GstRTSPMedia: GStreamer RTSP Server Reference Manual

## GstRTSPMedia GstRTSPMedia — The media pipeline |

## Functions

GstRTSPMedia * | gst_rtsp_media_new ()
GstElement * | gst_rtsp_media_get_element ()
void | gst_rtsp_media_take_pipeline ()
void | gst_rtsp_media_set_permissions ()
GstRTSPPermissions * | gst_rtsp_media_get_permissions ()
void | gst_rtsp_media_set_shared ()
gboolean | gst_rtsp_media_is_shared ()
void | gst_rtsp_media_set_reusable ()
gboolean | gst_rtsp_media_is_reusable ()
void | gst_rtsp_media_set_profiles ()
GstRTSPProfile | gst_rtsp_media_get_profiles ()
void | gst_rtsp_media_set_protocols ()
GstRTSPLowerTrans | gst_rtsp_media_get_protocols ()
void | gst_rtsp_media_set_eos_shutdown ()
gboolean | gst_rtsp_media_is_eos_shutdown ()
void | gst_rtsp_media_set_address_pool ()
GstRTSPAddressPool * | gst_rtsp_media_get_address_pool ()
void | gst_rtsp_media_set_buffer_size ()
guint | gst_rtsp_media_get_buffer_size ()
void | gst_rtsp_media_set_retransmission_time ()
GstClockTime | gst_rtsp_media_get_retransmission_time ()
void | gst_rtsp_media_set_do_retransmission ()
gboolean | gst_rtsp_media_get_do_retransmission ()
void | gst_rtsp_media_set_latency ()
guint | gst_rtsp_media_get_latency ()
gboolean | gst_rtsp_media_setup_sdp ()
gboolean | gst_rtsp_media_handle_sdp ()
gboolean | gst_rtsp_media_prepare ()
gboolean | gst_rtsp_media_unprepare ()
GstRTSPMediaStatus | gst_rtsp_media_get_status ()
void | gst_rtsp_media_set_suspend_mode ()
GstRTSPSuspendMode | gst_rtsp_media_get_suspend_mode ()
gboolean | gst_rtsp_media_suspend ()
gboolean | gst_rtsp_media_unsuspend ()
void | gst_rtsp_media_collect_streams ()
GstRTSPStream * | gst_rtsp_media_create_stream ()
guint | gst_rtsp_media_n_streams ()
GstRTSPStream * | gst_rtsp_media_get_stream ()
GstRTSPStream * | gst_rtsp_media_find_stream ()
gboolean | gst_rtsp_media_seek ()
gchar * | gst_rtsp_media_get_range_string ()
gboolean | gst_rtsp_media_set_state ()
void | gst_rtsp_media_set_pipeline_state ()
GstClock * | gst_rtsp_media_get_clock ()
void | gst_rtsp_media_set_clock ()
GstClockTime | gst_rtsp_media_get_base_time ()
void | gst_rtsp_media_use_time_provider ()
gboolean | gst_rtsp_media_is_time_provider ()
GstNetTimeProvider * | gst_rtsp_media_get_time_provider ()

## Properties

gboolean | bind-mcast-address | Read / Write
guint | buffer-size | Read / Write
GstClock * | clock | Read / Write
GstElement * | element | Read / Write / Construct Only
gboolean | eos-shutdown | Read / Write
guint | latency | Read / Write
guint | max-mcast-ttl | Read / Write
GstRTSPProfile | profiles | Read / Write
GstRTSPLowerTrans | protocols | Read / Write
gboolean | reusable | Read / Write
gboolean | shared | Read / Write
gboolean | stop-on-disconnect | Read / Write
GstRTSPSuspendMode | suspend-mode | Read / Write
gboolean | time-provider | Read / Write
GstRTSPTransportMode | transport-mode | Read / Write

## Signals

void | new-state | Run Last
void | new-stream | Run Last
void | prepared | Run Last
void | removed-stream | Run Last
void | target-state | Run Last
void | unprepared | Run Last

## Types and Values

struct | GstRTSPMedia
struct | GstRTSPMediaClass
enum | GstRTSPMediaStatus
enum | GstRTSPSuspendMode

## Object Hierarchy

        GObject
        ╰── GstRTSPMedia

## Description

a GstRTSPMedia contains the complete GStreamer pipeline to manage the streaming to the clients. The actual data transfer is done by the GstRTSPStream objects that are created and exposed by the GstRTSPMedia.

