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gstreamer/gst-libs/gst/rtp/gstrtpbasepayload.c
Stian Selnes f766b85b96 rtpbasepayload: rtpbasedepayload: Add source-info property 
Add a source-info property that will read/write meta to the buffers
about RTP source information. The GstRTPSourceMeta can be used to
transport information about the origin of a buffer, e.g. the sources
that is included in a mixed audio buffer.

A new function gst_rtp_base_payload_allocate_output_buffer() is added
for payloaders to use to allocate the output RTP buffer with the correct
number of CSRCs according to the meta and fill it.

RTPSourceMeta does not make sense on RTP buffers since the information
is in the RTP header. So the payloader will strip the meta from the
output buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=761947
2018-10-10 14:38:01 -04:00

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/* GStreamer
* Copyright (C) <2005> Wim Taymans <wim@fluendo.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
*/
/**
* SECTION:gstrtpbasepayload
* @title: GstRTPBasePayload
* @short_description: Base class for RTP payloader
*
* Provides a base class for RTP payloaders
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpbasepayload.h"
#include "gstrtpmeta.h"
GST_DEBUG_CATEGORY_STATIC (rtpbasepayload_debug);
#define GST_CAT_DEFAULT (rtpbasepayload_debug)
struct _GstRTPBasePayloadPrivate
{
gboolean ts_offset_random;
gboolean seqnum_offset_random;
gboolean ssrc_random;
guint16 next_seqnum;
gboolean perfect_rtptime;
gint notified_first_timestamp;
gboolean pt_set;
gboolean source_info;
GstBuffer *input_meta_buffer;
guint64 base_offset;
gint64 base_rtime;
guint64 base_rtime_hz;
guint64 running_time;
gint64 prop_max_ptime;
gint64 caps_max_ptime;
gboolean negotiated;
gboolean delay_segment;
GstEvent *pending_segment;
GstCaps *subclass_srccaps;
GstCaps *sinkcaps;
};
/* RTPBasePayload signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
/* FIXME 0.11, a better default is the Ethernet MTU of
* 1500 - sizeof(headers) as pointed out by marcelm in IRC:
* So an Ethernet MTU of 1500, minus 60 for the max IP, minus 8 for UDP, gives
* 1432 bytes or so. And that should be adjusted downward further for other
* encapsulations like PPPoE, so 1400 at most.
*/
#define DEFAULT_MTU 1400
#define DEFAULT_PT 96
#define DEFAULT_SSRC -1
#define DEFAULT_TIMESTAMP_OFFSET -1
#define DEFAULT_SEQNUM_OFFSET -1
#define DEFAULT_MAX_PTIME -1
#define DEFAULT_MIN_PTIME 0
#define DEFAULT_PERFECT_RTPTIME TRUE
#define DEFAULT_PTIME_MULTIPLE 0
#define DEFAULT_RUNNING_TIME GST_CLOCK_TIME_NONE
#define DEFAULT_SOURCE_INFO FALSE
enum
{
PROP_0,
PROP_MTU,
PROP_PT,
PROP_SSRC,
PROP_TIMESTAMP_OFFSET,
PROP_SEQNUM_OFFSET,
PROP_MAX_PTIME,
PROP_MIN_PTIME,
PROP_TIMESTAMP,
PROP_SEQNUM,
PROP_PERFECT_RTPTIME,
PROP_PTIME_MULTIPLE,
PROP_STATS,
PROP_SOURCE_INFO,
PROP_LAST
};
static void gst_rtp_base_payload_class_init (GstRTPBasePayloadClass * klass);
static void gst_rtp_base_payload_init (GstRTPBasePayload * rtpbasepayload,
gpointer g_class);
static void gst_rtp_base_payload_finalize (GObject * object);
static GstCaps *gst_rtp_base_payload_getcaps_default (GstRTPBasePayload *
rtpbasepayload, GstPad * pad, GstCaps * filter);
static gboolean gst_rtp_base_payload_sink_event_default (GstRTPBasePayload *
rtpbasepayload, GstEvent * event);
static gboolean gst_rtp_base_payload_sink_event (GstPad * pad,
GstObject * parent, GstEvent * event);
static gboolean gst_rtp_base_payload_src_event_default (GstRTPBasePayload *
rtpbasepayload, GstEvent * event);
static gboolean gst_rtp_base_payload_src_event (GstPad * pad,
GstObject * parent, GstEvent * event);
static gboolean gst_rtp_base_payload_query_default (GstRTPBasePayload *
rtpbasepayload, GstPad * pad, GstQuery * query);
static gboolean gst_rtp_base_payload_query (GstPad * pad, GstObject * parent,
GstQuery * query);
static GstFlowReturn gst_rtp_base_payload_chain (GstPad * pad,
GstObject * parent, GstBuffer * buffer);
static void gst_rtp_base_payload_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_base_payload_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstStateChangeReturn gst_rtp_base_payload_change_state (GstElement *
element, GstStateChange transition);
static gboolean gst_rtp_base_payload_negotiate (GstRTPBasePayload * payload);
static GstElementClass *parent_class = NULL;
static gint private_offset = 0;
GType
gst_rtp_base_payload_get_type (void)
{
static GType rtpbasepayload_type = 0;
if (g_once_init_enter ((gsize *) & rtpbasepayload_type)) {
static const GTypeInfo rtpbasepayload_info = {
sizeof (GstRTPBasePayloadClass),
NULL,
NULL,
(GClassInitFunc) gst_rtp_base_payload_class_init,
NULL,
NULL,
sizeof (GstRTPBasePayload),
0,
(GInstanceInitFunc) gst_rtp_base_payload_init,
};
GType _type;
_type = g_type_register_static (GST_TYPE_ELEMENT, "GstRTPBasePayload",
&rtpbasepayload_info, G_TYPE_FLAG_ABSTRACT);
private_offset =
g_type_add_instance_private (_type, sizeof (GstRTPBasePayloadPrivate));
g_once_init_leave ((gsize *) & rtpbasepayload_type, _type);
}
return rtpbasepayload_type;
}
static inline GstRTPBasePayloadPrivate *
gst_rtp_base_payload_get_instance_private (GstRTPBasePayload * self)
{
return (G_STRUCT_MEMBER_P (self, private_offset));
}
static void
gst_rtp_base_payload_class_init (GstRTPBasePayloadClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
if (private_offset != 0)
g_type_class_adjust_private_offset (klass, &private_offset);
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_rtp_base_payload_finalize;
gobject_class->set_property = gst_rtp_base_payload_set_property;
gobject_class->get_property = gst_rtp_base_payload_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MTU,
g_param_spec_uint ("mtu", "MTU",
"Maximum size of one packet",
28, G_MAXUINT, DEFAULT_MTU,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PT,
g_param_spec_uint ("pt", "payload type",
"The payload type of the packets", 0, 0x7f, DEFAULT_PT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SSRC,
g_param_spec_uint ("ssrc", "SSRC",
"The SSRC of the packets (default == random)", 0, G_MAXUINT32,
DEFAULT_SSRC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_TIMESTAMP_OFFSET, g_param_spec_uint ("timestamp-offset",
"Timestamp Offset",
"Offset to add to all outgoing timestamps (default = random)", 0,
G_MAXUINT32, DEFAULT_TIMESTAMP_OFFSET,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM_OFFSET,
g_param_spec_int ("seqnum-offset", "Sequence number Offset",
"Offset to add to all outgoing seqnum (-1 = random)", -1, G_MAXUINT16,
DEFAULT_SEQNUM_OFFSET, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MAX_PTIME,
g_param_spec_int64 ("max-ptime", "Max packet time",
"Maximum duration of the packet data in ns (-1 = unlimited up to MTU)",
-1, G_MAXINT64, DEFAULT_MAX_PTIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTPBasePayload:min-ptime:
*
* Minimum duration of the packet data in ns (can't go above MTU)
**/
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MIN_PTIME,
g_param_spec_int64 ("min-ptime", "Min packet time",
"Minimum duration of the packet data in ns (can't go above MTU)",
0, G_MAXINT64, DEFAULT_MIN_PTIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_TIMESTAMP,
g_param_spec_uint ("timestamp", "Timestamp",
"The RTP timestamp of the last processed packet",
0, G_MAXUINT32, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM,
g_param_spec_uint ("seqnum", "Sequence number",
"The RTP sequence number of the last processed packet",
0, G_MAXUINT16, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
/**
* GstRTPBasePayload:perfect-rtptime:
*
* Try to use the offset fields to generate perfect RTP timestamps. When this
* option is disabled, RTP timestamps are generated from GST_BUFFER_PTS of
* each payloaded buffer. The PTSes of buffers may not necessarily increment
* with the amount of data in each input buffer, consider e.g. the case where
* the buffer arrives from a network which means that the PTS is unrelated to
* the amount of data. Because the RTP timestamps are generated from
* GST_BUFFER_PTS this can result in RTP timestamps that also don't increment
* with the amount of data in the payloaded packet. To circumvent this it is
* possible to set the perfect rtptime option enabled. When this option is
* enabled the payloader will increment the RTP timestamps based on
* GST_BUFFER_OFFSET which relates to the amount of data in each packet
* rather than the GST_BUFFER_PTS of each buffer and therefore the RTP
* timestamps will more closely correlate with the amount of data in each
* buffer. Currently GstRTPBasePayload is limited to handling perfect RTP
* timestamps for audio streams.
