gstreamer/gst-libs/gst/rtp/gstrtpbasedepayload.c

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/* GStreamer
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* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
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* SECTION:gstrtpbasedepayload
* @short_description: Base class for RTP depayloader
*
* Provides a base class for RTP depayloaders
*/
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#include "gstrtpbasedepayload.h"
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GST_DEBUG_CATEGORY_STATIC (rtpbasedepayload_debug);
#define GST_CAT_DEFAULT (rtpbasedepayload_debug)
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#define GST_RTP_BASE_DEPAYLOAD_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BASE_DEPAYLOAD, GstRTPBaseDepayloadPrivate))
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struct _GstRTPBaseDepayloadPrivate
{
GstClockTime npt_start;
GstClockTime npt_stop;
gdouble play_speed;
gdouble play_scale;
gboolean discont;
GstClockTime pts;
GstClockTime dts;
GstClockTime duration;
guint32 last_seqnum;
guint32 last_rtptime;
guint32 next_seqnum;
gboolean negotiated;
GstCaps *last_caps;
};
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_STATS,
PROP_LAST
};
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static void gst_rtp_base_depayload_finalize (GObject * object);
static void gst_rtp_base_depayload_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_rtp_base_depayload_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
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static GstFlowReturn gst_rtp_base_depayload_chain (GstPad * pad,
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GstObject * parent, GstBuffer * in);
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static gboolean gst_rtp_base_depayload_handle_sink_event (GstPad * pad,
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GstObject * parent, GstEvent * event);
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static GstStateChangeReturn gst_rtp_base_depayload_change_state (GstElement *
element, GstStateChange transition);
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static gboolean gst_rtp_base_depayload_packet_lost (GstRTPBaseDepayload *
filter, GstEvent * event);
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static gboolean gst_rtp_base_depayload_handle_event (GstRTPBaseDepayload *
filter, GstEvent * event);
static GstElementClass *parent_class = NULL;
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static void gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass *
klass);
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static void gst_rtp_base_depayload_init (GstRTPBaseDepayload * rtpbasepayload,
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GstRTPBaseDepayloadClass * klass);
GType
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gst_rtp_base_depayload_get_type (void)
{
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static GType rtp_base_depayload_type = 0;
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if (g_once_init_enter ((gsize *) & rtp_base_depayload_type)) {
static const GTypeInfo rtp_base_depayload_info = {
sizeof (GstRTPBaseDepayloadClass),
NULL,
NULL,
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(GClassInitFunc) gst_rtp_base_depayload_class_init,
NULL,
NULL,
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sizeof (GstRTPBaseDepayload),
0,
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(GInstanceInitFunc) gst_rtp_base_depayload_init,
};
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g_once_init_leave ((gsize *) & rtp_base_depayload_type,
g_type_register_static (GST_TYPE_ELEMENT, "GstRTPBaseDepayload",
&rtp_base_depayload_info, G_TYPE_FLAG_ABSTRACT));
}
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return rtp_base_depayload_type;
}
static void
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gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass * 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);
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g_type_class_add_private (klass, sizeof (GstRTPBaseDepayloadPrivate));
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gobject_class->finalize = gst_rtp_base_depayload_finalize;
gobject_class->set_property = gst_rtp_base_depayload_set_property;
gobject_class->get_property = gst_rtp_base_depayload_get_property;
/**
* GstRTPBaseDepayload:stats:
*
* Various depayloader statistics retrieved atomically (and are therefore
* synchroized with each other). This property return a GstStructure named
* application/x-rtp-depayload-stats containing the following fields relating to
* the last processed buffer and current state of the stream being depayloaded:
*
* <variablelist>
* <varlistentry>
* <term>clock-rate</term>
* <listitem><para>#G_TYPE_UINT, clock-rate of the
* stream</para></listitem>
* </varlistentry>
* <varlistentry>
* <term>npt-start</term>
* <listitem><para>#G_TYPE_UINT64, time of playback start