The GstRTSPMedia is usually created from a GstRTSPMediaFactory when the client does a DESCRIBE or SETUP of a resource.

A media is created with `gst_rtsp_media_new()` that takes the element that will provide the streaming elements. For each of the streams, a new GstRTSPStream object needs to be made with the `gst_rtsp_media_create_stream()` which takes the payloader element and the source pad that produces the RTP stream.

The pipeline of the media is set to PAUSED with `gst_rtsp_media_prepare()`. The prepare method will add rtpbin and sinks and sources to send and receive RTP and RTCP packets from the clients. Each stream srcpad is connected to an input into the internal rtpbin.

It is also possible to dynamically create GstRTSPStream objects during the prepare phase. With `gst_rtsp_media_get_status()` you can check the status of the prepare phase.

After the media is prepared, it is ready for streaming. It will usually be managed in a session with `gst_rtsp_session_manage_media()`. See GstRTSPSession and GstRTSPSessionMedia.

The state of the media can be controlled with `gst_rtsp_media_set_state()`. Seeking can be done with `gst_rtsp_media_seek()`.

With `gst_rtsp_media_unprepare()` the pipeline is stopped and shut down. When `gst_rtsp_media_set_eos_shutdown()` an EOS will be sent to the pipeline to cleanly shut down.

With `gst_rtsp_media_set_shared()`, the media can be shared between multiple clients. With `gst_rtsp_media_set_reusable()` you can control if the pipeline can be prepared again after an unprepare.

Last reviewed on 2013-07-11 (1.0.0)

## Functions

### gst_rtsp_media_new ()

    GstRTSPMedia *
    gst_rtsp_media_new (GstElement *element);

Create a new GstRTSPMedia instance. `element` is the bin element that provides the different streams. The GstRTSPMedia object contains the element to produce RTP data for one or more related (audio/video/..) streams.

Ownership is taken of `element` .

#### Parameters

element | a GstElement. | [transfer full]

#### Returns

a new GstRTSPMedia object.

[transfer full]

### gst_rtsp_media_get_element ()

    GstElement *
    gst_rtsp_media_get_element (GstRTSPMedia *media);

Get the element that was used when constructing `media` .

#### Parameters

media | a GstRTSPMedia |

#### Returns

a GstElement. Unref after usage.

[transfer full]

### gst_rtsp_media_take_pipeline ()

    void
    gst_rtsp_media_take_pipeline (GstRTSPMedia *media,
                                  GstPipeline *pipeline);

Set `pipeline` as the GstPipeline for `media` . Ownership is taken of `pipeline` .

#### Parameters

media | a GstRTSPMedia |
pipeline | a GstPipeline. | [transfer full]

### gst_rtsp_media_set_permissions ()

    void
    gst_rtsp_media_set_permissions (GstRTSPMedia *media,
                                    GstRTSPPermissions *permissions);

Set `permissions` on `media` .

#### Parameters

media | a GstRTSPMedia |
permissions | a GstRTSPPermissions. | [transfer none][nullable]

### gst_rtsp_media_get_permissions ()

    GstRTSPPermissions *
    gst_rtsp_media_get_permissions (GstRTSPMedia *media);

Get the permissions object from `media` .

#### Parameters

media | a GstRTSPMedia |

#### Returns

a GstRTSPPermissions object, unref after usage.

[transfer full][nullable]

### gst_rtsp_media_set_shared ()

    void
    gst_rtsp_media_set_shared (GstRTSPMedia *media,
                               gboolean shared);

Set or unset if the pipeline for `media` can be shared will multiple clients. When `shared` is `TRUE`, client requests for this media will share the media pipeline.

