--- /dev/null
+/* GStreamer
+ * Copyright (C) <2005> Philippe Khalaf <burger@speedy.org>
+ * Copyright (C) <2005> Nokia Corporation <kai.vehmanen@nokia.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.
+ */
+
+/**
+ * SECTION:gstbasertpdepayload
+ * @short_description: Base class for RTP depayloader
+ *
+ * <refsect2>
+ * <para>
+ * Provides a base class for RTP depayloaders
+ * </para>
+ * </refsect2>
+ */
+
+#include "gstbasertpdepayload.h"
+
+#ifdef GST_DISABLE_DEPRECATED
+#define QUEUE_LOCK_INIT(base) (g_static_rec_mutex_init(&base->queuelock))
+#define QUEUE_LOCK_FREE(base) (g_static_rec_mutex_free(&base->queuelock))
+#define QUEUE_LOCK(base) (g_static_rec_mutex_lock(&base->queuelock))
+#define QUEUE_UNLOCK(base) (g_static_rec_mutex_unlock(&base->queuelock))
+#else
+/* otherwise it's already been defined in the header (FIXME 0.11)*/
+#endif
+
+GST_DEBUG_CATEGORY_STATIC (basertpdepayload_debug);
+#define GST_CAT_DEFAULT (basertpdepayload_debug)
+
+#define GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE(obj) \
+ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_RTP_DEPAYLOAD, GstBaseRTPDepayloadPrivate))
+
+struct _GstBaseRTPDepayloadPrivate
+{
+ GstClockTime npt_start;
+ GstClockTime npt_stop;
+ gdouble play_speed;
+ gdouble play_scale;
+
+ gboolean discont;
+ GstClockTime timestamp;
+ GstClockTime duration;
+
+ guint32 next_seqnum;
+
+ gboolean negotiated;
+};
+
+/* Filter signals and args */
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+#define DEFAULT_QUEUE_DELAY 0
+
+enum
+{
+ PROP_0,
+ PROP_QUEUE_DELAY,
+ PROP_LAST
+};
+
+static void gst_base_rtp_depayload_finalize (GObject * object);
+static void gst_base_rtp_depayload_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_base_rtp_depayload_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec);
+
+static gboolean gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps);
+static GstFlowReturn gst_base_rtp_depayload_chain (GstPad * pad,
+ GstBuffer * in);
+static gboolean gst_base_rtp_depayload_handle_sink_event (GstPad * pad,
+ GstEvent * event);
+
+static GstStateChangeReturn gst_base_rtp_depayload_change_state (GstElement *
+ element, GstStateChange transition);
+
+static void gst_base_rtp_depayload_set_gst_timestamp
+ (GstBaseRTPDepayload * filter, guint32 rtptime, GstBuffer * buf);
+static gboolean gst_base_rtp_depayload_packet_lost (GstBaseRTPDepayload *
+ filter, GstEvent * event);
+static gboolean gst_base_rtp_depayload_handle_event (GstBaseRTPDepayload *
+ filter, GstEvent * event);
+
+GST_BOILERPLATE (GstBaseRTPDepayload, gst_base_rtp_depayload, GstElement,
+ GST_TYPE_ELEMENT);
+
+static void
+gst_base_rtp_depayload_base_init (gpointer klass)
+{
+ /*GstElementClass *element_class = GST_ELEMENT_CLASS (klass); */
+}
+
+static void
+gst_base_rtp_depayload_class_init (GstBaseRTPDepayloadClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+
+ gobject_class = G_OBJECT_CLASS (klass);
+ gstelement_class = (GstElementClass *) klass;
+ parent_class = g_type_class_peek_parent (klass);
+
+ g_type_class_add_private (klass, sizeof (GstBaseRTPDepayloadPrivate));
+
+ gobject_class->finalize = gst_base_rtp_depayload_finalize;
+ gobject_class->set_property = gst_base_rtp_depayload_set_property;
+ gobject_class->get_property = gst_base_rtp_depayload_get_property;
+
+ /**
+ * GstBaseRTPDepayload::queue-delay
+ *
+ * Control the amount of packets to buffer.
