An address the #GstRTSPAddressPool owner of this address the address the port number number of ports TTL or 0 for unicast addresses Make a copy of @addr. a copy of @addr. a #GstRTSPAddress Free @addr and releasing it back into the pool when owned by a pool. a #GstRTSPAddress Flags used to control allocation of addresses no flags an IPv4 address and IPv6 address address with an even port a multicast address a unicast address An address pool, all member are private Make a new #GstRTSPAddressPool. a new #GstRTSPAddressPool Take an address and ports from @pool. @flags can be used to control the allocation. @n_ports consecutive ports will be allocated of which the first one can be found in @port. a #GstRTSPAddress that should be freed with gst_rtsp_address_free after use or %NULL when no address could be acquired. a #GstRTSPAddressPool flags the amount of ports Adds the addresses from @min_addess to @max_address (inclusive) to @pool. The valid port range for the addresses will be from @min_port to @max_port inclusive. When @ttl is 0, @min_address and @max_address should be unicast addresses. @min_address and @max_address can be set to #GST_RTSP_ADDRESS_POOL_ANY_IPV4 or #GST_RTSP_ADDRESS_POOL_ANY_IPV6 to bind to all available IPv4 or IPv6 addresses. When @ttl > 0, @min_address and @max_address should be multicast addresses. %TRUE if the addresses could be added. a #GstRTSPAddressPool a minimum address to add a maximum address to add the minimum port the maximum port a TTL or 0 for unicast addresses Clear all addresses in @pool. There should be no outstanding allocations. a #GstRTSPAddressPool Dump the free and allocated addresses to stdout. a #GstRTSPAddressPool Used to know if the pool includes any unicast addresses. %TRUE if the pool includes any unicast addresses, %FALSE otherwise a #GstRTSPAddressPool Take a specific address and ports from @pool. @n_ports consecutive ports will be allocated of which the first one can be found in @port. If @ttl is 0, @address should be a unicast address. If @ttl > 0, @address should be a valid multicast address. #GST_RTSP_ADDRESS_POOL_OK if an address was reserved. The address is returned in @address and should be freed with gst_rtsp_address_free after use. a #GstRTSPAddressPool The IP address to reserve The first port to reserve The number of ports The requested ttl storage for a #GstRTSPAddress the parent GObject Opaque Address pool class. Result codes from RTSP address pool functions. no error invalid arguments were provided to a function the addres has already been reserved the address is not in the pool last error The authentication structure. Create a new #GstRTSPAuth instance. a new #GstRTSPAuth Check if @check is allowed in the current context. FALSE if check failed. the item to check Construct a Basic authorisation token from @user and @pass. the base64 encoding of the string @user:@pass. g_free() after usage. a userid a password Add a basic token for the default authentication algorithm that enables the client with privileges listed in @token. a #GstRTSPAuth the basic token authorisation token Add a digest @user and @pass for the default authentication algorithm that enables the client with privileges listed in @token. a #GstRTSPAuth the digest user name the digest password authorisation token Get the default token for @auth. This token will be used for unauthenticated users. the #GstRTSPToken of @auth. gst_rtsp_token_unref() after usage. a #GstRTSPAuth the @realm of @auth Gets the supported authentication methods of @auth. The supported authentication methods a #GstRTSPAuth Get the #GTlsAuthenticationMode. the #GTlsAuthenticationMode. a #GstRTSPAuth Get the #GTlsCertificate used for negotiating TLS @auth. the #GTlsCertificate of @auth. g_object_unref() after usage. a #GstRTSPAuth Get the #GTlsDatabase used for verifying client certificate. the #GTlsDatabase of @auth. g_object_unref() after usage. a #GstRTSPAuth Parse the contents of the file at @path and enable the privileges listed in @token for the users it describes. The format of the file is expected to match the format described by <https://en.wikipedia.org/wiki/Digest_access_authentication#The_.htdigest_file>, as output by the `htdigest` command. %TRUE if the file was successfully parsed, %FALSE otherwise. Path to the htdigest file authorisation token Removes @basic authentication token. a #GstRTSPAuth the basic token Removes a digest user. a #GstRTSPAuth the digest user name Set the default #GstRTSPToken to @token in @auth. The default token will be used for unauthenticated users. a #GstRTSPAuth a #GstRTSPToken Set the @realm of @auth The realm to set Sets the supported authentication @methods for @auth. a #GstRTSPAuth supported methods The #GTlsAuthenticationMode to set on the underlying GTlsServerConnection. When set to another value than %G_TLS_AUTHENTICATION_NONE, #GstRTSPAuth::accept-certificate signal will be emitted and must be handled. a #GstRTSPAuth a #GTlsAuthenticationMode Set the TLS certificate for the auth. Client connections will only be accepted when TLS is negotiated. a #GstRTSPAuth a #GTlsCertificate Sets the certificate database that is used to verify peer certificates. If set to %NULL (the default), then peer certificate validation will always set the %G_TLS_CERTIFICATE_UNKNOWN_CA error. a #GstRTSPAuth a #GTlsDatabase Emitted during the TLS handshake after the client certificate has been received. See also gst_rtsp_auth_set_tls_authentication_mode(). %TRUE to accept @peer_cert (which will also immediately end the signal emission). %FALSE to allow the signal emission to continue, which will cause the handshake to fail if no one else overrides it. a #GTlsConnection the peer's #GTlsCertificate the problems with @peer_cert. The authentication class. The client object represents the connection and its state with a client. Create a new #GstRTSPClient instance. a new #GstRTSPClient Called before sending error response to give the application the possibility to adjust the error code. a #GstRTSPStatusCode, containing the adjusted error code. a #GstRTSPClient a #GstRTSPContext a #GstRTSPStatusCode Attaches @client to @context. When the mainloop for @context is run, the client will be dispatched. When @context is %NULL, the default context will be used). This function should be called when the client properties and urls are fully configured and the client is ready to start. the ID (greater than 0) for the source within the GMainContext. a #GstRTSPClient a #GMainContext Close the connection of @client and remove all media it was managing. a #GstRTSPClient Get the #GstRTSPAuth used as the authentication manager of @client. the #GstRTSPAuth of @client. g_object_unref() after usage. a #GstRTSPClient Get the #GstRTSPConnection of @client. the #GstRTSPConnection of @client. The connection object returned remains valid until the client is freed. a #GstRTSPClient Get the Content-Length limit of @client. the Content-Length limit. a #GstRTSPClient Get the #GstRTSPMountPoints object that @client uses to manage its sessions. a #GstRTSPMountPoints, unref after usage. a #GstRTSPClient Get the #GstRTSPSessionPool object that @client uses to manage its sessions. a #GstRTSPSessionPool, unref after usage. a #GstRTSPClient This is useful when providing a send function through gst_rtsp_client_set_send_func() when doing RTSP over TCP: the send function must call gst_rtsp_stream_transport_message_sent () on the appropriate transport when data has been received for streaming to continue. the #GstRTSPStreamTransport associated with @channel. Get the #GstRTSPThreadPool used as the thread pool of @client. the #GstRTSPThreadPool of @client. g_object_unref() after usage. a #GstRTSPClient Let the client handle @message. a #GstRTSPResult. a #GstRTSPClient an #GstRTSPMessage Send a message message to the remote end. @message must be a #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE. a #GstRTSPClient a #GstRTSPSession to send the message to or %NULL The #GstRTSPMessage to send Call @func for each session managed by @client. The result value of @func determines what happens to the session. @func will be called with @client locked so no further actions on @client can be performed from @func. If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from @client. If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client. If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but will also be added with an additional ref to the result #GList of this function.. When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session. a #GList with all sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each element in the #GList should be unreffed before the list is freed. a #GstRTSPClient a callback user data passed to @func configure @auth to be used as the authentication manager of @client. a #GstRTSPClient a #GstRTSPAuth Set the #GstRTSPConnection of @client. This function takes ownership of @conn. %TRUE on success. a #GstRTSPClient a #GstRTSPConnection Configure @client to use the specified Content-Length limit. Define an appropriate request size limit and reject requests exceeding the limit with response status 413 Request Entity Too Large a #GstRTSPClient Content-Length limit Set @mounts as the mount points for @client which it will use to map urls to media streams. These mount points are usually inherited from the server that created the client but can be overriden later. a #GstRTSPClient a #GstRTSPMountPoints Set @func as the callback that will be called when a new message needs to be sent to the client. @user_data is passed to @func and @notify is called when @user_data is no longer in use. By default, the client will send the messages on the #GstRTSPConnection that was configured with gst_rtsp_client_attach() was called. It is only allowed to set either a `send_func` or a `send_messages_func` but not both at the same time. a #GstRTSPClient a #GstRTSPClientSendFunc user data passed to @func called when @user_data is no longer in use Set @func as the callback that will be called when new messages needs to be sent to the client. @user_data is passed to @func and @notify is called when @user_data is no longer in use. By default, the client will send the messages on the #GstRTSPConnection that was configured with gst_rtsp_client_attach() was called. It is only allowed to set either a `send_func` or a `send_messages_func` but not both at the same time. a #GstRTSPClient a #GstRTSPClientSendMessagesFunc user data passed to @func called when @user_data is no longer in use Set @pool as the sessionpool for @client which it will use to find or allocate sessions. the sessionpool is usually inherited from the server that created the client but can be overridden later. a #GstRTSPClient a #GstRTSPSessionPool configure @pool to be used as the thread pool of @client. a #GstRTSPClient a #GstRTSPThreadPool a #GstRTSPContext a newly allocated string with comma-separated list of unsupported options. An empty string must be returned if all options are supported. a #GstRTSPContext a NULL-terminated array of strings a #GstRTSPContext a #GstRTSPContext a #GstRTSPContext a #GstRTSPContext a #GstRTSPContext a #GstRTSPContext a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code a #GstRTSPContext a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code a #GstRTSPContext a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code a #GstRTSPContext a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code a #GstRTSPContext a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code a #GstRTSPContext a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code a #GstRTSPContext a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code a #GstRTSPContext a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code a #GstRTSPContext a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code