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gstreamer/gst-libs/gst/rtp/gstrtpbasepayload.c
Matthew Waters a1e9f4e37b rtpbasepayload: place twcc-ext-id behind environment variable 
Adding properties for each and every rtp header extension is not
scalable and a new interface will be implemented for the general case
(https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/issues/777).

Set the environment variable "GST_RTP_ENABLE_EXPERIMENTAL_TWCC_PROPERTY"
to any value to reenable the short-lived twcc-ext-id property.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/761

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/756>
2020-07-21 11:57:55 +00:00

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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)
static gboolean enable_experimental_twcc = FALSE;
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;
guint8 twcc_ext_id;
guint64 base_offset;
gint64 base_rtime;
guint64 base_rtime_hz;
guint64 running_time;
gboolean scale_rtptime;
gint64 prop_max_ptime;
gint64 caps_max_ptime;
gboolean onvif_no_rate_control;
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
#define DEFAULT_ONVIF_NO_RATE_CONTROL FALSE
#define DEFAULT_TWCC_EXT_ID 0
#define DEFAULT_SCALE_RTPTIME TRUE
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_ONVIF_NO_RATE_CONTROL,
PROP_TWCC_EXT_ID,
PROP_SCALE_RTPTIME,
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 (g_getenv ("GST_RTP_ENABLE_EXPERIMENTAL_TWCC_PROPERTY"))
enable_experimental_twcc = TRUE;
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));
/**
* GstRTPBasePayload:onvif-no-rate-control:
*
* Make the payloader timestamp packets according to the Rate-Control=no
* behaviour specified in the ONVIF replay spec.
*
* Since: 1.16
*/
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_ONVIF_NO_RATE_CONTROL, g_param_spec_boolean ("onvif-no-rate-control",
"ONVIF no rate control",
"Enable ONVIF Rate-Control=no timestamping mode",
DEFAULT_ONVIF_NO_RATE_CONTROL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRTPBasePayload:twcc-ext-id:
*
* The RTP header-extension ID used for tagging buffers with Transport-Wide
* Congestion Control sequence-numbers.
*
* To use this across multiple bundled streams (transport wide), the
* GstRTPFunnel can mux TWCC sequence-numbers together.
*
* This is experimental and requires setting the
* 'GST_RTP_ENABLE_EXPERIMENTAL_TWCC_PROPERTY' environment variable as it is
* still a draft and not yet a standard. This property may also be removed
* in the future for 1.20.
*
* Since: 1.18
*/
if (enable_experimental_twcc) {
g_object_class_install_property (gobject_class, PROP_TWCC_EXT_ID,
g_param_spec_uint ("twcc-ext-id",
"Transport-wide Congestion Control Extension ID (experimental)",
"The RTP header-extension ID to use for tagging buffers with "
"Transport-wide Congestion Control sequencenumbers (0 = disable)",
0, 15, DEFAULT_TWCC_EXT_ID,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
/**
* GstRTPBasePayload:scale-rtptime:
*
* Make the RTP packets' timestamps be scaled with the segment's rate
* (corresponding to RTSP speed parameter). Disabling this property means
* the timestamps will not be affected by the set delivery speed (RTSP speed).
*
* Example: A server wants to allow streaming a recorded video in double
* speed but still have the timestamps correspond to the position in the
* video. This is achieved by the client setting RTSP Speed to 2 while the
* server has this property disabled.
*
* Since: 1.18
*/
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SCALE_RTPTIME,
g_param_spec_boolean ("scale-rtptime", "Scale RTP time",
"Whether the RTP timestamp should be scaled with the rate (speed)",
DEFAULT_SCALE_RTPTIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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->priv->onvif_no_rate_control = DEFAULT_ONVIF_NO_RATE_CONTROL;
rtpbasepayload->priv->scale_rtptime = DEFAULT_SCALE_RTPTIME;
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);
if (enable_experimental_twcc && payload->priv->twcc_ext_id > 0) {
/* TODO: put this as a separate utility-function for RTP extensions */
gchar *name = g_strdup_printf ("extmap-%u", payload->priv->twcc_ext_id);
gst_caps_set_simple (srccaps, name, G_TYPE_STRING,
"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01",
NULL);
g_free (name);
}
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;
guint8 twcc_ext_id;
} 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 void
_set_twcc_seq (GstRTPBuffer * rtp, guint16 seq, guint8 ext_id)
{
guint16 data;
if (ext_id == 0 || ext_id > 14)
return;
GST_WRITE_UINT16_BE (&data, seq);
gst_rtp_buffer_add_extension_onebyte_header (rtp, ext_id, &data, 2);
}
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);
if (enable_experimental_twcc)
_set_twcc_seq (&rtp, data->seqnum, data->twcc_ext_id);
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) 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,
(gpointer) 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;
data.twcc_ext_id = priv->twcc_ext_id;
/* 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 */
if (priv->onvif_no_rate_control || !priv->scale_rtptime)
rtime_ns = gst_segment_to_stream_time (&payload->segment,
GST_FORMAT_TIME, data.pts);
else
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);
/* sequence number has increased more if this was a buffer list */
payload->seqnum = data.seqnum - 1;
} 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 %s with size %"
G_GSIZE_FORMAT ", seq=%d, rtptime=%u, pts %" GST_TIME_FORMAT,
(is_list) ? "list" : "packet",
(is_list) ? gst_buffer_list_length (GST_BUFFER_LIST_CAST (obj)) :
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;
case PROP_ONVIF_NO_RATE_CONTROL:
priv->onvif_no_rate_control = g_value_get_boolean (value);
break;
case PROP_TWCC_EXT_ID:
priv->twcc_ext_id = g_value_get_uint (value);
break;
case PROP_SCALE_RTPTIME:
priv->scale_rtptime = 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;
case PROP_ONVIF_NO_RATE_CONTROL:
g_value_set_boolean (value, priv->onvif_no_rate_control);
break;
case PROP_TWCC_EXT_ID:
g_value_set_uint (value, priv->twcc_ext_id);
break;
case PROP_SCALE_RTPTIME:
g_value_set_boolean (value, priv->scale_rtptime);
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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