* rtspsrc currently understands SDP as the format of the session description.
* For each stream listed in the SDP a new rtp_stream%d pad will be created
* with caps derived from the SDP media description. This is a caps of mime type
- * "application/x-rtp" that can be connected to any available rtp depayloader
+ * "application/x-rtp" that can be connected to any available RTP depayloader
* element.
* </para>
* <para>
* <programlisting>
* gst-launch rtspsrc location=rtsp://some.server/url ! fakesink
* </programlisting>
- * Establish a connection to an RTSP server and send the stream to a fakesink.
+ * Establish a connection to an RTSP server and send the raw RTP packets to a fakesink.
* </para>
* </refsect2>
*
stream = (GstRTSPStream *) streams->data;
- /* first our rtp session manager */
+ /* first our RTP session manager */
if (stream->rtpdec) {
- if ((ret =
- gst_element_set_state (stream->rtpdec,
- state)) == GST_STATE_CHANGE_FAILURE)
+ ret = gst_element_set_state (stream->rtpdec, state);
+ if (ret == GST_STATE_CHANGE_FAILURE)
goto done;
}
/* then our sources */
if (stream->rtpsrc) {
- if ((ret =
- gst_element_set_state (stream->rtpsrc,
- state)) == GST_STATE_CHANGE_FAILURE)
+ ret = gst_element_set_state (stream->rtpsrc, state);
+ if (ret == GST_STATE_CHANGE_FAILURE)
goto done;
}
-
if (stream->rtcpsrc) {
- if ((ret =
- gst_element_set_state (stream->rtcpsrc,
- state)) == GST_STATE_CHANGE_FAILURE)
+ ret = gst_element_set_state (stream->rtcpsrc, state);
+ if (ret == GST_STATE_CHANGE_FAILURE)
goto done;
}
}
/*
* Mapping of caps to and from SDP fields:
*
- * m=<media> <
udp port> RTP/AVP <payload>
+ * m=<media> <
UDP port> RTP/AVP <payload>
* a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
* a=fmtp:<payload> <param>[=<value>];...
*/
}
pt = atoi (payload);
+ /* dynamic payloads need rtpmap */
if (pt >= 96) {
gint payload = 0;
gboolean ret;
if ((rtpmap = sdp_media_get_attribute_val (media, "rtpmap"))) {
- if ((ret =
- gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate,
- ¶ms))) {
+ ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
+ if (ret) {
if (payload != pt) {
g_warning ("rtpmap of wrong payload type");
name = NULL;
g_warning ("error parsing rtpmap");
}
} else {
- g_warning ("rtpmap type not given fot dynamic payload %d", pt);
+ g_warning ("rtpmap type not given for dynamic payload %d", pt);
return NULL;
}
}
{
GstStateChangeReturn ret;
GstRTSPSrc *src;
- GstCaps *caps;
- GstElement *tmp, *rtp, *rtcp;
+ GstElement *tmp, *rtpsrc, *rtcpsrc;
gint tmp_rtp, tmp_rtcp;
guint count;
src = stream->parent;
tmp = NULL;
- rtp = NULL;
- rtcp = NULL;
+ rtpsrc = NULL;
+ rtcpsrc = NULL;
count = 0;
- /* try to allocate 2 udp ports, the RTP port should be an even
+ /* try to allocate 2 UDP ports, the RTP port should be an even
* number and the RTCP port should be the next (uneven) port */
again:
- rtp = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0:0", NULL);
- if (rtp == NULL)
+ rtpsrc = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0:0", NULL);
+ if (rtpsrc == NULL)
goto no_udp_rtp_protocol;
- ret = gst_element_set_state (rtp, GST_STATE_PAUSED);
+ ret = gst_element_set_state (rtpsrc, GST_STATE_PAUSED);
if (ret == GST_STATE_CHANGE_FAILURE)
goto start_rtp_failure;
- g_object_get (G_OBJECT (rtp), "port", &tmp_rtp, NULL);
+ g_object_get (G_OBJECT (rtpsrc), "port", &tmp_rtp, NULL);
GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
/* check if port is even */
gst_element_set_state (tmp, GST_STATE_NULL);
gst_object_unref (tmp);
}
- tmp = rtp;
+ tmp = rtpsrc;
GST_DEBUG_OBJECT (src, "retry %d", count);
goto again;
}
}
/* allocate port+1 for RTCP now */
- rtcp = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", NULL);
- if (rtcp == NULL)
+ rtcpsrc = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", NULL);
+ if (rtcpsrc == NULL)
goto no_udp_rtcp_protocol;
/* set port */
tmp_rtcp = tmp_rtp + 1;
