static void gst_rtspsrc_init (GstRTSPSrc * rtspsrc);
static GstElementStateReturn gst_rtspsrc_change_state (GstElement * element);
-static gboolean gst_rtspsrc_activate (GstPad * pad, GstActivateMode mode);
static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_DEBUG,
g_param_spec_boolean ("debug", "Debug",
- "Dump request qnd response messages to stdout",
+ "Dump request and response messages to stdout",
DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
gstelement_class->change_state = gst_rtspsrc_change_state;
static void
gst_rtspsrc_init (GstRTSPSrc * src)
{
- /*
- src->srcpad =
- gst_pad_new_from_template (gst_static_pad_template_get (&srctemplate),
- "src");
- gst_pad_set_loop_function (src->srcpad, gst_rtspsrc_loop);
- gst_pad_set_activate_function (src->srcpad, gst_rtspsrc_activate);
- gst_element_add_pad (GST_ELEMENT (src), src->srcpad);
- */
}
static void
s = g_new0 (GstRTSPStream, 1);
s->parent = src;
+ s->id = src->numstreams++;
src->streams = g_list_append (src->streams, s);
}
static gboolean
-rtspsrc_add_element (GstRTSPSrc * src, GstElement * element)
+gst_rtspsrc_add_element (GstRTSPSrc * src, GstElement * element)
{
gst_object_set_parent (GST_OBJECT (element), GST_OBJECT (src));
gst_element_set_manager (element, GST_ELEMENT_MANAGER (src));
return TRUE;
}
+static GstElementStateReturn
+gst_rtspsrc_set_state (GstRTSPSrc * src, GstElementState state)
+{
+ GstElementStateReturn ret;
+ GList *streams;
+
+ /* for all streams */
+ for (streams = src->streams; streams; streams = g_list_next (streams)) {
+ GstRTSPStream *stream;
+
+ stream = (GstRTSPStream *) streams->data;
+
+ /* first our rtp session manager */
+ if ((ret =
+ gst_element_set_state (stream->rtpdec, state)) != GST_STATE_SUCCESS)
+ goto done;
+
+ /* then our sources */
+ if (stream->rtpsrc) {
+ if ((ret =
+ gst_element_set_state (stream->rtpsrc,
+ state)) != GST_STATE_SUCCESS)
+ goto done;
+ }
+ if (stream->rtcpsrc) {
+ if ((ret =
+ gst_element_set_state (stream->rtcpsrc,
+ state)) != GST_STATE_SUCCESS)
+ goto done;
+ }
+ }
+
+done:
+ return ret;
+}
+
static gboolean
gst_rtspsrc_stream_setup_rtp (GstRTSPStream * stream, gint * rtpport,
gint * rtcpport)
goto no_udp_rtp_protocol;
/* we manage this element */
- rtspsrc_add_element (src, stream->rtpsrc);
+ gst_rtspsrc_add_element (src, stream->rtpsrc);
if ((ret =
gst_element_set_state (stream->rtpsrc,
goto no_udp_rtcp_protocol;
/* we manage this element */
- rtspsrc_add_element (src, stream->rtcpsrc);
+ gst_rtspsrc_add_element (src, stream->rtcpsrc);
if ((ret =
gst_element_set_state (stream->rtcpsrc,
RTSPTransport * transport)
{
GstRTSPSrc *src;
+ GstPad *pad;
+ GstElementStateReturn ret;
+ gchar *name;
src = stream->parent;
- if (transport->lower_transport == RTSP_LOWER_TRANS_TCP) {
- GstPad *pad;
+ if (!(stream->rtpdec = gst_element_factory_make ("rtpdec", NULL)))
+ goto no_element;
- /* configure for interleaved delivery */
- if (!(stream->rtpdec = gst_element_factory_make ("rtpdec", NULL)))
- goto no_element;
+ /* we manage this element */
+ gst_rtspsrc_add_element (src, stream->rtpdec);
- /* we manage this element */
- rtspsrc_add_element (src, stream->rtpdec);
- stream->rtpdecrtp = gst_element_get_pad (stream->rtpdec, "sinkrtp");
- stream->rtpdecrtcp = gst_element_get_pad (stream->rtpdec, "sinkrtcp");
+ if ((ret =
+ gst_element_set_state (stream->rtpdec,
+ GST_STATE_PAUSED)) != GST_STATE_SUCCESS)
+ goto start_rtpdec_failure;
- /* FIXME, make sure it outputs the caps */
- pad = gst_element_get_pad (stream->rtpdec, "srcrtp");
- gst_element_add_ghost_pad (GST_ELEMENT (src), pad, "srcrtp");
- gst_object_unref (GST_OBJECT (pad));
+ stream->rtpdecrtp = gst_element_get_pad (stream->rtpdec, "sinkrtp");
+ stream->rtpdecrtcp = gst_element_get_pad (stream->rtpdec, "sinkrtcp");
+
+ /* FIXME, make sure it outputs the caps */
+ pad = gst_element_get_pad (stream->rtpdec, "srcrtp");
+ name = g_strdup_printf ("rtp_stream%d", stream->id);
+ gst_element_add_ghost_pad (GST_ELEMENT (src), pad, name);