a #GstRTSPContext a #GstRTSPStatusCode, GST_RTSP_STS_OK in case of success, otherwise an appropriate return code a #GstRTSPContext a #GstRTSPContext The session The message a #GstRTSPContext a #GstRTSPContext a #GstRTSPContext The client class structure. a #GstRTSPStatusCode, containing the adjusted error code. a #GstRTSPClient a #GstRTSPContext a #GstRTSPStatusCode This callback is called when @client wants to send @message. When @close is %TRUE, the connection should be closed when the message has been sent. %TRUE on success. a #GstRTSPClient a #GstRTSPMessage close the connection user data when registering the callback This callback is called when @client wants to send @messages. When @close is %TRUE, the connection should be closed when the message has been sent. %TRUE on success. a #GstRTSPClient #GstRTSPMessage number of messages close the connection user data when registering the callback This function will be called by the gst_rtsp_client_session_filter(). An implementation should return a value of #GstRTSPFilterResult. When this function returns #GST_RTSP_FILTER_REMOVE, @sess will be removed from @client. A return value of #GST_RTSP_FILTER_KEEP will leave @sess untouched in @client. A value of #GST_RTSP_FILTER_REF will add @sess to the result #GList of gst_rtsp_client_session_filter(). a #GstRTSPFilterResult. a #GstRTSPClient object a #GstRTSPSession in @client user data that has been given to gst_rtsp_client_session_filter() Information passed around containing the context of a request. the server the connection the client the complete request the complete url parsed from @request the parsed method of @uri the current auth object or %NULL authorisation token the session, can be %NULL the session media for the url can be %NULL the media factory for the url, can be %NULL the media for the url can be %NULL the stream for the url can be %NULL the response the stream transport, can be %NULL Pops @ctx off the context stack (verifying that @ctx is on the top of the stack). a #GstRTSPContext Pushes @ctx onto the context stack. The current context can then be received using gst_rtsp_context_get_current(). a #GstRTSPContext Set the token for @ctx. a #GstRTSPContext a #GstRTSPToken Get the current #GstRTSPContext. This object is retrieved from the current thread that is handling the request for a client. a #GstRTSPContext Possible return values for gst_rtsp_session_pool_filter(). Remove session Keep session in the pool Ref session in the result list Function registered with gst_rtsp_stream_transport_set_keepalive() and called when the stream is active. user data 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. 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. a new #GstRTSPMedia object. a #GstElement Configure an SDP on @media for receiving streams TRUE on success. a #GstRTSPMedia a #GstSDPMessage 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. %TRUE on success. a #GstRTSPMedia a #GstRTSPThread to run the bus handler or %NULL Add @media specific info to @sdp. @info is used to configure the connection information in the SDP. TRUE on success. a #GstRTSPMedia a #GstSDPMessage a #GstSDPInfo 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(); %TRUE on success. a #GstRTSPMedia 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. %TRUE on success. a #GstRTSPMedia Unsuspend @media if it was in a suspended state. This method does nothing when the media was not in the suspended state. %TRUE on success. a #GstRTSPMedia Check if the pipeline for @media can be shared between multiple clients. This checks if the media is shareable and whether it is either reusable or was never unprepared before. The function must be called with gst_rtsp_media_lock(). %TRUE if the media can be shared between clients. a #GstRTSPMedia 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. a #GstRTSPMedia Add a receiver and sender parts to the pipeline based on the transport from SETUP. %TRUE if the media pipeline has been sucessfully updated. a #GstRTSPMedia a list of #GstRTSPTransport Create a new stream in @media that provides RTP data on @pad. @pad should be a pad of an element inside @media->element. a new #GstRTSPStream that remains valid for as long as @media exists. a #GstRTSPMedia a #GstElement a #GstPad Find a stream in @media with @control as the control uri. the #GstRTSPStream with control uri @control or %NULL when a stream with that control did not exist. a #GstRTSPMedia the control of the stream Get the #GstRTSPAddressPool used as the address pool of @media. the #GstRTSPAddressPool of @media. g_object_unref() after usage. a #GstRTSPMedia Get the base_time that is used by the pipeline in @media. @media must be prepared before this method returns a valid base_time. the base_time used by @media. a #GstRTSPMedia Get the kernel UDP buffer size. the kernel UDP buffer size. a #GstRTSPMedia Get the clock that is used by the pipeline in @media. @media must be prepared before this method returns a valid clock object. the #GstClock used by @media. unref after usage. a #GstRTSPMedia Whether retransmission requests will be sent Get the configured DSCP QoS of attached media. the DSCP QoS value of attached streams or -1 if disabled. a #GstRTSPMedia Get the element that was used when constructing @media. a #GstElement. Unref after usage. a #GstRTSPMedia Get ensure-keyunit-on-start flag. The ensure-keyunit-on-start flag. a #GstRTSPMedia Get ensure-keyunit-on-start-timeout time. The ensure-keyunit-on-start-timeout time. a #GstRTSPMedia Get the latency that is used for receiving media. latency in milliseconds a #GstRTSPMedia Get the the maximum time-to-live value of outgoing multicast packets. the maximum time-to-live value of outgoing multicast packets. a #GstRTSPMedia Get the multicast interface used for @media. the multicast interface for @media. g_free() after usage. a #GstRTSPMedia Get the permissions object from @media. a #GstRTSPPermissions object, unref after usage. a #GstRTSPMedia Get the allowed profiles of @media. a #GstRTSPProfile a #GstRTSPMedia Get the allowed protocols of @media. a #GstRTSPLowerTrans a #GstRTSPMedia Gets if and how the media clock should be published according to RFC7273. The GstRTSPPublishClockMode a #GstRTSPMedia Get the current range as a string. @media must be prepared with gst_rtsp_media_prepare (). The range as a string, g_free() after usage. a #GstRTSPMedia for the PLAY request the unit to use for the string whether @media will follow the Rate-Control=no behaviour as specified in the ONVIF replay spec. Get the rate and applied_rate of the current segment. %FALSE if looking up the rate and applied rate failed. Otherwise %TRUE is returned and @rate and @applied_rate are set to the rate and applied_rate of the current segment. a #GstRTSPMedia the rate of the current segment the applied_rate of the current segment Get the amount of time to store retransmission data. the amount of time to store retransmission data. a #GstRTSPMedia Get the status of @media. When @media is busy preparing, this function waits until @media is prepared or in error. the status of @media. a #GstRTSPMedia Retrieve the stream with index @idx from @media. the #GstRTSPStream at index @idx or %NULL when a stream with that index did not exist. a #GstRTSPMedia the stream index Get how @media will be suspended. #GstRTSPSuspendMode. a #GstRTSPMedia Get the #GstNetTimeProvider for the clock used by @media. The time provider will listen on @address and @port for client time requests. the #GstNetTimeProvider of @media. a #GstRTSPMedia an address or %NULL a port or 0 Check if the pipeline for @media can be used for PLAY or RECORD methods. The transport mode. a #GstRTSPMedia Configure an SDP on @media for receiving streams TRUE on success. a #GstRTSPMedia a #GstSDPMessage See gst_rtsp_stream_is_complete(), gst_rtsp_stream_is_sender(). whether @media has at least one complete sender stream. Check if multicast sockets are configured to be bound to multicast addresses. %TRUE if multicast sockets are configured to be bound to multicast addresses. a #GstRTSPMedia Check if the pipeline for @media will send an EOS down the pipeline before unpreparing. %TRUE if the media will send EOS before unpreparing. a #GstRTSPMedia %TRUE if @media is receive-only, %FALSE otherwise. Check if the pipeline for @media can be reused after an unprepare. %TRUE if the media can be reused a #GstRTSPMedia Check if the pipeline for @media can be shared between multiple clients in theory. This simply returns the value set via gst_rtsp_media_set_shared(). To know if a media can be shared in practice, i.e. if it's shareable and either reusable or was never unprepared before, use gst_rtsp_media_can_be_shared(). %TRUE if the media can be shared between clients. a #GstRTSPMedia Check if the pipeline for @media will be stopped when a client disconnects without sending TEARDOWN. %TRUE if the media will be stopped when a client disconnects without sending TEARDOWN. a #GstRTSPMedia Check if @media can provide a #GstNetTimeProvider for its pipeline clock. Use gst_rtsp_media_get_time_provider() to get the network clock. %TRUE if @media can provide a #GstNetTimeProvider. a #GstRTSPMedia Lock the entire media. This is needed by callers such as rtsp_client to protect the media when it is shared by many clients. The lock prevents that concurrent clients alters the shared media, while one client already is working with it. Typically the lock is taken in external RTSP API calls that uses shared media such as DESCRIBE, SETUP, ANNOUNCE, TEARDOWN, PLAY, PAUSE. As best practice take the lock as soon as the function get hold of a shared media object. Release the lock right before the function returns. a #GstRTSPMedia Get the number of streams in this media. The number of streams. a #GstRTSPMedia 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. %TRUE on success. a #GstRTSPMedia a #GstRTSPThread to run the bus handler or %NULL Seek the pipeline of @media to @range. @media must be prepared with gst_rtsp_media_prepare(). %TRUE on success. a #GstRTSPMedia a #GstRTSPTimeRange Seek the pipeline of @media to @range with the given @flags. @media must be prepared with gst_rtsp_media_prepare(). %TRUE on success. a #GstRTSPMedia a #GstRTSPTimeRange The minimal set of #GstSeekFlags to use Seek the pipeline of @media to @range with the given @flags and @rate, and @trickmode_interval. @media must be prepared with gst_rtsp_media_prepare(). In order to perform the seek operation, the pipeline must contain all needed transport parts (transport sinks). %TRUE on success. a #GstRTSPMedia a #GstRTSPTimeRange The minimal set of #GstSeekFlags to use the rate to use in the seek The trickmode interval to use for KEY_UNITS trick mode Check if the pipeline for @media seek and up to what point in time, it can seek. -1 if the stream is not seekable, 0 if seekable only to the beginning and > 0 to indicate the longest duration between any two random access points. %G_MAXINT64 means any value is possible. a #GstRTSPMedia configure @pool to be used as the address pool of @media. a #GstRTSPMedia a #GstRTSPAddressPool Decide whether the multicast socket should be bound to a multicast address or INADDR_ANY. a #GstRTSPMedia the new value Set the kernel UDP buffer size. a #GstRTSPMedia the new value Configure the clock used for the media. a #GstRTSPMedia #GstClock to be used Set whether retransmission requests will be sent Configure the dscp qos of attached streams to @dscp_qos. a #GstRTSPMedia a new dscp qos value (0-63, or -1 to disable) Set whether or not a keyunit should be ensured when a client connects. It will also configure the streams to drop delta units to ensure that they start on a keyunit. Note that this will only affect non-shared medias for now. a #GstRTSPMedia the new value Sets the maximum