- g_object_set (G_OBJECT (rtcp), "port", tmp_rtcp, NULL);
+ g_object_set (G_OBJECT (rtcpsrc), "port", tmp_rtcp, NULL);
GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
- ret = gst_element_set_state (rtcp, GST_STATE_PAUSED);
+ ret = gst_element_set_state (rtcpsrc, GST_STATE_PAUSED);
/* FIXME, this could fail if the next port is not free, we
* should retry with another port then */
if (ret == GST_STATE_CHANGE_FAILURE)
goto start_rtcp_failure;
/* all fine, do port check */
- g_object_get (G_OBJECT (rtp), "port", rtpport, NULL);
- g_object_get (G_OBJECT (rtcp), "port", rtcpport, NULL);
+ g_object_get (G_OBJECT (rtpsrc), "port", rtpport, NULL);
+ g_object_get (G_OBJECT (rtcpsrc), "port", rtcpport, NULL);
/* this should not happen */
if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
goto port_error;
- /* we manage these elements */
- stream->rtpsrc = rtp;
+ /* we manage these elements, we set the caps in configure_transport */
+ stream->rtpsrc = rtpsrc;
gst_rtspsrc_add_element (src, stream->rtpsrc);
- stream->rtcpsrc = rtcp;
+ stream->rtcpsrc = rtcpsrc;
gst_rtspsrc_add_element (src, stream->rtcpsrc);
- caps = gst_rtspsrc_media_to_caps (media);
-
- /* set caps */
- g_object_set (G_OBJECT (stream->rtpsrc), "caps", caps, NULL);
-
return TRUE;
/* ERRORS */
gst_element_set_state (tmp, GST_STATE_NULL);
gst_object_unref (tmp);
}
- if (rtp) {
- gst_element_set_state (rtp, GST_STATE_NULL);
- gst_object_unref (rtp);
+ if (rtpsrc) {
+ gst_element_set_state (rtpsrc, GST_STATE_NULL);
+ gst_object_unref (rtpsrc);
}
- if (rtcp) {
- gst_element_set_state (rtcp, GST_STATE_NULL);
- gst_object_unref (rtcp);
+ if (rtcpsrc) {
+ gst_element_set_state (rtcpsrc, GST_STATE_NULL);
+ gst_object_unref (rtcpsrc);
}
return FALSE;
}
/* we manage this element */
gst_rtspsrc_add_element (src, stream->rtpdec);
- if ((ret =
- gst_element_set_state (stream->rtpdec,
- GST_STATE_PAUSED)) != GST_STATE_CHANGE_SUCCESS)
+ ret = gst_element_set_state (stream->rtpdec, GST_STATE_PAUSED);
+ if (ret != GST_STATE_CHANGE_SUCCESS)
goto start_rtpdec_failure;
stream->rtpdecrtp = gst_element_get_pad (stream->rtpdec, "sinkrtp");
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. */
+ * RTP session manager. */
stream->rtpchannel = transport->interleaved.min;
stream->rtcpchannel = transport->interleaved.max;
GST_DEBUG ("stream %p on channels %d-%d", stream,
stream->rtpchannel, stream->rtcpchannel);
- /* also store the caps in the stream */
+ /* also store the caps in the stream, we need this when setting caps on
+ * outgoing buffers */
stream->caps = gst_rtspsrc_media_to_caps (media);
} else {
- /* configure for UDP delivery, we need to connect the udp pads to
- * the rtp session plugin. */
+ /* multicast was selected, create UDP sources and connect to the multicast
+ * group. */
+ if (transport->multicast) {
+ gchar *uri;
+
+ /* creating RTP source */
+ uri =
+ g_strdup_printf ("udp://%s:%d", transport->destination,
+ transport->port.min);
+ stream->rtpsrc = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
+ g_free (uri);
+ if (stream->rtpsrc == NULL)
+ goto no_element;
+
+ /* creating RTCP source */
+ uri =
+ g_strdup_printf ("udp://%s:%d", transport->destination,
+ transport->port.max);
+ stream->rtcpsrc = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
+ g_free (uri);
+ if (stream->rtcpsrc == NULL)
+ goto no_element;
+
+
+ /* change state */
+ gst_element_set_state (stream->rtpsrc, GST_STATE_PAUSED);
+ gst_element_set_state (stream->rtcpsrc, GST_STATE_PAUSED);
+
+ /* we manage these elements */
+ gst_rtspsrc_add_element (src, stream->rtpsrc);
+ gst_rtspsrc_add_element (src, stream->rtcpsrc);
+ }
+
+ /* configure caps on the RTP source element */
+ stream->caps = gst_rtspsrc_media_to_caps (media);
+ g_object_set (G_OBJECT (stream->rtpsrc), "caps", stream->caps, NULL);
+
+ /* 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 (pad);
/* create OPTIONS */