+ *
+ * Deprecated: Use a jitterbuffer or RTP session manager to delay packet
+ * playback. This property has no effect anymore since 0.10.15.
+ */
+#ifndef GST_REMOVE_DEPRECATED
+ g_object_class_install_property (gobject_class, PROP_QUEUE_DELAY,
+ g_param_spec_uint ("queue-delay", "Queue Delay",
+ "Amount of ms to queue/buffer, deprecated", 0, G_MAXUINT,
+ DEFAULT_QUEUE_DELAY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+#endif
+
+ gstelement_class->change_state = gst_base_rtp_depayload_change_state;
+
+ klass->set_gst_timestamp = gst_base_rtp_depayload_set_gst_timestamp;
+ klass->packet_lost = gst_base_rtp_depayload_packet_lost;
+ klass->handle_event = gst_base_rtp_depayload_handle_event;
+
+ GST_DEBUG_CATEGORY_INIT (basertpdepayload_debug, "basertpdepayload", 0,
+ "Base class for RTP Depayloaders");
+}
+
+static void
+gst_base_rtp_depayload_init (GstBaseRTPDepayload * filter,
+ GstBaseRTPDepayloadClass * klass)
+{
+ GstPadTemplate *pad_template;
+ GstBaseRTPDepayloadPrivate *priv;
+
+ priv = GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE (filter);
+ filter->priv = priv;
+
+ GST_DEBUG_OBJECT (filter, "init");
+
+ pad_template =
+ gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "sink");
+ g_return_if_fail (pad_template != NULL);
+ filter->sinkpad = gst_pad_new_from_template (pad_template, "sink");
+ gst_pad_set_setcaps_function (filter->sinkpad,
+ gst_base_rtp_depayload_setcaps);
+ gst_pad_set_chain_function (filter->sinkpad, gst_base_rtp_depayload_chain);
+ gst_pad_set_event_function (filter->sinkpad,
+ gst_base_rtp_depayload_handle_sink_event);
+ gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad);
+
+ pad_template =
+ gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "src");
+ g_return_if_fail (pad_template != NULL);
+ filter->srcpad = gst_pad_new_from_template (pad_template, "src");
+ gst_pad_use_fixed_caps (filter->srcpad);
+ gst_element_add_pad (GST_ELEMENT (filter), filter->srcpad);
+
+ filter->queue = g_queue_new ();
+ filter->queue_delay = DEFAULT_QUEUE_DELAY;
+
+ gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
+}
+
+static void
+gst_base_rtp_depayload_finalize (GObject * object)
+{
+ GstBaseRTPDepayload *filter = GST_BASE_RTP_DEPAYLOAD (object);
+
+ g_queue_free (filter->queue);
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static gboolean
+gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps)
+{
+ GstBaseRTPDepayload *filter;
+ GstBaseRTPDepayloadClass *bclass;
+ GstBaseRTPDepayloadPrivate *priv;
+ gboolean res;
+ GstStructure *caps_struct;
+ const GValue *value;
+
+ filter = GST_BASE_RTP_DEPAYLOAD (gst_pad_get_parent (pad));
+ priv = filter->priv;
+
+ bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
+
+ GST_DEBUG_OBJECT (filter, "Set caps");
+
+ caps_struct = gst_caps_get_structure (caps, 0);
+
+ /* get other values for newsegment */
+ value = gst_structure_get_value (caps_struct, "npt-start");
+ if (value && G_VALUE_HOLDS_UINT64 (value))
+ priv->npt_start = g_value_get_uint64 (value);
+ else
+ priv->npt_start = 0;
+ GST_DEBUG_OBJECT (filter, "NPT start %" G_GUINT64_FORMAT, priv->npt_start);
+
+ value = gst_structure_get_value (caps_struct, "npt-stop");
+ if (value && G_VALUE_HOLDS_UINT64 (value))
+ priv->npt_stop = g_value_get_uint64 (value);
+ else
+ priv->npt_stop = -1;
+
+ GST_DEBUG_OBJECT (filter, "NPT stop %" G_GUINT64_FORMAT, priv->npt_stop);
+
+ value = gst_structure_get_value (caps_struct, "play-speed");
+ if (value && G_VALUE_HOLDS_DOUBLE (value))
