gstreamer/gst/rtp/gstrtpmp4gdepay.c
Wim Taymans 4a7cbe8489 fixes: #514889
Original commit message from CVS:
patch by:  Wim Taymans  <wim.taymans@collabora.co.uk>
fixes: #514889
* gst/rtp/gstrtph264pay.c:
* gst/rtp/gstrtpmp4gdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4gpay.h:
* gst/rtp/gstrtptheorapay.c:
* gst/rtp/gstrtpvorbispay.c:
Fix various leaks shown up in valgrind
- free sprops and buffer in error cases in H264 payloader
- fix leak in mp4g depayloader when construction the caps
- don't leak config string in the mp4g payloader
- don't leak buffers and headers in theora and vorbis payloaders
* tests/check/elements/rtp-payloading.c:
Fix the RTP data test
- Actually send valid amr data to the payloader instead of 20
zero-bytes
- The mp4g payloader expects codec_data on the caps
2008-02-12 23:38:19 +00:00

399 lines
12 KiB
C

/* GStreamer
* Copyright (C) <2005> Wim Taymans <wim@fluendo.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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <stdlib.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpmp4gdepay.h"
GST_DEBUG_CATEGORY_STATIC (rtpmp4gdepay_debug);
#define GST_CAT_DEFAULT (rtpmp4gdepay_debug)
/* elementfactory information */
static const GstElementDetails gst_rtp_mp4gdepay_details =
GST_ELEMENT_DETAILS ("RTP packet depayloader",
"Codec/Depayloader/Network",
"Extracts MPEG4 elementary streams from RTP packets (RFC 3640)",
"Wim Taymans <wim@fluendo.com>");
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")
);
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\" }, "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) [1, MAX ], "
"encoding-name = (string) \"MPEG4-GENERIC\", "
/* required string params */
"streamtype = (string) { \"4\", \"5\" }, " /* 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, 16], " *//* max 16 bits, should be enough... */
/* "indexlength = (string) [1, 8], " */
/* "indexdeltalength = (string) [1, 8], " */
/* "ctsdeltalength = (string) [1, 64], " */
/* "dtsdeltalength = (string) [1, 64], " */
/* "randomaccessindication = (string) {0, 1}, " */
/* "streamstateindication = (string) [0, 64], " */
/* "auxiliarydatasizelength = (string) [0, 64]" */ )
);
GST_BOILERPLATE (GstRtpMP4GDepay, gst_rtp_mp4g_depay, GstBaseRTPDepayload,
GST_TYPE_BASE_RTP_DEPAYLOAD);
static void gst_rtp_mp4g_depay_finalize (GObject * object);
static gboolean gst_rtp_mp4g_depay_setcaps (GstBaseRTPDepayload * depayload,
GstCaps * caps);
static GstBuffer *gst_rtp_mp4g_depay_process (GstBaseRTPDepayload * depayload,
GstBuffer * buf);
static GstStateChangeReturn gst_rtp_mp4g_depay_change_state (GstElement *
element, GstStateChange transition);
static void
gst_rtp_mp4g_depay_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_mp4g_depay_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_mp4g_depay_sink_template));
gst_element_class_set_details (element_class, &gst_rtp_mp4gdepay_details);
}
static void
gst_rtp_mp4g_depay_class_init (GstRtpMP4GDepayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_rtp_mp4g_depay_finalize;
gstelement_class->change_state = gst_rtp_mp4g_depay_change_state;
gstbasertpdepayload_class->process = gst_rtp_mp4g_depay_process;
gstbasertpdepayload_class->set_caps = gst_rtp_mp4g_depay_setcaps;
GST_DEBUG_CATEGORY_INIT (rtpmp4gdepay_debug, "rtpmp4gdepay", 0,
"MP4-generic RTP Depayloader");
}
static void
gst_rtp_mp4g_depay_init (GstRtpMP4GDepay * rtpmp4gdepay,
GstRtpMP4GDepayClass * klass)
{
