static gboolean
start_stop_sending_dtmf (gpointer data)
{
GstState state;
GstStateChangeReturn ret;
if (!dat || !dat->pipeline || !dat->session)
return TRUE;
ret = gst_element_get_state (dat->pipeline, &state, NULL, 0);
ts_fail_if (ret == GST_STATE_CHANGE_FAILURE);
if (ret != GST_STATE_CHANGE_SUCCESS || state != GST_STATE_PLAYING)
return TRUE;
if (!ready_to_send)
return TRUE;
if (sending)
{
ts_fail_unless (fs_session_stop_telephony_event (dat->session, method),
"Could not stop telephony event");
sending = FALSE;
ts_fail_unless (received == TRUE, "Did not receive any buffer for digit %d",
digit);
if (digit >= FS_DTMF_EVENT_D && !change_codec)
{
g_main_loop_quit (loop);
return FALSE;
}
}
else
{
if (digit >= FS_DTMF_EVENT_D)
{
digit = 0;
dtmf_id++;
ready_to_send = FALSE;
change_codec = FALSE;
set_codecs (dat, stream);
return TRUE;
}
digit++;
received = FALSE;
ts_fail_unless (fs_session_start_telephony_event (dat->session,
digit, digit, method),
"Could not start telephony event");
sending = TRUE;
}
return TRUE;
}
gboolean
skype_audio_stream_send_dtmf (SkypeAudioStream *self,
int event,
int type)
{
SkypeAudioStreamPrivate *priv = self->priv;
FsuSession *session;
FsSession *fssession;
FsDTMFMethod method;
gboolean ret = TRUE;
GstStructure *structure;
GstEvent *gstevent;
g_debug ("Sending telephony event: %d, type: %d", event, type);
structure = gst_structure_new ("dtmf-event",
"type", G_TYPE_INT, 1,
"number", G_TYPE_INT, event,
"volume", G_TYPE_INT, 8,
"start", G_TYPE_BOOLEAN, TRUE,
NULL);
gstevent = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure);
if (!gst_element_send_event (priv->dtmfsrc, gstevent))
g_warning ("DTMF start event wasn't applied");
structure = gst_structure_new ("dtmf-event",
"type", G_TYPE_INT, 1,
"number", G_TYPE_INT, event,
"volume", G_TYPE_INT, 8,
"start", G_TYPE_BOOLEAN, FALSE,
NULL);
gstevent = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure);
if (!gst_element_send_event (priv->dtmfsrc, gstevent))
g_warning ("DTMF start event wasn't applied");
if (g_queue_is_empty (priv->event_queue))
{
/* The type argument doubles as both the type of event
* (RTP or in-band) as well as the PT to use if it's RTP */
if (type == SKYPE_DTMF_EVENT_TYPE_IN_BAND)
{
method = FS_DTMF_METHOD_IN_BAND;
}
else if (type == priv->last_telephony_pt)
{
/* Play now only if the current PT matches the desired one */
method = FS_DTMF_METHOD_RTP_RFC4733;
}
else
{
FsCodec *codec;
/* Can't play the DTMF event now, queue the event and update codecs */
DTMFQueueEvent *dtmf_event = g_slice_new0 (DTMFQueueEvent);
dtmf_event->event_nr = event;
dtmf_event->rtp_type = type;
g_queue_push_tail (priv->event_queue, dtmf_event);
priv->pending_telephony_pt = type;
/* Trigger a codec update */
g_object_get (self, "codec", &codec, NULL);
g_object_set (self, "codec", codec, NULL);
fs_codec_destroy (codec);
return ret;
}
}
else
{
/* Other events are pending, queue this one */
DTMFQueueEvent *dtmf_event = g_slice_new0 (DTMFQueueEvent);
dtmf_event->event_nr = event;
dtmf_event->rtp_type = type;
g_queue_push_tail (priv->event_queue, dtmf_event);
return ret;
}
g_object_get (priv->stream, "fsu-session", &session, NULL);
if (session == NULL)
{
g_warning ("SendDTMF: session == NULL");
return FALSE;
}
g_object_get (session, "fs-session", &fssession, NULL);
g_object_unref (session);
if (fssession == NULL)
{
g_warning ("SendDTMF: fssession == NULL");
return FALSE;
}
/* Play the event */
if (!fs_session_start_telephony_event (fssession, event, 0, method))
{
g_warning ("Telephony event type not supported");
ret = FALSE;
}
else
{
fs_session_stop_telephony_event (fssession, method);
}
g_object_unref (fssession);
return ret;
}
gboolean
skype_audio_stream_handle_message (SkypeBaseStream *self, GstMessage *message)
{
SkypeAudioStreamPrivate *priv = SKYPE_AUDIO_STREAM (self)->priv;
GstStructure *structure;
FsSession *session = NULL;
GList *secondary_codecs = NULL;
GList *iter;
if (GST_MESSAGE_TYPE (message) != GST_MESSAGE_ELEMENT)
return FALSE;
structure = (GstStructure *)gst_message_get_structure (message);
if (!gst_structure_has_name (structure, "farsight-send-codec-changed"))
return FALSE;
gst_structure_get (structure,
"session", FS_TYPE_SESSION, &session,
"secondary-codecs", FS_TYPE_CODEC_LIST, &secondary_codecs,
NULL);
/* Update the current telephony PT */
for (iter = secondary_codecs; iter != NULL; iter = g_list_next (iter))
{
FsCodec *codec = (FsCodec *)iter->data;
if (codec != NULL && !g_strcmp0 (codec->encoding_name, "telephone-event"))
{
priv->last_telephony_pt = codec->id;
if (priv->pending_telephony_pt != FS_CODEC_ID_DISABLE &&
priv->pending_telephony_pt == priv->last_telephony_pt)
priv->pending_telephony_pt = FS_CODEC_ID_DISABLE;
break;
}
}
while (!g_queue_is_empty (priv->event_queue))
{
DTMFQueueEvent *event = (DTMFQueueEvent *)
g_queue_pop_head (priv->event_queue);
FsDTMFMethod method = FS_DTMF_METHOD_AUTO;
if (event->rtp_type == SKYPE_DTMF_EVENT_TYPE_IN_BAND)
{
method = FS_DTMF_METHOD_IN_BAND;
}
else if (event->rtp_type == priv->last_telephony_pt)
{
method = FS_DTMF_METHOD_RTP_RFC4733;
}
else
{
/* New PT needed. Update codecs */
FsCodec *codec;
g_queue_push_head (priv->event_queue, event);
priv->pending_telephony_pt = event->rtp_type;
/* Trigger a codec update */
g_object_get (self, "codec", &codec, NULL);
g_object_set (self, "codec", codec, NULL);
fs_codec_destroy (codec);
break;
}
if (!fs_session_start_telephony_event (session,
event->event_nr, 0, method))
{
g_warning ("Telephony event type not supported");
g_queue_push_head (priv->event_queue, event);
}
else
{
fs_session_stop_telephony_event (session, method);
}
g_slice_free (DTMFQueueEvent, event);
}
g_object_unref (session);
fs_codec_list_destroy (secondary_codecs);
return TRUE;
}