+ priv->play_speed = g_value_get_double (value);
+ else
+ priv->play_speed = 1.0;
+
+ value = gst_structure_get_value (caps_struct, "play-scale");
+ if (value && G_VALUE_HOLDS_DOUBLE (value))
+ priv->play_scale = g_value_get_double (value);
+ else
+ priv->play_scale = 1.0;
+
+ if (bclass->set_caps) {
+ res = bclass->set_caps (filter, caps);
+ if (!res) {
+ GST_WARNING_OBJECT (filter, "Subclass rejected caps %" GST_PTR_FORMAT,
+ caps);
+ }
+ } else {
+ res = TRUE;
+ }
+
+ priv->negotiated = res;
+
+ gst_object_unref (filter);
+
+ return res;
+}
+
+static GstFlowReturn
+gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in)
+{
+ GstBaseRTPDepayload *filter;
+ GstBaseRTPDepayloadPrivate *priv;
+ GstBaseRTPDepayloadClass *bclass;
+ GstFlowReturn ret = GST_FLOW_OK;
+ GstBuffer *out_buf;
+ GstClockTime timestamp;
+ guint16 seqnum;
+ guint32 rtptime;
+ gboolean discont;
+ gint gap;
+
+ filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
+ priv = filter->priv;
+
+ /* we must have a setcaps first */
+ if (G_UNLIKELY (!priv->negotiated))
+ goto not_negotiated;
+
+ /* we must validate, it's possible that this element is plugged right after a
+ * network receiver and we don't want to operate on invalid data */
+ if (G_UNLIKELY (!gst_rtp_buffer_validate (in)))
+ goto invalid_buffer;
+
+ if (!priv->discont)
+ priv->discont = GST_BUFFER_IS_DISCONT (in);
+
+ timestamp = GST_BUFFER_TIMESTAMP (in);
+ /* convert to running_time and save the timestamp, this is the timestamp
+ * we put on outgoing buffers. */
+ timestamp = gst_segment_to_running_time (&filter->segment, GST_FORMAT_TIME,
+ timestamp);
+ priv->timestamp = timestamp;
+ priv->duration = GST_BUFFER_DURATION (in);
+
+ seqnum = gst_rtp_buffer_get_seq (in);
+ rtptime = gst_rtp_buffer_get_timestamp (in);
+ discont = FALSE;
+
+ GST_LOG_OBJECT (filter, "discont %d, seqnum %u, rtptime %u, timestamp %"
+ GST_TIME_FORMAT, priv->discont, seqnum, rtptime,
+ GST_TIME_ARGS (timestamp));
+
+ /* Check seqnum. This is a very simple check that makes sure that the seqnums
+ * are striclty increasing, dropping anything that is out of the ordinary. We
+ * can only do this when the next_seqnum is known. */
+ if (G_LIKELY (priv->next_seqnum != -1)) {
+ gap = gst_rtp_buffer_compare_seqnum (seqnum, priv->next_seqnum);
+
+ /* if we have no gap, all is fine */
+ if (G_UNLIKELY (gap != 0)) {
+ GST_LOG_OBJECT (filter, "got packet %u, expected %u, gap %d", seqnum,
+ priv->next_seqnum, gap);
+ if (gap < 0) {
+ /* seqnum > next_seqnum, we are missing some packets, this is always a
+ * DISCONT. */
+ GST_LOG_OBJECT (filter, "%d missing packets", gap);
+ discont = TRUE;
+ } else {
+ /* seqnum < next_seqnum, we have seen this packet before or the sender
+ * could be restarted. If the packet is not too old, we throw it away as
+ * a duplicate, otherwise we mark discont and continue. 100 misordered
+ * packets is a good threshold. See also RFC 4737. */
+ if (gap < 100)
+ goto dropping;
+
+ GST_LOG_OBJECT (filter,
+ "%d > 100, packet too old, sender likely restarted", gap);
+ discont = TRUE;
+ }
+ }
+ }
+ priv->next_seqnum = (seqnum + 1) & 0xffff;
+
+ if (G_UNLIKELY (discont && !priv->discont)) {
+ GST_LOG_OBJECT (filter, "mark DISCONT on input buffer");
+ /* we detected a seqnum discont but the buffer was not flagged with a discont,
+ * set the discont flag so that the subclass can throw away old data. */
+ priv->discont = TRUE;
+ in = gst_buffer_make_metadata_writable (in);
+ GST_BUFFER_FLAG_SET (in, GST_BUFFER_FLAG_DISCONT);
