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Farsight2 Reference Manual | ![]() |
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#include <gst/farsight/fs-conference-iface.h> enum FsDTMFEvent; enum FsDTMFMethod; FsSession; FsSessionClass; FsStream* fs_session_new_stream (FsSession *session, FsParticipant *participant, FsStreamDirection direction, const gchar *transmitter, guint stream_transmitter_n_parameters, GParameter *stream_transmitter_parameters, GError **error); gboolean fs_session_start_telephony_event (FsSession *session, guint8 event, guint8 volume, FsDTMFMethod method); gboolean fs_session_stop_telephony_event (FsSession *session, FsDTMFMethod method); gboolean fs_session_set_send_codec (FsSession *session, FsCodec *send_codec, GError **error); gboolean fs_session_set_codec_preferences (FsSession *session, GList *codec_preferences, GError **error); gchar** fs_session_list_transmitters (FsSession *session); GType fs_session_get_stream_transmitter_type (FsSession *session, const gchar *transmitter); void fs_session_emit_error (FsSession *session, gint error_no, const gchar *error_msg, const gchar *debug_msg);
"codec-preferences" FsCodecGList* : Read "codecs" FsCodecGList* : Read "codecs-ready" gboolean : Read "codecs-without-config" FsCodecGList* : Read "current-send-codec" FsCodec* : Read "id" guint : Read / Write / Construct Only "media-type" FsMediaType : Read / Write / Construct Only "sink-pad" GstPad* : Read "tos" guint : Read / Write
This object is the base implementation of a Farsight Session. It needs to be derived and implemented by a farsight conference gstreamer element. A Farsight 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.
farsight-send-codec-changed
"
messageThis message is sent on the bus when the value of the "current-send-codec" property changes.
farsight-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, once should check the value of "codecs-ready"
property to make sure all of the codecs configuration are ready and have been
discovered before using the codecs. If its not TRUE
, one should wait for the
next "farsight-codecs-changed" message until reading the codecs.
typedef enum { FS_DTMF_EVENT_0 = 0, FS_DTMF_EVENT_1 = 1, FS_DTMF_EVENT_2 = 2, FS_DTMF_EVENT_3 = 3, FS_DTMF_EVENT_4 = 4, FS_DTMF_EVENT_5 = 5, FS_DTMF_EVENT_6 = 6, FS_DTMF_EVENT_7 = 7, FS_DTMF_EVENT_8 = 8, FS_DTMF_EVENT_9 = 9, FS_DTMF_EVENT_STAR = 10, FS_DTMF_EVENT_POUND = 11, FS_DTMF_EVENT_A = 12, FS_DTMF_EVENT_B = 13, FS_DTMF_EVENT_C = 14, FS_DTMF_EVENT_D = 15 } FsDTMFEvent;
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
typedef enum { FS_DTMF_METHOD_AUTO = 0, FS_DTMF_METHOD_RTP_RFC4733, FS_DTMF_METHOD_IN_BAND } FsDTMFMethod;
An enum that represents the different ways a DTMF event can be sent
typedef struct _FsSession FsSession;
All members are private, access them using methods and properties
typedef struct { GObjectClass parent_class; /*virtual functions */ FsStream *(* new_stream) (FsSession *session, FsParticipant *participant, FsStreamDirection direction, const gchar *transmitter, guint stream_transmitter_n_parameters, GParameter *stream_transmitter_parameters, GError **error); gboolean (* start_telephony_event) (FsSession *session, guint8 event, guint8 volume, FsDTMFMethod method); gboolean (* stop_telephony_event) (FsSession *session, FsDTMFMethod method); 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); } FsSessionClass;
You must override at least new_stream in a subclass.