allowed time before the first keyunit is considered expired. Note that this will only have an effect when ensure-keyunit-on-start is enabled. a #GstRTSPMedia the new value Set or unset if an EOS event will be sent to the pipeline for @media before it is unprepared. a #GstRTSPMedia the new value Configure the latency used for receiving media. a #GstRTSPMedia latency in milliseconds Set the maximum time-to-live value of outgoing multicast packets. %TRUE if the requested ttl has been set successfully. a #GstRTSPMedia the new multicast ttl value configure @multicast_iface to be used for @media. a #GstRTSPMedia a multicast interface name Set @permissions on @media. a #GstRTSPMedia a #GstRTSPPermissions Set the state of the pipeline managed by @media to @state a #GstRTSPMedia the target state of the pipeline Configure the allowed lower transport for @media. a #GstRTSPMedia the new flags Configure the allowed lower transport for @media. a #GstRTSPMedia the new flags Sets if and how the media clock should be published according to RFC7273. a #GstRTSPMedia the clock publish mode Define whether @media will follow the Rate-Control=no behaviour as specified in the ONVIF replay spec. Set the amount of time to store retransmission packets. a #GstRTSPMedia the new value Set or unset if the pipeline for @media can be reused after the pipeline has been unprepared. a #GstRTSPMedia the new value 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. a #GstRTSPMedia the new value Set the state of @media to @state and for the transports in @transports. @media must be prepared with gst_rtsp_media_prepare(); %TRUE on success. a #GstRTSPMedia the target state of the media a #GPtrArray of #GstRTSPStreamTransport pointers Set or unset if the pipeline for @media should be stopped when a client disconnects without sending TEARDOWN. a #GstRTSPMedia the new value 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. a #GstRTSPMedia the new #GstRTSPSuspendMode Sets if the media pipeline can work in PLAY or RECORD mode a #GstRTSPMedia the new value Add @media specific info to @sdp. @info is used to configure the connection information in the SDP. TRUE on success. a #GstRTSPMedia a #GstSDPMessage a #GstSDPInfo 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(); %TRUE on success. a #GstRTSPMedia Set @pipeline as the #GstPipeline for @media. Ownership is taken of @pipeline. a #GstRTSPMedia a #GstPipeline Unlock the media. a #GstRTSPMedia 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. %TRUE on success. a #GstRTSPMedia Unsuspend @media if it was in a suspended state. This method does nothing when the media was not in the suspended state. %TRUE on success. a #GstRTSPMedia Set @media to provide a #GstNetTimeProvider. a #GstRTSPMedia if a #GstNetTimeProvider should be used Whether or not a keyunit should be ensured when a client connects. It will also configure the streams to drop delta units to ensure that they start on a keyunit. Note that this will only affect non-shared medias for now. The maximum allowed time before the first keyunit is considered expired. Note that this will only have an effect when ensure-keyunit-on-start is enabled. Will be emitted when a message appears on the pipeline bus. a #gboolean indicating if the call was successful or not. a #GstMessage The RTSP media class %TRUE on success. a #GstRTSPMedia a #GstRTSPThread to run the bus handler or %NULL %TRUE on success. a #GstRTSPMedia %TRUE on success. a #GstRTSPMedia %TRUE on success. a #GstRTSPMedia TRUE on success. a #GstRTSPMedia a #GstSDPMessage a #GstSDPInfo TRUE on success. a #GstRTSPMedia a #GstSDPMessage The definition and logic for constructing the pipeline for a media. The media can contain multiple streams like audio and video. Create a new #GstRTSPMediaFactory instance. a new #GstRTSPMediaFactory object. Construct the media object and create its streams. Implementations should create the needed gstreamer elements and add them to the result object. No state changes should be performed on them yet. One or more GstRTSPStream objects should be created from the result with gst_rtsp_media_create_stream (). After the media is constructed, it can be configured and then prepared with gst_rtsp_media_prepare (). The returned media will be locked and must be unlocked afterwards. a new #GstRTSPMedia if the media could be prepared. a #GstRTSPMediaFactory the url used Construct and return a #GstElement that is a #GstBin containing the elements to use for streaming the media. The bin should contain payloaders pay\%d for each stream. The default implementation of this function returns the bin created from the launch parameter. a new #GstElement. a #GstRTSPMediaFactory the url used A convenience method to add @role with @fieldname and additional arguments to the permissions of @factory. If @factory had no permissions, new permissions will be created and the role will be added to it. a #GstRTSPMediaFactory a role the first field name additional arguments A convenience wrapper around gst_rtsp_permissions_add_role_from_structure(). If @factory had no permissions, new permissions will be created and the role will be added to it. Construct the media object and create its streams. Implementations should create the needed gstreamer elements and add them to the result object. No state changes should be performed on them yet. One or more GstRTSPStream objects should be created from the result with gst_rtsp_media_create_stream (). After the media is constructed, it can be configured and then prepared with gst_rtsp_media_prepare (). The returned media will be locked and must be unlocked afterwards. a new #GstRTSPMedia if the media could be prepared. a #GstRTSPMediaFactory the url used Construct and return a #GstElement that is a #GstBin containing the elements to use for streaming the media. The bin should contain payloaders pay\%d for each stream. The default implementation of this function returns the bin created from the launch parameter. a new #GstElement. a #GstRTSPMediaFactory the url used Get the #GstRTSPAddressPool used as the address pool of @factory. the #GstRTSPAddressPool of @factory. g_object_unref() after usage. a #GstRTSPMediaFactory Get the kernel UDP buffer size. the kernel UDP buffer size. a #GstRTSPMedia Returns the clock that is going to be used by the pipelines of all medias created from this factory. The GstClock a #GstRTSPMediaFactory Whether retransmission requests will be sent for receiving media Get the configured media DSCP QoS. the media DSCP QoS value or -1 if disabled. a #GstRTSPMediaFactory Get ensure-keyunit-on-start flag. The ensure-keyunit-on-start flag. a #GstRTSPMediaFactory Get ensure-keyunit-on-start-timeout time. The ensure-keyunit-on-start-timeout time. a #GstRTSPMediaFactory Get the latency that is used for receiving media latency in milliseconds a #GstRTSPMediaFactory Get the gst_parse_launch() pipeline description that will be used in the default prepare vmethod. the configured launch description. g_free() after usage. a #GstRTSPMediaFactory Get the the maximum time-to-live value of outgoing multicast packets. the maximum time-to-live value of outgoing multicast packets. a #GstRTSPMedia Return the GType of the GstRTSPMedia subclass this factory will create. a #GstRTSPMediaFactory Get the multicast interface used for @factory. the multicast interface for @factory. g_free() after usage. a #GstRTSPMediaFactory Get the permissions object from @factory. a #GstRTSPPermissions object, unref after usage. a #GstRTSPMediaFactory Get the allowed profiles of @factory. a #GstRTSPProfile a #GstRTSPMediaFactory Get the allowed protocols of @factory. a #GstRTSPLowerTrans a #GstRTSPMediaFactory Gets if and how the media clock should be published according to RFC7273. The GstRTSPPublishClockMode a #GstRTSPMediaFactory Get the time that is stored for retransmission purposes a #GstClockTime a #GstRTSPMediaFactory Get how media created from this factory will be suspended. a #GstRTSPSuspendMode. a #GstRTSPMediaFactory Get if media created from this factory can be used for PLAY or RECORD methods. The transport mode. a #GstRTSPMediaFactory Check if multicast sockets are configured to be bound to multicast addresses. %TRUE if multicast sockets are configured to be bound to multicast addresses. a #GstRTSPMediaFactory Check if created media will send and receive RTCP %TRUE if created media will send and receive RTCP a #GstRTSPMediaFactory Get if media created from this factory will have an EOS event sent to the pipeline before shutdown. %TRUE if the media will receive EOS before shutdown. a #GstRTSPMediaFactory Get if media created from this factory can be shared between clients. %TRUE if the media will be shared between clients. a #GstRTSPMediaFactory configure @pool to be used as the address pool of @factory. a #GstRTSPMediaFactory a #GstRTSPAddressPool Decide whether the multicast socket should be bound to a multicast address or INADDR_ANY. a #GstRTSPMediaFactory the new value Set the kernel UDP buffer size. a #GstRTSPMedia the new value Configures a specific clock to be used by the pipelines of all medias created from this factory. a #GstRTSPMediaFactory the clock to be used by the media factory Set whether retransmission requests will be sent for receiving media Configure the media dscp qos to @dscp_qos. a #GstRTSPMediaFactory a new dscp qos value (0-63, or -1 to disable) Decide whether the created media should send and receive RTCP a #GstRTSPMediaFactory the new value If media from this factory should ensure a key unit when a client connects. a #GstRTSPMediaFactory the new value Configures medias from this factory to consider keyunits older than timeout to be expired. Expired keyunits will be discarded. a #GstRTSPMediaFactory the new value Configure if media created from this factory will have an EOS sent to the pipeline before shutdown. a #GstRTSPMediaFactory the new value Configure the latency used for receiving media a #GstRTSPMediaFactory latency in milliseconds The gst_parse_launch() line to use for constructing the pipeline in the default prepare vmethod. The pipeline description should return a GstBin as the toplevel element which can be accomplished by enclosing the description with brackets '(' ')'. The description should return a pipeline with payloaders named pay0, pay1, etc.. Each of the payloaders will result in a stream. a #GstRTSPMediaFactory the launch description Set the maximum time-to-live value of outgoing multicast packets. %TRUE if the requested ttl has been set successfully. a #GstRTSPMedia the new multicast ttl value Configure the GType of the GstRTSPMedia subclass to create (by default, overridden construct vmethods may of course do something different) a #GstRTSPMediaFactory the GType of the class to create configure @multicast_iface to be used for @factory. a #GstRTSPMediaFactory a multicast interface name Set @permissions on @factory. a #GstRTSPMediaFactory a #GstRTSPPermissions Configure the allowed profiles for @factory. a #GstRTSPMediaFactory the new flags Configure the allowed lower transport for @factory. a #GstRTSPMediaFactory the new flags Sets if and how the media clock should be published according to RFC7273. a #GstRTSPMediaFactory the clock publish mode Configure the time to store for possible retransmission a #GstRTSPMediaFactory a #GstClockTime Configure if media created from this factory can be shared between clients. a #GstRTSPMediaFactory the new value Configure if media created from this factory should be stopped when a client disconnects without sending TEARDOWN. a #GstRTSPMediaFactory the new value Configure how media created from this factory will be suspended. a #GstRTSPMediaFactory the new #GstRTSPSuspendMode Configure if this factory creates media for PLAY or RECORD modes. a #GstRTSPMediaFactory the new value Whether the created