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GstCaps * | allowed-sink-caps | Read |
GstCaps * | allowed-src-caps | Read |
FsCodecGList * | codec-preferences | Read |
FsCodecGList * | codecs | Read |
FsCodecGList * | codecs-without-config | Read |
FsConference * | conference | Read / Write / Construct Only |
FsCodec * | current-send-codec | Read |
GstStructure * | encryption-parameters | Read |
guint | id | Read / Write / Construct Only |
FsMediaType | media-type | Read / Write / Construct Only |
GstPad * | sink-pad | Read |
guint | tos | Read / Write |
This object is the base implementation of a Farstream Session. It needs to be derived and implemented by a farstream conference gstreamer element. A Farstream session is defined in the same way as an RTP session. It can contain one or more participants but represents only one media stream (i.e. One session for video and one session for audio in an AV conference). Sessions contained in the same conference will be synchronised together during playback.
This will communicate asynchronous events to the user through GstMessage of type GST_MESSAGE_ELEMENT sent over the GstBus.
farstream-send-codec-changed
"
message"session" |
FsSession | The session that emits the message |
"codec" |
FsCodec | The new send codec |
"secondary-codecs" |
GList | A GList of FsCodec (to be freed with fs_codec_list_destroy() )
|
This message is sent on the bus when the value of the “current-send-codec” property changes.
farstream-codecs-changed
"
message"session" |
FsSession | The session that emits the message |
This message is sent on the bus when the value of the “codecs” or “codecs-without-config” properties change. If one is using codecs that have configuration data that needs to be transmitted reliably, one should fetch “codecs”, otherwise, “codecs-without-config” should be enough.
farstream-telephony-event-started
"
message"session" |
FsSession | The session that emits the message |
"method" |
FsDTMFMethod | The method used to send the DTMF |
"event" |
FSDTMFEvent | The event number |
"volume" |
guchar | The volume of the event |
This message is emitted after a succesful call to
fs_session_start_telephony_event()
to inform the application that the
telephony event has started.
farstream-telephony-event-stopped
"
message"session" |
FsSession | The session that emits the message |
"method" |
FsDTMFMethod | The method used to send the DTMF |
This message is emitted after a succesful call to
fs_session_stop_telephony_event()
to inform the application that the
telephony event has stopped.
FsStream * fs_session_new_stream (FsSession *session
,FsParticipant *participant
,FsStreamDirection direction
,GError **error
);
This function creates a stream for the given participant into the active session.
session |
||
participant |
FsParticipant of a participant for the new stream |
|
direction |
FsStreamDirection describing the direction of the new stream that will be created for this participant |
|
error |
gboolean fs_session_set_codec_preferences (FsSession *session
,GList *codec_preferences
,GError **error
);
Set the list of desired codec preferences. The user may change this value during an ongoing session. Note that doing this can cause the codecs to change. Therefore this requires the user to fetch the new codecs and renegotiate them with the peers. It is a GList of FsCodec. The changes are immediately effective. The function does not take ownership of the list.
The payload type may be a valid dynamic PT (96-127), FS_CODEC_ID_DISABLE
or FS_CODEC_ID_ANY
. If the encoding name is "reserve-pt", then the
payload type of the codec will be "reserved" and not be used by any
dynamically assigned payload type.
If the list of specifications would invalidate all codecs, an error will be returned.
void
fs_session_destroy (FsSession *session
);
This will cause the session to remove all links to other objects and to remove itself from the FsConference, it will also destroy all FsStream inside this FsSession Once a FsSession has been destroyed, it can not be used anymore.
It is strongly recommended to call this function from the main thread because releasing the application's reference to a session.
gboolean fs_session_start_telephony_event (FsSession *session
,guint8 event
,guint8 volume
);
This function will start sending a telephony event (such as a DTMF
tone) on the FsSession. You have to call the function
fs_session_stop_telephony_event()
to stop it.
If this function returns TRUE
, a "farstream-telephony-event-started" will
always be emitted when the event is actually played out.
gboolean
fs_session_stop_telephony_event (FsSession *session
);
This function will stop sending a telephony event started by
fs_session_start_telephony_event()
. If the event was being sent
for less than 50ms, it will be sent for 50ms minimum. If the
duration was a positive and the event is not over, it will cut it
short.
If this function returns TRUE
, a "farstream-telephony-event-stopped" will
always be emitted when the event is actually stopped.
gboolean fs_session_set_send_codec (FsSession *session
,FsCodec *send_codec
,GError **error
);
This function will set the currently being sent codec for all streams in this
session. The given FsCodec must be taken directly from the codecs
property of the session. If the given codec is not in the codecs
list, error
will be set and FALSE
will be returned. The send_codec
will be
copied so it must be free'd using fs_codec_destroy()
when done.
gchar **
fs_session_list_transmitters (FsSession *session
);
Get the list of all available transmitters for this session.
a newly-allocagted NULL
terminated array of
named of transmitters or NULL
if no transmitter is needed for this type of
session. It should be freed with g_strfreev()
.