#### Parameters

media | a GstRTSPMedia |
shared | the new value |

### gst_rtsp_media_is_shared ()

    gboolean
    gst_rtsp_media_is_shared (GstRTSPMedia *media);

Check if the pipeline for `media` can be shared between multiple clients.

#### Parameters

media | a GstRTSPMedia |

#### Returns

`TRUE` if the media can be shared between clients.

### gst_rtsp_media_set_reusable ()

    void
    gst_rtsp_media_set_reusable (GstRTSPMedia *media,
                                 gboolean reusable);

Set or unset if the pipeline for `media` can be reused after the pipeline has been unprepared.

#### Parameters

media | a GstRTSPMedia |
reusable | the new value |

### gst_rtsp_media_is_reusable ()

    gboolean
    gst_rtsp_media_is_reusable (GstRTSPMedia *media);

Check if the pipeline for `media` can be reused after an unprepare.

#### Parameters

media | a GstRTSPMedia |

#### Returns

`TRUE` if the media can be reused

### gst_rtsp_media_set_profiles ()

    void
    gst_rtsp_media_set_profiles (GstRTSPMedia *media,
                                 GstRTSPProfile profiles);

Configure the allowed lower transport for `media` .

#### Parameters

media | a GstRTSPMedia |
profiles | the new flags |

### gst_rtsp_media_get_profiles ()

    GstRTSPProfile
    gst_rtsp_media_get_profiles (GstRTSPMedia *media);

Get the allowed profiles of `media` .

#### Parameters

media | a GstRTSPMedia |

#### Returns

a GstRTSPProfile

### gst_rtsp_media_set_protocols ()

    void
    gst_rtsp_media_set_protocols (GstRTSPMedia *media,
                                  GstRTSPLowerTrans protocols);

Configure the allowed lower transport for `media` .

#### Parameters

media | a GstRTSPMedia |
protocols | the new flags |

### gst_rtsp_media_get_protocols ()

    GstRTSPLowerTrans
    gst_rtsp_media_get_protocols (GstRTSPMedia *media);

Get the allowed protocols of `media` .

#### Parameters

media | a GstRTSPMedia |

#### Returns

a GstRTSPLowerTrans

### gst_rtsp_media_set_eos_shutdown ()

    void
    gst_rtsp_media_set_eos_shutdown (GstRTSPMedia *media,
                                     gboolean eos_shutdown);

Set or unset if an EOS event will be sent to the pipeline for `media` before it is unprepared.

#### Parameters

media | a GstRTSPMedia |
eos_shutdown | the new value |

### gst_rtsp_media_is_eos_shutdown ()

    gboolean
    gst_rtsp_media_is_eos_shutdown (GstRTSPMedia *media);

Check if the pipeline for `media` will send an EOS down the pipeline before unpreparing.

#### Parameters

media | a GstRTSPMedia |

#### Returns

`TRUE` if the media will send EOS before unpreparing.

### gst_rtsp_media_set_address_pool ()

    void
    gst_rtsp_media_set_address_pool (GstRTSPMedia *media,
                                     GstRTSPAddressPool *pool);

configure `pool` to be used as the address pool of `media` .

#### Parameters

media | a GstRTSPMedia |
pool | a GstRTSPAddressPool. | [transfer none][nullable]

### gst_rtsp_media_get_address_pool ()

    GstRTSPAddressPool *
    gst_rtsp_media_get_address_pool (GstRTSPMedia *media);

Get the GstRTSPAddressPool used as the address pool of `media` .

#### Parameters

media | a GstRTSPMedia |

#### Returns

the GstRTSPAddressPool of `media` . `g_object_unref()` after usage.

[transfer full][nullable]

### gst_rtsp_media_set_buffer_size ()

    void
    gst_rtsp_media_set_buffer_size (GstRTSPMedia *media,
                                    guint size);

Set the kernel UDP buffer size.