GST_DEBUG_OBJECT (src, "create options...");
- if ((res =
- rtsp_message_init_request (RTSP_OPTIONS, src->location,
- &request)) < 0)
+ res = rtsp_message_init_request (RTSP_OPTIONS, src->location, &request);
+ if (res < 0)
goto create_request_failed;
/* send OPTIONS */
/* create DESCRIBE */
GST_DEBUG_OBJECT (src, "create describe...");
- if ((res =
- rtsp_message_init_request (RTSP_DESCRIBE, src->location,
- &request)) < 0)
+ res = rtsp_message_init_request (RTSP_DESCRIBE, src->location, &request);
+ if (res < 0)
goto create_request_failed;
- /* we accept SDP for now */
+
+ /* we only accept SDP for now */
rtsp_message_add_header (&request, RTSP_HDR_ACCEPT, "application/sdp");
/* send DESCRIBE */
}
}
- /* parse SDP */
+ /* get message body and parse as SDP */
rtsp_message_get_body (&response, &data, &size);
GST_DEBUG_OBJECT (src, "parse sdp...");
if (src->debug)
sdp_message_dump (&sdp);
- /* we allow all configured protocols */
+ /* we initially allow all configured protocols. based on the replies from the
+ * server we narrow them down. */
protocols = src->protocols;
+
/* setup streams */
{
gint i;
}
GST_DEBUG_OBJECT (src, "setup %s", setup_url);
+
/* create SETUP request */
- if ((res =
- rtsp_message_init_request (RTSP_SETUP, setup_url,
- &request)) < 0) {
- g_free (setup_url);
- goto create_request_failed;
- }
+ res = rtsp_message_init_request (RTSP_SETUP, setup_url, &request);
g_free (setup_url);
+ if (res < 0)
+ goto create_request_failed;
transports = g_strdup ("");
if (protocols & GST_RTSP_PROTO_UDP_UNICAST) {
gint rtpport, rtcpport;
gchar *trxparams;
- /* allocate two udp ports */
+ /* allocate two UDP ports */
if (!gst_rtspsrc_stream_setup_rtp (stream, media, &rtpport, &rtcpport))
goto setup_rtp_failed;
GST_DEBUG_OBJECT (src, "setting up MULTICAST");
+ /* we don't hav to allocate any UDP ports yet, if the selected transport
+ * turns out to be multicast we can create them and join the multicast
+ * group indicated in the transport reply */
new =
g_strconcat (transports, transports[0] ? "," : "",
"RTP/AVP/UDP;multicast", NULL);
/* parse transport */
rtsp_transport_parse (resptrans, &transport);
- /* update allowed transports for other streams */
+
+ /* update allowed transports for other streams. once the transport of
+ * one stream has been determined, we make sure that all other streams
+ * are configured in the same way */
if (transport.lower_transport == RTSP_LOWER_TRANS_TCP) {
GST_DEBUG_OBJECT (src, "stream %d as TCP", i);
protocols = GST_RTSP_PROTO_TCP;
src->interleaved = TRUE;
} else {
if (transport.multicast) {
- /* disable unicast */
+ /* only allow multicast for other streams */
GST_DEBUG_OBJECT (src, "stream %d as MULTICAST", i);
protocols = GST_RTSP_PROTO_UDP_MULTICAST;
} else {
- /* disable multicast */
+ /* only allow unicast for other streams */
GST_DEBUG_OBJECT (src, "stream %d as UNICAST", i);
protocols = GST_RTSP_PROTO_UDP_UNICAST;
}
if (src->options & RTSP_PLAY) {
/* do TEARDOWN */
- if ((res =
- rtsp_message_init_request (RTSP_TEARDOWN, src->location,
- &request)) < 0)
+ res = rtsp_message_init_request (RTSP_TEARDOWN, src->location, &request);
+ if (res < 0)
goto create_request_failed;
if (!gst_rtspsrc_send (src, &request, &response, NULL))
GST_DEBUG_OBJECT (src, "PLAY...");
/* do play */
- if ((res =
- rtsp_message_init_request (RTSP_PLAY, src->location, &request)) < 0)
+ res = rtsp_message_init_request (RTSP_PLAY, src->location, &request);
+ if (res < 0)
goto create_request_failed;
if (!gst_rtspsrc_send (src, &request, &response, NULL))
GST_DEBUG_OBJECT (src, "PAUSE...");
/* do pause */
- if ((res =
- rtsp_message_init_request (RTSP_PAUSE, src->location, &request)) < 0)
+ res = rtsp_message_init_request (RTSP_PAUSE, src->location, &request);
+ if (res < 0)
goto create_request_failed;
if (!gst_rtspsrc_send (src, &request, &response, NULL))
projects / platform / upstream / gst-plugins-good.git / commitdiff