media should send and receive RTCP If media from this factory should ensure a key unit when a client connects. This property will ensure that the stream always starts on a key unit instead of a delta unit which the client would not be able to decode. Note that this will only affect non-shared medias for now. Timeout in milliseconds used to determine if a keyunit should be discarded when a client connects. If the timeout has been reached a new keyframe will be forced, otherwise the currently blocking keyframe will be used. This options is only relevant when ensure-keyunit-on-start is enabled. The #GstRTSPMediaFactory class structure. a new #GstElement. a #GstRTSPMediaFactory the url used a new #GstRTSPMedia if the media could be prepared. a #GstRTSPMediaFactory the url used A media factory that creates a pipeline to play any uri. Create a new #GstRTSPMediaFactoryURI instance. a new #GstRTSPMediaFactoryURI object. Get the URI that will provide media for this factory. the configured URI. g_free() after usage. a #GstRTSPMediaFactory Set the URI of the resource that will be streamed by this factory. a #GstRTSPMediaFactory the uri the stream The #GstRTSPMediaFactoryURI class structure. The state of the media pipeline. media pipeline not prerolled media pipeline is busy doing a clean shutdown. media pipeline is prerolling media pipeline is prerolled media is suspended media pipeline is in error Function registered with gst_rtsp_stream_transport_set_message_sent() and called when a message has been sent on the transport. user data Function registered with gst_rtsp_stream_transport_set_message_sent_full() and called when a message has been sent on the transport. user data Creates a #GstRTSPMediaFactory object for a given url. Make a new mount points object. a new #GstRTSPMountPoints Make a path string from @url. a path string for @url, g_free() after usage. a #GstRTSPMountPoints a #GstRTSPUrl Attach @factory to the mount point @path in @mounts. @path is either of the form (/node)+ or the root path '/'. (An empty path is not allowed.) Any previous mount point will be freed. Ownership is taken of the reference on @factory so that @factory should not be used after calling this function. a #GstRTSPMountPoints a mount point a #GstRTSPMediaFactory Make a path string from @url. a path string for @url, g_free() after usage. a #GstRTSPMountPoints a #GstRTSPUrl Find the factory in @mounts that has the longest match with @path. If @matched is %NULL, @path will match the factory exactly otherwise the amount of characters that matched is returned in @matched. the #GstRTSPMediaFactory for @path. g_object_unref() after usage. a #GstRTSPMountPoints a mount point the amount of @path matched Remove the #GstRTSPMediaFactory associated with @path in @mounts. a #GstRTSPMountPoints a mount point The class for the media mounts object. a path string for @url, g_free() after usage. a #GstRTSPMountPoints a #GstRTSPUrl Create a new #GstRTSPOnvifClient instance. a new #GstRTSPOnvifClient Find the ONVIF backchannel depayloader element. It should be named 'depay_backchannel', be placed in a bin called 'onvif-backchannel' and return all supported RTP caps on a caps query. Complete RTP caps with at least the payload type, clock-rate and encoding-name are required. A new #GstRTSPStream is created for the backchannel if found. %TRUE if a backchannel stream could be found and created a #GstRTSPOnvifMedia Get the configured/supported bandwidth of the ONVIF backchannel pipeline in bits per second. the configured/supported backchannel bandwidth. a #GstRTSPMedia Set the configured/supported bandwidth of the ONVIF backchannel pipeline in bits per second. a #GstRTSPMedia the bandwidth in bits per second Create a new #GstRTSPOnvifMediaFactory A new #GstRTSPOnvifMediaFactory Checks whether the client request requires backchannel. %TRUE if the client request requires backchannel. a #GstRTSPMediaFactory Returns %TRUE if an ONVIF backchannel is supported by the media factory. %TRUE if an ONVIF backchannel is supported by the media factory. a #GstRTSPMediaFactory Get the configured/supported bandwidth of the ONVIF backchannel pipeline in bits per second. the configured/supported backchannel bandwidth. a #GstRTSPMediaFactory Get the gst_parse_launch() pipeline description that will be used in the default prepare vmethod for generating the ONVIF backchannel pipeline. the configured backchannel launch description. g_free() after usage. a #GstRTSPMediaFactory Returns %TRUE if an ONVIF backchannel is supported by the media factory. %TRUE if an ONVIF backchannel is supported by the media factory. a #GstRTSPMediaFactory %TRUE if ONVIF replay is supported by the media factory. Set the configured/supported bandwidth of the ONVIF backchannel pipeline in bits per second. a #GstRTSPMediaFactory the bandwidth in bits per second The gst_parse_launch() line to use for constructing the ONVIF backchannel pipeline in the default prepare vmethod if requested by the client. The pipeline description should return a GstBin as the toplevel element which can be accomplished by enclosing the description with brackets '(' ')'. The description should return a pipeline with a single depayloader named depay_backchannel. A caps query on the depayloader's sinkpad should return all possible, complete RTP caps that are going to be supported. At least the payload type, clock-rate and encoding-name need to be specified. Note: The pipeline part passed here must end in sinks that are not waiting until pre-rolling before reaching the PAUSED state, i.e. setting async=false on #GstBaseSink. Otherwise the whole media will not be able to prepare. a #GstRTSPMediaFactory the launch description Set to %TRUE if ONVIF replay is supported by the media factory. %TRUE if an ONVIF backchannel is supported by the media factory. a #GstRTSPMediaFactory Create a new #GstRTSPOnvifServer instance. a new #GstRTSPOnvifServer The opaque permissions structure. It is used to define the permissions of objects in different roles. Create a new empty Authorization permissions. a new empty authorization permissions. Add a new @permission for @role to @permissions with the access in @allowed. a #GstRTSPPermissions a role the permission whether the role has this permission or not Add a new @role to @permissions with the given variables. The fields are the same layout as gst_structure_new(). a #GstRTSPPermissions a role the first field name additional arguments Add a new @role to @permissions without any permissions. You can add permissions for the role with gst_rtsp_permissions_add_permission_for_role(). a #GstRTSPPermissions a role Add a new role to @permissions based on @structure, for example given a role named `tester`, which should be granted a permission named `permission1`, the structure could be created with: ``` gst_structure_new ("tester", "permission1", G_TYPE_BOOLEAN, TRUE, NULL); ``` Add a new @role to @permissions with the given variables. Structure fields are set according to the varargs in a manner similar to gst_structure_new(). a #GstRTSPPermissions a role the first field name additional fields to add Get all permissions for @role in @permissions. the structure with permissions for @role. It remains valid for as long as @permissions is valid. a #GstRTSPPermissions a role Check if @role in @permissions is given permission for @permission. %TRUE if @role is allowed @permission. a #GstRTSPPermissions a role a permission Remove all permissions for @role in @permissions. a #GstRTSPPermissions a role Whether the clock and possibly RTP/clock offset should be published according to RFC7273. Publish nothing Publish the clock but not the offset Publish the clock and offset Function registered with gst_rtsp_stream_transport_set_callbacks() and called when @buffer must be sent on @channel. %TRUE on success a #GstBuffer a channel user data Function registered with gst_rtsp_stream_transport_set_callbacks() and called when @buffer_list must be sent on @channel. %TRUE on success a #GstBufferList a channel user data This object listens on a port, creates and manages the clients connected to it. Create a new #GstRTSPServer instance. a new #GstRTSPServer A default #GSocketSourceFunc that creates a new #GstRTSPClient to accept and handle a new connection on @socket or @server. TRUE if the source could be connected, FALSE if an error occurred. a #GSocket the condition on @source a #GstRTSPServer Attaches @server to @context. When the mainloop for @context is run, the server will be dispatched. When @context is %NULL, the default context will be used). This function should be called when the server properties and urls are fully configured and the server is ready to start. This takes a reference on @server until the source is destroyed. Note that if @context is not the default main context as returned by g_main_context_default() (or %NULL), g_source_remove() cannot be used to destroy the source. In that case it is recommended to use gst_rtsp_server_create_source() and attach it to @context manually. the ID (greater than 0) for the source within the GMainContext. a #GstRTSPServer a #GMainContext Call @func for each client managed by @server. The result value of @func determines what happens to the client. @func will be called with @server locked so no further actions on @server can be performed from @func. If @func returns #GST_RTSP_FILTER_REMOVE, the client will be removed from @server. If @func returns #GST_RTSP_FILTER_KEEP, the client will remain in @server. If @func returns #GST_RTSP_FILTER_REF, the client will remain in @server but will also be added with an additional ref to the result #GList of this function.. When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each client. a #GList with all clients for which @func returned #GST_RTSP_FILTER_REF. After usage, each element in the #GList should be unreffed before the list is freed. a #GstRTSPServer a callback user data passed to @func Create a #GSocket for @server. The socket will listen on the configured service. the #GSocket for @server or %NULL when an error occurred. a #GstRTSPServer a #GCancellable Create a #GSource for @server. The new source will have a default #GSocketSourceFunc of gst_rtsp_server_io_func(). @cancellable if not %NULL can be used to cancel the source, which will cause the source to trigger, reporting the current condition (which is likely 0 unless cancellation happened at the same time as a condition change). You can check for this in the callback using g_cancellable_is_cancelled(). This takes a reference on @server until @source is destroyed. the #GSource for @server or %NULL when an error occurred. Free with g_source_unref () a #GstRTSPServer a #GCancellable or %NULL. Get the address on which the server will accept connections. the server address. g_free() after usage. a #GstRTSPServer Get the #GstRTSPAuth used as the authentication manager of @server. the #GstRTSPAuth of @server. g_object_unref() after usage. a #GstRTSPServer The maximum amount of queued requests for the server. the server backlog. a #GstRTSPServer Get the port number where the server was bound to. the port number a #GstRTSPServer Get the Content-Length limit of @server. the Content-Length limit. a #GstRTSPServer Get the #GstRTSPMountPoints used as the mount points of @server. the #GstRTSPMountPoints of @server. g_object_unref() after usage. a #GstRTSPServer Get the service on which the server will accept connections. the service. use g_free() after usage. a #GstRTSPServer Get the #GstRTSPSessionPool used as the session pool of @server. the #GstRTSPSessionPool used for sessions. g_object_unref() after usage. a #GstRTSPServer Get the #GstRTSPThreadPool used as the thread pool of @server. the #GstRTSPThreadPool of @server. g_object_unref() after usage. a #GstRTSPServer Configure @server to accept