+ }
+
+ bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
+
+ if (G_UNLIKELY (bclass->process == NULL))
+ goto no_process;
+
+ /* let's send it out to processing */
+ out_buf = bclass->process (filter, in);
+ if (out_buf) {
+ /* we pass rtptime as backward compatibility, in reality, the incomming
+ * buffer timestamp is always applied to the outgoing packet. */
+ ret = gst_base_rtp_depayload_push_ts (filter, rtptime, out_buf);
+ }
+ gst_buffer_unref (in);
+
+ return ret;
+
+ /* ERRORS */
+not_negotiated:
+ {
+ /* this is not fatal but should be filtered earlier */
+ if (GST_BUFFER_CAPS (in) == NULL) {
+ GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION,
+ ("No RTP format was negotiated."),
+ ("Input buffers need to have RTP caps set on them. This is usually "
+ "achieved by setting the 'caps' property of the upstream source "
+ "element (often udpsrc or appsrc), or by putting a capsfilter "
+ "element before the depayloader and setting the 'caps' property "
+ "on that. Also see http://cgit.freedesktop.org/gstreamer/"
+ "gst-plugins-good/tree/gst/rtp/README"));
+ } else {
+ GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION,
+ ("No RTP format was negotiated."),
+ ("RTP caps on input buffer were rejected, most likely because they "
+ "were incomplete or contained wrong values. Check the debug log "
+ "for more information."));
+ }
+ gst_buffer_unref (in);
+ return GST_FLOW_NOT_NEGOTIATED;
+ }
+invalid_buffer:
+ {
+ /* this is not fatal but should be filtered earlier */
+ GST_ELEMENT_WARNING (filter, STREAM, DECODE, (NULL),
+ ("Received invalid RTP payload, dropping"));
+ gst_buffer_unref (in);
+ return GST_FLOW_OK;
+ }
+dropping:
+ {
+ GST_WARNING_OBJECT (filter, "%d <= 100, dropping old packet", gap);
+ gst_buffer_unref (in);
+ return GST_FLOW_OK;
+ }
+no_process:
+ {
+ /* this is not fatal but should be filtered earlier */
+ GST_ELEMENT_ERROR (filter, STREAM, NOT_IMPLEMENTED, (NULL),
+ ("The subclass does not have a process method"));
+ gst_buffer_unref (in);
+ return GST_FLOW_ERROR;
+ }
+}
+
+static gboolean
+gst_base_rtp_depayload_handle_event (GstBaseRTPDepayload * filter,
+ GstEvent * event)
+{
+ gboolean res = TRUE;
+ gboolean forward = TRUE;
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_FLUSH_STOP:
+ gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
+ filter->need_newsegment = TRUE;
+ filter->priv->next_seqnum = -1;
+ break;
+ case GST_EVENT_NEWSEGMENT:
+ {
+ gboolean update;
+ gdouble rate;
+ GstFormat fmt;
+ gint64 start, stop, position;
+
+ gst_event_parse_new_segment (event, &update, &rate, &fmt, &start, &stop,
+ &position);
+
+ gst_segment_set_newsegment (&filter->segment, update, rate, fmt,
+ start, stop, position);
+
+ /* don't pass the event downstream, we generate our own segment including
+ * the NTP time and other things we receive in caps */
+ forward = FALSE;
+ break;
+ }
+ case GST_EVENT_CUSTOM_DOWNSTREAM:
+ {
+ GstBaseRTPDepayloadClass *bclass;
+
+ bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
+
+ if (gst_event_has_name (event, "GstRTPPacketLost")) {
+ /* we get this event from the jitterbuffer when it considers a packet as
+ * being lost. We send it to our packet_lost vmethod. The default
+ * implementation will make time progress by pushing out a NEWSEGMENT
+ * update event. Subclasses can override and to one of the following:
+ * - Adjust timestamp/duration to something more accurate before
+ * calling the parent (default) packet_lost method.