*/
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PERFECT_RTPTIME,
g_param_spec_boolean ("perfect-rtptime", "Perfect RTP Time",
"Generate perfect RTP timestamps when possible",
DEFAULT_PERFECT_RTPTIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTPBasePayload:ptime-multiple:
*
* Force buffers to be multiples of this duration in ns (0 disables)
**/
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PTIME_MULTIPLE,
g_param_spec_int64 ("ptime-multiple", "Packet time multiple",
"Force buffers to be multiples of this duration in ns (0 disables)",
0, G_MAXINT64, DEFAULT_PTIME_MULTIPLE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTPBasePayload:stats:
*
* Various payloader statistics retrieved atomically (and are therefore
* synchroized with each other), these can be used e.g. to generate an
* RTP-Info header. This property return a GstStructure named
* application/x-rtp-payload-stats containing the following fields relating to
* the last processed buffer and current state of the stream being payloaded:
*
* * `clock-rate` :#G_TYPE_UINT, clock-rate of the stream
* * `running-time` :#G_TYPE_UINT64, running time
* * `seqnum` :#G_TYPE_UINT, sequence number, same as #GstRTPBasePayload:seqnum
* * `timestamp` :#G_TYPE_UINT, RTP timestamp, same as #GstRTPBasePayload:timestamp
* * `ssrc` :#G_TYPE_UINT, The SSRC in use
* * `pt` :#G_TYPE_UINT, The Payload type in use, same as #GstRTPBasePayload:pt
* * `seqnum-offset` :#G_TYPE_UINT, The current offset added to the seqnum
* * `timestamp-offset` :#G_TYPE_UINT, The current offset added to the timestamp
**/
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));
/**
* GstRTPBasePayload:source-info:
*
* Enable writing the CSRC field in allocated RTP header based on RTP source
* information found in the input buffer's #GstRTPSourceMeta.
*
* Since: 1.16
**/
g_object_class_install_property (gobject_class, PROP_SOURCE_INFO,
g_param_spec_boolean ("source-info", "RTP source information",
"Write CSRC based on buffer meta RTP source information",
DEFAULT_SOURCE_INFO, G_PARAM_READWRITE));
gstelement_class->change_state = gst_rtp_base_payload_change_state;
klass->get_caps = gst_rtp_base_payload_getcaps_default;
klass->sink_event = gst_rtp_base_payload_sink_event_default;
klass->src_event = gst_rtp_base_payload_src_event_default;
klass->query = gst_rtp_base_payload_query_default;
GST_DEBUG_CATEGORY_INIT (rtpbasepayload_debug, "rtpbasepayload", 0,
"Base class for RTP Payloaders");
}
static void
gst_rtp_base_payload_init (GstRTPBasePayload * rtpbasepayload, gpointer g_class)
{
GstPadTemplate *templ;
GstRTPBasePayloadPrivate *priv;
rtpbasepayload->priv = priv =
gst_rtp_base_payload_get_instance_private (rtpbasepayload);
templ =
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "src");
g_return_if_fail (templ != NULL);
rtpbasepayload->srcpad = gst_pad_new_from_template (templ, "src");
gst_pad_set_event_function (rtpbasepayload->srcpad,
gst_rtp_base_payload_src_event);
gst_element_add_pad (GST_ELEMENT (rtpbasepayload), rtpbasepayload->srcpad);
templ =
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "sink");
g_return_if_fail (templ != NULL);
rtpbasepayload->sinkpad = gst_pad_new_from_template (templ, "sink");
gst_pad_set_chain_function (rtpbasepayload->sinkpad,
gst_rtp_base_payload_chain);
gst_pad_set_event_function (rtpbasepayload->sinkpad,
gst_rtp_base_payload_sink_event);
gst_pad_set_query_function (rtpbasepayload->sinkpad,
gst_rtp_base_payload_query);
gst_element_add_pad (GST_ELEMENT (rtpbasepayload), rtpbasepayload->sinkpad);
rtpbasepayload->mtu = DEFAULT_MTU;
rtpbasepayload->pt = DEFAULT_PT;
rtpbasepayload->seqnum_offset = DEFAULT_SEQNUM_OFFSET;
rtpbasepayload->ssrc = DEFAULT_SSRC;
rtpbasepayload->ts_offset = DEFAULT_TIMESTAMP_OFFSET;
priv->running_time = DEFAULT_RUNNING_TIME;
priv->seqnum_offset_random = (rtpbasepayload->seqnum_offset == -1);
priv->ts_offset_random = (rtpbasepayload->ts_offset == -1);
priv->ssrc_random = (rtpbasepayload->ssrc == -1);
priv->pt_set = FALSE;
priv->source_info = DEFAULT_SOURCE_INFO;
rtpbasepayload->max_ptime = DEFAULT_MAX_PTIME;
rtpbasepayload->min_ptime = DEFAULT_MIN_PTIME;
rtpbasepayload->priv->perfect_rtptime = DEFAULT_PERFECT_RTPTIME;
rtpbasepayload->ptime_multiple = DEFAULT_PTIME_MULTIPLE;
rtpbasepayload->priv->base_offset = GST_BUFFER_OFFSET_NONE;
rtpbasepayload->priv->base_rtime_hz = GST_BUFFER_OFFSET_NONE;
rtpbasepayload->media = NULL;
rtpbasepayload->encoding_name = NULL;
rtpbasepayload->clock_rate = 0;
rtpbasepayload->priv->caps_max_ptime = DEFAULT_MAX_PTIME;
rtpbasepayload->priv->prop_max_ptime = DEFAULT_MAX_PTIME;
}
static void
gst_rtp_base_payload_finalize (GObject * object)
{
GstRTPBasePayload *rtpbasepayload;
rtpbasepayload = GST_RTP_BASE_PAYLOAD (object);
g_free (rtpbasepayload->media);
rtpbasepayload->media = NULL;
g_free (rtpbasepayload->encoding_name);
rtpbasepayload->encoding_name = NULL;
gst_caps_replace (&rtpbasepayload->priv->subclass_srccaps, NULL);
gst_caps_replace (&rtpbasepayload->priv->sinkcaps, NULL);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstCaps *
gst_rtp_base_payload_getcaps_default (GstRTPBasePayload * rtpbasepayload,
GstPad * pad, GstCaps * filter)
{
GstCaps *caps;
caps = GST_PAD_TEMPLATE_CAPS (GST_PAD_PAD_TEMPLATE (pad));
GST_DEBUG_OBJECT (pad,
"using pad template %p with caps %p %" GST_PTR_FORMAT,
GST_PAD_PAD_TEMPLATE (pad), caps, caps);
if (filter)
caps = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
else
caps = gst_caps_ref (caps);
return caps;
}
static gboolean
gst_rtp_base_payload_sink_event_default (GstRTPBasePayload * rtpbasepayload,
GstEvent * event)
{
GstObject *parent = GST_OBJECT_CAST (rtpbasepayload);
gboolean res = FALSE;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
res = gst_pad_event_default (rtpbasepayload->sinkpad, parent, event);