connections on the given address. This function must be called before the server is bound. a #GstRTSPServer the address configure @auth to be used as the authentication manager of @server. a #GstRTSPServer a #GstRTSPAuth configure the maximum amount of requests that may be queued for the server. This function must be called before the server is bound. a #GstRTSPServer the backlog Define an appropriate request size limit and reject requests exceeding the limit. a #GstRTSPServer Configure @server to use the specified Content-Length limit. configure @mounts to be used as the mount points of @server. a #GstRTSPServer a #GstRTSPMountPoints Configure @server to accept connections on the given service. @service should be a string containing the service name (see services(5)) or a string containing a port number between 1 and 65535. When @service is set to "0", the server will listen on a random free port. The actual used port can be retrieved with gst_rtsp_server_get_bound_port(). This function must be called before the server is bound. a #GstRTSPServer the service configure @pool to be used as the session pool of @server. a #GstRTSPServer a #GstRTSPSessionPool configure @pool to be used as the thread pool of @server. a #GstRTSPServer a #GstRTSPThreadPool Take an existing network socket and use it for an RTSP connection. This is used when transferring a socket from an HTTP server which should be used as an RTSP over HTTP tunnel. The @initial_buffer contains any remaining data that the HTTP server read from the socket while parsing the HTTP header. TRUE if all was ok, FALSE if an error occurred. a #GstRTSPServer a network socket the IP address of the remote client the port used by the other end any initial data that was already read from the socket The RTSP server class structure This function will be called by the gst_rtsp_server_client_filter(). An implementation should return a value of #GstRTSPFilterResult. When this function returns #GST_RTSP_FILTER_REMOVE, @client will be removed from @server. A return value of #GST_RTSP_FILTER_KEEP will leave @client untouched in @server. A value of #GST_RTSP_FILTER_REF will add @client to the result #GList of gst_rtsp_server_client_filter(). a #GstRTSPFilterResult. a #GstRTSPServer object a #GstRTSPClient in @server user data that has been given to gst_rtsp_server_client_filter() Session information kept by the server for a specific client. One client session, identified with a session id, can handle multiple medias identified with the url of a media. Create a new #GstRTSPSession instance with @sessionid. a new #GstRTSPSession a session id Allow @session to expire. This method must be called an equal amount of time as gst_rtsp_session_prevent_expire(). a #GstRTSPSession Gets the session media for @path, increasing its reference count. @matched will contain the number of matched characters of @path. the configuration for @path in @sess, should be unreferenced when no longer needed. a #GstRTSPSession the path for the media the amount of matched characters Call @func for each media in @sess. The result value of @func determines what happens to the media. @func will be called with @sess locked so no further actions on @sess can be performed from @func. If @func returns #GST_RTSP_FILTER_REMOVE, the media will be removed from @sess. If @func returns #GST_RTSP_FILTER_KEEP, the media will remain in @sess. If @func returns #GST_RTSP_FILTER_REF, the media will remain in @sess but will also be added with an additional ref to the result #GList of this function.. When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for all media. a GList with all media for which @func returned #GST_RTSP_FILTER_REF. After usage, each element in the #GList should be unreffed before the list is freed. a #GstRTSPSession a callback user data passed to @func Get the string that can be placed in the Session header field. the Session header of @session. g_free() after usage. a #GstRTSPSession Gets the session media for @path. @matched will contain the number of matched characters of @path. the configuration for @path in @sess. a #GstRTSPSession the path for the media the amount of matched characters Get the sessionid of @session. the sessionid of @session. The value remains valid as long as @session is alive. a #GstRTSPSession Get the timeout value of @session. the timeout of @session in seconds. a #GstRTSPSession Check if @session timeout out. Use gst_rtsp_session_is_expired_usec() instead. %TRUE if @session timed out a #GstRTSPSession the current system time Check if @session timeout out. %TRUE if @session timed out a #GstRTSPSession the current monotonic time Manage the media object @obj in @sess. @path will be used to retrieve this media from the session with gst_rtsp_session_get_media(). Ownership is taken from @media. a new @GstRTSPSessionMedia object. a #GstRTSPSession the path for the media a #GstRTSPMedia Get the amount of milliseconds till the session will expire. Use gst_rtsp_session_next_timeout_usec() instead. the amount of milliseconds since the session will time out. a #GstRTSPSession the current system time Get the amount of milliseconds till the session will expire. the amount of milliseconds since the session will time out. a #GstRTSPSession the current monotonic time Prevent @session from expiring. a #GstRTSPSession Release the managed @media in @sess, freeing the memory allocated by it. %TRUE if there are more media session left in @sess. a #GstRTSPSession a #GstRTSPMedia Configure @session for a timeout of @timeout seconds. The session will be cleaned up when there is no activity for @timeout seconds. a #GstRTSPSession the new timeout Update the last_access time of the session to the current time. a #GstRTSPSession This function will be called by the gst_rtsp_session_filter(). An implementation should return a value of #GstRTSPFilterResult. When this function returns #GST_RTSP_FILTER_REMOVE, @media will be removed from @sess. A return value of #GST_RTSP_FILTER_KEEP will leave @media untouched in @sess. A value of GST_RTSP_FILTER_REF will add @media to the result #GList of gst_rtsp_session_filter(). a #GstRTSPFilterResult. a #GstRTSPSession object a #GstRTSPSessionMedia in @sess user data that has been given to gst_rtsp_session_filter() State of a client session regarding a specific media identified by path. Create a new #GstRTSPSessionMedia that manages the streams in @media for @path. @media should be prepared. Ownership is taken of @media. a new #GstRTSPSessionMedia. the path the #GstRTSPMedia Fill @range with the next available min and max channels for interleaved transport. %TRUE on success. a #GstRTSPSessionMedia a #GstRTSPRange Get the base_time of the #GstRTSPMedia in @media the base_time of the media. a #GstRTSPSessionMedia Get the #GstRTSPMedia that was used when constructing @media the #GstRTSPMedia of @media. Remains valid as long as @media is valid. a #GstRTSPSessionMedia Retrieve the RTP-Info header string for all streams in @media with configured transports. The RTP-Info as a string or %NULL when no RTP-Info could be generated, g_free() after usage. a #GstRTSPSessionMedia Get the current RTSP state of @media. the current RTSP state of @media. a #GstRTSPSessionMedia Get a previously created #GstRTSPStreamTransport for the stream at @idx. a #GstRTSPStreamTransport that is valid until the session of @media is unreffed. a #GstRTSPSessionMedia the stream index Get a list of all available #GstRTSPStreamTransport in this session. a list of #GstRTSPStreamTransport, g_ptr_array_unref () after usage. a #GstRTSPSessionMedia Check if the path of @media matches @path. It @path matches, the amount of matched characters is returned in @matched. %TRUE when @path matches the path of @media. a #GstRTSPSessionMedia a path the amount of matched characters of @path Set the RTSP state of @media to @state. a #GstRTSPSessionMedia a #GstRTSPState Tell the media object @media to change to @state. %TRUE on success. a #GstRTSPSessionMedia the new state Configure the transport for @stream to @tr in @media. the new or updated #GstRTSPStreamTransport for @stream. a #GstRTSPSessionMedia a #GstRTSPStream a #GstRTSPTransport An object that keeps track of the active sessions. This object is usually attached to a #GstRTSPServer object to manage the sessions in that server. Create a new #GstRTSPSessionPool instance. A new #GstRTSPSessionPool. g_object_unref() after usage. Inspect all the sessions in @pool and remove the sessions that are inactive for more than their timeout. the amount of sessions that got removed. a #GstRTSPSessionPool Create a new #GstRTSPSession object in @pool. a new #GstRTSPSession. a #GstRTSPSessionPool Create a #GSource that will be dispatched when the session should be cleaned up. a #GSource a #GstRTSPSessionPool Call @func for each session in @pool. The result value of @func determines what happens to the session. @func will be called with the session pool locked so no further actions on @pool can be performed from @func. If @func returns #GST_RTSP_FILTER_REMOVE, the session will be set to the expired state and removed from @pool. If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @pool. If @func returns #GST_RTSP_FILTER_REF, the session will remain in @pool but will also be added with an additional ref to the result GList of this function.. When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for all sessions. a GList with all sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each element in the GList should be unreffed before the list is freed. a #GstRTSPSessionPool a callback user data passed to @func Find the session with @sessionid in @pool. The access time of the session will be updated with gst_rtsp_session_touch(). the #GstRTSPSession with @sessionid or %NULL when the session did not exist. g_object_unref() after usage. the pool to search the session id Get the maximum allowed number of sessions in @pool. 0 means an unlimited amount of sessions. the maximum allowed number of sessions. a #GstRTSPSessionPool Get the amount of active sessions in @pool. the amount of active sessions in @pool. a #GstRTSPSessionPool Remove @sess from @pool, releasing the ref that the pool has on @sess. %TRUE if the session was found and removed. a #GstRTSPSessionPool a #GstRTSPSession Configure the maximum allowed number of sessions in @pool to @max. A value of 0 means an unlimited amount of sessions. a #GstRTSPSessionPool the maximum number of sessions This function will be called by the gst_rtsp_session_pool_filter(). An implementation should return a value of #GstRTSPFilterResult. When this function returns #GST_RTSP_FILTER_REMOVE, @session will be removed from @pool. A return value of #GST_RTSP_FILTER_KEEP will leave @session untouched in @pool. A value of GST_RTSP_FILTER_REF will add @session to the result #GList of gst_rtsp_session_pool_filter(). a #GstRTSPFilterResult. a #GstRTSPSessionPool object a #GstRTSPSession in @pool user data that has been given to gst_rtsp_session_pool_filter() The function that will be called from the GSource watch on the session pool. The function will be called when the pool must be cleaned up because one or more sessions timed out. %FALSE if the source should be removed. a #GstRTSPSessionPool object user data that has been given when registering the handler The definition of a media stream. Create a new media stream with index @idx that handles RTP data on @pad and has a payloader element @payloader if @pad is a source pad or a depayloader element @payloader if @pad is a sink pad. a new #GstRTSPStream an index a #GstElement a #GstPad Add multicast client address to stream. At this point, the sockets that will stream RTP and RTCP data to @destination are supposed to be allocated. %TRUE if @destination can be addedd and handled by @stream. a #GstRTSPStream a multicast address