* </para></listitem>
* </varlistentry>
* <varlistentry>
* <term>npt-stop</term>
* <listitem><para>#G_TYPE_UINT64, time of playback stop
* </para></listitem>
* </varlistentry>
* <varlistentry>
* <term>play-speed</term>
* <listitem><para>#G_TYPE_DOUBLE, the playback speed
* </para></listitem>
* </varlistentry>
* <varlistentry>
* <term>play-scale</term>
* <listitem><para>#G_TYPE_DOUBLE, the playback scale
* </para></listitem>
* </varlistentry>
* <varlistentry>
* <term>running-time-dts</term>
* <listitem><para>#G_TYPE_UINT64, the last running-time of the
* last DTS
* </para></listitem>
* </varlistentry>
* <varlistentry>
* <term>running-time-pts</term>
* <listitem><para>#G_TYPE_UINT64, the last running-time of the
* last PTS
* </para></listitem>
* </varlistentry>
* <varlistentry>
* <term>seqnum</term>
* <listitem><para>#G_TYPE_UINT, the last seen seqnum
* </para></listitem>
* </varlistentry>
* <varlistentry>
* <term>timestamp</term>
* <listitem><para>#G_TYPE_UINT, the last seen RTP timestamp
* </para></listitem>
* </varlistentry>
* </variablelist>
**/
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_STATS,
g_param_spec_boxed ("stats", "Statistics", "Various statistics",
GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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gstelement_class->change_state = gst_rtp_base_depayload_change_state;
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klass->packet_lost = gst_rtp_base_depayload_packet_lost;
klass->handle_event = gst_rtp_base_depayload_handle_event;
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GST_DEBUG_CATEGORY_INIT (rtpbasedepayload_debug, "rtpbasedepayload", 0,
"Base class for RTP Depayloaders");
}
static void
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gst_rtp_base_depayload_init (GstRTPBaseDepayload * filter,
GstRTPBaseDepayloadClass * klass)
{
GstPadTemplate *pad_template;
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GstRTPBaseDepayloadPrivate *priv;
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priv = GST_RTP_BASE_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");
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gst_pad_set_chain_function (filter->sinkpad, gst_rtp_base_depayload_chain);
gst_pad_set_event_function (filter->sinkpad,
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gst_rtp_base_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);
priv->npt_start = 0;
priv->npt_stop = -1;
priv->play_speed = 1.0;
priv->play_scale = 1.0;
priv->dts = -1;
priv->pts = -1;
priv->duration = -1;
gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
}
static void
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gst_rtp_base_depayload_finalize (GObject * object)
{
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
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gst_rtp_base_depayload_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps)
{
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GstRTPBaseDepayloadClass *bclass;
GstRTPBaseDepayloadPrivate *priv;
gboolean res;
GstStructure *caps_struct;
const GValue *value;
priv = filter->priv;
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bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter);
GST_DEBUG_OBJECT (filter, "Set caps %" GST_PTR_FORMAT, caps);
if (priv->last_caps) {
if (gst_caps_is_equal (priv->last_caps, caps)) {
res = TRUE;
goto caps_not_changed;
} else {
gst_caps_unref (priv->last_caps);
priv->last_caps = NULL;
}
}
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;
if (priv->negotiated)
priv->last_caps = gst_caps_ref (caps);
return res;
caps_not_changed:
{
GST_DEBUG_OBJECT (filter, "Caps did not change");
return res;
}
}
static GstFlowReturn
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gst_rtp_base_depayload_chain (GstPad * pad, GstObject * parent, GstBuffer * in)
{
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GstRTPBaseDepayload *filter;
GstRTPBaseDepayloadPrivate *priv;
GstRTPBaseDepayloadClass *bclass;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
GstClockTime pts, dts;
guint16 seqnum;
guint32 rtptime;
gboolean discont, buf_discont;
gint gap;
GstRTPBuffer rtp = { NULL };
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filter = GST_RTP_BASE_DEPAYLOAD (parent);
priv = filter->priv;
/* we must have a setcaps first */
if (G_UNLIKELY (!priv->negotiated))
goto not_negotiated;
if (G_UNLIKELY (!gst_rtp_buffer_map (in, GST_MAP_READ, &rtp)))
goto invalid_buffer;
buf_discont = GST_BUFFER_IS_DISCONT (in);
pts = GST_BUFFER_PTS (in);
dts = GST_BUFFER_DTS (in);
/* convert to running_time and save the timestamp, this is the timestamp
* we put on outgoing buffers. */
pts = gst_segment_to_running_time (&filter->segment, GST_FORMAT_TIME, pts);
dts = gst_segment_to_running_time (&filter->segment, GST_FORMAT_TIME, dts);
priv->pts = pts;
priv->dts = dts;
priv->duration = GST_BUFFER_DURATION (in);
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seqnum = gst_rtp_buffer_get_seq (&rtp);