break;
case GST_EVENT_FLUSH_STOP:
res = gst_pad_event_default (rtpbasepayload->sinkpad, parent, event);
gst_segment_init (&rtpbasepayload->segment, GST_FORMAT_UNDEFINED);
gst_event_replace (&rtpbasepayload->priv->pending_segment, NULL);
break;
case GST_EVENT_CAPS:
{
GstRTPBasePayloadClass *rtpbasepayload_class;
GstCaps *caps;
gst_event_parse_caps (event, &caps);
GST_DEBUG_OBJECT (rtpbasepayload, "setting caps %" GST_PTR_FORMAT, caps);
gst_caps_replace (&rtpbasepayload->priv->sinkcaps, caps);
rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
if (rtpbasepayload_class->set_caps)
res = rtpbasepayload_class->set_caps (rtpbasepayload, caps);
else
res = gst_rtp_base_payload_negotiate (rtpbasepayload);
rtpbasepayload->priv->negotiated = res;
gst_event_unref (event);
break;
}
case GST_EVENT_SEGMENT:
{
GstSegment *segment;
segment = &rtpbasepayload->segment;
gst_event_copy_segment (event, segment);
rtpbasepayload->priv->base_offset = GST_BUFFER_OFFSET_NONE;
GST_DEBUG_OBJECT (rtpbasepayload,
"configured SEGMENT %" GST_SEGMENT_FORMAT, segment);
if (rtpbasepayload->priv->delay_segment) {
gst_event_replace (&rtpbasepayload->priv->pending_segment, event);
gst_event_unref (event);
res = TRUE;
} else {
res = gst_pad_event_default (rtpbasepayload->sinkpad, parent, event);
}
break;
}
default:
res = gst_pad_event_default (rtpbasepayload->sinkpad, parent, event);
break;
}
return res;
}
static gboolean
gst_rtp_base_payload_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstRTPBasePayload *rtpbasepayload;
GstRTPBasePayloadClass *rtpbasepayload_class;
gboolean res = FALSE;
rtpbasepayload = GST_RTP_BASE_PAYLOAD (parent);
rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
if (rtpbasepayload_class->sink_event)
res = rtpbasepayload_class->sink_event (rtpbasepayload, event);
else
gst_event_unref (event);
return res;
}
static gboolean
gst_rtp_base_payload_src_event_default (GstRTPBasePayload * rtpbasepayload,
GstEvent * event)
{
GstObject *parent = GST_OBJECT_CAST (rtpbasepayload);
gboolean res = TRUE, forward = TRUE;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CUSTOM_UPSTREAM:
{
const GstStructure *s = gst_event_get_structure (event);
if (gst_structure_has_name (s, "GstRTPCollision")) {
guint ssrc = 0;
if (!gst_structure_get_uint (s, "ssrc", &ssrc))
ssrc = -1;
GST_DEBUG_OBJECT (rtpbasepayload, "collided ssrc: %" G_GUINT32_FORMAT,
ssrc);
/* choose another ssrc for our stream */
if (ssrc == rtpbasepayload->current_ssrc) {
GstCaps *caps;
guint suggested_ssrc = 0;
if (gst_structure_get_uint (s, "suggested-ssrc", &suggested_ssrc))
rtpbasepayload->current_ssrc = suggested_ssrc;
while (ssrc == rtpbasepayload->current_ssrc)
rtpbasepayload->current_ssrc = g_random_int ();
caps = gst_pad_get_current_caps (rtpbasepayload->srcpad);
if (caps) {
caps = gst_caps_make_writable (caps);
gst_caps_set_simple (caps,
"ssrc", G_TYPE_UINT, rtpbasepayload->current_ssrc, NULL);
res = gst_pad_set_caps (rtpbasepayload->srcpad, caps);
gst_caps_unref (caps);
}
/* the event was for us */
forward = FALSE;
}
}
break;
}
default:
break;
}
if (forward)
res = gst_pad_event_default (rtpbasepayload->srcpad, parent, event);
else
gst_event_unref (event);
return res;
}
static gboolean
gst_rtp_base_payload_src_event (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstRTPBasePayload *rtpbasepayload;
GstRTPBasePayloadClass *rtpbasepayload_class;
gboolean res = FALSE;
rtpbasepayload = GST_RTP_BASE_PAYLOAD (parent);
rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
if (rtpbasepayload_class->src_event)
res = rtpbasepayload_class->src_event (rtpbasepayload, event);
else
gst_event_unref (event);
return res;
}
static gboolean
gst_rtp_base_payload_query_default (GstRTPBasePayload * rtpbasepayload,
GstPad * pad, GstQuery * query)
{
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CAPS:
{
GstRTPBasePayloadClass *rtpbasepayload_class;
GstCaps *filter, *caps;
gst_query_parse_caps (query, &filter);
GST_DEBUG_OBJECT (rtpbasepayload, "getting caps with filter %"
GST_PTR_FORMAT, filter);
rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
if (rtpbasepayload_class->get_caps) {
caps = rtpbasepayload_class->get_caps (rtpbasepayload, pad, filter);
gst_query_set_caps_result (query, caps);
gst_caps_unref (caps);
res = TRUE;
}
break;
}
default:
res =
gst_pad_query_default (pad, GST_OBJECT_CAST (rtpbasepayload), query);
break;
}
return res;
}
static gboolean
gst_rtp_base_payload_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
GstRTPBasePayload *rtpbasepayload;
GstRTPBasePayloadClass *rtpbasepayload_class;
gboolean res = FALSE;
rtpbasepayload = GST_RTP_BASE_PAYLOAD (parent);
rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
if (rtpbasepayload_class->query)
res = rtpbasepayload_class->query (rtpbasepayload, pad, query);
return res;
}
static GstFlowReturn
gst_rtp_base_payload_chain (GstPad * pad, GstObject * parent,
GstBuffer * buffer)
{
GstRTPBasePayload *rtpbasepayload;
GstRTPBasePayloadClass *rtpbasepayload_class;
GstFlowReturn ret;
rtpbasepayload = GST_RTP_BASE_PAYLOAD (parent);
rtpbasepayload_class = GST_RTP_BASE_PAYLOAD_GET_CLASS (rtpbasepayload);
if (!rtpbasepayload_class->handle_buffer)
goto no_function;
if (!rtpbasepayload->priv->negotiated)
goto not_negotiated;
if (rtpbasepayload->priv->source_info) {
/* Save a copy of meta (instead of taking an extra reference before
* handle_buffer) to make the meta available when allocating a output
* buffer. */
rtpbasepayload->priv->input_meta_buffer = gst_buffer_new ();
gst_buffer_copy_into (rtpbasepayload->priv->input_meta_buffer, buffer,
GST_BUFFER_COPY_META, 0, -1);
}
if (gst_pad_check_reconfigure (GST_RTP_BASE_PAYLOAD_SRCPAD (rtpbasepayload))) {
if (!gst_rtp_base_payload_negotiate (rtpbasepayload)) {