#### Parameters

media | a GstRTSPMedia |
size | the new value |

### gst_rtsp_media_get_buffer_size ()

    guint
    gst_rtsp_media_get_buffer_size (GstRTSPMedia *media);

Get the kernel UDP buffer size.

#### Parameters

media | a GstRTSPMedia |

#### Returns

the kernel UDP buffer size.

### gst_rtsp_media_set_retransmission_time ()

    void
    gst_rtsp_media_set_retransmission_time
                                   (GstRTSPMedia *media,
                                    GstClockTime time);

Set the amount of time to store retransmission packets.

#### Parameters

media | a GstRTSPMedia |
time | the new value |

### gst_rtsp_media_get_retransmission_time ()

    GstClockTime
    gst_rtsp_media_get_retransmission_time
                                   (GstRTSPMedia *media);

Get the amount of time to store retransmission data.

#### Parameters

media | a GstRTSPMedia |

#### Returns

the amount of time to store retransmission data.

### gst_rtsp_media_set_do_retransmission ()

    void
    gst_rtsp_media_set_do_retransmission (GstRTSPMedia *media,
                                          gboolean do_retransmission);

Set whether retransmission requests will be sent

Since: 1.16

### gst_rtsp_media_get_do_retransmission ()

    gboolean
    gst_rtsp_media_get_do_retransmission (GstRTSPMedia *media);

#### Returns

Whether retransmission requests will be sent

Since: 1.16

### gst_rtsp_media_set_latency ()

    void
    gst_rtsp_media_set_latency (GstRTSPMedia *media,
                                guint latency);

Configure the latency used for receiving media.

#### Parameters

media | a GstRTSPMedia |
latency | latency in milliseconds |

### gst_rtsp_media_get_latency ()

    guint
    gst_rtsp_media_get_latency (GstRTSPMedia *media);

Get the latency that is used for receiving media.

#### Parameters

media | a GstRTSPMedia |

#### Returns

latency in milliseconds

### gst_rtsp_media_setup_sdp ()

    gboolean
    gst_rtsp_media_setup_sdp (GstRTSPMedia *media,
                              GstSDPMessage *sdp,
                              GstSDPInfo *info);

Add `media` specific info to `sdp` . `info` is used to configure the connection information in the SDP.

#### Parameters

media | a GstRTSPMedia |
sdp | a GstSDPMessage. | [transfer none]
info | a GstSDPInfo. | [transfer none]

#### Returns

TRUE on success.

### gst_rtsp_media_handle_sdp ()

    gboolean
    gst_rtsp_media_handle_sdp (GstRTSPMedia *media,
                               GstSDPMessage *sdp);

Configure an SDP on `media` for receiving streams

#### Parameters

media | a GstRTSPMedia |
sdp | a GstSDPMessage. | [transfer none]

#### Returns

TRUE on success.

### gst_rtsp_media_prepare ()

    gboolean
    gst_rtsp_media_prepare (GstRTSPMedia *media,
                            GstRTSPThread *thread);

Prepare `media` for streaming. This function will create the objects to manage the streaming. A pipeline must have been set on `media` with `gst_rtsp_media_take_pipeline()`.

It will preroll the pipeline and collect vital information about the streams such as the duration.

#### Parameters

media | a GstRTSPMedia |
thread | a GstRTSPThread to run the bus handler or `NULL`. | [transfer full][allow-none]

#### Returns

`TRUE` on success.

### gst_rtsp_media_unprepare ()

    gboolean
    gst_rtsp_media_unprepare (GstRTSPMedia *media);

Unprepare `media` . After this call, the media should be prepared again before it can be used again. If the media is set to be non-reusable, a new instance must be created.

#### Parameters

media | a GstRTSPMedia |

#### Returns

`TRUE` on success.

### gst_rtsp_media_get_status ()

    GstRTSPMediaStatus
    gst_rtsp_media_get_status (GstRTSPMedia *media);

Get the status of `media` . When `media` is busy preparing, this function waits until `media` is prepared or in error.