GType fs_session_get_stream_transmitter_type (FsSession *session
,const gchar *transmitter
);
Returns the GType of the stream transmitter, bindings can use it
to validate/convert the parameters passed to fs_session_new_stream()
.
GList * fs_session_codecs_need_resend (FsSession *session
,GList *old_codecs
,GList *new_codecs
);
Some codec updates need to be reliably transmitted to the other side because they contain important parameters required to decode the media. Other codec updates, caused by user action, don't.
session |
||
old_codecs |
Codecs previously retrieved from the “codecs” property. |
[element-type FsCodec][transfer none][allow-none] |
new_codecs |
Codecs recently retrieved from the “codecs” property. |
[element-type FsCodec][transfer none][allow-none] |
A new GList of
FsCodec that need to be resent or NULL
if there are none. This
list must be freed with fs_codec_list_destroy()
.
[element-type FsCodec][transfer full]
gboolean fs_session_set_allowed_caps (FsSession *session
,GstCaps *sink_caps
,GstCaps *src_caps
,GError **error
);
Sets the allowed caps for the sink and source pads for this FsSession.
Only codecs that can take the input specified by the sink_caps
and
can produce output as specified by the src_caps
will be produced
in the “codecs” property and so only those will be negotiated.
If NULL
is passed to either src_caps
or sink_caps
, it is not changed.
The default is "video/x-raw" for a video stream, "audio/x-raw" for an audio stream and "ANY" for an application stream.
The values can be retrived using the “allowed-src-caps” and “allowed-sink-caps” properties.
session |
||
sink_caps |
Caps for the sink pad or |
[allow-none] |
src_caps |
Caps for the src pad or |
[allow-none] |
error |
Since: UNRELEASED
gboolean fs_session_set_encryption_parameters (FsSession *session
,GstStructure *parameters
,GError **error
);
Sets encryption parameters. The exact parameters depend on the type of plugin being used.
session |
||
parameters |
a GstStructure containing the
encryption parameters or |
[transfer none][allow-none] |
error |
Since: UNRELEASED
void fs_session_emit_error (FsSession *session
,gint error_no
,const gchar *error_msg
);
This function emit the "error" signal on a FsSession, it should only be called by subclasses.
gboolean fs_session_parse_codecs_changed (FsSession *session
,GstMessage *message
);
Parses a "farstream-codecs-changed" message and checks if it matches
the session
parameters.
gboolean fs_session_parse_send_codec_changed (FsSession *session
,GstMessage *message
,FsCodec **codec
,GList **secondary_codecs
);
Parses a "farstream-send-codec-changed" message and checks if it matches
the session
parameters.
session |
a FsSession to match against the message |
|
message |
a GstMessage to parse |
|
codec |
[out][transfer none] | |
secondary_codecs |
[out][transfer none][element-type FsCodec] |
gboolean fs_session_parse_telephony_event_started (FsSession *session
,GstMessage *message
,FsDTMFMethod *method
,FsDTMFEvent *event
,guint8 *volume
);
Parses a "farstream-telephony-event-started" message and checks if it matches
the session
parameters.
session |
a FsSession to match against the message |
|
message |
a GstMessage to parse |
|
method |
Returns the FsDTMFMethod in the message if not |
[out] |
event |
Returns the FsDTMFEvent in the message if not |
[out] |
volume |
Returns the volume in the message if not |
[out] |
gboolean fs_session_parse_telephony_event_stopped (FsSession *session
,GstMessage *message
,FsDTMFMethod *method
);
Parses a "farstream-telephony-event-stopped" message and checks if it matches
the session
parameters.
session |
a FsSession to match against the message |
|
message |
a GstMessage to parse |
|
method |
Returns the FsDTMFMethod in the message if not |
[out] |
struct FsSession;
All members are private, access them using methods and properties
struct FsSessionClass { GObjectClass parent_class; /*virtual functions */ FsStream *(* new_stream) (FsSession *session, FsParticipant *participant, FsStreamDirection direction, GError **error); gboolean (* start_telephony_event) (FsSession *session, guint8 event, guint8 volume); gboolean (* stop_telephony_event) (FsSession *session); gboolean (* set_send_codec) (FsSession *session, FsCodec *send_codec, GError **error); gboolean (* set_codec_preferences) (FsSession *session, GList *codec_preferences, GError **error); gchar** (* list_transmitters) (FsSession *session); GType (* get_stream_transmitter_type) (FsSession *session, const gchar *transmitter); GList* (* codecs_need_resend) (FsSession *session, GList *old_codecs, GList *new_codecs); gboolean (* set_allowed_caps) (FsSession *session, GstCaps *sink_caps, GstCaps *src_caps, GError **error); gboolean (* set_encryption_parameters) (FsSession *session, GstStructure *parameters, GError **error); };
You must override at least new_stream in a subclass.