gst_pad_mark_reconfigure (GST_RTP_BASE_PAYLOAD_SRCPAD (rtpbasepayload));
if (GST_PAD_IS_FLUSHING (GST_RTP_BASE_PAYLOAD_SRCPAD (rtpbasepayload))) {
goto flushing;
} else {
goto negotiate_failed;
}
}
}
ret = rtpbasepayload_class->handle_buffer (rtpbasepayload, buffer);
gst_buffer_replace (&rtpbasepayload->priv->input_meta_buffer, NULL);
return ret;
/* ERRORS */
no_function:
{
GST_ELEMENT_ERROR (rtpbasepayload, STREAM, NOT_IMPLEMENTED, (NULL),
("subclass did not implement handle_buffer function"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
not_negotiated:
{
GST_ELEMENT_ERROR (rtpbasepayload, CORE, NEGOTIATION, (NULL),
("No input format was negotiated, i.e. no caps event was received. "
"Perhaps you need a parser or typefind element before the payloader"));
gst_buffer_unref (buffer);
return GST_FLOW_NOT_NEGOTIATED;
}
negotiate_failed:
{
GST_DEBUG_OBJECT (rtpbasepayload, "Not negotiated");
gst_buffer_unref (buffer);
return GST_FLOW_NOT_NEGOTIATED;
}
flushing:
{
GST_DEBUG_OBJECT (rtpbasepayload, "we are flushing");
gst_buffer_unref (buffer);
return GST_FLOW_FLUSHING;
}
}
/**
* gst_rtp_base_payload_set_options:
* @payload: a #GstRTPBasePayload
* @media: the media type (typically "audio" or "video")
* @dynamic: if the payload type is dynamic
* @encoding_name: the encoding name
* @clock_rate: the clock rate of the media
*
* Set the rtp options of the payloader. These options will be set in the caps
* of the payloader. Subclasses must call this method before calling
* gst_rtp_base_payload_push() or gst_rtp_base_payload_set_outcaps().
*/
void
gst_rtp_base_payload_set_options (GstRTPBasePayload * payload,
const gchar * media, gboolean dynamic, const gchar * encoding_name,
guint32 clock_rate)
{
g_return_if_fail (payload != NULL);
g_return_if_fail (clock_rate != 0);
g_free (payload->media);
payload->media = g_strdup (media);
payload->dynamic = dynamic;
g_free (payload->encoding_name);
payload->encoding_name = g_strdup (encoding_name);
payload->clock_rate = clock_rate;
}
static gboolean
copy_fixed (GQuark field_id, const GValue * value, GstStructure * dest)
{
if (gst_value_is_fixed (value)) {
gst_structure_id_set_value (dest, field_id, value);
}
return TRUE;
}
static void
update_max_ptime (GstRTPBasePayload * rtpbasepayload)
{
if (rtpbasepayload->priv->caps_max_ptime != -1 &&
rtpbasepayload->priv->prop_max_ptime != -1)
rtpbasepayload->max_ptime = MIN (rtpbasepayload->priv->caps_max_ptime,
rtpbasepayload->priv->prop_max_ptime);
else if (rtpbasepayload->priv->caps_max_ptime != -1)
rtpbasepayload->max_ptime = rtpbasepayload->priv->caps_max_ptime;
else if (rtpbasepayload->priv->prop_max_ptime != -1)
rtpbasepayload->max_ptime = rtpbasepayload->priv->prop_max_ptime;
else
rtpbasepayload->max_ptime = DEFAULT_MAX_PTIME;
}
/**
* gst_rtp_base_payload_set_outcaps:
* @payload: a #GstRTPBasePayload
* @fieldname: the first field name or %NULL
* @...: field values
*
* Configure the output caps with the optional parameters.
*
* Variable arguments should be in the form field name, field type
* (as a GType), value(s). The last variable argument should be NULL.
*
* Returns: %TRUE if the caps could be set.
*/
gboolean
gst_rtp_base_payload_set_outcaps (GstRTPBasePayload * payload,
const gchar * fieldname, ...)
{
GstCaps *srccaps;
/* fill in the defaults, their properties cannot be negotiated. */
srccaps = gst_caps_new_simple ("application/x-rtp",
"media", G_TYPE_STRING, payload->media,
"clock-rate", G_TYPE_INT, payload->clock_rate,
"encoding-name", G_TYPE_STRING, payload->encoding_name, NULL);
GST_DEBUG_OBJECT (payload, "defaults: %" GST_PTR_FORMAT, srccaps);
if (fieldname) {
va_list varargs;
/* override with custom properties */
va_start (varargs, fieldname);
gst_caps_set_simple_valist (srccaps, fieldname, varargs);
va_end (varargs);
GST_DEBUG_OBJECT (payload, "custom added: %" GST_PTR_FORMAT, srccaps);
}
gst_caps_replace (&payload->priv->subclass_srccaps, srccaps);
gst_caps_unref (srccaps);
return gst_rtp_base_payload_negotiate (payload);
}
static gboolean
gst_rtp_base_payload_negotiate (GstRTPBasePayload * payload)
{
GstCaps *templ, *peercaps, *srccaps;
GstStructure *s, *d;
gboolean res;
payload->priv->caps_max_ptime = DEFAULT_MAX_PTIME;
payload->ptime = 0;
gst_pad_check_reconfigure (payload->srcpad);
templ = gst_pad_get_pad_template_caps (payload->srcpad);
if (payload->priv->subclass_srccaps) {
GstCaps *tmp = gst_caps_intersect (payload->priv->subclass_srccaps,
templ);
gst_caps_unref (templ);
templ = tmp;
}
peercaps = gst_pad_peer_query_caps (payload->srcpad, templ);
if (peercaps == NULL) {
/* no peer caps, just add the other properties */
srccaps = gst_caps_copy (templ);
gst_caps_set_simple (srccaps,
"payload", G_TYPE_INT, GST_RTP_BASE_PAYLOAD_PT (payload),
"ssrc", G_TYPE_UINT, payload->current_ssrc,
"timestamp-offset", G_TYPE_UINT, payload->ts_base,
"seqnum-offset", G_TYPE_UINT, payload->seqnum_base, NULL);
GST_DEBUG_OBJECT (payload, "no peer caps: %" GST_PTR_FORMAT, srccaps);
} else {
GstCaps *temp;
const GValue *value;
gboolean have_pt = FALSE;
gboolean have_ts_offset = FALSE;
gboolean have_seqnum_offset = FALSE;
guint max_ptime, ptime;
/* peer provides caps we can use to fixate. They are already intersected
* with our srccaps, just make them writable */
temp = gst_caps_make_writable (peercaps);
peercaps = NULL;
if (gst_caps_is_empty (temp)) {
gst_caps_unref (temp);
gst_caps_unref (templ);
res = FALSE;
goto out;
}
/* We prefer the pt, timestamp-offset, seqnum-offset from the
* property (if set), or any previously configured value over what
* downstream prefers. Only if downstream can't accept that, or the
* properties were not set, we fall back to choosing downstream's
* preferred value
*
* For ssrc we prefer any value downstream suggests, otherwise
* the property value or as a last resort a random value.