rtptime = gst_rtp_buffer_get_timestamp (&rtp);
gst_rtp_buffer_unmap (&rtp);
priv->last_seqnum = seqnum;
priv->last_rtptime = rtptime;
discont = buf_discont;
GST_LOG_OBJECT (filter, "discont %d, seqnum %u, rtptime %u, pts %"
GST_TIME_FORMAT ", dts %" GST_TIME_FORMAT, buf_discont, seqnum, rtptime,
GST_TIME_ARGS (pts), GST_TIME_ARGS (dts));
/* 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 = TRUE;
if (!buf_discont) {
/* 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. */
GST_LOG_OBJECT (filter, "mark DISCONT on input buffer");
in = gst_buffer_make_writable (in);
GST_BUFFER_FLAG_SET (in, GST_BUFFER_FLAG_DISCONT);
}
}
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bclass = GST_RTP_BASE_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) {
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ret = gst_rtp_base_depayload_push (filter, out_buf);
}
gst_buffer_unref (in);
return ret;
/* ERRORS */
not_negotiated:
{
/* this is not fatal but should be filtered earlier */
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"));
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
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gst_rtp_base_depayload_handle_event (GstRTPBaseDepayload * filter,
GstEvent * event)
{
gboolean res = TRUE;
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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_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
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res = gst_rtp_base_depayload_setcaps (filter, caps);
forward = FALSE;
break;
}
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case GST_EVENT_SEGMENT:
{
gst_event_copy_segment (event, &filter->segment);
/* don't pass the event downstream, we generate our own segment including
* the NTP time and other things we receive in caps */
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forward = FALSE;
break;
}
case GST_EVENT_CUSTOM_DOWNSTREAM:
{
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GstRTPBaseDepayloadClass *bclass;
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bclass = GST_RTP_BASE_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 GAP event.
* Subclasses can override and do 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);
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forward = FALSE;
}
break;
}
default:
break;
}
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if (forward)
res = gst_pad_push_event (filter->srcpad, event);
else
gst_event_unref (event);
return res;
}
static gboolean
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gst_rtp_base_depayload_handle_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
gboolean res = FALSE;
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GstRTPBaseDepayload *filter;
GstRTPBaseDepayloadClass *bclass;
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filter = GST_RTP_BASE_DEPAYLOAD (parent);
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bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter);
if (bclass->handle_event)
res = bclass->handle_event (filter, event);
else
gst_event_unref (event);
return res;
}
static GstEvent *
create_segment_event (GstRTPBaseDepayload * filter, GstClockTime position)
{
GstEvent *event;
GstClockTime stop, running_time;
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GstRTPBaseDepayloadPrivate *priv;
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GstSegment segment;
priv = filter->priv;
if (priv->npt_stop != -1)
stop = position + priv->npt_stop - priv->npt_start;
else
stop = -1;
if (position == -1)
position = 0;
running_time = gst_segment_to_running_time (&filter->segment,
GST_FORMAT_TIME, position);
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gst_segment_init (&segment, GST_FORMAT_TIME);
segment.rate = priv->play_speed;
segment.applied_rate = priv->play_scale;
segment.start = position;
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segment.stop = stop;
segment.time = priv->npt_start;
segment.position = position;
segment.base = running_time;
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event = gst_event_new_segment (&segment);
return event;
}
typedef struct
{
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GstRTPBaseDepayload *depayload;
GstRTPBaseDepayloadClass *bclass;
} HeaderData;
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static gboolean
set_headers (GstBuffer ** buffer, guint idx, HeaderData * data)
{
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GstRTPBaseDepayload *depayload = data->depayload;
GstRTPBaseDepayloadPrivate *priv = depayload->priv;
GstClockTime pts, dts, duration;
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*buffer = gst_buffer_make_writable (*buffer);
pts = GST_BUFFER_PTS (*buffer);
dts = GST_BUFFER_DTS (*buffer);
duration = GST_BUFFER_DURATION (*buffer);
/* apply last incomming timestamp and duration to outgoing buffer if
* not otherwise set. */
if (!GST_CLOCK_TIME_IS_VALID (pts))
GST_BUFFER_PTS (*buffer) = priv->pts;
if (!GST_CLOCK_TIME_IS_VALID (dts))
