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#define | GST_RTP_BASE_PAYLOAD_MTU() |
#define | GST_RTP_BASE_PAYLOAD_PT() |
#define | GST_RTP_BASE_PAYLOAD_SINKPAD() |
#define | GST_RTP_BASE_PAYLOAD_SRCPAD() |
gboolean | gst_rtp_base_payload_is_filled () |
GstFlowReturn | gst_rtp_base_payload_push () |
GstFlowReturn | gst_rtp_base_payload_push_list () |
void | gst_rtp_base_payload_set_options () |
gboolean | gst_rtp_base_payload_set_outcaps () |
GstBuffer * | gst_rtp_base_payload_allocate_output_buffer () |
guint | gst_rtp_base_payload_get_source_count () |
gboolean | gst_rtp_base_payload_is_source_info_enabled () |
void | gst_rtp_base_payload_set_source_info_enabled () |
gint64 | max-ptime | Read / Write |
gint64 | min-ptime | Read / Write |
guint | mtu | Read / Write |
gboolean | onvif-no-rate-control | Read / Write |
gboolean | perfect-rtptime | Read / Write |
guint | pt | Read / Write |
gint64 | ptime-multiple | Read / Write |
guint | seqnum | Read |
gint | seqnum-offset | Read / Write |
gboolean | source-info | Read / Write |
guint | ssrc | Read / Write |
GstStructure * | stats | Read |
guint | timestamp | Read |
guint | timestamp-offset | Read / Write |
GObject ╰── GInitiallyUnowned ╰── GstObject ╰── GstElement ╰── GstRTPBasePayload ╰── GstRTPBaseAudioPayload
#define GST_RTP_BASE_PAYLOAD_MTU(payload) (GST_RTP_BASE_PAYLOAD (payload)->mtu)
Get access to the configured MTU of payload
.
#define GST_RTP_BASE_PAYLOAD_PT(payload) (GST_RTP_BASE_PAYLOAD (payload)->pt)
Get access to the configured payload type of payload
.
#define GST_RTP_BASE_PAYLOAD_SINKPAD(payload) (GST_RTP_BASE_PAYLOAD (payload)->sinkpad)
Get access to the sinkpad of payload
.
#define GST_RTP_BASE_PAYLOAD_SRCPAD(payload) (GST_RTP_BASE_PAYLOAD (payload)->srcpad)
Get access to the srcpad of payload
.
gboolean gst_rtp_base_payload_is_filled (GstRTPBasePayload *payload
,guint size
,GstClockTime duration
);
Check if the packet with size
and duration
would exceed the configured
maximum size.
GstFlowReturn gst_rtp_base_payload_push (GstRTPBasePayload *payload
,GstBuffer *buffer
);
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
.
GstFlowReturn gst_rtp_base_payload_push_list (GstRTPBasePayload *payload
,GstBufferList *list
);
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
.
void gst_rtp_base_payload_set_options (GstRTPBasePayload *payload
,const gchar *media
,gboolean dynamic
,const gchar *encoding_name
,guint32 clock_rate
);
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()
.
gboolean gst_rtp_base_payload_set_outcaps (GstRTPBasePayload *payload
,const gchar *fieldname
,...
);
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.
GstBuffer * gst_rtp_base_payload_allocate_output_buffer (GstRTPBasePayload *payload
,guint payload_len
,guint8 pad_len
,guint8 csrc_count
);
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 “source-info” TRUE
additional
CSRCs may be allocated and filled with RTP source information.
payload |
||
payload_len |
the length of the payload |
|
pad_len |
the amount of padding |
|
csrc_count |
the minimum number of CSRC entries |
Since: 1.16
guint gst_rtp_base_payload_get_source_count (GstRTPBasePayload *payload
,GstBuffer *buffer
);
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 “source-info” is FALSE
the count will be 0.
Since: 1.16
gboolean
gst_rtp_base_payload_is_source_info_enabled
(GstRTPBasePayload *payload
);
Queries whether the payloader will add contributing sources (CSRCs) to the RTP header from GstRTPSourceMeta.
