gstreamer/gst/rtp/gstrtpmp4gdepay.c
Tim-Philipp Müller 506e080a15 rtpmp4gdepay: detect broken senders who send AAC with ADTS frames
Strip ADTS headers if we detect any, apparently some Sony cameras
send AAC with ADTS headers. We could also change the stream-format
in the output caps, but that would be unexpected to pipeline builders
and would not exactly be backwards compatible.
2018-09-26 12:25:24 +01:00

810 lines
27 KiB
C

/* 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 "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\" } "
/* 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);
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;
} 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 higest 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) {
GST_WARNING_OBJECT (rtpmp4gdepay, "Detected ADTS header of "
"%u bytes, skipping", adts_hdr_len);
gst_adapter_flush (rtpmp4gdepay->adapter, adts_hdr_len);
avail -= adts_hdr_len;
}
} else {
rtpmp4gdepay->check_adts = FALSE;
}
}
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);
}