GST_BUFFER_DTS (*buffer) = priv->dts;
if (!GST_CLOCK_TIME_IS_VALID (duration))
GST_BUFFER_DURATION (*buffer) = priv->duration;
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;
}
/* make sure we only set the timestamp on the first packet */
priv->pts = GST_CLOCK_TIME_NONE;
priv->dts = GST_CLOCK_TIME_NONE;
priv->duration = GST_CLOCK_TIME_NONE;
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return TRUE;
}
static GstFlowReturn
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gst_rtp_base_depayload_prepare_push (GstRTPBaseDepayload * filter,
gboolean is_list, gpointer obj)
{
HeaderData data;
data.depayload = filter;
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data.bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (filter);
if (is_list) {
GstBufferList **blist = obj;
gst_buffer_list_foreach (*blist, (GstBufferListFunc) set_headers, &data);
} else {
GstBuffer **buf = obj;
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set_headers (buf, 0, &data);
}
/* if this is the first buffer send a NEWSEGMENT */
if (G_UNLIKELY (filter->need_newsegment)) {
GstEvent *event;
GstClockTime pts;
if (is_list) {
GstBufferList **blist = obj;
GstBuffer *buf = gst_buffer_list_get (*blist, 0);
pts = GST_BUFFER_PTS (buf);
} else {
GstBuffer **buf = obj;
set_headers (buf, 0, &data);
pts = GST_BUFFER_PTS (*buf);
}
event = create_segment_event (filter, pts);
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;
}
/**
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* gst_rtp_base_depayload_push:
* @filter: a #GstRTPBaseDepayload
* @out_buf: a #GstBuffer
*
* Push @out_buf to the peer of @filter. This function takes ownership of
* @out_buf.
*
* This function will by default apply the last incomming timestamp on
* the outgoing buffer when it didn't have a timestamp already.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
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gst_rtp_base_depayload_push (GstRTPBaseDepayload * filter, GstBuffer * out_buf)
{
GstFlowReturn res;
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res = gst_rtp_base_depayload_prepare_push (filter, 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;
}
/**
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* gst_rtp_base_depayload_push_list:
* @filter: a #GstRTPBaseDepayload
* @out_list: a #GstBufferList
*
* Push @out_list to the peer of @filter. This function takes ownership of
* @out_list.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
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gst_rtp_base_depayload_push_list (GstRTPBaseDepayload * filter,
GstBufferList * out_list)
{
GstFlowReturn res;
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res = gst_rtp_base_depayload_prepare_push (filter, 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 from a jitterbuffer to a GAP event.
* subclasses can override this. */
static gboolean
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gst_rtp_base_depayload_packet_lost (GstRTPBaseDepayload * filter,
GstEvent * event)
{
GstClockTime timestamp, duration;
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", &timestamp);
gst_structure_get_clock_time (s, "duration", &duration);
/* send GAP event */
sevent = gst_event_new_gap (timestamp, duration);
return gst_pad_push_event (filter->srcpad, sevent);
}
static GstStateChangeReturn
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gst_rtp_base_depayload_change_state (GstElement * element,
GstStateChange transition)
{
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GstRTPBaseDepayload *filter;
GstRTPBaseDepayloadPrivate *priv;
GstStateChangeReturn ret;
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filter = GST_RTP_BASE_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:
gst_caps_replace (&priv->last_caps, NULL);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
static GstStructure *
gst_rtp_base_depayload_create_stats (GstRTPBaseDepayload * depayload)
{
GstRTPBaseDepayloadPrivate *priv;
GstStructure *s;
priv = depayload->priv;
s = gst_structure_new ("application/x-rtp-depayload-stats",
"clock_rate", G_TYPE_UINT, depayload->clock_rate,
"npt-start", G_TYPE_UINT64, priv->npt_start,
"npt-stop", G_TYPE_UINT64, priv->npt_stop,
"play-speed", G_TYPE_DOUBLE, priv->play_speed,
"play-scale", G_TYPE_DOUBLE, priv->play_scale,
"running-time-dts", G_TYPE_UINT64, priv->dts,
"running-time-pts", G_TYPE_UINT64, priv->pts,
"seqnum", G_TYPE_UINT, (guint) priv->last_seqnum,
"timestamp", G_TYPE_UINT, (guint) priv->last_rtptime, NULL);
return s;
}
static void
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gst_rtp_base_depayload_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
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gst_rtp_base_depayload_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRTPBaseDepayload *depayload;
depayload = GST_RTP_BASE_DEPAYLOAD (object);
switch (prop_id) {
case PROP_STATS:
g_value_take_boxed (value,
gst_rtp_base_depayload_create_stats (depayload));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}