+ g_free (name);
+ gst_object_unref (GST_OBJECT (pad));
+
+ if (transport->lower_transport == RTSP_LOWER_TRANS_TCP) {
+ /* configure for interleaved delivery, nothing needs to be done
+ * here, the loop function will call the chain functions of the
+ * rtp session manager. */
} else {
- /* configure for UDP delivery, FIXME */
+ /* configure for UDP delivery, we need to connect the udp pads to
+ * the rtp session plugin. */
+ pad = gst_element_get_pad (stream->rtpsrc, "src");
+ gst_pad_link (pad, stream->rtpdecrtp);
+ gst_object_unref (GST_OBJECT (pad));
+
+ pad = gst_element_get_pad (stream->rtcpsrc, "src");
+ gst_pad_link (pad, stream->rtpdecrtcp);
+ gst_object_unref (GST_OBJECT (pad));
}
return TRUE;
GST_DEBUG ("no rtpdec element found");
return FALSE;
}
+start_rtpdec_failure:
+ {
+ GST_DEBUG ("could not start RTP session");
+ return FALSE;
+ }
}
static gint
rtsp_message_add_header (&request, RTSP_HDR_TRANSPORT, transports);
g_free (transports);
- rtsp_message_dump (&request);
-
if (!gst_rtspsrc_send (src, &request, &response))
goto send_error;
rtsp_message_get_header (&response, RTSP_HDR_TRANSPORT, &resptrans);
- /* update allowed transports for other streams */
+ /* parse transport */
rtsp_transport_parse (resptrans, &transport);
+ /* update allowed transports for other streams */
if (transport.lower_transport == RTSP_LOWER_TRANS_TCP) {
protocols = GST_RTSP_PROTO_TCP;
src->interleaved = TRUE;
protocols = GST_RTSP_PROTO_UDP_UNICAST;
}
}
- gst_rtspsrc_stream_configure_transport (stream, &transport);
+ /* now configure the stream with the transport */
+ if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
+ GST_DEBUG ("could not configure stream transport, skipping stream");
+ }
+ /* clean up our transport struct */
rtsp_transport_init (&transport);
}
}
RTSPResult res;
GST_DEBUG ("TEARDOWN...");
+
+ /* stop task if any */
+ if (src->task) {
+ gst_task_stop (src->task);
+ gst_object_unref (GST_OBJECT (src->task));
+ src->task = NULL;
+ }
+
/* do TEARDOWN */
if ((res =
rtsp_message_init_request (RTSP_TEARDOWN, src->location,
RTSPResult res;
GST_DEBUG ("PLAY...");
+
/* do play */
if ((res =
rtsp_message_init_request (RTSP_PLAY, src->location, &request)) < 0)
}
}
-static gboolean
-gst_rtspsrc_activate (GstPad * pad, GstActivateMode mode)
-{
- gboolean result;
- GstRTSPSrc *rtspsrc;
-
- rtspsrc = GST_RTSPSRC (GST_OBJECT_PARENT (pad));
-
- switch (mode) {
- case GST_ACTIVATE_PUSH:
- /* if we have a scheduler we can start the task */
- if (GST_ELEMENT_SCHEDULER (rtspsrc) && rtspsrc->interleaved) {
- GST_STREAM_LOCK (pad);
- GST_RPAD_TASK (pad) =
- gst_scheduler_create_task (GST_ELEMENT_SCHEDULER (rtspsrc),
- (GstTaskFunction) gst_rtspsrc_loop, pad);
-
- gst_task_start (GST_RPAD_TASK (pad));
- GST_STREAM_UNLOCK (pad);
- result = TRUE;
- }
- break;
- case GST_ACTIVATE_PULL:
- result = FALSE;
- break;
- case GST_ACTIVATE_NONE:
- /* step 1, unblock clock sync (if any) */
-
- /* step 2, make sure streaming finishes */
- GST_STREAM_LOCK (pad);
- gst_rtspsrc_close (rtspsrc);
-
- /* step 3, stop the task */
- if (GST_RPAD_TASK (pad)) {
- gst_task_stop (GST_RPAD_TASK (pad));
- gst_object_unref (GST_OBJECT (GST_RPAD_TASK (pad)));
- GST_RPAD_TASK (pad) = NULL;
- }
- GST_STREAM_UNLOCK (pad);
-
- result = TRUE;
- break;
- }
- return result;
-}
-
static GstElementStateReturn
gst_rtspsrc_change_state (GstElement * element)
{
gst_rtspsrc_play (rtspsrc);
break;
case GST_STATE_PAUSED_TO_PLAYING:
+ gst_rtspsrc_play (rtspsrc);
break;
default:
break;
}
+ ret = gst_rtspsrc_set_state (rtspsrc, GST_STATE_PENDING (rtspsrc));
+ if (ret != GST_STATE_SUCCESS)
+ goto error;
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element);
switch (transition) {
case GST_STATE_PLAYING_TO_PAUSED:
+ gst_rtspsrc_pause (rtspsrc);
break;
case GST_STATE_PAUSED_TO_READY:
- gst_rtspsrc_pause (rtspsrc);
break;
case GST_STATE_READY_TO_NULL:
break;
break;
}
+error:
return ret;
}
projects / platform / upstream / gstreamer.git / commitdiff