|  |  |  | GStreamer Base Plugins 0.10 Library Reference Manual |  | 
|---|---|---|---|---|
| Top | Description | Object Hierarchy | Properties | ||||
#include <gst/rtp/gstbasertppayload.h> struct GstBaseRTPPayload; struct GstBaseRTPPayloadClass; #define GST_BASE_RTP_PAYLOAD_MTU (payload) #define GST_BASE_RTP_PAYLOAD_PT (payload) #define GST_BASE_RTP_PAYLOAD_SINKPAD (payload) #define GST_BASE_RTP_PAYLOAD_SRCPAD (payload) gboolean gst_basertppayload_is_filled (GstBaseRTPPayload *payload,guint size,GstClockTime duration); GstFlowReturn gst_basertppayload_push (GstBaseRTPPayload *payload,GstBuffer *buffer); GstFlowReturn gst_basertppayload_push_list (GstBaseRTPPayload *payload,GstBufferList *list); void gst_basertppayload_set_options (GstBaseRTPPayload *payload,const gchar *media,gboolean dynamic,const gchar *encoding_name,guint32 clock_rate); gboolean gst_basertppayload_set_outcaps (GstBaseRTPPayload *payload,const gchar *fieldname,...);
GObject +----GstObject +----GstElement +----GstBaseRTPPayload +----GstBaseRTPAudioPayload
"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 GstBaseRTPPayloadClass {
  GstElementClass parent_class;
  /* receive caps on the sink pad, configure the payloader. */
  gboolean      (*set_caps)             (GstBaseRTPPayload *payload, GstCaps *caps);
  /* handle a buffer, perform 0 or more gst_basertppayload_push() on
   * the RTP buffers. This function takes ownership of the buffer. */
  GstFlowReturn (*handle_buffer)        (GstBaseRTPPayload *payload,
                                         GstBuffer *buffer);
  gboolean      (*handle_event)         (GstPad * pad, GstEvent * event);
  GstCaps *     (*get_caps)             (GstBaseRTPPayload *payload, GstPad * pad);
};
Base class for audio RTP payloader.
| GstElementClass  | the parent class | 
| configure the payloader | |
| process data | |
| custom event handling | |
| get desired caps | 
#define GST_BASE_RTP_PAYLOAD_MTU(payload) (GST_BASE_RTP_PAYLOAD (payload)->mtu)
Get access to the configured MTU of payload.
| 
 | a GstBaseRTPPayload | 
#define GST_BASE_RTP_PAYLOAD_PT(payload) (GST_BASE_RTP_PAYLOAD (payload)->pt)
Get access to the configured payload type of payload.
| 
 | a GstBaseRTPPayload | 
#define GST_BASE_RTP_PAYLOAD_SINKPAD(payload) (GST_BASE_RTP_PAYLOAD (payload)->sinkpad)
Get access to the sinkpad of payload.
| 
 | a GstBaseRTPPayload | 
#define GST_BASE_RTP_PAYLOAD_SRCPAD(payload) (GST_BASE_RTP_PAYLOAD (payload)->srcpad)
Get access to the srcpad of payload.
| 
 | a GstBaseRTPPayload | 
gboolean gst_basertppayload_is_filled (GstBaseRTPPayload *payload,guint size,GstClockTime duration);
Check if the packet with size and duration would exceed the configured
maximum size.
| 
 | a GstBaseRTPPayload | 
| 
 | the size of the packet | 
| 
 | the duration of the packet | 
| Returns : | TRUEif the packet ofsizeanddurationwould exceed the
configured MTU or max_ptime. | 
GstFlowReturn gst_basertppayload_push (GstBaseRTPPayload *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 GstBaseRTPPayload | 
| 
 | a GstBuffer | 
| Returns : | a GstFlowReturn. | 
GstFlowReturn gst_basertppayload_push_list (GstBaseRTPPayload *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 GstBaseRTPPayload | 
| 
 | a GstBufferList | 
| Returns : | a GstFlowReturn. | 
Since 0.10.24
void gst_basertppayload_set_options (GstBaseRTPPayload *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_basertppayload_push() or gst_basertppayload_set_outcaps().
| 
 | a GstBaseRTPPayload | 
| 
 | the media type (typically "audio" or "video") | 
| 
 | if the payload type is dynamic | 
| 
 | the encoding name | 
| 
 | the clock rate of the media | 
gboolean gst_basertppayload_set_outcaps (GstBaseRTPPayload *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 GstBaseRTPPayload | 
| 
 | the first field name or NULL | 
| 
 | field values | 
| Returns : | TRUEif 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: [G_MAXULONG,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