to add RTP port RTCP port socket family Add the transport in @trans to @stream. The media of @stream will then also be send to the values configured in @trans. Adding the same transport twice will not add it a second time. @stream must be joined to a bin. @trans must contain a valid #GstRTSPTransport. %TRUE if @trans was added a #GstRTSPStream a #GstRTSPStreamTransport Allocates RTP and RTCP ports. %TRUE if the RTP and RTCP sockets have been succeccully allocated. a #GstRTSPStream protocol family transport method Whether to use client settings or not Add a receiver and sender part to the pipeline based on the transport from SETUP. %TRUE if the stream has been successfully updated. a #GstRTSPStream a #GstRTSPTransport Get the #GstRTSPAddressPool used as the address pool of @stream. the #GstRTSPAddressPool of @stream. g_object_unref() after usage. a #GstRTSPStream Get the size of the UDP transmission buffer (in bytes) the size of the UDP TX buffer a #GstRTSPStream Retrieve the current caps of @stream. the #GstCaps of @stream. use gst_caps_unref() after usage. a #GstRTSPStream Get the control string to identify this stream. the control string. g_free() after usage. a #GstRTSPStream Get the configured DSCP QoS in of the outgoing sockets. the DSCP QoS value of the outgoing sockets, or -1 if disbled. a #GstRTSPStream Get the stream index. the stream index. a #GstRTSPStream Get the previous joined bin with gst_rtsp_stream_join_bin() or NULL. the joined bin or NULL. a #GstRTSPStream Get the the maximum time-to-live value of outgoing multicast packets. the maximum time-to-live value of outgoing multicast packets. a #GstRTSPStream Get the configured MTU in the payloader of @stream. the MTU of the payloader. a #GstRTSPStream Get the multicast address of @stream for @family. The original #GstRTSPAddress is cached and copy is returned, so freeing the return value won't release the address from the pool. the #GstRTSPAddress of @stream or %NULL when no address could be allocated. gst_rtsp_address_free() after usage. a #GstRTSPStream the #GSocketFamily Get all multicast client addresses that RTP data will be sent to A comma separated list of host:port pairs with destinations a #GstRTSPStream Get the multicast interface used for @stream. the multicast interface for @stream. g_free() after usage. a #GstRTSPStream Get the allowed profiles of @stream. a #GstRTSPProfile a #GstRTSPStream Get the allowed protocols of @stream. a #GstRTSPLowerTrans a #GstRTSPStream Get the stream payload type. the stream payload type. a #GstRTSPStream Gets if and how the stream clock should be published according to RFC7273. The GstRTSPPublishClockMode a #GstRTSPStream whether @stream will follow the Rate-Control=no behaviour as specified in the ONVIF replay spec. Retrieve the current rate and/or applied_rate. %TRUE if rate and/or applied_rate could be determined. a #GstRTSPStream the configured rate the configured applied_rate Get the payload-type used for retransmission of this stream The retransmission PT. a #GstRTSPStream Get the amount of time to store retransmission data. the amount of time to store retransmission data. a #GstRTSPStream Get the multicast RTCP socket from @stream for a @family. the multicast RTCP socket or %NULL if no socket could be allocated for @family. Unref after usage a #GstRTSPStream the socket family Get the RTCP socket from @stream for a @family. @stream must be joined to a bin. the RTCP socket or %NULL if no socket could be allocated for @family. Unref after usage a #GstRTSPStream the socket family Get the multicast RTP socket from @stream for a @family. the multicast RTP socket or %NULL if no socket could be allocated for @family. Unref after usage a #GstRTSPStream the socket family Get the RTP socket from @stream for a @family. @stream must be joined to a bin. the RTP socket or %NULL if no socket could be allocated for @family. Unref after usage a #GstRTSPStream the socket family Retrieve the current rtptime, seq and running-time. This is used to construct a RTPInfo reply header. %TRUE when rtptime, seq and running-time could be determined. a #GstRTSPStream result RTP timestamp result RTP seqnum the clock rate result running-time Get the RTP session of this stream. The RTP session of this stream. Unref after usage. a #GstRTSPStream Fill @server_port with the port pair used by the server. This function can only be called when @stream has been joined. a #GstRTSPStream result server port the port family to get Get the sinkpad associated with @stream. the sinkpad. Unref after usage. a #GstRTSPStream Get the srcpad associated with @stream. the srcpad. Unref after usage. a #GstRTSPStream Get the SRTP encoder for this stream. The SRTP encoder for this stream. Unref after usage. a #GstRTSPStream Get the SSRC used by the RTP session of this stream. This function can only be called when @stream has been joined. a #GstRTSPStream result ssrc the amount of redundancy applied when creating ULPFEC protection packets. the payload type used for ULPFEC protection packets Parse and handle a KeyMgmt header. a #GstRTSPStream a keymgmt header Check if @stream has the control string @control. %TRUE is @stream has @control as the control string a #GstRTSPStream a control string Check if multicast sockets are configured to be bound to multicast addresses. %TRUE if multicast sockets are configured to be bound to multicast addresses. a #GstRTSPStream Check if @stream is blocking on a #GstBuffer. %TRUE if @stream is blocking a #GstRTSPStream See gst_rtsp_stream_set_client_side() TRUE if this #GstRTSPStream is client-side. a #GstRTSPStream Checks whether the stream is complete, contains the receiver and the sender parts. As the stream contains sink(s) element(s), it's possible to perform seek operations on it. %TRUE if the stream contains at least one sink element. a #GstRTSPStream Checks whether the stream is a receiver. %TRUE if the stream is a receiver and %FALSE otherwise. a #GstRTSPStream Checks whether the stream is a sender. %TRUE if the stream is a sender and %FALSE otherwise. a #GstRTSPStream Check if @transport can be handled by stream %TRUE if @transport can be handled by @stream. a #GstRTSPStream a #GstRTSPTransport Join the #GstBin @bin that contains the element @rtpbin. @stream will link to @rtpbin, which must be inside @bin. The elements added to @bin will be set to the state given in @state. %TRUE on success. a #GstRTSPStream a #GstBin to join a rtpbin element in @bin the target state of the new elements Remove the elements of @stream from @bin. %TRUE on success. a #GstRTSPStream a #GstBin a rtpbin #GstElement Query the position of the stream in %GST_FORMAT_TIME. This only considers the RTP parts of the pipeline and not the RTCP parts. %TRUE if the position could be queried a #GstRTSPStream current position of a #GstRTSPStream Query the stop of the stream in %GST_FORMAT_TIME. This only considers the RTP parts of the pipeline and not the RTCP parts. %TRUE if the stop could be queried a #GstRTSPStream current stop of a #GstRTSPStream Handle an RTCP buffer for the stream. This method is usually called when a message has been received from a client using the TCP transport. This function takes ownership of @buffer. a GstFlowReturn. a #GstRTSPStream a #GstBuffer Handle an RTP buffer for the stream. This method is usually called when a message has been received from a client using the TCP transport. This function takes ownership of @buffer. a GstFlowReturn. a #GstRTSPStream a #GstBuffer Remove the transport in @trans from @stream. The media of @stream will not be sent to the values configured in @trans. @stream must be joined to a bin. @trans must contain a valid #GstRTSPTransport. %TRUE if @trans was removed a #GstRTSPStream a #GstRTSPStreamTransport Creating a rtxreceive bin a #GstElement. a #GstRTSPStream the session id Creating a rtxsend bin a #GstElement. a #GstRTSPStream the session id Creating a rtpulpfecdec element a #GstElement. Creating a rtpulpfecenc element a #GstElement. Reserve @address and @port as the address and port of @stream. The original #GstRTSPAddress is cached and copy is returned, so freeing the return value won't release the address from the pool. the #GstRTSPAddress of @stream or %NULL when the address could not be reserved. gst_rtsp_address_free() after usage. a #GstRTSPStream an address a port n_ports a TTL Checks whether the individual @stream is seekable. %TRUE if @stream is seekable, else %FALSE. a #GstRTSPStream configure @pool to be used as the address pool of @stream. a #GstRTSPStream a #GstRTSPAddressPool Decide whether the multicast socket should be bound to a multicast address or INADDR_ANY. a #GstRTSPStream, the new value Blocks or unblocks the dataflow on @stream. %TRUE on success a #GstRTSPStream boolean indicating we should block or unblock Set the size of the UDP transmission buffer (in bytes) Needs to be set before the stream is joined to a bin. a #GstRTSPStream the buffer size Sets the #GstRTSPStream as a 'client side' stream - used for sending streams to an RTSP server via RECORD. This has the practical effect of changing which UDP port numbers are used when setting up the local side of the stream sending to be either the 'server' or 'client' pair of a configured UDP transport. a #GstRTSPStream TRUE if this #GstRTSPStream is running on the 'client' side of an RTSP connection. Set the control string in @stream. a #GstRTSPStream a control string Configure the dscp qos of the outgoing sockets to @dscp_qos. a #GstRTSPStream a new dscp qos value (0-63, or -1 to disable) Set the maximum time-to-live value of outgoing multicast packets. %TRUE if the requested ttl has been set successfully. a #GstRTSPStream the new multicast ttl value Configure the mtu in the payloader of @stream to @mtu. a #GstRTSPStream a new MTU configure @multicast_iface to be used for @stream. a #GstRTSPStream a multicast interface name Configure the allowed profiles for @stream. a #GstRTSPStream the new profiles Configure the allowed lower transport for @stream. a #GstRTSPStream the new flags Configure a pt map between @pt and @caps. a #GstRTSPStream the pt a #GstCaps Sets if and how the stream clock should be published according to RFC7273. a #GstRTSPStream the clock publish mode Define whether @stream will follow the Rate-Control=no behaviour as specified in the ONVIF replay spec. Set the payload type (pt) for retransmission of this stream. a #GstRTSPStream a #guint Set the amount of time to store retransmission packets. a #GstRTSPStream a #GstClockTime Sets the amount of redundancy to apply when creating ULPFEC protection packets. Set the payload type to be used for ULPFEC protection packets Call @func for each transport managed by @stream. The result value of @func determines what happens to the transport. @func will be called with @stream locked so no further actions on @stream can be performed from @func. If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from @stream. If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream. If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but will also be added with an additional ref to the result #GList of this function.. When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport. a #GList with all transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each element in the #GList should be unreffed before the list is freed. a #GstRTSPStream a callback user data passed to @func Remove blocking probe from the RTCP source. When creating an UDP source for RTCP it is initially blocked until this function is called. This functions should be called once the pipeline is ready for handling RTCP packets. Update the new crypto information for @ssrc in @stream. If information for @ssrc did not exist, it will be added. If information for @ssrc existed, it will be replaced. If @crypto is %NULL, it