Since: 1.16
void gst_rtp_base_payload_set_source_info_enabled (GstRTPBasePayload *payload
,gboolean enable
);
Enable or disable adding contributing sources to RTP packets from GstRTPSourceMeta.
Since: 1.16
struct GstRTPBasePayloadClass { GstElementClass parent_class; /* query accepted caps */ GstCaps * (*get_caps) (GstRTPBasePayload *payload, GstPad * pad, GstCaps * filter); /* receive caps on the sink pad, configure the payloader. */ gboolean (*set_caps) (GstRTPBasePayload *payload, GstCaps *caps); /* handle a buffer, perform 0 or more gst_rtp_base_payload_push() on * the RTP buffers. This function takes ownership of the buffer. */ GstFlowReturn (*handle_buffer) (GstRTPBasePayload *payload, GstBuffer *buffer); /* handle events and queries */ gboolean (*sink_event) (GstRTPBasePayload *payload, GstEvent * event); gboolean (*src_event) (GstRTPBasePayload *payload, GstEvent * event); gboolean (*query) (GstRTPBasePayload *payload, GstPad *pad, GstQuery * query); };
Base class for audio RTP payloader.
“max-ptime”
property “max-ptime” gint64
Maximum duration of the packet data in ns (-1 = unlimited up to MTU).
Owner: GstRTPBasePayload
Flags: Read / Write
Allowed values: >= -1
Default value: -1
“min-ptime”
property “min-ptime” gint64
Minimum duration of the packet data in ns (can't go above MTU)
Owner: GstRTPBasePayload
Flags: Read / Write
Allowed values: >= 0
Default value: 0
“mtu”
property “mtu” guint
Maximum size of one packet.
Owner: GstRTPBasePayload
Flags: Read / Write
Allowed values: >= 28
Default value: 1400
“onvif-no-rate-control”
property “onvif-no-rate-control” gboolean
Make the payloader timestamp packets according to the Rate-Control=no behaviour specified in the ONVIF replay spec.
Owner: GstRTPBasePayload
Flags: Read / Write
Default value: FALSE
Since: 1.16
“perfect-rtptime”
property “perfect-rtptime” gboolean
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.
Owner: GstRTPBasePayload
Flags: Read / Write
Default value: TRUE
“pt”
property “pt” guint
The payload type of the packets.
Owner: GstRTPBasePayload
Flags: Read / Write
Allowed values: <= 127
Default value: 96
“ptime-multiple”
property “ptime-multiple” gint64
Force buffers to be multiples of this duration in ns (0 disables)
Owner: GstRTPBasePayload
Flags: Read / Write
Allowed values: >= 0
Default value: 0
“seqnum”
property “seqnum” guint
The RTP sequence number of the last processed packet.
Owner: GstRTPBasePayload
Flags: Read
Allowed values: <= 65535
Default value: 0
“seqnum-offset”
property “seqnum-offset” gint
Offset to add to all outgoing seqnum (-1 = random).
Owner: GstRTPBasePayload
Flags: Read / Write
Allowed values: [-1,65535]
Default value: -1
“source-info”
property “source-info” gboolean
Enable writing the CSRC field in allocated RTP header based on RTP source information found in the input buffer's GstRTPSourceMeta.
Owner: GstRTPBasePayload
Flags: Read / Write
Default value: FALSE
Since: 1.16
“ssrc”
property “ssrc” guint
The SSRC of the packets (default == random).
Owner: GstRTPBasePayload
Flags: Read / Write
Default value: 4294967295
“stats”
property “stats” GstStructure *
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 “seqnum”
timestamp
:G_TYPE_UINT, RTP timestamp, same as “timestamp”
ssrc
:G_TYPE_UINT, The SSRC in use
pt
:G_TYPE_UINT, The Payload type in use, same as “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
Owner: GstRTPBasePayload
Flags: Read
“timestamp”
property “timestamp” guint
The RTP timestamp of the last processed packet.
Owner: GstRTPBasePayload
Flags: Read
Default value: 0