+ * - do some more advanced error concealing on the already received
+ * (fragmented) packets.
+ * - ignore the packet lost.
+ */
+ if (bclass->packet_lost)
+ res = bclass->packet_lost (filter, event);
+ forward = FALSE;
+ }
+ break;
+ }
+ default:
+ break;
+ }
+
+ if (forward)
+ res = gst_pad_push_event (filter->srcpad, event);
+ else
+ gst_event_unref (event);
+
+ return res;
+}
+
+static gboolean
+gst_base_rtp_depayload_handle_sink_event (GstPad * pad, GstEvent * event)
+{
+ gboolean res = FALSE;
+ GstBaseRTPDepayload *filter;
+ GstBaseRTPDepayloadClass *bclass;
+
+ filter = GST_BASE_RTP_DEPAYLOAD (gst_pad_get_parent (pad));
+ if (G_UNLIKELY (filter == NULL)) {
+ gst_event_unref (event);
+ return FALSE;
+ }
+
+ bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
+ if (bclass->handle_event)
+ res = bclass->handle_event (filter, event);
+ else
+ gst_event_unref (event);
+
+ gst_object_unref (filter);
+ return res;
+}
+
+static GstEvent *
+create_segment_event (GstBaseRTPDepayload * filter, gboolean update,
+ GstClockTime position)
+{
+ GstEvent *event;
+ GstClockTime stop;
+ GstBaseRTPDepayloadPrivate *priv;
+
+ priv = filter->priv;
+
+ if (priv->npt_stop != -1)
+ stop = priv->npt_stop - priv->npt_start;
+ else
+ stop = -1;
+
+ event = gst_event_new_new_segment_full (update, priv->play_speed,
+ priv->play_scale, GST_FORMAT_TIME, position, stop,
+ position + priv->npt_start);
+
+ return event;
+}
+
+typedef struct
+{
+ GstBaseRTPDepayload *depayload;
+ GstBaseRTPDepayloadClass *bclass;
+ GstCaps *caps;
+ gboolean do_ts;
+ gboolean rtptime;
+} HeaderData;
+
+static GstBufferListItem
+set_headers (GstBuffer ** buffer, guint group, guint idx, HeaderData * data)
+{
+ GstBaseRTPDepayload *depayload = data->depayload;
+
+ *buffer = gst_buffer_make_metadata_writable (*buffer);
+ gst_buffer_set_caps (*buffer, data->caps);
+
+ /* set the timestamp if we must and can */
+ if (data->bclass->set_gst_timestamp && data->do_ts)
+ data->bclass->set_gst_timestamp (depayload, data->rtptime, *buffer);
+
+ if (G_UNLIKELY (depayload->priv->discont)) {
+ GST_LOG_OBJECT (depayload, "Marking DISCONT on output buffer");
+ GST_BUFFER_FLAG_SET (*buffer, GST_BUFFER_FLAG_DISCONT);
+ depayload->priv->discont = FALSE;
+ }
+
+ return GST_BUFFER_LIST_SKIP_GROUP;
+}
+
+static GstFlowReturn
+gst_base_rtp_depayload_prepare_push (GstBaseRTPDepayload * filter,
+ gboolean do_ts, guint32 rtptime, gboolean is_list, gpointer obj)
+{
+ HeaderData data;
+
+ data.depayload = filter;
+ data.caps = GST_PAD_CAPS (filter->srcpad);
+ data.rtptime = rtptime;
+ data.do_ts = do_ts;
