FsSession

FsSession — A session in a conference

Synopsis

#include <farstream/fs-conference.h>

                    FsSession;
struct              FsSessionClass;
enum                FsDTMFEvent;
enum                FsDTMFMethod;
FsStream *          fs_session_new_stream               (FsSession *session,
                                                         FsParticipant *participant,
                                                         FsStreamDirection direction,
                                                         GError **error);
gboolean            fs_session_set_codec_preferences    (FsSession *session,
                                                         GList *codec_preferences,
                                                         GError **error);
void                fs_session_destroy                  (FsSession *session);
gboolean            fs_session_start_telephony_event    (FsSession *session,
                                                         guint8 event,
                                                         guint8 volume);
gboolean            fs_session_stop_telephony_event     (FsSession *session);
gboolean            fs_session_set_send_codec           (FsSession *session,
                                                         FsCodec *send_codec,
                                                         GError **error);
gchar **            fs_session_list_transmitters        (FsSession *session);
GType               fs_session_get_stream_transmitter_type
                                                        (FsSession *session,
                                                         const gchar *transmitter);
GList *             fs_session_codecs_need_resend       (FsSession *session,
                                                         GList *old_codecs,
                                                         GList *new_codecs);
void                fs_session_emit_error               (FsSession *session,
                                                         gint error_no,
                                                         const gchar *error_msg);
gboolean            fs_session_parse_codecs_changed     (FsSession *session,
                                                         GstMessage *message);
gboolean            fs_session_parse_send_codec_changed (FsSession *session,
                                                         GstMessage *message,
                                                         FsCodec **codec,
                                                         GList **secondary_codecs);
gboolean            fs_session_parse_telephony_event_started
                                                        (FsSession *session,
                                                         GstMessage *message,
                                                         FsDTMFMethod *method,
                                                         FsDTMFEvent *event,
                                                         guint8 *volume);
gboolean            fs_session_parse_telephony_event_stopped
                                                        (FsSession *session,
                                                         GstMessage *message,
                                                         FsDTMFMethod *method);

Object Hierarchy

  GObject
   +----GstObject
         +----FsSession

Properties

  "codec-preferences"        FsCodecGList*         : Read
  "codecs"                   FsCodecGList*         : Read
  "codecs-without-config"    FsCodecGList*         : Read
  "conference"               FsConference*         : Read / Write / Construct Only
  "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

Signals

  "error"                                          : Run Last

Description

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.

The "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.


The "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.


The "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.


The "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.

Details

FsSession

typedef struct _FsSession FsSession;

All members are private, access them using methods and properties


struct FsSessionClass

struct FsSessionClass {
  GstObjectClass 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);
};

You must override at least new_stream in a subclass.

GstObjectClass 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

codecs_need_resend ()

Returns the list of codecs that need resending

enum FsDTMFEvent

typedef enum {
} 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


enum FsDTMFMethod

typedef enum {
  FS_DTMF_METHOD_RTP_RFC4733 = 1,
  FS_DTMF_METHOD_SOUND = 2
} FsDTMFMethod;

An enum that represents the different ways a DTMF event can be sent

FS_DTMF_METHOD_RTP_RFC4733

Send as a special payload type defined by RFC 4733 (which obsoletes RFC 2833)

FS_DTMF_METHOD_SOUND

Send as tones as in-band audio sound

fs_session_new_stream ()

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 :

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

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. [transfer full]

fs_session_set_codec_preferences ()

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.

session :

a FsSession

codec_preferences :

a GList of FsCodec with the desired configuration. [element-type FsCodec]

error :

location of a GError, or NULL if no error occured

Returns :

TRUE on success, FALSE on error.

fs_session_destroy ()

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.

session :

a FsSession

fs_session_start_telephony_event ()

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.

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

Returns :

TRUE if sucessful, it can return FALSE if the FsStream does not support this telephony event.

fs_session_stop_telephony_event ()

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.

session :

an FsSession

Returns :

TRUE if sucessful, it can return FALSE if the FsSession does not support telephony events or if no telephony event is being sent

fs_session_set_send_codec ()

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.

session :

a FsSession

send_codec :

a FsCodec representing the codec to send

error :

location of a GError, or NULL if no error occured

Returns :

FALSE if the send codec couldn't be set.

fs_session_list_transmitters ()

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(). [transfer full]

fs_session_get_stream_transmitter_type ()

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

fs_session_codecs_need_resend ()

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 :

a FsSession

old_codecs :

Codecs previously retrieved from the "codecs" property. [element-type FsCodec][transfer none]

new_codecs :

Codecs recently retrieved from the "codecs" property. [element-type FsCodec][transfer none]

Returns :

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]

fs_session_emit_error ()

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.

session :

FsSession on which to emit the error signal

error_no :

The number of the error of type FsError

error_msg :

Error message

fs_session_parse_codecs_changed ()

gboolean            fs_session_parse_codecs_changed     (FsSession *session,
                                                         GstMessage *message);

Parses a "farstream-codecs-changed" message and checks if it matches the session parameters.

session :

a FsSession to match against the message

message :

a GstMessage to parse

Returns :

TRUE if the message matches the session and is valid.

fs_session_parse_send_codec_changed ()

gboolean            fs_session_parse_send_codec_changed (FsSession *session,
                                                         GstMessage *message,
                                                         FsCodec **codec,
                                                         GList **secondary_codecs);

Returns a GList of FsCodec of the message if not NULL

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 :

Returns the FsCodec in the message if not NULL. [out][transfer none]

secondary_codecs :

. [out][transfer none][element-type FsCodec]

Returns :

TRUE if the message matches the session and is valid.

fs_session_parse_telephony_event_started ()

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 NULL. [out]

event :

Returns the FsDTMFEvent in the message if not NULL. [out]

volume :

Returns the volume in the message if not NULL. [out]

Returns :

TRUE if the message matches the session and is valid.

fs_session_parse_telephony_event_stopped ()

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 NULL. [out]

Returns :

TRUE if the message matches the session and is valid.

Property Details

The "codec-preferences" property

  "codec-preferences"        FsCodecGList*         : Read

Type: GLib.List(FsCodec) Transfer: full

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.


The "codecs" property

  "codecs"                   FsCodecGList*         : Read

Type: GLib.List(FsCodec) Transfer: full

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.


The "codecs-without-config" property

  "codecs-without-config"    FsCodecGList*         : Read

Type: GLib.List(FsCodec) Transfer: full

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.


The "conference" property

  "conference"               FsConference*         : Read / Write / Construct Only

The FsConference parent of this session. This property is a construct param and is read-only.


The "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 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.


The "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


The "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


The "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.


The "tos" property

  "tos"                      guint                 : Read / Write

Sets the IP ToS field (and if possible the IPv6 TCLASS field

Allowed values: <= 255

Default value: 0

Signal Details

The "error" signal

void                user_function                      (FsSession *self,
                                                        GObject   *object,
                                                        FsError    error_no,
                                                        gchar     *error_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

user_data :

user data set when the signal handler was connected.