*
* 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 <gst/rtp/gstrtpbuffer.h>
#include "gstrtpmp4gdepay.h"
+#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpmp4gdepay_debug);
#define GST_CAT_DEFAULT (rtpmp4gdepay_debug)
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\" } "
+ "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 */
}
-GST_BOILERPLATE (GstRtpMP4GDepay, gst_rtp_mp4g_depay, GstBaseRTPDepayload,
- GST_TYPE_BASE_RTP_DEPAYLOAD);
+#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 (GstBaseRTPDepayload * depayload,
+static gboolean gst_rtp_mp4g_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
-static GstBuffer *gst_rtp_mp4g_depay_process (GstBaseRTPDepayload * depayload,
- GstBuffer * buf);
+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_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_simple (element_class,
- "RTP MPEG4 ES depayloader", "Codec/Depayloader/Network",
- "Extracts MPEG4 elementary streams from RTP packets (RFC 3640)",
- "Wim Taymans <wim.taymans@gmail.com>");
-}
-
-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;
+ gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) 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;
+ 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,
- GstRtpMP4GDepayClass * klass)
+gst_rtp_mp4g_depay_init (GstRtpMP4GDepay * rtpmp4gdepay)
{
rtpmp4gdepay->adapter = gst_adapter_new ();
rtpmp4gdepay->packets = g_queue_new ();
}
static gboolean
-gst_rtp_mp4g_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
+gst_rtp_mp4g_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
{
GstStructure *structure;
GstRtpMP4GDepay *rtpmp4gdepay;
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,
}
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;
- gboolean discont = FALSE;
guint AU_index;
while ((outbuf = g_queue_pop_head (rtpmp4gdepay->packets))) {
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_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpmp4gdepay), outbuf);
- rtpmp4gdepay->next_AU_index = AU_index + 1;
+ gst_rtp_mp4g_depay_push_outbuf (rtpmp4gdepay, outbuf, AU_index);
}
}
/* we received the expected packet, push it and flush as much as we can from
* the queue */
- gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpmp4gdepay), outbuf);
- rtpmp4gdepay->next_AU_index++;
+ 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) {
- GST_DEBUG_OBJECT (rtpmp4gdepay, "pushing expected AU_index %u",
- AU_index);
outbuf = g_queue_pop_head (rtpmp4gdepay->packets);
- gst_base_rtp_depayload_push (GST_BASE_RTP_DEPAYLOAD (rtpmp4gdepay),
- outbuf);
- rtpmp4gdepay->next_AU_index++;
+ 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);
}
static GstBuffer *
-gst_rtp_mp4g_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+gst_rtp_mp4g_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
GstRtpMP4GDepay *rtpmp4gdepay;
- GstBuffer *outbuf;
+ GstBuffer *outbuf = NULL;
GstClockTime timestamp;
rtpmp4gdepay = GST_RTP_MP4G_DEPAY (depayload);
/* flush remaining data on discont */
- if (GST_BUFFER_IS_DISCONT (buf)) {
+ if (GST_BUFFER_IS_DISCONT (rtp->buffer)) {
GST_DEBUG_OBJECT (rtpmp4gdepay, "received DISCONT");
gst_adapter_clear (rtpmp4gdepay->adapter);
}
- timestamp = GST_BUFFER_TIMESTAMP (buf);
+ timestamp = GST_BUFFER_PTS (rtp->buffer);
{
gint payload_len, payload_AU;
guint AU_size, AU_index, AU_index_delta, payload_AU_size;
gboolean M;
- payload_len = gst_rtp_buffer_get_payload_len (buf);
- payload = gst_rtp_buffer_get_payload (buf);
+ 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 (buf);
- M = gst_rtp_buffer_get_marker (buf);
+ 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;
/* 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 */
rtpmp4gdepay->last_AU_index = AU_index;
}
- /* keep track of the higest 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");
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;
/* collect stuff in the adapter, strip header from payload and push in
* the adapter */
outbuf =
- gst_rtp_buffer_get_payload_subbuffer (buf, payload_AU, AU_size);
+ 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);
- gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
/* 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_PTS (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;
+ 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 %d",
- GST_BUFFER_SIZE (outbuf));
+ GST_DEBUG_OBJECT (depayload,
+ "pushing buffer of size %" G_GSIZE_FORMAT,
+ gst_buffer_get_size (outbuf));
gst_rtp_mp4g_depay_queue (rtpmp4gdepay, outbuf);
}
} else {
/* push complete buffer in adapter */
- outbuf = gst_rtp_buffer_get_payload_subbuffer (buf, 0, payload_len);
+ 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 */
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;
}
}
}
+
return NULL;
/* ERRORS */
}
}
+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)
projects / platform / upstream / gst-plugins-good.git / blobdiff