+ data.bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
+
+ if (is_list) {
+ GstBufferList **blist = obj;
+ gst_buffer_list_foreach (*blist, (GstBufferListFunc) set_headers, &data);
+ } else {
+ GstBuffer **buf = obj;
+ set_headers (buf, 0, 0, &data);
+ }
+
+ /* if this is the first buffer send a NEWSEGMENT */
+ if (G_UNLIKELY (filter->need_newsegment)) {
+ GstEvent *event;
+
+ event = create_segment_event (filter, FALSE, 0);
+
+ gst_pad_push_event (filter->srcpad, event);
+
+ filter->need_newsegment = FALSE;
+ GST_DEBUG_OBJECT (filter, "Pushed newsegment event on this first buffer");
+ }
+
+ return GST_FLOW_OK;
+}
+
+/**
+ * gst_base_rtp_depayload_push_ts:
+ * @filter: a #GstBaseRTPDepayload
+ * @timestamp: an RTP timestamp to apply
+ * @out_buf: a #GstBuffer
+ *
+ * Push @out_buf to the peer of @filter. This function takes ownership of
+ * @out_buf.
+ *
+ * Unlike gst_base_rtp_depayload_push(), this function will by default apply
+ * the last incomming timestamp on the outgoing buffer when it didn't have a
+ * timestamp already. The set_get_timestamp vmethod can be overwritten to change
+ * this behaviour (and take, for example, @timestamp into account).
+ *
+ * Returns: a #GstFlowReturn.
+ */
+GstFlowReturn
+gst_base_rtp_depayload_push_ts (GstBaseRTPDepayload * filter, guint32 timestamp,
+ GstBuffer * out_buf)
+{
+ GstFlowReturn res;
+
+ res =
+ gst_base_rtp_depayload_prepare_push (filter, TRUE, timestamp, FALSE,
+ &out_buf);
+
+ if (G_LIKELY (res == GST_FLOW_OK))
+ res = gst_pad_push (filter->srcpad, out_buf);
+ else
+ gst_buffer_unref (out_buf);
+
+ return res;
+}
+
+/**
+ * gst_base_rtp_depayload_push:
+ * @filter: a #GstBaseRTPDepayload
+ * @out_buf: a #GstBuffer
+ *
+ * Push @out_buf to the peer of @filter. This function takes ownership of
+ * @out_buf.
+ *
+ * Unlike gst_base_rtp_depayload_push_ts(), this function will not apply
+ * any timestamp on the outgoing buffer. Subclasses should therefore timestamp
+ * outgoing buffers themselves.
+ *
+ * Returns: a #GstFlowReturn.
+ */
+GstFlowReturn
+gst_base_rtp_depayload_push (GstBaseRTPDepayload * filter, GstBuffer * out_buf)
+{
+ GstFlowReturn res;
+
+ res = gst_base_rtp_depayload_prepare_push (filter, FALSE, 0, FALSE, &out_buf);
+
+ if (G_LIKELY (res == GST_FLOW_OK))
+ res = gst_pad_push (filter->srcpad, out_buf);
+ else
+ gst_buffer_unref (out_buf);
+
+ return res;
+}
+
+/**
+ * gst_base_rtp_depayload_push_list:
+ * @filter: a #GstBaseRTPDepayload
+ * @out_list: a #GstBufferList
+ *
+ * Push @out_list to the peer of @filter. This function takes ownership of
+ * @out_list.
+ *
+ * Returns: a #GstFlowReturn.