* This difference for ssrc is implemented for retaining backwards
* compatibility with changing rtpsession's internal-ssrc property.
*
* FIXME 2.0: All these properties should go away and be negotiated
* via caps only!
*/
/* try to use the previously set pt, or the one from the property */
if (payload->priv->pt_set || gst_pad_has_current_caps (payload->srcpad)) {
GstCaps *probe_caps = gst_caps_copy (templ);
GstCaps *intersection;
gst_caps_set_simple (probe_caps, "payload", G_TYPE_INT,
GST_RTP_BASE_PAYLOAD_PT (payload), NULL);
intersection = gst_caps_intersect (probe_caps, temp);
if (!gst_caps_is_empty (intersection)) {
GST_LOG_OBJECT (payload, "Using selected pt %d",
GST_RTP_BASE_PAYLOAD_PT (payload));
have_pt = TRUE;
gst_caps_unref (temp);
temp = intersection;
} else {
GST_WARNING_OBJECT (payload, "Can't use selected pt %d",
GST_RTP_BASE_PAYLOAD_PT (payload));
gst_caps_unref (intersection);
}
gst_caps_unref (probe_caps);
}
/* If we got no pt above, select one now */
if (!have_pt) {
gint pt;
/* get first structure */
s = gst_caps_get_structure (temp, 0);
if (gst_structure_get_int (s, "payload", &pt)) {
/* use peer pt */
GST_RTP_BASE_PAYLOAD_PT (payload) = pt;
GST_LOG_OBJECT (payload, "using peer pt %d", pt);
} else {
if (gst_structure_has_field (s, "payload")) {
/* can only fixate if there is a field */
gst_structure_fixate_field_nearest_int (s, "payload",
GST_RTP_BASE_PAYLOAD_PT (payload));
gst_structure_get_int (s, "payload", &pt);
GST_RTP_BASE_PAYLOAD_PT (payload) = pt;
GST_LOG_OBJECT (payload, "using peer pt %d", pt);
} else {
/* no pt field, use the internal pt */
pt = GST_RTP_BASE_PAYLOAD_PT (payload);
gst_structure_set (s, "payload", G_TYPE_INT, pt, NULL);
GST_LOG_OBJECT (payload, "using internal pt %d", pt);
}
}
s = NULL;
}
/* If we got no ssrc above, select one now */
/* get first structure */
s = gst_caps_get_structure (temp, 0);
if (gst_structure_has_field_typed (s, "ssrc", G_TYPE_UINT)) {
value = gst_structure_get_value (s, "ssrc");
payload->current_ssrc = g_value_get_uint (value);
GST_LOG_OBJECT (payload, "using peer ssrc %08x", payload->current_ssrc);
} else {
/* FIXME, fixate_nearest_uint would be even better but we
* don't support uint ranges so how likely is it that anybody
* uses a list of possible ssrcs */
gst_structure_set (s, "ssrc", G_TYPE_UINT, payload->current_ssrc, NULL);
GST_LOG_OBJECT (payload, "using internal ssrc %08x",
payload->current_ssrc);
}
s = NULL;
/* try to select the previously used timestamp-offset, or the one from the property */
if (!payload->priv->ts_offset_random
|| gst_pad_has_current_caps (payload->srcpad)) {
GstCaps *probe_caps = gst_caps_copy (templ);
GstCaps *intersection;
gst_caps_set_simple (probe_caps, "timestamp-offset", G_TYPE_UINT,
payload->ts_base, NULL);
intersection = gst_caps_intersect (probe_caps, temp);
if (!gst_caps_is_empty (intersection)) {
GST_LOG_OBJECT (payload, "Using selected timestamp-offset %u",
payload->ts_base);
gst_caps_unref (temp);
temp = intersection;
have_ts_offset = TRUE;
} else {
GST_WARNING_OBJECT (payload, "Can't use selected timestamp-offset %u",
payload->ts_base);
gst_caps_unref (intersection);
}
gst_caps_unref (probe_caps);
}
/* If we got no timestamp-offset above, select one now */
if (!have_ts_offset) {
/* get first structure */
s = gst_caps_get_structure (temp, 0);
if (gst_structure_has_field_typed (s, "timestamp-offset", G_TYPE_UINT)) {
value = gst_structure_get_value (s, "timestamp-offset");
payload->ts_base = g_value_get_uint (value);
GST_LOG_OBJECT (payload, "using peer timestamp-offset %u",
payload->ts_base);
} else {
/* FIXME, fixate_nearest_uint would be even better but we
* don't support uint ranges so how likely is it that anybody
* uses a list of possible timestamp-offsets */
gst_structure_set (s, "timestamp-offset", G_TYPE_UINT, payload->ts_base,
NULL);
GST_LOG_OBJECT (payload, "using internal timestamp-offset %u",
payload->ts_base);
}
s = NULL;
}
/* try to select the previously used seqnum-offset, or the one from the property */
if (!payload->priv->seqnum_offset_random
|| gst_pad_has_current_caps (payload->srcpad)) {
GstCaps *probe_caps = gst_caps_copy (templ);
GstCaps *intersection;
gst_caps_set_simple (probe_caps, "seqnum-offset", G_TYPE_UINT,
payload->seqnum_base, NULL);
intersection = gst_caps_intersect (probe_caps, temp);
if (!gst_caps_is_empty (intersection)) {
GST_LOG_OBJECT (payload, "Using selected seqnum-offset %u",
payload->seqnum_base);
gst_caps_unref (temp);
temp = intersection;
have_seqnum_offset = TRUE;
} else {
GST_WARNING_OBJECT (payload, "Can't use selected seqnum-offset %u",
payload->seqnum_base);
gst_caps_unref (intersection);
}
gst_caps_unref (probe_caps);
}
/* If we got no seqnum-offset above, select one now */
if (!have_seqnum_offset) {
/* get first structure */
s = gst_caps_get_structure (temp, 0);
if (gst_structure_has_field_typed (s, "seqnum-offset", G_TYPE_UINT)) {
value = gst_structure_get_value (s, "seqnum-offset");
payload->seqnum_base = g_value_get_uint (value);
GST_LOG_OBJECT (payload, "using peer seqnum-offset %u",
payload->seqnum_base);
payload->priv->next_seqnum = payload->seqnum_base;
payload->seqnum = payload->seqnum_base;
payload->priv->seqnum_offset_random = FALSE;
} else {
/* FIXME, fixate_nearest_uint would be even better but we
* don't support uint ranges so how likely is it that anybody
* uses a list of possible seqnum-offsets */
gst_structure_set (s, "seqnum-offset", G_TYPE_UINT,
payload->seqnum_base, NULL);
GST_LOG_OBJECT (payload, "using internal seqnum-offset %u",
payload->seqnum_base);
}
s = NULL;
}
/* now fixate, start by taking the first caps */
temp = gst_caps_truncate (temp);
/* get first structure */
s = gst_caps_get_structure (temp, 0);
if (gst_structure_get_uint (s, "maxptime", &max_ptime))
payload->priv->caps_max_ptime = max_ptime * GST_MSECOND;
if (gst_structure_get_uint (s, "ptime", &ptime))
payload->ptime = ptime * GST_MSECOND;
/* make the target caps by copying over all the fixed fields, removing the
* unfixed fields. */
srccaps = gst_caps_new_empty_simple (gst_structure_get_name (s));
d = gst_caps_get_structure (srccaps, 0);
gst_structure_foreach (s, (GstStructureForeachFunc) copy_fixed, d);
gst_caps_unref (temp);
GST_DEBUG_OBJECT (payload, "with peer caps: %" GST_PTR_FORMAT, srccaps);
}
if (payload->priv->sinkcaps != NULL) {
s = gst_caps_get_structure (payload->priv->sinkcaps, 0);
if (g_str_has_prefix (gst_structure_get_name (s), "video")) {
gboolean has_framerate;
gint num, denom;
GST_DEBUG_OBJECT (payload, "video caps: %" GST_PTR_FORMAT,
payload->priv->sinkcaps);
has_framerate = gst_structure_get_fraction (s, "framerate", &num, &denom);
if (has_framerate && num == 0 && denom == 1) {
has_framerate =
gst_structure_get_fraction (s, "max-framerate", &num, &denom);
}
if (has_framerate) {
gchar str[G_ASCII_DTOSTR_BUF_SIZE];
gdouble framerate;
gst_util_fraction_to_double (num, denom, &framerate);
g_ascii_dtostr (str, G_ASCII_DTOSTR_BUF_SIZE, framerate);
d = gst_caps_get_structure (srccaps, 0);
gst_structure_set (d, "a-framerate", G_TYPE_STRING, str, NULL);
}
GST_DEBUG_OBJECT (payload, "with video caps: %" GST_PTR_FORMAT, srccaps);
}
}
update_max_ptime (payload);
res = gst_pad_set_caps (GST_RTP_BASE_PAYLOAD_SRCPAD (payload), srccaps);
gst_caps_unref (srccaps);
gst_caps_unref (templ);
out:
if (!res)
gst_pad_mark_reconfigure (GST_RTP_BASE_PAYLOAD_SRCPAD (payload));
return res;
}
/**
* gst_rtp_base_payload_is_filled:
* @payload: a #GstRTPBasePayload
* @size: the size of the packet
* @duration: the duration of the packet
*
* Check if the packet with @size and @duration would exceed the configured
* maximum size.