#### Parameters

media | a GstRTSPMedia |

#### Returns

the status of `media` .

### gst_rtsp_media_set_suspend_mode ()

    void
    gst_rtsp_media_set_suspend_mode (GstRTSPMedia *media,
                                     GstRTSPSuspendMode mode);

Control how @ media will be suspended after the SDP has been generated and after a PAUSE request has been performed.

Media must be unprepared when setting the suspend mode.

#### Parameters

media | a GstRTSPMedia |
mode | the new GstRTSPSuspendMode |

### gst_rtsp_media_get_suspend_mode ()

    GstRTSPSuspendMode
    gst_rtsp_media_get_suspend_mode (GstRTSPMedia *media);

Get how `media` will be suspended.

#### Parameters

media | a GstRTSPMedia |

#### Returns

GstRTSPSuspendMode.

### gst_rtsp_media_suspend ()

    gboolean
    gst_rtsp_media_suspend (GstRTSPMedia *media);

Suspend `media` . The state of the pipeline managed by `media` is set to GST_STATE_NULL but all streams are kept. `media` can be prepared again with `gst_rtsp_media_unsuspend()`

`media` must be prepared with `gst_rtsp_media_prepare()`;

#### Parameters

media | a GstRTSPMedia |

#### Returns

`TRUE` on success.

### gst_rtsp_media_unsuspend ()

    gboolean
    gst_rtsp_media_unsuspend (GstRTSPMedia *media);

Unsuspend `media` if it was in a suspended state. This method does nothing when the media was not in the suspended state.

#### Parameters

media | a GstRTSPMedia |

#### Returns

`TRUE` on success.

### gst_rtsp_media_collect_streams ()

    void
    gst_rtsp_media_collect_streams (GstRTSPMedia *media);

Find all payloader elements, they should be named pay%d in the element of `media` , and create GstRTSPStreams for them.

Collect all dynamic elements, named dynpay%d, and add them to the list of dynamic elements.

Find all depayloader elements, they should be named depay%d in the element of `media` , and create GstRTSPStreams for them.

#### Parameters

media | a GstRTSPMedia |

### gst_rtsp_media_create_stream ()

    GstRTSPStream *
    gst_rtsp_media_create_stream (GstRTSPMedia *media,
                                  GstElement *payloader,
                                  GstPad *pad);

Create a new stream in `media` that provides RTP data on `pad` . `pad` should be a pad of an element inside `media->element` .

#### Parameters

media | a GstRTSPMedia |
payloader | a GstElement |
pad | a GstPad |

#### Returns

a new GstRTSPStream that remains valid for as long as `media` exists.

[transfer none]

### gst_rtsp_media_n_streams ()

    guint
    gst_rtsp_media_n_streams (GstRTSPMedia *media);

Get the number of streams in this media.

#### Parameters

media | a GstRTSPMedia |

#### Returns

The number of streams.

### gst_rtsp_media_get_stream ()

    GstRTSPStream *
    gst_rtsp_media_get_stream (GstRTSPMedia *media,
                               guint idx);

Retrieve the stream with index `idx` from `media` .

#### Parameters

media | a GstRTSPMedia |
idx | the stream index |

#### Returns

the GstRTSPStream at index `idx` or `NULL` when a stream with that index did not exist.

[nullable][transfer none]

### gst_rtsp_media_find_stream ()

    GstRTSPStream *
    gst_rtsp_media_find_stream (GstRTSPMedia *media,
                                const gchar *control);

Find a stream in `media` with `control` as the control uri.

#### Parameters

media | a GstRTSPMedia |
control | the control of the stream |

#### Returns

the GstRTSPStream with control uri `control` or `NULL` when a stream with that control did not exist.

[nullable][transfer none]

### gst_rtsp_media_seek ()

    gboolean
    gst_rtsp_media_seek (GstRTSPMedia *media,
                         GstRTSPTimeRange *range);

Seek the pipeline of `media` to `range` . `media` must be prepared with `gst_rtsp_media_prepare()`.

