gstreamer/gst-libs/gst/rtp/gstrtpbasedepayload.c
Tim-Philipp Müller bc309a100f rtpbasedepayload: provide chain_list function on sink pad
Implement a chain_list function, which avoids lots of locking
compared to the default fallback implementation in GstPad.
We may also want to do some more sophisticated timestamp
tracking here at some point, but for now leave it up to the
jitterbuffer and/or subclasses (in case buffers in the
buffer list have no timestamp set on them, there may only
be a timestamp for the whole list on the first buffer).
This provides the exact same behaviour as the default
fallback implementation.
2015-06-01 19:00:55 +01:00

940 lines
27 KiB
C

/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:gstrtpbasedepayload
* @short_description: Base class for RTP depayloader
*
* Provides a base class for RTP depayloaders
*/
#include "gstrtpbasedepayload.h"
GST_DEBUG_CATEGORY_STATIC (rtpbasedepayload_debug);
#define GST_CAT_DEFAULT (rtpbasedepayload_debug)
#define GST_RTP_BASE_DEPAYLOAD_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BASE_DEPAYLOAD, GstRTPBaseDepayloadPrivate))
struct _GstRTPBaseDepayloadPrivate
{
GstClockTime npt_start;
GstClockTime npt_stop;
gdouble play_speed;
gdouble play_scale;
guint clock_base;
gboolean discont;
GstClockTime pts;
GstClockTime dts;
GstClockTime duration;
guint32 last_seqnum;
guint32 last_rtptime;
guint32 next_seqnum;
gboolean negotiated;
GstCaps *last_caps;
GstEvent *segment_event;
};
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_STATS,
PROP_LAST
};
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);
static void gst_rtp_base_depayload_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstFlowReturn gst_rtp_base_depayload_chain (GstPad * pad,
GstObject * parent, GstBuffer * in);
static GstFlowReturn gst_rtp_base_depayload_chain_list (GstPad * pad,
GstObject * parent, GstBufferList * list);
static gboolean gst_rtp_base_depayload_handle_sink_event (GstPad * pad,
GstObject * parent, GstEvent * event);
static GstStateChangeReturn gst_rtp_base_depayload_change_state (GstElement *
element, GstStateChange transition);
static gboolean gst_rtp_base_depayload_packet_lost (GstRTPBaseDepayload *
filter, GstEvent * event);
static gboolean gst_rtp_base_depayload_handle_event (GstRTPBaseDepayload *
filter, GstEvent * event);
static GstElementClass *parent_class = NULL;
static void gst_rtp_base_depayload_class_init (GstRTPBaseDepayloadClass *
klass);
static void gst_rtp_base_depayload_init (GstRTPBaseDepayload * rtpbasepayload,
GstRTPBaseDepayloadClass * klass);
static GstEvent *create_segment_event (GstRTPBaseDepayload * filter,
guint rtptime, GstClockTime position);
GType
gst_rtp_base_depayload_get_type (void)
{
static GType rtp_base_depayload_type = 0;
if (g_once_init_enter ((gsize *) & rtp_base_depayload_type)) {
static const GTypeInfo rtp_base_depayload_info = {
sizeof (GstRTPBaseDepayloadClass),
NULL,
NULL,
(GClassInitFunc) gst_rtp_base_depayload_class_init,
NULL,
NULL,
sizeof (GstRTPBaseDepayload),
0,
(GInstanceInitFunc) gst_rtp_base_depayload_init,
};
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));
}
return rtp_base_depayload_type;
}
static void
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);
g_type_class_add_private (klass, sizeof (GstRTPBaseDepayloadPrivate));
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));
gstelement_class->change_state = gst_rtp_base_depayload_change_state;
klass->packet_lost = gst_rtp_base_depayload_packet_lost;
klass->handle_event = gst_rtp_base_depayload_handle_event;
GST_DEBUG_CATEGORY_INIT (rtpbasedepayload_debug, "rtpbasedepayload", 0,
"Base class for RTP Depayloaders");
}
static void
gst_rtp_base_depayload_init (GstRTPBaseDepayload * filter,
GstRTPBaseDepayloadClass * klass)
{
GstPadTemplate *pad_template;
GstRTPBaseDepayloadPrivate *priv;
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");