*
* Returns: %TRUE if the packet of @size and @duration would exceed the
* configured MTU or max_ptime.
*/
gboolean
gst_rtp_base_payload_is_filled (GstRTPBasePayload * payload,
guint size, GstClockTime duration)
{
if (size > payload->mtu)
return TRUE;
if (payload->max_ptime != -1 && duration >= payload->max_ptime)
return TRUE;
return FALSE;
}
typedef struct
{
GstRTPBasePayload *payload;
guint32 ssrc;
guint16 seqnum;
guint8 pt;
GstClockTime dts;
GstClockTime pts;
guint64 offset;
guint32 rtptime;
} HeaderData;
static gboolean
find_timestamp (GstBuffer ** buffer, guint idx, gpointer user_data)
{
HeaderData *data = user_data;
data->dts = GST_BUFFER_DTS (*buffer);
data->pts = GST_BUFFER_PTS (*buffer);
data->offset = GST_BUFFER_OFFSET (*buffer);
/* stop when we find a timestamp. We take whatever offset is associated with
* the timestamp (if any) to do perfect timestamps when we need to. */
if (data->pts != -1)
return FALSE;
else
return TRUE;
}
static gboolean
set_headers (GstBuffer ** buffer, guint idx, gpointer user_data)
{
HeaderData *data = user_data;
GstRTPBuffer rtp = { NULL, };
if (!gst_rtp_buffer_map (*buffer, GST_MAP_WRITE, &rtp))
goto map_failed;
gst_rtp_buffer_set_ssrc (&rtp, data->ssrc);
gst_rtp_buffer_set_payload_type (&rtp, data->pt);
gst_rtp_buffer_set_seq (&rtp, data->seqnum);
gst_rtp_buffer_set_timestamp (&rtp, data->rtptime);
gst_rtp_buffer_unmap (&rtp);
/* increment the seqnum for each buffer */
data->seqnum++;
return TRUE;
/* ERRORS */
map_failed:
{
GST_ERROR ("failed to map buffer %p", *buffer);
return FALSE;
}
}
static gboolean
foreach_metadata_drop (GstBuffer * buffer, GstMeta ** meta, gpointer user_data)
{
GType drop_api_type = (GType) GPOINTER_TO_INT (user_data);
const GstMetaInfo *info = (*meta)->info;
if (info->api == drop_api_type)
*meta = NULL;
return TRUE;
}
static gboolean
filter_meta (GstBuffer ** buffer, guint idx, gpointer user_data)
{
return gst_buffer_foreach_meta (*buffer, foreach_metadata_drop,
GINT_TO_POINTER (GST_RTP_SOURCE_META_API_TYPE));
}
/* Updates the SSRC, payload type, seqnum and timestamp of the RTP buffer
* before the buffer is pushed. */
static GstFlowReturn
gst_rtp_base_payload_prepare_push (GstRTPBasePayload * payload,
gpointer obj, gboolean is_list)
{
GstRTPBasePayloadPrivate *priv;
HeaderData data;
if (payload->clock_rate == 0)
goto no_rate;
priv = payload->priv;
/* update first, so that the property is set to the last
* seqnum pushed */
payload->seqnum = priv->next_seqnum;
/* fill in the fields we want to set on all headers */
data.payload = payload;
data.seqnum = payload->seqnum;
data.ssrc = payload->current_ssrc;
data.pt = payload->pt;
/* find the first buffer with a timestamp */
if (is_list) {
data.dts = -1;
data.pts = -1;
data.offset = GST_BUFFER_OFFSET_NONE;
gst_buffer_list_foreach (GST_BUFFER_LIST_CAST (obj), find_timestamp, &data);
} else {
data.dts = GST_BUFFER_DTS (GST_BUFFER_CAST (obj));
data.pts = GST_BUFFER_PTS (GST_BUFFER_CAST (obj));
data.offset = GST_BUFFER_OFFSET (GST_BUFFER_CAST (obj));
}
/* convert to RTP time */
if (priv->perfect_rtptime && data.offset != GST_BUFFER_OFFSET_NONE &&
priv->base_offset != GST_BUFFER_OFFSET_NONE) {
/* generate perfect RTP time by adding together the base timestamp, the
* running time of the first buffer and difference between the offset of the
* first buffer and the offset of the current buffer. */
guint64 offset = data.offset - priv->base_offset;
data.rtptime = payload->ts_base + priv->base_rtime_hz + offset;
GST_LOG_OBJECT (payload,
"Using offset %" G_GUINT64_FORMAT " for RTP timestamp", data.offset);
/* store buffer's running time */
GST_LOG_OBJECT (payload,
"setting running-time to %" G_GUINT64_FORMAT,
data.offset - priv->base_offset);
priv->running_time = priv->base_rtime + data.offset - priv->base_offset;
} else if (GST_CLOCK_TIME_IS_VALID (data.pts)) {
guint64 rtime_ns;
guint64 rtime_hz;
/* no offset, use the gstreamer pts */
rtime_ns = gst_segment_to_running_time (&payload->segment, GST_FORMAT_TIME,
data.pts);
if (!GST_CLOCK_TIME_IS_VALID (rtime_ns)) {
GST_LOG_OBJECT (payload, "Clipped pts, using base RTP timestamp");
rtime_hz = 0;
} else {
GST_LOG_OBJECT (payload,
"Using running_time %" GST_TIME_FORMAT " for RTP timestamp",
GST_TIME_ARGS (rtime_ns));
rtime_hz =
gst_util_uint64_scale_int (rtime_ns, payload->clock_rate, GST_SECOND);
priv->base_offset = data.offset;
priv->base_rtime_hz = rtime_hz;
}
/* add running_time in clock-rate units to the base timestamp */
data.rtptime = payload->ts_base + rtime_hz;
/* store buffer's running time */