#### Parameters

media | a GstRTSPMedia |
range | a GstRTSPTimeRange. | [transfer none]

#### Returns

`TRUE` on success.

### gst_rtsp_media_get_range_string ()

    gchar *
    gst_rtsp_media_get_range_string (GstRTSPMedia *media,
                                     gboolean play,
                                     GstRTSPRangeUnit unit);

Get the current range as a string. `media` must be prepared with `gst_rtsp_media_prepare()`.

#### Parameters

media | a GstRTSPMedia |
play | for the PLAY request |
unit | the unit to use for the string |

#### Returns

The range as a string, `g_free()` after usage.

[transfer full][nullable]

### gst_rtsp_media_set_state ()

    gboolean
    gst_rtsp_media_set_state (GstRTSPMedia *media,
                              GstState state,
                              GPtrArray *transports);

Set the state of `media` to `state` and for the transports in `transports` .

`media` must be prepared with `gst_rtsp_media_prepare()`;

#### Parameters

media | a GstRTSPMedia |
state | the target state of the media |
transports | a GPtrArray of GstRTSPStreamTransport pointers. | [transfer none][element-type GstRtspServer.RTSPStreamTransport]

#### Returns

`TRUE` on success.

### gst_rtsp_media_set_pipeline_state ()

    void
    gst_rtsp_media_set_pipeline_state (GstRTSPMedia *media,
                                       GstState state);

Set the state of the pipeline managed by `media` to `state`

#### Parameters

media | a GstRTSPMedia |
state | the target state of the pipeline |

### gst_rtsp_media_get_clock ()

    GstClock *
    gst_rtsp_media_get_clock (GstRTSPMedia *media);

Get the clock that is used by the pipeline in `media` .

`media` must be prepared before this method returns a valid clock object.

#### Parameters

media | a GstRTSPMedia |

#### Returns

the GstClock used by `media` . unref after usage.

[transfer full][nullable]

### gst_rtsp_media_set_clock ()

    void
    gst_rtsp_media_set_clock (GstRTSPMedia *media,
                              GstClock *clock);

Configure the clock used for the media.

#### Parameters

media | a GstRTSPMedia |
clock | GstClock to be used. | [nullable]

### gst_rtsp_media_get_base_time ()

    GstClockTime
    gst_rtsp_media_get_base_time (GstRTSPMedia *media);

Get the base_time that is used by the pipeline in `media` .

`media` must be prepared before this method returns a valid base_time.

#### Parameters

media | a GstRTSPMedia |

#### Returns

the base_time used by `media` .

### gst_rtsp_media_use_time_provider ()

    void
    gst_rtsp_media_use_time_provider (GstRTSPMedia *media,
                                      gboolean time_provider);

Set `media` to provide a GstNetTimeProvider.

#### Parameters

media | a GstRTSPMedia |
time_provider | if a GstNetTimeProvider should be used |

### gst_rtsp_media_is_time_provider ()

    gboolean
    gst_rtsp_media_is_time_provider (GstRTSPMedia *media);

Check if `media` can provide a GstNetTimeProvider for its pipeline clock.

Use `gst_rtsp_media_get_time_provider()` to get the network clock.

#### Parameters

media | a GstRTSPMedia |

#### Returns

`TRUE` if `media` can provide a GstNetTimeProvider.

### gst_rtsp_media_get_time_provider ()

    GstNetTimeProvider *
    gst_rtsp_media_get_time_provider (GstRTSPMedia *media,
                                      const gchar *address,
                                      guint16 port);

Get the GstNetTimeProvider for the clock used by `media` . The time provider will listen on `address` and `port` for client time requests.

#### Parameters

media | a GstRTSPMedia |
address | an address or `NULL`. | [allow-none]
port | a port or 0 |

#### Returns

the GstNetTimeProvider of `media` .