GObjectClass parent_class ; |
Our parent |
new_stream () |
Create a new FsStream |
start_telephony_event () |
Starts a telephony event |
stop_telephony_event () |
Stops a telephony event |
set_send_codec () |
Forces sending with a specific codec |
set_codec_preferences () |
Specifies the codec preferences |
list_transmitters () |
Returns a list of the available transmitters |
get_stream_transmitter_type () |
Returns the GType of the stream transmitter |
FsStream* fs_session_new_stream (FsSession *session, FsParticipant *participant, FsStreamDirection direction, const gchar *transmitter, guint stream_transmitter_n_parameters, GParameter *stream_transmitter_parameters, GError **error);
This function creates a stream for the given participant into the active session.
session : |
a FsSession |
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 |
transmitter : |
Name of the type of transmitter to use for this session |
stream_transmitter_n_parameters : |
Number of parametrs passed to the stream transmitter |
stream_transmitter_parameters : |
an array of n_parameters GParameter struct that will be passed to the newly-create FsStreamTransmitter |
error : |
location of a GError, or NULL if no error occured
|
Returns : | the new FsStream that has been created. User must unref the FsStream when the stream is ended. If an error occured, returns NULL. |
gboolean fs_session_start_telephony_event (FsSession *session, guint8 event, guint8 volume, FsDTMFMethod method);
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.
This function will use any available method, if you want to use a specific
method only, use fs_session_start_telephony_event_full()
session : |
a FsSession |
event : |
A FsStreamDTMFEvent or another number defined at http://www.iana.org/assignments/audio-telephone-event-registry |
volume : |
The volume in dBm0 without the negative sign. Should be between 0 and 36. Higher values mean lower volume |
method : |
The method used to send the event |
Returns : | TRUE if sucessful, it can return FALSE if the FsStream
does not support this telephony event.
|
gboolean fs_session_stop_telephony_event (FsSession *session, FsDTMFMethod method);
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.
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.
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.
gchar** fs_session_list_transmitters (FsSession *session);
Get the list of all available transmitters for this session.
session : |
A FsSession |
Returns : | 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()
.
session : |
A FsSession |
transmitter : |
The name of the transmitter |
Returns : | The GType of the stream transmitter |
"codec-preferences"
property"codec-preferences" FsCodecGList* : Read
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.
"codecs"
property"codecs" FsCodecGList* : Read
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.
You can only assume that the configuration parameters are valid when
the "codecs-ready" property is TRUE
.
The "farsight-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.
"codecs-ready"
property"codecs-ready" gboolean : Read
Some codecs that have configuration data that needs to be sent reliably
may need to be initialized from actual data before being ready. If your
application uses such codecs, wait until this property is TRUE
before
using the "codecs"
property. If the value if not TRUE
, the "farsight-codecs-changed"
message will be emitted when it becomes TRUE
. You should re-check
the value of this property when you receive the message.
Default value: TRUE
"codecs-without-config"
property"codecs-without-config" FsCodecGList* : Read
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 "farsight-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.
"current-send-codec"
property"current-send-codec" FsCodec* : Read
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 Farsight. This property is an
FsCodec. User must free the codec using fs_codec_destroy()
when done.
The "farsight-send-codec-changed" message is emitted on the bus when
the value of this property changes.
"id"
property"id" guint : Read / Write / Construct Only
The ID of the session, the first number of the pads linked to this session will be this id
Default value: 0
"media-type"
property"media-type" FsMediaType : Read / Write / Construct Only
The media-type of the session. This is either Audio, Video or both. This is a constructor parameter that cannot be changed.
Default value: FS_MEDIA_TYPE_AUDIO
"sink-pad"
property"sink-pad" GstPad* : Read
The Gstreamer sink pad that must be used to send media data on this session. User must unref this GstPad when done with it.
"error"
signalvoid user_function (FsSession *self, GObject *object, FsError error_no, gchar *error_msg, gchar *debug_msg, gpointer user_data) : Run Last
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 to be displayed to user |
debug_msg : |
Debugging error message |
user_data : |
user data set when the signal handler was connected. |