if (priv->perfect_rtptime) {
GST_LOG_OBJECT (payload,
"setting running-time to %" G_GUINT64_FORMAT, rtime_hz);
priv->running_time = rtime_hz;
} else {
GST_LOG_OBJECT (payload,
"setting running-time to %" GST_TIME_FORMAT,
GST_TIME_ARGS (rtime_ns));
priv->running_time = rtime_ns;
}
} else {
GST_LOG_OBJECT (payload,
"Using previous RTP timestamp %" G_GUINT32_FORMAT, payload->timestamp);
/* no timestamp to convert, take previous timestamp */
data.rtptime = payload->timestamp;
}
/* set ssrc, payload type, seq number, caps and rtptime */
/* remove unwanted meta */
if (is_list) {
gst_buffer_list_foreach (GST_BUFFER_LIST_CAST (obj), set_headers, &data);
gst_buffer_list_foreach (GST_BUFFER_LIST_CAST (obj), filter_meta, NULL);
} else {
GstBuffer *buf = GST_BUFFER_CAST (obj);
set_headers (&buf, 0, &data);
filter_meta (&buf, 0, NULL);
}
priv->next_seqnum = data.seqnum;
payload->timestamp = data.rtptime;
GST_LOG_OBJECT (payload, "Preparing to push packet with size %"
G_GSIZE_FORMAT ", seq=%d, rtptime=%u, pts %" GST_TIME_FORMAT,
(is_list) ? -1 : gst_buffer_get_size (GST_BUFFER (obj)),
payload->seqnum, data.rtptime, GST_TIME_ARGS (data.pts));
if (g_atomic_int_compare_and_exchange (&payload->priv->
notified_first_timestamp, 1, 0)) {
g_object_notify (G_OBJECT (payload), "timestamp");
g_object_notify (G_OBJECT (payload), "seqnum");
}
return GST_FLOW_OK;
/* ERRORS */
no_rate:
{
GST_ELEMENT_ERROR (payload, STREAM, NOT_IMPLEMENTED, (NULL),
("subclass did not specify clock-rate"));
return GST_FLOW_ERROR;
}
}
/**
* gst_rtp_base_payload_push_list:
* @payload: a #GstRTPBasePayload
* @list: a #GstBufferList
*
* Push @list to the peer element of the payloader. The SSRC, payload type,
* seqnum and timestamp of the RTP buffer will be updated first.
*
* This function takes ownership of @list.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
gst_rtp_base_payload_push_list (GstRTPBasePayload * payload,
GstBufferList * list)
{
GstFlowReturn res;
res = gst_rtp_base_payload_prepare_push (payload, list, TRUE);
if (G_LIKELY (res == GST_FLOW_OK)) {
if (G_UNLIKELY (payload->priv->pending_segment)) {
gst_pad_push_event (payload->srcpad, payload->priv->pending_segment);
payload->priv->pending_segment = FALSE;
payload->priv->delay_segment = FALSE;
}
res = gst_pad_push_list (payload->srcpad, list);
} else {
gst_buffer_list_unref (list);
}
return res;
}
/**
* gst_rtp_base_payload_push:
* @payload: a #GstRTPBasePayload
* @buffer: a #GstBuffer
*
* Push @buffer to the peer element of the payloader. The SSRC, payload type,
* seqnum and timestamp of the RTP buffer will be updated first.
*
* This function takes ownership of @buffer.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
gst_rtp_base_payload_push (GstRTPBasePayload * payload, GstBuffer * buffer)
{
GstFlowReturn res;
res = gst_rtp_base_payload_prepare_push (payload, buffer, FALSE);
if (G_LIKELY (res == GST_FLOW_OK)) {
if (G_UNLIKELY (payload->priv->pending_segment)) {
gst_pad_push_event (payload->srcpad, payload->priv->pending_segment);
payload->priv->pending_segment = FALSE;
payload->priv->delay_segment = FALSE;
}
res = gst_pad_push (payload->srcpad, buffer);
} else {
gst_buffer_unref (buffer);
}
return res;
}
/**
* gst_rtp_base_payload_allocate_output_buffer:
* @payload: a #GstRTPBasePayload
* @payload_len: the length of the payload
* @pad_len: the amount of padding
* @csrc_count: the minimum number of CSRC entries
*
* Allocate a new #GstBuffer with enough data to hold an RTP packet with
* minimum @csrc_count CSRCs, a payload length of @payload_len and padding of
* @pad_len. If @payload has #GstRTPBasePayload:source-info %TRUE additional
* CSRCs may be allocated and filled with RTP source information.
*
* Returns: A newly allocated buffer that can hold an RTP packet with given
* parameters.
*
* Since: 1.16
*/
GstBuffer *
gst_rtp_base_payload_allocate_output_buffer (GstRTPBasePayload * payload,
guint payload_len, guint8 pad_len, guint8 csrc_count)
{
GstBuffer *buffer = NULL;
if (payload->priv->input_meta_buffer != NULL) {
GstRTPSourceMeta *meta =
gst_buffer_get_rtp_source_meta (payload->priv->input_meta_buffer);
if (meta != NULL) {
guint total_csrc_count, idx, i;
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
total_csrc_count = csrc_count + meta->csrc_count +
(meta->ssrc_valid ? 1 : 0);
total_csrc_count = MIN (total_csrc_count, 15);
buffer = gst_rtp_buffer_new_allocate (payload_len, pad_len,
total_csrc_count);
gst_rtp_buffer_map (buffer, GST_MAP_READWRITE, &rtp);
/* Skip CSRC fields requested by derived class and fill CSRCs from meta.