[transfer full]

## Types and Values

### struct GstRTSPMedia

    struct GstRTSPMedia;

A class that contains the GStreamer element along with a list of GstRTSPStream objects that can produce data.

This object is usually created from a GstRTSPMediaFactory.

### struct GstRTSPMediaClass

    struct GstRTSPMediaClass {
      GObjectClass  parent_class;

      /* vmethods */
      gboolean        (*handle_message)  (GstRTSPMedia *media, GstMessage *message);
      gboolean        (*prepare)         (GstRTSPMedia *media, GstRTSPThread *thread);
      gboolean        (*unprepare)       (GstRTSPMedia *media);
      gboolean        (*suspend)         (GstRTSPMedia *media);
      gboolean        (*unsuspend)       (GstRTSPMedia *media);
      gboolean        (*convert_range)   (GstRTSPMedia *media, GstRTSPTimeRange *range,
                                          GstRTSPRangeUnit unit);
      gboolean        (*query_position)  (GstRTSPMedia *media, gint64 *position);
      gboolean        (*query_stop)      (GstRTSPMedia *media, gint64 *stop);
      GstElement *    (*create_rtpbin)   (GstRTSPMedia *media);
      gboolean        (*setup_rtpbin)    (GstRTSPMedia *media, GstElement *rtpbin);
      gboolean        (*setup_sdp)       (GstRTSPMedia *media, GstSDPMessage *sdp, GstSDPInfo *info);

      /* signals */
      void            (*new_stream)      (GstRTSPMedia *media, GstRTSPStream * stream);
      void            (*removed_stream)  (GstRTSPMedia *media, GstRTSPStream * stream);

      void            (*prepared)        (GstRTSPMedia *media);
      void            (*unprepared)      (GstRTSPMedia *media);

      void            (*target_state)    (GstRTSPMedia *media, GstState state);
      void            (*new_state)       (GstRTSPMedia *media, GstState state);

      gboolean        (*handle_sdp)      (GstRTSPMedia *media, GstSDPMessage *sdp);
    };

The RTSP media class

#### Members

`handle_message` () | handle a message |
`prepare` () | the default implementation adds all elements and sets the pipeline's state to GST_STATE_PAUSED (or GST_STATE_PLAYING in case of NO_PREROLL elements). |
`unprepare` () | the default implementation sets the pipeline's state to GST_STATE_NULL and removes all elements. |
`suspend` () | the default implementation sets the pipeline's state to GST_STATE_NULL GST_STATE_PAUSED depending on the selected suspend mode. |
`unsuspend` () | the default implementation reverts the suspend operation. The pipeline will be prerolled again if it's state was set to GST_STATE_NULL in suspend. |
`convert_range` () | convert a range to the given unit |
`query_position` () | query the current position in the pipeline |
`query_stop` () | query when playback will stop |
`create_rtpbin` () |  |
`setup_rtpbin` () |  |
`setup_sdp` () |  |
`new_stream` () |  |
`removed_stream` () |  |
`prepared` () |  |
`unprepared` () |  |
`target_state` () |  |
`new_state` () |  |
`handle_sdp` () |  |

### enum GstRTSPMediaStatus

The state of the media pipeline.

#### Members

GST_RTSP_MEDIA_STATUS_UNPREPARED | media pipeline not prerolled |
GST_RTSP_MEDIA_STATUS_UNPREPARING | media pipeline is busy doing a clean shutdown. |
GST_RTSP_MEDIA_STATUS_PREPARING | media pipeline is prerolling |
GST_RTSP_MEDIA_STATUS_PREPARED | media pipeline is prerolled |
GST_RTSP_MEDIA_STATUS_SUSPENDED | media is suspended |
GST_RTSP_MEDIA_STATUS_ERROR | media pipeline is in error |

### enum GstRTSPSuspendMode

The suspend mode of the media pipeline. A media pipeline is suspended right after creating the SDP and when the client performs a PAUSED request.