will be removed from @stream. %TRUE if @crypto could be updated a #GstRTSPStream the SSRC a #GstCaps with crypto info Check if the requested multicast ttl value is allowed. TRUE if the requested ttl value is allowed. a #GstRTSPStream a requested multicast ttl A Transport description for a stream Create a new #GstRTSPStreamTransport that can be used to manage @stream with transport @tr. a new #GstRTSPStreamTransport a #GstRTSPStream a GstRTSPTransport Get the RTP-Info string for @trans and @start_time. the RTPInfo string for @trans and @start_time or %NULL when the RTP-Info could not be determined. g_free() after usage. a #GstRTSPStreamTransport a star time Get the #GstRTSPStream used when constructing @trans. the stream used when constructing @trans. a #GstRTSPStreamTransport Get the transport configured in @trans. the transport configured in @trans. It remains valid for as long as @trans is valid. a #GstRTSPStreamTransport Get the url configured in @trans. the url configured in @trans. It remains valid for as long as @trans is valid. a #GstRTSPStreamTransport Check if @trans is timed out. %TRUE if @trans timed out. a #GstRTSPStreamTransport Signal the installed keep_alive callback for @trans. a #GstRTSPStreamTransport Signal the installed message_sent / message_sent_full callback for @trans. a #GstRTSPStreamTransport Receive @buffer on @channel @trans. a #GstFlowReturn. Returns GST_FLOW_NOT_LINKED when @channel is not configured in the transport of @trans. a #GstRTSPStreamTransport a channel a #GstBuffer Send @buffer to the installed RTCP callback for @trans. %TRUE on success a #GstRTSPStreamTransport a #GstBuffer Send @buffer_list to the installed RTCP callback for @trans. %TRUE on success a #GstRTSPStreamTransport a #GstBuffer Send @buffer to the installed RTP callback for @trans. %TRUE on success a #GstRTSPStreamTransport a #GstBuffer Send @buffer_list to the installed RTP callback for @trans. %TRUE on success a #GstRTSPStreamTransport a #GstBufferList Activate or deactivate datatransfer configured in @trans. %TRUE when the state was changed. a #GstRTSPStreamTransport new state of @trans Install callbacks that will be called when data for a stream should be sent to a client. This is usually used when sending RTP/RTCP over TCP. a #GstRTSPStreamTransport a callback called when RTP should be sent a callback called when RTCP should be sent user data passed to callbacks called with the user_data when no longer needed. Install callbacks that will be called when RTCP packets are received from the receiver of @trans. a #GstRTSPStreamTransport a callback called when the receiver is active user data passed to callback called with the user_data when no longer needed. Install callbacks that will be called when data for a stream should be sent to a client. This is usually used when sending RTP/RTCP over TCP. a #GstRTSPStreamTransport a callback called when RTP should be sent a callback called when RTCP should be sent user data passed to callbacks called with the user_data when no longer needed. Install a callback that will be called when a message has been sent on @trans. a #GstRTSPStreamTransport a callback called when a message has been sent user data passed to callback called with the user_data when no longer needed Install a callback that will be called when a message has been sent on @trans. a #GstRTSPStreamTransport a callback called when a message has been sent user data passed to callback called with the user_data when no longer needed Set the timed out state of @trans to @timedout a #GstRTSPStreamTransport timed out value Set @tr as the client transport. This function takes ownership of the passed @tr. a #GstRTSPStreamTransport a client #GstRTSPTransport Set @url as the client url. a #GstRTSPStreamTransport a client #GstRTSPUrl parent instance This function will be called by the gst_rtsp_stream_transport_filter(). An implementation should return a value of #GstRTSPFilterResult. When this function returns #GST_RTSP_FILTER_REMOVE, @trans will be removed from @stream. A return value of #GST_RTSP_FILTER_KEEP will leave @trans untouched in @stream. A value of #GST_RTSP_FILTER_REF will add @trans to the result #GList of gst_rtsp_stream_transport_filter(). a #GstRTSPFilterResult. a #GstRTSPStream object a #GstRTSPStreamTransport in @stream user data that has been given to gst_rtsp_stream_transport_filter() 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. Media is not suspended Media is PAUSED in suspend The media is set to NULL when suspended Structure holding info about a mainloop running in a thread parent #GstMiniObject the thread type a #GMainContext a #GMainLoop Create a new thread object that can run a mainloop. a #GstRTSPThread. the thread type Reuse the mainloop of @thread %TRUE if the mainloop could be reused a #GstRTSPThread Stop and unref @thread. When no threads are using the mainloop, the thread will be stopped and the final ref to @thread will be released. a #GstRTSPThread The thread pool structure. Create a new #GstRTSPThreadPool instance. a new #GstRTSPThreadPool Wait for all tasks to be stopped and free all allocated resources. This is mainly used in test suites to ensure proper cleanup of internal data structures. Get a new #GstRTSPThread for @type and @ctx. a new #GstRTSPThread, gst_rtsp_thread_stop() after usage a #GstRTSPThreadPool the #GstRTSPThreadType a #GstRTSPContext Get the maximum number of threads used for client connections. See gst_rtsp_thread_pool_set_max_threads(). the maximum number of threads. a #GstRTSPThreadPool Get a new #GstRTSPThread for @type and @ctx. a new #GstRTSPThread, gst_rtsp_thread_stop() after usage a #GstRTSPThreadPool the #GstRTSPThreadType a #GstRTSPContext Set the maximum threads used by the pool to handle client requests. A value of 0 will use the pool mainloop, a value of -1 will use an unlimited number of threads. a #GstRTSPThreadPool maximum threads Class for managing threads. a #GThreadPool used internally a new #GstRTSPThread, gst_rtsp_thread_stop() after usage a #GstRTSPThreadPool the #GstRTSPThreadType a #GstRTSPContext Different thread types a thread to handle the client communication a thread to handle media An opaque object used for checking authorisations. It is generated after successful authentication. Create a new Authorization token with the given fieldnames and values. Arguments are given similar to gst_structure_new(). a new authorization token. the first fieldname additional arguments Create a new empty Authorization token. a new empty authorization token. Create a new Authorization token with the given fieldnames and values. Arguments are given similar to gst_structure_new_valist(). a new authorization token. the first fieldname additional arguments Get the string value of @field in @token. the string value of @field in @token or %NULL when @field is not defined in @token. The string becomes invalid when you free @token. a #GstRTSPToken a field name Access the structure of the token. The structure of the token. The structure is still owned by the token, which means that you should not free it and that the pointer becomes invalid when you free the token. MT safe. The #GstRTSPToken. Check if @token has a boolean @field and if it is set to %TRUE. %TRUE if @token has a boolean field named @field set to %TRUE. a #GstRTSPToken a field name Sets a boolean value on @token. The #GstRTSPToken. field to set boolean value to set Sets a string value on @token. The #GstRTSPToken. field to set string value to set Get a writable version of the structure. The structure of the token. The structure is still owned by the token, which means that you should not free it and that the pointer becomes invalid when you free the token. This function ensures that @token is writable, and if so, will never return %NULL. MT safe. A writable #GstRTSPToken. The supported modes of the media. Transport supports PLAY mode Transport supports RECORD mode Used with gst_rtsp_address_pool_add_range() to bind to all IPv4 addresses Used with gst_rtsp_address_pool_add_range() to bind to all IPv6 addresses Check a new connection Check if access is allowed to a factory. When access is not allowed an 404 Not Found is sent in the response. Check if media can be constructed from a media factory A response should be sent on error. Check if the client can specify TTL, destination and port pair in multicast. No response is sent when the check returns %FALSE. Check the URL and methods G_TYPE_BOOLEAN, %TRUE if the media can be accessed, %FALSE will return a 404 Not Found error when trying to access the media. G_TYPE_BOOLEAN, %TRUE if the media can be constructed, %FALSE will return a 404 Not Found error when trying to access the media. G_TYPE_STRING, the role to use when dealing with media factories The default #GstRTSPAuth object uses this string in the token to find the role of the media factory. It will then retrieve the #GstRTSPPermissions of the media factory and retrieve the role with the same name. G_TYPE_BOOLEAN, %TRUE if the client can specify TTL, destination and port pair in multicast. The #GstRTSPAddressPool is an object that maintains a collection of network addresses. It is used to allocate server ports and server multicast addresses but also to reserve client provided destination addresses. A range of addresses can be added with gst_rtsp_address_pool_add_range(). Both multicast and unicast addresses can be added. With gst_rtsp_address_pool_acquire_address() an unused address and port range can be acquired from the pool. With gst_rtsp_address_pool_reserve_address() a specific address can be retrieved. Both methods return a boxed #GstRTSPAddress that should be freed with gst_rtsp_address_free() after usage, which brings the address back into the pool. Last reviewed on 2013-07-16 (1.0.0) The #GstRTSPAuth object is responsible for checking if the current user is allowed to perform requested actions. The default implementation has some reasonable checks but subclasses can implement custom security policies. A new auth object is made with gst_rtsp_auth_new(). It is usually configured on the #GstRTSPServer object. The RTSP server will call gst_rtsp_auth_check() with a string describing the check to perform. The possible checks are prefixed with GST_RTSP_AUTH_CHECK_*. Depending on the check, the default implementation will use the current #GstRTSPToken, #GstRTSPContext and #GstRTSPPermissions on the object to check if an operation is allowed. The default #GstRTSPAuth object has support for basic authentication. With gst_rtsp_auth_add_basic() you can add a basic authentication string together with the #GstRTSPToken that will become active when successfully authenticated. When a TLS certificate has been set with gst_rtsp_auth_set_tls_certificate(), the default auth object will require the client to connect with a TLS connection. Last reviewed on 2013-07-16 (1.0.0) The client object handles the connection with a client for as long as a TCP connection is open. A #GstRTSPClient is created by #GstRTSPServer when a new connection is accepted and it inherits the #GstRTSPMountPoints, #GstRTSPSessionPool, #GstRTSPAuth and #GstRTSPThreadPool from the server. The client connection should be configured with the #GstRTSPConnection using gst_rtsp_client_set_connection() before it can be attached to a #GMainContext using gst_rtsp_client_attach(). From then on the client will handle requests on the connection. Use gst_rtsp_client_session_filter() to iterate or modify all the #GstRTSPSession objects managed by the client object. Last reviewed on 2013-07-11 (1.0.0) Last reviewed on 2013-07-11 (1.0.0) 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(), or gst_rtsp_media_seek_full() or gst_rtsp_media_seek_trickmode() for finer control of the 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) The #GstRTSPMediaFactory