diff options
| -rw-r--r-- | docs/libs/farstream-libs-sections.txt | 3 | ||||
| -rw-r--r-- | farstream/fs-session.c | 32 | ||||
| -rw-r--r-- | farstream/fs-session.h | 15 | ||||
| -rw-r--r-- | gst/fsrtpconference/fs-rtp-session.c | 8 |
4 files changed, 52 insertions, 6 deletions
diff --git a/docs/libs/farstream-libs-sections.txt b/docs/libs/farstream-libs-sections.txt index 4614573b..a8745969 100644 --- a/docs/libs/farstream-libs-sections.txt +++ b/docs/libs/farstream-libs-sections.txt @@ -49,11 +49,12 @@ FsStreamPrivate <SECTION> <FILE>fs-session</FILE> -FsDTMFEvent <TITLE>FsSession</TITLE> <INCLUDE>farstream/fs-conference.h</INCLUDE> FsSession FsSessionClass +FsDTMFEvent +FsDTMFMethod fs_session_new_stream fs_session_set_codec_preferences fs_session_destroy diff --git a/farstream/fs-session.c b/farstream/fs-session.c index 36faf983..e22a8e1e 100644 --- a/farstream/fs-session.c +++ b/farstream/fs-session.c @@ -66,6 +66,32 @@ * #FsSession:codecs-without-config should be enough. * </para> * </refsect2> + * <refsect2><title>The "<literal>farstream-telephony-event-started</literal>" + * message</title> + * |[ + * "session" #FsSession The session that emits the message + * "method" #FsDTMFMethod The method used to send the DTMF + * "event" #FSDTMFEvent The event number + * "volume" gint The volume of the event + * ]| + * <para> + * This message is emitted after a succesful call to + * fs_session_start_telephony_event() to inform the application that the + * telephony event has started. + * </para> + * </refsect2> + * <refsect2><title>The "<literal>farstream-telephony-event-stopped</literal>" + * message</title> + * |[ + * "session" #FsSession The session that emits the message + * "method" #FsDTMFMethod The method used to send the DTMF + * ]| + * <para> + * This message is emitted after a succesful call to + * fs_session_stop_telephony_event() to inform the application that the + * telephony event has stopped. + * </para> + * </refsect2> * <para> */ @@ -423,6 +449,9 @@ fs_session_new_stream (FsSession *session, * 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. + * * Returns: %TRUE if sucessful, it can return %FALSE if the #FsStream * does not support this telephony event. */ @@ -454,6 +483,9 @@ fs_session_start_telephony_event (FsSession *session, guint8 event, * 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. + * Returns: %TRUE if sucessful, it can return %FALSE if the #FsSession * does not support telephony events or if no telephony event is being sent */ diff --git a/farstream/fs-session.h b/farstream/fs-session.h index 4a7cf928..bc063afb 100644 --- a/farstream/fs-session.h +++ b/farstream/fs-session.h @@ -84,6 +84,21 @@ typedef enum _FsDTMFEvent } FsDTMFEvent; /** + * FsDTMFMethod: + * @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 + * + * An enum that represents the different ways a DTMF event can be sent + * + */ +typedef enum _FsDTMFMethod +{ + FS_DTMF_METHOD_RTP_RFC4733 = 1, + FS_DTMF_METHOD_SOUND = 2 +} FsDTMFMethod; + +/** * FsSessionClass: * @parent_class: Our parent * @new_stream: Create a new #FsStream diff --git a/gst/fsrtpconference/fs-rtp-session.c b/gst/fsrtpconference/fs-rtp-session.c index 30863bfc..9976ed2d 100644 --- a/gst/fsrtpconference/fs-rtp-session.c +++ b/gst/fsrtpconference/fs-rtp-session.c @@ -4990,9 +4990,8 @@ fs_rtp_session_handle_dtmf_event_message (FsRtpSession *self, GST_OBJECT (self->priv->conference), gst_structure_new ("farstream-telephony-event-started", "session", FS_TYPE_SESSION, self, - "type", G_TYPE_INT, 1, - "method", G_TYPE_INT, m_method, - "event", G_TYPE_INT, m_number, + "method", FS_TYPE_DTMF_METHOD, m_method, + "event", FS_TYPE_DTMF_EVENT, m_number, "volume", G_TYPE_INT, m_volume, NULL)); else @@ -5000,8 +4999,7 @@ fs_rtp_session_handle_dtmf_event_message (FsRtpSession *self, GST_OBJECT (self->priv->conference), gst_structure_new ("farstream-telephony-event-stopped", "session", FS_TYPE_SESSION, self, - "type", G_TYPE_INT, 1, - "method", G_TYPE_INT, m_method, + "method", FS_TYPE_DTMF_METHOD, m_method, NULL)); } else |