+ *
+ * Since: 0.10.32
+ */
+GstFlowReturn
+gst_base_rtp_depayload_push_list (GstBaseRTPDepayload * filter,
+ GstBufferList * out_list)
+{
+ GstFlowReturn res;
+
+ res = gst_base_rtp_depayload_prepare_push (filter, TRUE, 0, TRUE, &out_list);
+
+ if (G_LIKELY (res == GST_FLOW_OK))
+ res = gst_pad_push_list (filter->srcpad, out_list);
+ else
+ gst_buffer_list_unref (out_list);
+
+ return res;
+}
+
+/* convert the PacketLost event form a jitterbuffer to a segment update.
+ * subclasses can override this. */
+static gboolean
+gst_base_rtp_depayload_packet_lost (GstBaseRTPDepayload * filter,
+ GstEvent * event)
+{
+ GstClockTime timestamp, duration, position;
+ GstEvent *sevent;
+ const GstStructure *s;
+
+ s = gst_event_get_structure (event);
+
+ /* first start by parsing the timestamp and duration */
+ timestamp = -1;
+ duration = -1;
+
+ gst_structure_get_clock_time (s, "timestamp", ×tamp);
+ gst_structure_get_clock_time (s, "duration", &duration);
+
+ position = timestamp;
+ if (duration != -1)
+ position += duration;
+
+ /* update the current segment with the elapsed time */
+ sevent = create_segment_event (filter, TRUE, position);
+
+ return gst_pad_push_event (filter->srcpad, sevent);
+}
+
+static void
+gst_base_rtp_depayload_set_gst_timestamp (GstBaseRTPDepayload * filter,
+ guint32 rtptime, GstBuffer * buf)
+{
+ GstBaseRTPDepayloadPrivate *priv;
+ GstClockTime timestamp, duration;
+
+ priv = filter->priv;
+
+ timestamp = GST_BUFFER_TIMESTAMP (buf);
+ duration = GST_BUFFER_DURATION (buf);
+
+ /* apply last incomming timestamp and duration to outgoing buffer if
+ * not otherwise set. */
+ if (!GST_CLOCK_TIME_IS_VALID (timestamp))
+ GST_BUFFER_TIMESTAMP (buf) = priv->timestamp;
+ if (!GST_CLOCK_TIME_IS_VALID (duration))
+ GST_BUFFER_DURATION (buf) = priv->duration;
+}
+
+static GstStateChangeReturn
+gst_base_rtp_depayload_change_state (GstElement * element,
+ GstStateChange transition)
+{
+ GstBaseRTPDepayload *filter;
+ GstBaseRTPDepayloadPrivate *priv;
+ GstStateChangeReturn ret;
+
+ filter = GST_BASE_RTP_DEPAYLOAD (element);
+ priv = filter->priv;
+
+ switch (transition) {
+ case GST_STATE_CHANGE_NULL_TO_READY:
+ break;
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ filter->need_newsegment = TRUE;
+ priv->npt_start = 0;
+ priv->npt_stop = -1;
+ priv->play_speed = 1.0;
+ priv->play_scale = 1.0;
+ priv->next_seqnum = -1;
+ priv->negotiated = FALSE;
+ priv->discont = FALSE;
+ break;
+ case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+ break;
+ default:
+ break;
+ }
+
+ ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+ break;
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ break;
+ case GST_STATE_CHANGE_READY_TO_NULL:
+ break;
+ default:
+ break;
+ }
+ return ret;
+}
+
+static void
+gst_base_rtp_depayload_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstBaseRTPDepayload *filter;
+
+ filter = GST_BASE_RTP_DEPAYLOAD (object);
+
+ switch (prop_id) {
+ case PROP_QUEUE_DELAY:
+ filter->queue_delay = g_value_get_uint (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_base_rtp_depayload_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstBaseRTPDepayload *filter;
+
+ filter = GST_BASE_RTP_DEPAYLOAD (object);
+
+ switch (prop_id) {
+ case PROP_QUEUE_DELAY:
+ g_value_set_uint (value, filter->queue_delay);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
projects / mafwsubrenderer / blobdiff