is responsible for creating or recycling #GstRTSPMedia objects based on the passed URL. The default implementation of the object can create #GstRTSPMedia objects containing a pipeline created from a launch description set with gst_rtsp_media_factory_set_launch(). Media from a factory can be shared by setting the shared flag with gst_rtsp_media_factory_set_shared(). When a factory is shared, gst_rtsp_media_factory_construct() will return the same #GstRTSPMedia when the url matches. Last reviewed on 2013-07-11 (1.0.0) This specialized #GstRTSPMediaFactory constructs media pipelines from a URI, given with gst_rtsp_media_factory_uri_set_uri(). It will automatically demux and payload the different streams found in the media at URL. Last reviewed on 2013-07-11 (1.0.0) A #GstRTSPMountPoints object maintains a relation between paths and #GstRTSPMediaFactory objects. This object is usually given to #GstRTSPClient and used to find the media attached to a path. With gst_rtsp_mount_points_add_factory () and gst_rtsp_mount_points_remove_factory(), factories can be added and removed. With gst_rtsp_mount_points_match() you can find the #GstRTSPMediaFactory object that completely matches the given path. Last reviewed on 2013-07-11 (1.0.0) a #GstRTSPOnvifMedia 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. On top of #GstRTSPMedia this subclass adds special ONVIF features. Special ONVIF features that are currently supported is a backchannel for the client to send back media to the server in a normal PLAY media. To handle the ONVIF backchannel, a #GstRTSPOnvifMediaFactory and #GstRTSPOnvifServer has to be used. The #GstRTSPOnvifMediaFactory is responsible for creating or recycling #GstRTSPMedia objects based on the passed URL. Different to #GstRTSPMediaFactory, this supports special ONVIF features and can create #GstRTSPOnvifMedia in addition to normal #GstRTSPMedia. Special ONVIF features that are currently supported is a backchannel for the client to send back media to the server in a normal PLAY media, see gst_rtsp_onvif_media_factory_set_backchannel_launch() and gst_rtsp_onvif_media_factory_set_backchannel_bandwidth(). The server object is the object listening for connections on a port and creating #GstRTSPOnvifClient objects to handle those connections. The only different to #GstRTSPServer is that #GstRTSPOnvifServer creates #GstRTSPOnvifClient that have special handling for ONVIF specific features, like a backchannel that allows clients to send back media to the server. Last reviewed on 2013-07-11 (1.0.0) The #GstRTSPPermissions object contains an array of roles and associated permissions. The roles are represented with a string and the permissions with a generic #GstStructure. The permissions are deliberately kept generic. The possible values of the roles and #GstStructure keys and values are only determined by the #GstRTSPAuth object that performs the checks on the permissions and the current #GstRTSPToken. As a convenience function, gst_rtsp_permissions_is_allowed() can be used to check if the permissions contains a role that contains the boolean value %TRUE for the the given key. Last reviewed on 2013-07-15 (1.0.0) Last reviewed on 2013-07-11 (1.0.0) The server object is the object listening for connections on a port and creating #GstRTSPClient objects to handle those connections. The server will listen on the address set with gst_rtsp_server_set_address() and the port or service configured with gst_rtsp_server_set_service(). Use gst_rtsp_server_set_backlog() to configure the amount of pending requests that the server will keep. By default the server listens on the current network (0.0.0.0) and port 8554. The server will require an SSL connection when a TLS certificate has been set in the auth object with gst_rtsp_auth_set_tls_certificate(). To start the server, use gst_rtsp_server_attach() to attach it to a #GMainContext. For more control, gst_rtsp_server_create_source() and gst_rtsp_server_create_socket() can be used to get a #GSource and #GSocket respectively. gst_rtsp_server_transfer_connection() can be used to transfer an existing socket to the RTSP server, for example from an HTTP server. Once the server socket is attached to a mainloop, it will start accepting connections. When a new connection is received, a new #GstRTSPClient object is created to handle the connection. The new client will be configured with the server #GstRTSPAuth, #GstRTSPMountPoints, #GstRTSPSessionPool and #GstRTSPThreadPool. The server uses the configured #GstRTSPThreadPool object to handle the remainder of the communication with this client. Last reviewed on 2013-07-11 (1.0.0) The #GstRTSPSession is identified by an id, unique in the #GstRTSPSessionPool that created the session and manages media and its configuration. A #GstRTSPSession has a timeout that can be retrieved with gst_rtsp_session_get_timeout(). You can check if the sessions is expired with gst_rtsp_session_is_expired(). gst_rtsp_session_touch() will reset the expiration counter of the session. When a client configures a media with SETUP, a session will be created to keep track of the configuration of that media. With gst_rtsp_session_manage_media(), the media is added to the managed media in the session. With gst_rtsp_session_release_media() the media can be released again from the session. Managed media is identified in the sessions with a url. Use gst_rtsp_session_get_media() to get the media that matches (part of) the given url. The media in a session can be iterated with gst_rtsp_session_filter(). Last reviewed on 2013-07-11 (1.0.0) The #GstRTSPSessionMedia object manages a #GstRTSPMedia with a given path. With gst_rtsp_session_media_get_transport() and gst_rtsp_session_media_set_transport() the transports of a #GstRTSPStream of the managed #GstRTSPMedia can be retrieved and configured. Use gst_rtsp_session_media_set_state() to control the media state and transports. Last reviewed on 2013-07-16 (1.0.0) The #GstRTSPSessionPool object manages a list of #GstRTSPSession objects. The maximum number of sessions can be configured with gst_rtsp_session_pool_set_max_sessions(). The current number of sessions can be retrieved with gst_rtsp_session_pool_get_n_sessions(). Use gst_rtsp_session_pool_create() to create a new #GstRTSPSession object. The session object can be found again with its id and gst_rtsp_session_pool_find(). All sessions can be iterated with gst_rtsp_session_pool_filter(). Run gst_rtsp_session_pool_cleanup() periodically to remove timed out sessions or use gst_rtsp_session_pool_create_watch() to be notified when session cleanup should be performed. Last reviewed on 2013-07-11 (1.0.0) The #GstRTSPStream object manages the data transport for one stream. It is created from a payloader element and a source pad that produce the RTP packets for the stream. With gst_rtsp_stream_join_bin() the streaming elements are added to the bin and rtpbin. gst_rtsp_stream_leave_bin() removes the elements again. The #GstRTSPStream will use the configured addresspool, as set with gst_rtsp_stream_set_address_pool(), to allocate multicast addresses for the stream. With gst_rtsp_stream_get_multicast_address() you can get the configured address. With gst_rtsp_stream_get_server_port () you can get the port that the server will use to receive RTCP. This is the part that the clients will use to send RTCP to. With gst_rtsp_stream_add_transport() destinations can be added where the stream should be sent to. Use gst_rtsp_stream_remove_transport() to remove the destination again. Each #GstRTSPStreamTransport spawns one queue that will serve as a backlog of a controllable maximum size when the reflux from the TCP connection's backpressure starts spilling all over. Unlike the backlog in rtspconnection, which we have decided should only contain at most one RTP and one RTCP data message in order to allow control messages to go through unobstructed, this backlog only consists of data messages, allowing us to fill it up without concern. When multiple TCP transports exist, for example in the context of a shared media, we only pop samples from our appsinks when at least one of the transports doesn't experience back pressure: this allows us to pace our sample popping to the speed of the fastest client. When a sample is popped, it is either sent directly on transports that don't experience backpressure, or queued on the transport's backlog otherwise. Samples are then popped from that backlog when the transport reports it has sent the message. Once the backlog reaches an overly large duration, the transport is dropped as the client was deemed too slow. The #GstRTSPStreamTransport configures the transport used by a #GstRTSPStream. It is usually manages by a #GstRTSPSessionMedia object. With gst_rtsp_stream_transport_set_callbacks(), callbacks can be configured to handle the RTP and RTCP packets from the stream, for example when they need to be sent over TCP. With gst_rtsp_stream_transport_set_active() the transports are added and removed from the stream. A #GstRTSPStream will call gst_rtsp_stream_transport_keep_alive() when RTCP is received from the client. It will also call gst_rtsp_stream_transport_set_timed_out() when a receiver has timed out. A #GstRTSPClient will call gst_rtsp_stream_transport_message_sent() when it has sent a data message for the transport. Last reviewed on 2013-07-16 (1.0.0) A #GstRTSPThreadPool manages reusable threads for various server tasks. Currently the defined thread types can be found in #GstRTSPThreadType. Threads of type #GST_RTSP_THREAD_TYPE_CLIENT are used to handle requests from a connected client. With gst_rtsp_thread_pool_get_max_threads() a maximum number of threads can be set after which the pool will start to reuse the same thread for multiple clients. Threads of type #GST_RTSP_THREAD_TYPE_MEDIA will be used to perform the state changes of the media pipelines and handle its bus messages. gst_rtsp_thread_pool_get_thread() can be used to create a #GstRTSPThread object of the right type. The thread object contains a mainloop and context that run in a seperate thread and can be used to attached sources to. gst_rtsp_thread_reuse() can be used to reuse a thread for multiple purposes. If all gst_rtsp_thread_reuse() calls are matched with a gst_rtsp_thread_stop() call, the mainloop will be quit and the thread will stop. To configure the threads, a subclass of this object should be made and the virtual methods should be overriden to implement the desired functionality. Last reviewed on 2013-07-11 (1.0.0) A #GstRTSPToken contains the permissions and roles of the user performing the current request. A token is usually created when a user is authenticated by the #GstRTSPAuth object and is then placed as the current token for the current request. #GstRTSPAuth can use the token and its contents to check authorization for various operations by comparing the token to the #GstRTSPPermissions of the object. The accepted values of the token are entirely defined by the #GstRTSPAuth object that implements the security policy. Last reviewed on 2013-07-15 (1.0.0) Get the current #GstRTSPContext. This object is retrieved from the current thread that is handling the request for a client. a #GstRTSPContext Get parameters (not implemented yet) a #GstRTSPResult a #GstRTSPClient a #GstRTSPContext Set parameters (not implemented yet) a #GstRTSPResult a #GstRTSPClient a #GstRTSPContext Add @media specific info to @sdp. @info is used to configure the connection information in the SDP. TRUE on success. a #GstSDPMessage a #GstSDPInfo a #GstRTSPMedia Add info from @stream to @sdp. TRUE on success. a #GstSDPMessage a #GstSDPInfo a #GstRTSPStream Creates a #GstSDPMedia from the parameters and stores it in @sdp. %TRUE on success a #GstRTSPMessage a #GstSDPInfo a #GstRTSPStream a #GstCaps a #GstRTSPProfile