#### Members

GST_RTSP_SUSPEND_MODE_NONE | Media is not suspended |
GST_RTSP_SUSPEND_MODE_PAUSE | Media is PAUSED in suspend |
GST_RTSP_SUSPEND_MODE_RESET | The media is set to NULL when suspended |

## Property Details

### The `“bind-mcast-address”` property

      “bind-mcast-address”       gboolean

Whether the multicast sockets should be bound to multicast addresses or INADDR_ANY.

Flags: Read / Write

Default value: FALSE

### The `“buffer-size”` property

      “buffer-size”              guint

The kernel UDP buffer size to use.

Flags: Read / Write

Default value: 524288

### The `“clock”` property

      “clock”                    GstClock *

Clock to be used by the media pipeline.

Flags: Read / Write

### The `“element”` property

      “element”                  GstElement *

The GstBin to use for streaming the media.

Flags: Read / Write / Construct Only

### The `“eos-shutdown”` property

      “eos-shutdown”             gboolean

Send an EOS event to the pipeline before unpreparing.

Flags: Read / Write

Default value: FALSE

### The `“latency”` property

      “latency”                  guint

Latency used for receiving media in milliseconds.

Flags: Read / Write

Default value: 524288

### The `“max-mcast-ttl”` property

      “max-mcast-ttl”            guint

The maximum time-to-live value of outgoing multicast packets.

Flags: Read / Write

Allowed values: [1,255]

Default value: 255

### The `“profiles”` property

      “profiles”                 GstRTSPProfile

Allowed transfer profiles.

Flags: Read / Write

Default value: GST_RTSP_PROFILE_AVP

### The `“protocols”` property

      “protocols”                GstRTSPLowerTrans

Allowed lower transport protocols.

Flags: Read / Write

Default value: GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP

### The `“reusable”` property

      “reusable”                 gboolean

If this media pipeline can be reused after an unprepare.

Flags: Read / Write

Default value: FALSE

### The `“shared”` property

      “shared”                   gboolean

If this media pipeline can be shared.

Flags: Read / Write

Default value: FALSE

### The `“stop-on-disconnect”` property

      “stop-on-disconnect”       gboolean

If this media pipeline should be stopped when a client disconnects without TEARDOWN.

Flags: Read / Write

Default value: TRUE

### The `“suspend-mode”` property

      “suspend-mode”             GstRTSPSuspendMode

How to suspend the media in PAUSED.

Flags: Read / Write

Default value: GST_RTSP_SUSPEND_MODE_NONE

### The `“time-provider”` property

      “time-provider”            gboolean

Use a NetTimeProvider for clients.

Flags: Read / Write

Default value: FALSE

### The `“transport-mode”` property

      “transport-mode”           GstRTSPTransportMode

If this media pipeline can be used for PLAY or RECORD.

Flags: Read / Write

Default value: GST_RTSP_TRANSPORT_MODE_PLAY

## Signal Details

### The `“new-state”` signal

    void
    user_function (GstRTSPMedia *gstrtspmedia,
                   gint          arg1,
                   gpointer      user_data)

Flags: Run Last

### The `“new-stream”` signal

    void
    user_function (GstRTSPMedia  *gstrtspmedia,
                   GstRTSPStream *arg1,
                   gpointer       user_data)

Flags: Run Last

### The `“prepared”` signal

    void
    user_function (GstRTSPMedia *gstrtspmedia,
                   gpointer      user_data)

Flags: Run Last

### The `“removed-stream”` signal

    void
    user_function (GstRTSPMedia  *gstrtspmedia,
                   GstRTSPStream *arg1,
                   gpointer       user_data)

Flags: Run Last

### The `“target-state”` signal

    void
    user_function (GstRTSPMedia *gstrtspmedia,
                   gint          arg1,
                   gpointer      user_data)

Flags: Run Last

### The `“unprepared”` signal

    void
    user_function (GstRTSPMedia *gstrtspmedia,
                   gpointer      user_data)

Flags: Run Last

## See Also

GstRTSPMediaFactory, GstRTSPStream, GstRTSPSession, GstRTSPSessionMedia