/* GStreamer
- * Copyright (C) <2005> Wim Taymans <wim@fluendo.com>
+ * 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
*
* 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.
+ * 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 <string.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>");
-
-/* RtpMP4GDepay signals and args */
-enum
-{
- /* FILL ME */
- LAST_SIGNAL
-};
-
-enum
-{
- ARG_0,
-};
-
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")
+ "systemstream=(boolean)false;"
+ "audio/mpeg," "mpegversion=(int) 4, " "stream-format=(string)raw")
);
static GstStaticPadTemplate gst_rtp_mp4g_depay_sink_template =
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 */
+ /* "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\" } "
/* "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], " */
+ /* "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, 64], " */
- /* "auxiliarydatasizelength = (string) [0, 64]" */ )
+ /* "streamstateindication = (string) [0, 32], " */
+ /* "auxiliarydatasizelength = (string) [0, 32]" */ )
);
-GST_BOILERPLATE (GstRtpMP4GDepay, gst_rtp_mp4g_depay, GstBaseRTPDepayload,
- GST_TYPE_BASE_RTP_DEPAYLOAD);
+/* 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_rtp_mp4g_depay_finalize (GObject * object);
+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 gboolean gst_rtp_mp4g_depay_setcaps (GstBaseRTPDepayload * depayload,
- GstCaps * caps);
-static GstBuffer *gst_rtp_mp4g_depay_process (GstBaseRTPDepayload * depayload,
- GstBuffer * buf);
+static guint32
+gst_bs_parse_read (GstBsParse * bs, guint n)
+{
+ guint32 res = 0;
+ gint shift;
-static void gst_rtp_mp4g_depay_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec);
-static void gst_rtp_mp4g_depay_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec);
+ if (n == 0)
+ return res;
-static GstStateChangeReturn gst_rtp_mp4g_depay_change_state (GstElement *
- element, GstStateChange transition);
+ /* 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;
-static void
-gst_rtp_mp4g_depay_base_init (gpointer klass)
-{
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ /* mask out required bits */
+ if (n < 32)
+ res &= (1 << n) - 1;
- 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));
+ bs->head = shift;
- gst_element_class_set_details (element_class, &gst_rtp_mp4gdepay_details);
+ 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,
+ GstBuffer * buf);
+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;
- GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+ GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
- gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
-
- parent_class = g_type_class_peek_parent (klass);
+ gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
gobject_class->finalize = gst_rtp_mp4g_depay_finalize;
- gobject_class->set_property = gst_rtp_mp4g_depay_set_property;
- gobject_class->get_property = gst_rtp_mp4g_depay_get_property;
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;
+ gstrtpbasedepayload_class->process = 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_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_rtp_mp4g_depay_src_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&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,
- GstRtpMP4GDepayClass * klass)
+gst_rtp_mp4g_depay_init (GstRtpMP4GDepay * rtpmp4gdepay)
{
rtpmp4gdepay->adapter = gst_adapter_new ();
+ rtpmp4gdepay->packets = g_queue_new ();
}
static void
g_object_unref (rtpmp4gdepay->adapter);
rtpmp4gdepay->adapter = NULL;
+ g_queue_free (rtpmp4gdepay->packets);
+ rtpmp4gdepay->packets = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
}
static gboolean
-gst_rtp_mp4g_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
+gst_rtp_mp4g_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
-
GstStructure *structure;
GstRtpMP4GDepay *rtpmp4gdepay;
GstCaps *srccaps = NULL;
const gchar *str;
- gint clock_rate = 90000; /* default */
+ gint clock_rate;
gint someint;
+ gboolean res;
rtpmp4gdepay = GST_RTP_MP4G_DEPAY (depayload);
structure = gst_caps_get_structure (caps, 0);
- gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ clock_rate = 90000; /* default */
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);
+ "mpegversion", G_TYPE_INT, 4, "stream-format", G_TYPE_STRING, "raw",
+ NULL);
} else if (strcmp (str, "video") == 0) {
srccaps = gst_caps_new_simple ("video/mpeg",
"mpegversion", G_TYPE_INT, 4,
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"))) {
GstBuffer *buffer;
buffer = gst_value_get_buffer (&v);
- gst_buffer_ref (buffer);
- g_value_unset (&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);
+ res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
- return TRUE;
+ return res;
/* ERRORS */
unknown_media:
}
}
+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_flush_queue (GstRtpMP4GDepay * rtpmp4gdepay)
+{
+ GstBuffer *outbuf;
+ gboolean discont = FALSE;
+ 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);
+
+ if (rtpmp4gdepay->next_AU_index != AU_index) {
+ GST_DEBUG_OBJECT (rtpmp4gdepay, "discont, expected AU_index %u",
+ rtpmp4gdepay->next_AU_index);
+ discont = TRUE;
+ }
+
+ if (discont) {
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ discont = FALSE;
+ }
+
+ GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing AU_index %u", AU_index);
+ gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpmp4gdepay), outbuf);
+ rtpmp4gdepay->next_AU_index = AU_index + 1;
+ }
+}
+
+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_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpmp4gdepay), outbuf);
+ rtpmp4gdepay->next_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) {
+ GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing expected AU_index %u",
+ AU_index);
+ outbuf = g_queue_pop_head (rtpmp4gdepay->packets);
+ gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpmp4gdepay),
+ outbuf);
+ rtpmp4gdepay->next_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 (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+gst_rtp_mp4g_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstRtpMP4GDepay *rtpmp4gdepay;
- GstBuffer *outbuf;
+ GstBuffer *outbuf = NULL;
+ GstClockTime timestamp;
+ GstRTPBuffer rtp = { NULL };
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_DEBUG_OBJECT (rtpmp4gdepay, "received DISCONT");
gst_adapter_clear (rtpmp4gdepay->adapter);
}
+ timestamp = GST_BUFFER_TIMESTAMP (buf);
+
{
- gint payload_len, payload_header;
+ gint payload_len, payload_AU;
guint8 *payload;
- guint32 timestamp;
+ guint32 rtptime;
guint AU_headers_len;
- guint AU_size, AU_index;
+ guint AU_size, AU_index, AU_index_delta, payload_AU_size;
+ gboolean M;
+
+ gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
+ payload_len = gst_rtp_buffer_get_payload_len (&rtp);
+ payload = gst_rtp_buffer_get_payload (&rtp);
- payload_len = gst_rtp_buffer_get_payload_len (buf);
- payload = gst_rtp_buffer_get_payload (buf);
- payload_header = 0;
+ 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 lenght is 2 bytes and contains the length of the following
+ * 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_header += 2;
payload_len -= 2;
- /* FIXME, use bits */
- AU_size = ((payload[0] << 8) | payload[1]) >> 3;
- AU_index = payload[1] & 0x7;
-
- GST_DEBUG_OBJECT (rtpmp4gdepay, "len, %d, size %d, index %d",
- AU_headers_len, AU_size, AU_index);
-
- /* skip special headers */
- payload += (AU_headers_len + 7) / 8;
- payload_header += (AU_headers_len + 7) / 8;
- payload_len = AU_size;
- }
-
- timestamp = gst_rtp_buffer_get_timestamp (buf);
-
- /* strip header from payload and push in the adapter */
- outbuf =
- gst_rtp_buffer_get_payload_subbuffer (buf, payload_header, payload_len);
- gst_adapter_push (rtpmp4gdepay->adapter, outbuf);
+ 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);
+ } 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) {
+ guint avail;
+
+ /* packet is complete, flush */
+ avail = gst_adapter_available (rtpmp4gdepay->adapter);
+
+ 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_TIMESTAMP (outbuf) = timestamp;
+ GST_BUFFER_OFFSET (outbuf) = AU_index;
+
+ /* make sure we don't use the timestamp again for other AUs in this
+ * RTP packet. */
+ timestamp = -1;
+
+ 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 (gst_rtp_buffer_get_marker (buf)) {
- guint avail;
+ /* if this was the last packet of the VOP, create and push a buffer */
+ if (M) {
+ guint avail;
- avail = gst_adapter_available (rtpmp4gdepay->adapter);
+ 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_BUFFER_TIMESTAMP (outbuf) = gst_util_uint64_scale_int
- (timestamp, GST_SECOND, depayload->clock_rate);
+ outbuf = gst_adapter_take_buffer (rtpmp4gdepay->adapter, avail);
- GST_DEBUG ("gst_rtp_mp4g_depay_chain: pushing buffer of size %d",
- GST_BUFFER_SIZE (outbuf));
+ GST_DEBUG ("gst_rtp_mp4g_depay_chain: pushing buffer of size %"
+ G_GSIZE_FORMAT, gst_buffer_get_size (outbuf));
- return outbuf;
- } else {
- return NULL;
+ gst_rtp_buffer_unmap (&rtp);
+ return outbuf;
+ }
}
}
+
+ gst_rtp_buffer_unmap (&rtp);
return NULL;
/* ERRORS */
-bad_packet:
+short_payload:
{
GST_ELEMENT_WARNING (rtpmp4gdepay, STREAM, DECODE,
- ("Packet did not validate."), (NULL));
+ ("Packet payload was too short."), (NULL));
+ gst_rtp_buffer_unmap (&rtp);
return NULL;
}
}
-static void
-gst_rtp_mp4g_depay_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec)
+static gboolean
+gst_rtp_mp4g_depay_handle_event (GstRTPBaseDepayload * filter, GstEvent * event)
{
+ gboolean ret;
GstRtpMP4GDepay *rtpmp4gdepay;
- rtpmp4gdepay = GST_RTP_MP4G_DEPAY (object);
+ rtpmp4gdepay = GST_RTP_MP4G_DEPAY (filter);
- switch (prop_id) {
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_FLUSH_STOP:
+ gst_rtp_mp4g_depay_reset (rtpmp4gdepay);
+ break;
default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
-}
-
-static void
-gst_rtp_mp4g_depay_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec)
-{
- GstRtpMP4GDepay *rtpmp4gdepay;
- rtpmp4gdepay = GST_RTP_MP4G_DEPAY (object);
+ ret =
+ GST_RTP_BASE_DEPAYLOAD_CLASS (parent_class)->handle_event (filter, event);
- switch (prop_id) {
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
+ return ret;
}
static GstStateChangeReturn
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
- gst_adapter_clear (rtpmp4gdepay->adapter);
+ 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;
}
gst_rtp_mp4g_depay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpmp4gdepay",
- GST_RANK_MARGINAL, GST_TYPE_RTP_MP4G_DEPAY);
+ GST_RANK_SECONDARY, GST_TYPE_RTP_MP4G_DEPAY);
}
projects / platform / upstream / gstreamer.git / blobdiff