gst_pad_set_chain_function (filter->sinkpad, gst_rtp_base_depayload_chain);
gst_pad_set_chain_list_function (filter->sinkpad,
gst_rtp_base_depayload_chain_list);
gst_pad_set_event_function (filter->sinkpad,
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->clock_base = -1;
priv->dts = -1;
priv->pts = -1;
priv->duration = -1;
gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
}
static void
gst_rtp_base_depayload_finalize (GObject * object)
{
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtp_base_depayload_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps)
{
GstRTPBaseDepayloadClass *bclass;
GstRTPBaseDepayloadPrivate *priv;
gboolean res;
GstStructure *caps_struct;
const GValue *value;
priv = filter->priv;
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;
value = gst_structure_get_value (caps_struct, "clock-base");
if (value && G_VALUE_HOLDS_UINT (value))
priv->clock_base = g_value_get_uint (value);
else
priv->clock_base = -1;
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
gst_rtp_base_depayload_handle_buffer (GstRTPBaseDepayload * filter,
GstRTPBaseDepayloadClass * bclass, GstBuffer * in)
{
GstRTPBaseDepayloadPrivate *priv;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
GstClockTime pts, dts;
guint16 seqnum;
guint32 rtptime;
gboolean discont, buf_discont;
gint gap;
GstRTPBuffer rtp = { NULL };
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);
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);
}
}
/* prepare segment event if needed */
if (filter->need_newsegment) {
priv->segment_event = create_segment_event (filter, rtptime,
GST_BUFFER_PTS (in));
filter->need_newsegment = FALSE;
}
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) {
ret = gst_rtp_base_depayload_push (filter, out_buf);
}
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"));
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"));
return GST_FLOW_OK;
}
dropping:
{
GST_WARNING_OBJECT (filter, "%d <= 100, dropping old packet", gap);
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"));
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_rtp_base_depayload_chain (GstPad * pad, GstObject * parent, GstBuffer * in)
{
GstRTPBaseDepayloadClass *bclass;
GstRTPBaseDepayload *basedepay;
GstFlowReturn flow_ret;
basedepay = GST_RTP_BASE_DEPAYLOAD_CAST (parent);
bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (basedepay);
flow_ret = gst_rtp_base_depayload_handle_buffer (basedepay, bclass, in);
gst_buffer_unref (in);
return flow_ret;
}
static GstFlowReturn
gst_rtp_base_depayload_chain_list (GstPad * pad, GstObject * parent,
GstBufferList * list)
{
GstRTPBaseDepayloadClass *bclass;
GstRTPBaseDepayload *basedepay;
GstFlowReturn flow_ret;
GstBuffer *buffer;
guint i, len;
basedepay = GST_RTP_BASE_DEPAYLOAD_CAST (parent);
bclass = GST_RTP_BASE_DEPAYLOAD_GET_CLASS (basedepay);
flow_ret = GST_FLOW_OK;
/* chain each buffer in list individually */
len = gst_buffer_list_length (list);
if (len == 0)
goto done;
for (i = 0; i < len; i++) {
buffer = gst_buffer_list_get (list, i);
/* Should we fix up any missing timestamps for list buffers here
* (e.g. set to first or previous timestamp in list) or just assume
* the's a jitterbuffer that will have done that for us? */
flow_ret = gst_rtp_base_depayload_handle_buffer (basedepay, bclass, buffer);
if (flow_ret != GST_FLOW_OK)
break;
}
done:
gst_buffer_list_unref (list);
return flow_ret;
}
static gboolean
gst_rtp_base_depayload_handle_event (GstRTPBaseDepayload * 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;
gst_event_replace (&filter->priv->segment_event, NULL);
break;
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
res = gst_rtp_base_depayload_setcaps (filter, caps);
forward = FALSE;
break;
}
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 */
forward = FALSE;
break;
}
case GST_EVENT_CUSTOM_DOWNSTREAM:
{
GstRTPBaseDepayloadClass *bclass;
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);
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_rtp_base_depayload_handle_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
gboolean res = FALSE;
GstRTPBaseDepayload *filter;
GstRTPBaseDepayloadClass *bclass;
filter = GST_RTP_BASE_DEPAYLOAD (parent);
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, guint rtptime,
GstClockTime position)
{
GstEvent *event;
GstClockTime start, stop, running_time;
GstRTPBaseDepayloadPrivate *priv;
GstSegment segment;
priv = filter->priv;
/* determining the start of the segment */
start = 0;
if (priv->clock_base != -1 && position != -1) {
GstClockTime exttime, gap;
exttime = priv->clock_base;
gst_rtp_buffer_ext_timestamp (&exttime, rtptime);
gap = gst_util_uint64_scale_int (exttime - priv->clock_base,
filter->clock_rate, GST_SECOND);
/* account for lost packets */
if (position > gap) {
GST_DEBUG_OBJECT (filter,
"Found gap of %" GST_TIME_FORMAT ", adjusting start: %"
GST_TIME_FORMAT " = %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
GST_TIME_ARGS (gap), GST_TIME_ARGS (position - gap),
GST_TIME_ARGS (position), GST_TIME_ARGS (gap));
start = position - gap;
}
}
/* determining the stop of the segment */
stop = -1;
if (priv->npt_stop != -1)
stop = start + (priv->npt_stop - priv->npt_start);
if (position == -1)
position = 0;
running_time = gst_segment_to_running_time (&filter->segment,
GST_FORMAT_TIME, start);
gst_segment_init (&segment, GST_FORMAT_TIME);
segment.rate = priv->play_speed;
segment.applied_rate = priv->play_scale;
segment.start = start;
segment.stop = stop;
segment.time = priv->npt_start;
segment.position = position;
segment.base = running_time;
GST_DEBUG_OBJECT (filter, "Creating segment event %" GST_SEGMENT_FORMAT,
&segment);
event = gst_event_new_segment (&segment);
return event;
}
typedef struct
{
GstRTPBaseDepayload *depayload;
GstRTPBaseDepayloadClass *bclass;
} HeaderData;
static gboolean
set_headers (GstBuffer ** buffer, guint idx, HeaderData * data)
{
GstRTPBaseDepayload *depayload = data->depayload;
GstRTPBaseDepayloadPrivate *priv = depayload->priv;
GstClockTime pts, dts, duration;
*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;
return TRUE;
}
static GstFlowReturn
gst_rtp_base_depayload_prepare_push (GstRTPBaseDepayload * filter,
gboolean is_list, gpointer obj)
{
HeaderData data;
data.depayload = filter;
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;
set_headers (buf, 0, &data);
}
/* if this is the first buffer send a NEWSEGMENT */
if (G_UNLIKELY (filter->priv->segment_event)) {
gst_pad_push_event (filter->srcpad, filter->priv->segment_event);
filter->priv->segment_event = NULL;
GST_DEBUG_OBJECT (filter, "Pushed newsegment event on this first buffer");
}
return GST_FLOW_OK;
}
/**
* 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
gst_rtp_base_depayload_push (GstRTPBaseDepayload * filter, GstBuffer * out_buf)
{
GstFlowReturn res;
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;
}
/**
* 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
gst_rtp_base_depayload_push_list (GstRTPBaseDepayload * filter,
GstBufferList * out_list)
{
GstFlowReturn res;
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
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
gst_rtp_base_depayload_change_state (GstElement * element,
GstStateChange transition)
{
GstRTPBaseDepayload *filter;
GstRTPBaseDepayloadPrivate *priv;
GstStateChangeReturn ret;
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->clock_base = -1;
priv->next_seqnum = -1;
priv->negotiated = FALSE;
priv->discont = FALSE;
gst_event_replace (&filter->priv->segment_event, NULL);
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
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
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;
}
}