| GStreamer Base Plugins 1.0 Library Reference Manual | ||||
|---|---|---|---|---|
| Top | Description | Object Hierarchy | Properties | ||||
#include <gst/rtp/gstrtpbasepayload.h> struct GstRTPBasePayload; struct GstRTPBasePayloadClass; #define GST_RTP_BASE_PAYLOAD_MTU (payload) #define GST_RTP_BASE_PAYLOAD_PT (payload) #define GST_RTP_BASE_PAYLOAD_SINKPAD (payload) #define GST_RTP_BASE_PAYLOAD_SRCPAD (payload) gboolean gst_rtp_base_payload_is_filled (GstRTPBasePayload *payload,guint size,GstClockTime duration); GstFlowReturn gst_rtp_base_payload_push (GstRTPBasePayload *payload,GstBuffer *buffer); GstFlowReturn gst_rtp_base_payload_push_list (GstRTPBasePayload *payload,GstBufferList *list); void gst_rtp_base_payload_set_options (GstRTPBasePayload *payload,const gchar *media,gboolean dynamic,const gchar *encoding_name,guint32 clock_rate); gboolean gst_rtp_base_payload_set_outcaps (GstRTPBasePayload *payload,const gchar *fieldname,...);
GObject +----GInitiallyUnowned +----GstObject +----GstElement +----GstRTPBasePayload +----GstRTPBaseAudioPayload
"max-ptime" gint64 : Read / Write "min-ptime" gint64 : Read / Write "mtu" guint : Read / Write "perfect-rtptime" gboolean : Read / Write "pt" guint : Read / Write "ptime-multiple" gint64 : Read / Write "seqnum" guint : Read "seqnum-offset" gint : Read / Write "ssrc" guint : Read / Write "timestamp" guint : Read "timestamp-offset" guint : Read / Write
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.
GstElementClass |
the parent class |
| get desired caps | |
| configure the payloader | |
| process data | |
| custom event handling on the sinkpad | |
| custom event handling on the srcpad | |
| custom query handling |
#define GST_RTP_BASE_PAYLOAD_MTU(payload) (GST_RTP_BASE_PAYLOAD (payload)->mtu)
Get access to the configured MTU of payload.
|
a GstRTPBasePayload |
#define GST_RTP_BASE_PAYLOAD_PT(payload) (GST_RTP_BASE_PAYLOAD (payload)->pt)
Get access to the configured payload type of payload.
|
a GstRTPBasePayload |
#define GST_RTP_BASE_PAYLOAD_SINKPAD(payload) (GST_RTP_BASE_PAYLOAD (payload)->sinkpad)
Get access to the sinkpad of payload.
|
a GstRTPBasePayload |
#define GST_RTP_BASE_PAYLOAD_SRCPAD(payload) (GST_RTP_BASE_PAYLOAD (payload)->srcpad)
Get access to the srcpad of payload.
|
a GstRTPBasePayload |
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.
|
a GstRTPBasePayload |
|
the size of the packet |
|
the duration of the packet |
Returns : |
TRUE if the packet of size and duration would exceed the
configured MTU or max_ptime. |
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.
|
a GstRTPBasePayload |
|
a GstBuffer |
Returns : |
a GstFlowReturn. |
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.
|
a GstRTPBasePayload |
|
a GstBufferList |
Returns : |
a GstFlowReturn. |
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().
|
a GstRTPBasePayload |
|
the media type (typically "audio" or "video") |
|
if the payload type is dynamic |
|
the encoding name |
|
the clock rate of the media |
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.
|
a GstRTPBasePayload |
|
the first field name or NULL
|
|
field values |
Returns : |
TRUE if the caps could be set. |
"max-ptime" property "max-ptime" gint64 : Read / Write
Maximum duration of the packet data in ns (-1 = unlimited up to MTU).
Allowed values: >= -1
Default value: -1
"min-ptime" property "min-ptime" gint64 : Read / Write
Minimum duration of the packet data in ns (can't go above MTU).
Allowed values: >= 0
Default value: 0
"mtu" property"mtu" guint : Read / Write
Maximum size of one packet.
Allowed values: >= 28
Default value: 1400
"perfect-rtptime" property"perfect-rtptime" gboolean : Read / Write
Generate perfect RTP timestamps when possible.
Default value: TRUE
"pt" property"pt" guint : Read / Write
The payload type of the packets.
Allowed values: <= 128
Default value: 96
"ptime-multiple" property "ptime-multiple" gint64 : Read / Write
Force buffers to be multiples of this duration in ns (0 disables).
Allowed values: >= 0
Default value: 0
"seqnum" property"seqnum" guint : Read
The RTP sequence number of the last processed packet.
Allowed values: <= 65535
Default value: 0
"seqnum-offset" property"seqnum-offset" gint : Read / Write
Offset to add to all outgoing seqnum (-1 = random).
Allowed values: [-1,65535]
Default value: -1
"ssrc" property"ssrc" guint : Read / Write
The SSRC of the packets (default == random).
Default value: 4294967295
"timestamp" property"timestamp" guint : Read
The RTP timestamp of the last processed packet.
Default value: 0
"timestamp-offset" property"timestamp-offset" guint : Read / Write
Offset to add to all outgoing timestamps (default = random).
Default value: 4294967295