| GStreamer Base Plugins 1.0 Library Reference Manual | ||||
|---|---|---|---|---|
| Top | Description | ||||
#include <gst/rtsp/gstrtsptransport.h> enum GstRTSPTransMode; enum GstRTSPProfile; struct GstRTSPRange; enum GstRTSPLowerTrans; struct GstRTSPTransport; GstRTSPResult gst_rtsp_transport_new (GstRTSPTransport **transport); GstRTSPResult gst_rtsp_transport_init (GstRTSPTransport *transport); GstRTSPResult gst_rtsp_transport_parse (const gchar *str,GstRTSPTransport *transport); gchar * gst_rtsp_transport_as_text (GstRTSPTransport *transport); GstRTSPResult gst_rtsp_transport_get_mime (GstRTSPTransMode trans,const gchar **mime); GstRTSPResult gst_rtsp_transport_get_manager (GstRTSPTransMode trans,const gchar **manager,guint option); GstRTSPResult gst_rtsp_transport_free (GstRTSPTransport *transport);
typedef enum {
GST_RTSP_TRANS_UNKNOWN = 0,
GST_RTSP_TRANS_RTP = (1 << 0),
GST_RTSP_TRANS_RDT = (1 << 1)
} GstRTSPTransMode;
The transfer mode to use.
typedef enum {
GST_RTSP_PROFILE_UNKNOWN = 0,
GST_RTSP_PROFILE_AVP = (1 << 0),
GST_RTSP_PROFILE_SAVP = (1 << 1),
GST_RTSP_PROFILE_AVPF = (1 << 2),
GST_RTSP_PROFILE_SAVPF = (1 << 3),
} GstRTSPProfile;
The transfer profile to use.
| invalid profile | |
| the Audio/Visual profile (RFC 3551) | |
| the secure Audio/Visual profile (RFC 3711) | |
| the Audio/Visual profile with feedback (RFC 4585) | |
| the secure Audio/Visual profile with feedback (RFC 5124) |
typedef enum {
GST_RTSP_LOWER_TRANS_UNKNOWN = 0,
GST_RTSP_LOWER_TRANS_UDP = (1 << 0),
GST_RTSP_LOWER_TRANS_UDP_MCAST = (1 << 1),
GST_RTSP_LOWER_TRANS_TCP = (1 << 2),
GST_RTSP_LOWER_TRANS_HTTP = (1 << 4),
GST_RTSP_LOWER_TRANS_TLS = (1 << 5)
} GstRTSPLowerTrans;
The different transport methods.
| invalid transport flag | |
| stream data over UDP | |
| stream data over UDP multicast | |
| stream data over TCP | |
| stream data tunneled over HTTP. | |
| encrypt TCP and HTTP with TLS |
struct GstRTSPTransport {
GstRTSPTransMode trans;
GstRTSPProfile profile;
GstRTSPLowerTrans lower_transport;
gchar *destination;
gchar *source;
guint layers;
gboolean mode_play;
gboolean mode_record;
gboolean append;
GstRTSPRange interleaved;
/* multicast specific */
guint ttl;
GstRTSPRange port;
/* UDP/TCP specific */
GstRTSPRange client_port;
GstRTSPRange server_port;
/* RTP specific */
guint ssrc;
};
A structure holding the RTSP transport values.
GstRTSPTransMode |
the transport mode |
GstRTSPProfile |
the tansport profile |
GstRTSPLowerTrans |
the lower transport |
| the destination ip/hostname | |
| the source ip/hostname | |
| the number of layers | |
| if play mode was selected | |
| if record mode was selected | |
| is append mode was selected | |
GstRTSPRange |
the interleave range |
| the time to live for multicast UDP | |
GstRTSPRange |
the port pair for multicast sessions |
GstRTSPRange |
the client port pair for receiving data. For TCP based transports, applications can use this field to store the sender and receiver ports of the client. |
GstRTSPRange |
the server port pair for receiving data. For TCP based transports, applications can use this field to store the sender and receiver ports of the server. |
| the ssrc that the sender/receiver will use |
GstRTSPResult gst_rtsp_transport_new (GstRTSPTransport **transport);
Allocate a new initialized GstRTSPTransport. Use gst_rtsp_transport_free()
after usage.
|
location to hold the new GstRTSPTransport |
Returns : |
a GstRTSPResult. |
GstRTSPResult gst_rtsp_transport_init (GstRTSPTransport *transport);
Initialize transport so that it can be used.
|
a GstRTSPTransport |
Returns : |
GST_RTSP_OK. |
GstRTSPResult gst_rtsp_transport_parse (const gchar *str,GstRTSPTransport *transport);
Parse the RTSP transport string str into transport.
|
a transport string |
|
a GstRTSPTransport |
Returns : |
a GstRTSPResult. |
gchar * gst_rtsp_transport_as_text (GstRTSPTransport *transport);
Convert transport into a string that can be used to signal the transport in
an RTSP SETUP response.
|
a GstRTSPTransport |
Returns : |
a string describing the RTSP transport or NULL when the transport is invalid. |
GstRTSPResult gst_rtsp_transport_get_mime (GstRTSPTransMode trans,const gchar **mime);
gst_rtsp_transport_get_mime is deprecated and should not be used in newly-written code. This functions only deals with the GstRTSPTransMode and only
Get the mime type of the transport mode trans. This mime type is typically
used to generate GstCaps events.
returns the mime type for GST_RTSP_PROFILE_AVP. Use
gst_rtsp_transport_get_media_type() instead.
|
a GstRTSPTransMode |
|
location to hold the result |
Returns : |
GST_RTSP_OK. |
GstRTSPResult gst_rtsp_transport_get_manager (GstRTSPTransMode trans,const gchar **manager,guint option);
Get the GstElement that can handle the buffers transported over trans.
It is possible that there are several managers available, use option to
selected one.
manager will contain an element name or NULL when no manager is
needed/available for trans.
|
a GstRTSPTransMode |
|
location to hold the result |
|
option index. |
Returns : |
GST_RTSP_OK. |
GstRTSPResult gst_rtsp_transport_free (GstRTSPTransport *transport);
Free the memory used by transport.
|
a GstRTSPTransport |
Returns : |
GST_RTSP_OK. |