* Finally append the SSRC as a new CSRC. */
idx = csrc_count;
for (i = 0; i < meta->csrc_count && idx < 15; i++, idx++)
gst_rtp_buffer_set_csrc (&rtp, idx, meta->csrc[i]);
if (meta->ssrc_valid && idx < 15)
gst_rtp_buffer_set_csrc (&rtp, idx, meta->ssrc);
gst_rtp_buffer_unmap (&rtp);
}
}
if (buffer == NULL)
buffer = gst_rtp_buffer_new_allocate (payload_len, pad_len, csrc_count);
return buffer;
}
static GstStructure *
gst_rtp_base_payload_create_stats (GstRTPBasePayload * rtpbasepayload)
{
GstRTPBasePayloadPrivate *priv;
GstStructure *s;
priv = rtpbasepayload->priv;
s = gst_structure_new ("application/x-rtp-payload-stats",
"clock-rate", G_TYPE_UINT, (guint) rtpbasepayload->clock_rate,
"running-time", G_TYPE_UINT64, priv->running_time,
"seqnum", G_TYPE_UINT, (guint) rtpbasepayload->seqnum,
"timestamp", G_TYPE_UINT, (guint) rtpbasepayload->timestamp,
"ssrc", G_TYPE_UINT, rtpbasepayload->current_ssrc,
"pt", G_TYPE_UINT, rtpbasepayload->pt,
"seqnum-offset", G_TYPE_UINT, (guint) rtpbasepayload->seqnum_base,
"timestamp-offset", G_TYPE_UINT, (guint) rtpbasepayload->ts_base, NULL);
return s;
}
static void
gst_rtp_base_payload_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRTPBasePayload *rtpbasepayload;
GstRTPBasePayloadPrivate *priv;
gint64 val;
rtpbasepayload = GST_RTP_BASE_PAYLOAD (object);
priv = rtpbasepayload->priv;
switch (prop_id) {
case PROP_MTU:
rtpbasepayload->mtu = g_value_get_uint (value);
break;
case PROP_PT:
rtpbasepayload->pt = g_value_get_uint (value);
priv->pt_set = TRUE;
break;
case PROP_SSRC:
val = g_value_get_uint (value);
rtpbasepayload->ssrc = val;
priv->ssrc_random = FALSE;
break;
case PROP_TIMESTAMP_OFFSET:
val = g_value_get_uint (value);
rtpbasepayload->ts_offset = val;
priv->ts_offset_random = FALSE;
break;
case PROP_SEQNUM_OFFSET:
val = g_value_get_int (value);
rtpbasepayload->seqnum_offset = val;
priv->seqnum_offset_random = (val == -1);
GST_DEBUG_OBJECT (rtpbasepayload, "seqnum offset 0x%04x, random %d",
rtpbasepayload->seqnum_offset, priv->seqnum_offset_random);
break;
case PROP_MAX_PTIME:
rtpbasepayload->priv->prop_max_ptime = g_value_get_int64 (value);
update_max_ptime (rtpbasepayload);
break;
case PROP_MIN_PTIME:
rtpbasepayload->min_ptime = g_value_get_int64 (value);
break;
case PROP_PERFECT_RTPTIME:
priv->perfect_rtptime = g_value_get_boolean (value);
break;
case PROP_PTIME_MULTIPLE:
rtpbasepayload->ptime_multiple = g_value_get_int64 (value);
break;
case PROP_SOURCE_INFO:
gst_rtp_base_payload_set_source_info_enabled (rtpbasepayload,
g_value_get_boolean (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_base_payload_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRTPBasePayload *rtpbasepayload;
GstRTPBasePayloadPrivate *priv;
rtpbasepayload = GST_RTP_BASE_PAYLOAD (object);
priv = rtpbasepayload->priv;
switch (prop_id) {
case PROP_MTU:
g_value_set_uint (value, rtpbasepayload->mtu);
break;
case PROP_PT:
g_value_set_uint (value, rtpbasepayload->pt);
break;
case PROP_SSRC:
if (priv->ssrc_random)
g_value_set_uint (value, -1);
else
g_value_set_uint (value, rtpbasepayload->ssrc);
break;
case PROP_TIMESTAMP_OFFSET:
if (priv->ts_offset_random)
g_value_set_uint (value, -1);
else
g_value_set_uint (value, (guint32) rtpbasepayload->ts_offset);
break;
case PROP_SEQNUM_OFFSET:
if (priv->seqnum_offset_random)
g_value_set_int (value, -1);
else
g_value_set_int (value, (guint16) rtpbasepayload->seqnum_offset);
break;
case PROP_MAX_PTIME:
g_value_set_int64 (value, rtpbasepayload->max_ptime);
break;
case PROP_MIN_PTIME:
g_value_set_int64 (value, rtpbasepayload->min_ptime);
break;
case PROP_TIMESTAMP:
g_value_set_uint (value, rtpbasepayload->timestamp);
break;
case PROP_SEQNUM:
g_value_set_uint (value, rtpbasepayload->seqnum);
break;
case PROP_PERFECT_RTPTIME:
g_value_set_boolean (value, priv->perfect_rtptime);
break;
case PROP_PTIME_MULTIPLE:
g_value_set_int64 (value, rtpbasepayload->ptime_multiple);
break;
case PROP_STATS:
g_value_take_boxed (value,
gst_rtp_base_payload_create_stats (rtpbasepayload));
break;
case PROP_SOURCE_INFO:
g_value_set_boolean (value,
gst_rtp_base_payload_is_source_info_enabled (rtpbasepayload));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_rtp_base_payload_change_state (GstElement * element,
GstStateChange transition)
{
GstRTPBasePayload *rtpbasepayload;
GstRTPBasePayloadPrivate *priv;
GstStateChangeReturn ret;
rtpbasepayload = GST_RTP_BASE_PAYLOAD (element);
priv = rtpbasepayload->priv;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_segment_init (&rtpbasepayload->segment, GST_FORMAT_UNDEFINED);
rtpbasepayload->priv->delay_segment = TRUE;
gst_event_replace (&rtpbasepayload->priv->pending_segment, NULL);
if (priv->seqnum_offset_random)
rtpbasepayload->seqnum_base = g_random_int_range (0, G_MAXINT16);
else
rtpbasepayload->seqnum_base = rtpbasepayload->seqnum_offset;
priv->next_seqnum = rtpbasepayload->seqnum_base;
rtpbasepayload->seqnum = rtpbasepayload->seqnum_base;
if (priv->ssrc_random)
rtpbasepayload->current_ssrc = g_random_int ();
else
rtpbasepayload->current_ssrc = rtpbasepayload->ssrc;
if (priv->ts_offset_random)
rtpbasepayload->ts_base = g_random_int ();
else
rtpbasepayload->ts_base = rtpbasepayload->ts_offset;
rtpbasepayload->timestamp = rtpbasepayload->ts_base;
priv->running_time = DEFAULT_RUNNING_TIME;
g_atomic_int_set (&rtpbasepayload->priv->notified_first_timestamp, 1);
priv->base_offset = GST_BUFFER_OFFSET_NONE;
priv->negotiated = FALSE;
gst_caps_replace (&rtpbasepayload->priv->subclass_srccaps, NULL);
gst_caps_replace (&rtpbasepayload->priv->sinkcaps, NULL);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
g_atomic_int_set (&rtpbasepayload->priv->notified_first_timestamp, 1);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_event_replace (&rtpbasepayload->priv->pending_segment, NULL);
break;
default:
break;
}
return ret;
}
/**
* gst_rtp_base_payload_set_source_info_enabled:
* @payload: a #GstRTPBasePayload
* @enable: whether to add contributing sources to RTP packets
*
* Enable or disable adding contributing sources to RTP packets from
* #GstRTPSourceMeta.
*
* Since: 1.16
**/
void
gst_rtp_base_payload_set_source_info_enabled (GstRTPBasePayload * payload,
gboolean enable)
{
payload->priv->source_info = enable;
}
/**
* gst_rtp_base_payload_is_source_info_enabled:
* @payload: a #GstRTPBasePayload
*
* Queries whether the payloader will add contributing sources (CSRCs) to the
* RTP header from #GstRTPSourceMeta.
*
* Returns: %TRUE if source-info is enabled.
*
* Since: 1.16
**/
gboolean
gst_rtp_base_payload_is_source_info_enabled (GstRTPBasePayload * payload)
{
return payload->priv->source_info;
}
/**
* gst_rtp_base_payload_get_source_count:
* @payload: a #GstRTPBasePayload
* @buffer: (transfer none): a #GstBuffer, typically the buffer to payload
*
* Count the total number of RTP sources found in the meta of @buffer, which
* will be automically added by gst_rtp_base_payload_allocate_output_buffer().
* If #GstRTPBasePayload:source-info is %FALSE the count will be 0.
*
* Returns: The number of sources.
*
* Since: 1.16
**/
guint
gst_rtp_base_payload_get_source_count (GstRTPBasePayload * payload,
GstBuffer * buffer)
{
guint count = 0;
g_return_val_if_fail (buffer != NULL, 0);
if (gst_rtp_base_payload_is_source_info_enabled (payload)) {
GstRTPSourceMeta *meta = gst_buffer_get_rtp_source_meta (buffer);
if (meta != NULL)
count = gst_rtp_source_meta_get_source_count (meta);
}
return count;
}
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