_caps_set_simple (srccaps,
"codec_data", GST_TYPE_BUFFER, buffer, NULL);
g_value_unset (&v);
} else {
g_warning ("cannot convert config to buffer");
}
}
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
return res;
/* ERRORS */
unknown_media:
{
GST_DEBUG_OBJECT (rtpmp4gdepay, "Unknown media type");
return FALSE;
}
}
static void
gst_rtp_mp4g_depay_clear_queue (GstRtpMP4GDepay * rtpmp4gdepay)
{
GstBuffer *outbuf;
while ((outbuf = g_queue_pop_head (rtpmp4gdepay->packets)))
gst_buffer_unref (outbuf);
}
static void
gst_rtp_mp4g_depay_reset (GstRtpMP4GDepay * rtpmp4gdepay)
{
gst_adapter_clear (rtpmp4gdepay->adapter);
rtpmp4gdepay->max_AU_index = -1;
rtpmp4gdepay->next_AU_index = -1;
rtpmp4gdepay->prev_AU_index = -1;
rtpmp4gdepay->prev_rtptime = -1;
rtpmp4gdepay->last_AU_index = -1;
gst_rtp_mp4g_depay_clear_queue (rtpmp4gdepay);
}
static void
gst_rtp_mp4g_depay_push_outbuf (GstRtpMP4GDepay * rtpmp4gdepay,
GstBuffer * outbuf, guint AU_index)
{
gboolean discont = FALSE;
if (AU_index != rtpmp4gdepay->next_AU_index) {
GST_DEBUG_OBJECT (rtpmp4gdepay, "discont, expected AU_index %u",
rtpmp4gdepay->next_AU_index);
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
discont = TRUE;
}
GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing %sAU_index %u",
discont ? "" : "expected ", AU_index);
gst_rtp_drop_meta (GST_ELEMENT_CAST (rtpmp4gdepay), outbuf, 0);
gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpmp4gdepay), outbuf);
rtpmp4gdepay->next_AU_index = AU_index + 1;
}
static void
gst_rtp_mp4g_depay_flush_queue (GstRtpMP4GDepay * rtpmp4gdepay)
{
GstBuffer *outbuf;
guint AU_index;
while ((outbuf = g_queue_pop_head (rtpmp4gdepay->packets))) {
AU_index = GST_BUFFER_OFFSET (outbuf);
GST_DEBUG_OBJECT (rtpmp4gdepay, "next available AU_index %u", AU_index);
gst_rtp_mp4g_depay_push_outbuf (rtpmp4gdepay, outbuf, AU_index);
}
}
static void
gst_rtp_mp4g_depay_queue (GstRtpMP4GDepay * rtpmp4gdepay, GstBuffer * outbuf)
{
guint AU_index = GST_BUFFER_OFFSET (outbuf);
if (rtpmp4gdepay->next_AU_index == -1) {
GST_DEBUG_OBJECT (rtpmp4gdepay, "Init AU counter %u", AU_index);
rtpmp4gdepay->next_AU_index = AU_index;
}
if (rtpmp4gdepay->next_AU_index == AU_index) {
GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing expected AU_index %u", AU_index);
/* we received the expected packet, push it and flush as much as we can from
* the queue */
gst_rtp_mp4g_depay_push_outbuf (rtpmp4gdepay, outbuf, AU_index);
while ((outbuf = g_queue_peek_head (rtpmp4gdepay->packets))) {
AU_index = GST_BUFFER_OFFSET (outbuf);
GST_DEBUG_OBJECT (rtpmp4gdepay, "next available AU_index %u", AU_index);
if (rtpmp4gdepay->next_AU_index == AU_index) {
outbuf = g_queue_pop_head (rtpmp4gdepay->packets);
gst_rtp_mp4g_depay_push_outbuf (rtpmp4gdepay, outbuf, AU_index);
} else {
GST_DEBUG_OBJECT (rtpmp4gdepay, "waiting for next AU_index %u",
rtpmp4gdepay->next_AU_index);
break;
}
}
} else {
GList *list;
GST_DEBUG_OBJECT (rtpmp4gdepay, "queueing AU_index %u", AU_index);
/* loop the list to skip strictly smaller AU_index buffers */
for (list = rtpmp4gdepay->packets->head; list; list = g_list_next (list)) {
guint idx;
gint gap;
idx = GST_BUFFER_OFFSET (GST_BUFFER_CAST (list->data));
/* compare the new seqnum to the one in the buffer */
gap = (gint) (idx - AU_index);
GST_DEBUG_OBJECT (rtpmp4gdepay, "compare with AU_index %u, gap %d", idx,
gap);
/* AU_index <= idx, we can stop looking */
if (G_LIKELY (gap > 0))
break;
}
if (G_LIKELY (list))
g_queue_insert_before (rtpmp4gdepay->packets, list, outbuf);
else
g_queue_push_tail (rtpmp4gdepay->packets, outbuf);
}
}
static GstBuffer *
gst_rtp_mp4g_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
GstRtpMP4GDepay *rtpmp4gdepay;
GstBuffer *outbuf = NULL;
GstClockTime timestamp;
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;
}
Reference in a new issue View git blame Copy permalink