GObjectClass |
Our parent |
|
Create a new FsStream |
||
Starts a telephony event |
||
Stops a telephony event |
||
Forces sending with a specific codec |
||
Specifies the codec preferences |
||
Returns a list of the available transmitters |
||
Returns the GType of the stream transmitter |
||
Returns the list of codecs that need resending |
||
Set the possible allowed src and sink caps |
||
Set encryption parameters |
An enum that represents the different DTMF event that can be sent to a FsSession. The values corresponds those those defined in RFC 4733 The rest of the possibles values are in the IANA registry at: http://www.iana.org/assignments/audio-telephone-event-registry
“allowed-sink-caps”
property“allowed-sink-caps” GstCaps *
These are the GstCaps that can be fed into the session, they are used to filter the codecs to only those that can accepted those caps as input.
Flags: Read
“allowed-src-caps”
property“allowed-src-caps” GstCaps *
These are the GstCaps that the session can produce, they are used to filter the codecs to only those that can accepted those caps as output.
Flags: Read
“codec-preferences”
property “codec-preferences” FsCodecGList *
This is the current preferences list for the local codecs. It is
set by the user to specify the codec options and priorities. The user may
change its value with fs_session_set_codec_preferences()
at any time
during a session. It is a GList of FsCodec.
The user must free this codec list using fs_codec_list_destroy()
when done.
The payload type may be a valid dynamic PT (96-127), FS_CODEC_ID_DISABLE
or FS_CODEC_ID_ANY
. If the encoding name is "reserve-pt", then the
payload type of the codec will be "reserved" and not be used by any
dynamically assigned payload type.
Flags: Read
“codecs”
property “codecs” FsCodecGList *
This is the list of codecs used for this session. It will include the codecs and payload type used to receive media on this session. It will also include any configuration parameter that must be transmitted reliably for the other end to decode the content.
It may change when the codec preferences are set, when codecs are set on a FsStream in this session, when a FsStream is destroyed or asynchronously when new config data is discovered.
If any configuration parameter needs to be discovered, this property
will be NULL
until they have been discovered. One can always get
the codecs from “codecs-without-config”.
The "farstream-codecs-changed" message will be emitted whenever the value
of this property changes.
It is a GList of FsCodec. User must free this codec list using
fs_codec_list_destroy()
when done.
Flags: Read
“codecs-without-config”
property “codecs-without-config” FsCodecGList *
This is the same list of codecs as “codecs” without the configuration information that describes the data sent. It is suitable for configurations where a list of codecs is shared by many senders. If one is using codecs such as Theora, Vorbis or H.264 that require such information to be transmitted, the configuration data should be included in the stream and retransmitted regularly.
It may change when the codec preferences are set, when codecs are set on a FsStream in this session, when a FsStream is destroyed or asynchronously when new config data is discovered.
The "farstream-codecs-changed" message will be emitted whenever the value of this property changes.
It is a GList of FsCodec. User must free this codec list using
fs_codec_list_destroy()
when done.
Flags: Read
“conference”
property“conference” FsConference *
The FsConference parent of this session. This property is a construct param and is read-only.
Flags: Read / Write / Construct Only
“current-send-codec”
property“current-send-codec” FsCodec *
Indicates the currently active send codec. A user can change the active
send codec by calling fs_session_set_send_codec()
. The send codec could
also be automatically changed by Farstream. This property is an
FsCodec. User must free the codec using fs_codec_destroy()
when done.
The "farstream-send-codec-changed" message is emitted on the bus when
the value of this property changes.
Flags: Read
“encryption-parameters”
property“encryption-parameters” GstStructure *
Retrieves previously set encryption parameters
Flags: Read
“id”
property“id” guint
The ID of the session, the first number of the pads linked to this session will be this id
Flags: Read / Write / Construct Only
Default value: 0
“media-type”
property“media-type” FsMediaType
The media-type of the session. This is either Audio, Video or both. This is a constructor parameter that cannot be changed.
Flags: Read / Write / Construct Only
Default value: FS_MEDIA_TYPE_AUDIO
“sink-pad”
property“sink-pad” GstPad *
The Gstreamer sink pad that must be used to send media data on this session. User must unref this GstPad when done with it.
Flags: Read
“tos”
property“tos” guint
Sets the IP ToS field (and if possible the IPv6 TCLASS field
Flags: Read / Write
Allowed values: <= 255
Default value: 0
“error”
signalvoid user_function (FsSession *self, GObject *object, FsError error_no, gchar *error_msg, gpointer user_data)
This signal is emitted in any error condition, it can be emitted on any thread. Applications should listen to the GstBus for errors.
self |
FsSession that emitted the signal |
|
object |
The Gobject that emitted the signal |
|
error_no |
The number of the error |
|
error_msg |
Error message |
|
user_data |
user data set when the signal handler was connected. |
Flags: Run Last