rtpmp4gdepay->adapter = gst_adapter_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_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 (GstBaseRTPDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpMP4GDepay *rtpmp4gdepay;
GstCaps *srccaps = NULL;
const gchar *str;
gint clock_rate = 90000; /* default */
gint someint;
rtpmp4gdepay = GST_RTP_MP4G_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
gst_structure_get_int (structure, "clock-rate", &clock_rate);
depayload->clock_rate = clock_rate;
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, NULL);
} 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);
/* 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");
}
}
gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
return TRUE;
/* ERRORS */
unknown_media:
{
GST_DEBUG_OBJECT (rtpmp4gdepay, "Unknown media type");
return FALSE;
}
}
static GstBuffer *
gst_rtp_mp4g_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{
GstRtpMP4GDepay *rtpmp4gdepay;
GstBuffer *outbuf;
rtpmp4gdepay = GST_RTP_MP4G_DEPAY (depayload);
if (!gst_rtp_buffer_validate (buf))
goto bad_packet;
/* flush remaining data on discont */
if (GST_BUFFER_IS_DISCONT (buf)) {
gst_adapter_clear (rtpmp4gdepay->adapter);
}
{
gint payload_len, payload_header;
guint8 *payload;
guint32 timestamp;
guint AU_headers_len;
guint AU_size, AU_index;
gboolean M;
payload_len = gst_rtp_buffer_get_payload_len (buf);
payload = gst_rtp_buffer_get_payload (buf);
payload_header = 0;
timestamp = gst_rtp_buffer_get_timestamp (buf);
M = gst_rtp_buffer_get_marker (buf);
if (rtpmp4gdepay->sizelength > 0) {
gint num_AU_headers, AU_headers_bytes, i;
/* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+
* |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;
/* skip special headers */
payload_header = 2 + AU_headers_bytes;
for (i = 0; i < num_AU_headers; i++) {
/* FIXME, use bits */
AU_size = ((payload[0] << 8) | payload[1]) >> 3;
AU_index = payload[1] & 0x7;
payload += 2;
GST_DEBUG_OBJECT (rtpmp4gdepay, "len, %d, size %d, index %d",
AU_headers_len, AU_size, AU_index);
/* collect stuff in the adapter, strip header from payload and push in
* the adapter */
outbuf =
gst_rtp_buffer_get_payload_subbuffer (buf, payload_header, AU_size);
gst_adapter_push (rtpmp4gdepay->adapter, outbuf);
if (M) {
guint avail;
/* packet is complete, flush */
avail = gst_adapter_available (rtpmp4gdepay->adapter);
outbuf = gst_adapter_take_buffer (rtpmp4gdepay->adapter, avail);
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
GST_DEBUG ("gst_rtp_mp4g_depay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
/* only apply the timestamp for the first buffer */
if (i == 0)
gst_base_rtp_depayload_push_ts (depayload, timestamp, outbuf);
else
gst_base_rtp_depayload_push (depayload, outbuf);
}
payload_header += AU_size;
}
} else {
/* push complete buffer in adapter */
outbuf = gst_rtp_buffer_get_payload_subbuffer (buf, 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_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
GST_DEBUG ("gst_rtp_mp4g_depay_chain: pushing buffer of size %d",
GST_BUFFER_SIZE (outbuf));
return outbuf;
}
}
}
return NULL;
/* ERRORS */
bad_packet:
{
GST_ELEMENT_WARNING (rtpmp4gdepay, STREAM, DECODE,
("Packet did not validate."), (NULL));
return NULL;
}
}
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_adapter_clear (rtpmp4gdepay->adapter);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
default:
break;
}
return ret;
}
gboolean
gst_rtp_mp4g_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpmp4gdepay",
GST_RANK_MARGINAL, GST_TYPE_RTP_MP4G_DEPAY);
}