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gst/rtsp/: Small cleanups. when multicast is selected as the transport, create UDP sources and connect to the multica...
Original commit message from CVS: Based on patch by: Thijs Vermeir <thijs dot vermeir at barco dot com> * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause): * gst/rtsp/rtspconnection.c: (inet_aton): Small cleanups. when multicast is selected as the transport, create UDP sources and connect to the multicast group. Move parsing and setting of caps to a common place. Fixes #349894.
This commit is contained in:
Thijs Vermeir
Wim Taymans
parent
af06a16852
commit
7484c92dfe
17
ChangeLog
17
ChangeLog
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@ -1,3 +1,18 @@
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2006-09-18 Wim Taymans <wim@fluendo.com>
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Based on patch by: Thijs Vermeir <thijs dot vermeir at barco dot com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
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(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
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(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open),
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(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause):
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* gst/rtsp/rtspconnection.c: (inet_aton):
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Small cleanups.
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when multicast is selected as the transport, create UDP sources and
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connect to the multicast group.
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Move parsing and setting of caps to a common place.
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Fixes #349894.
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2006-09-17 Stefan Kost <ensonic@users.sf.net>
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* ext/flac/gstflactag.c:
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@ -20,7 +35,7 @@
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* gst/videofilter/gstvideotemplate.c:
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* gst/videomixer/videomixer.c:
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* sys/sunaudio/gstsunaudiosrc.h:
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More G_OBJECT macro fixing.
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More G_OBJECT macro fixing.
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2006-09-16 Wim Taymans <wim@fluendo.com>
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@ -35,7 +35,7 @@
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* rtspsrc currently understands SDP as the format of the session description.
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* For each stream listed in the SDP a new rtp_stream%d pad will be created
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* with caps derived from the SDP media description. This is a caps of mime type
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* "application/x-rtp" that can be connected to any available rtp depayloader
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* "application/x-rtp" that can be connected to any available RTP depayloader
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* element.
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* </para>
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* <para>
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@ -53,7 +53,7 @@
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* <programlisting>
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* gst-launch rtspsrc location=rtsp://some.server/url ! fakesink
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* </programlisting>
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* Establish a connection to an RTSP server and send the stream to a fakesink.
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* Establish a connection to an RTSP server and send the raw RTP packets to a fakesink.
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* </para>
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* </refsect2>
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*
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@ -370,26 +370,22 @@ gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
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stream = (GstRTSPStream *) streams->data;
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/* first our rtp session manager */
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/* first our RTP session manager */
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if (stream->rtpdec) {
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if ((ret =
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gst_element_set_state (stream->rtpdec,
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state)) == GST_STATE_CHANGE_FAILURE)
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ret = gst_element_set_state (stream->rtpdec, state);
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if (ret == GST_STATE_CHANGE_FAILURE)
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goto done;
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}
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/* then our sources */
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if (stream->rtpsrc) {
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if ((ret =
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gst_element_set_state (stream->rtpsrc,
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state)) == GST_STATE_CHANGE_FAILURE)
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ret = gst_element_set_state (stream->rtpsrc, state);
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if (ret == GST_STATE_CHANGE_FAILURE)
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goto done;
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}
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if (stream->rtcpsrc) {
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if ((ret =
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gst_element_set_state (stream->rtcpsrc,
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state)) == GST_STATE_CHANGE_FAILURE)
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ret = gst_element_set_state (stream->rtcpsrc, state);
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if (ret == GST_STATE_CHANGE_FAILURE)
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goto done;
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}
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}
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@ -469,7 +465,7 @@ gst_rtspsrc_parse_rtpmap (gchar * rtpmap, gint * payload, gchar ** name,
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/*
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* Mapping of caps to and from SDP fields:
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*
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* m=<media> <udp port> RTP/AVP <payload>
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* m=<media> <UDP port> RTP/AVP <payload>
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* a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
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* a=fmtp:<payload> <param>[=<value>];...
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*/
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@ -493,14 +489,14 @@ gst_rtspsrc_media_to_caps (SDPMedia * media)
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}
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pt = atoi (payload);
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/* dynamic payloads need rtpmap */
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if (pt >= 96) {
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gint payload = 0;
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gboolean ret;
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if ((rtpmap = sdp_media_get_attribute_val (media, "rtpmap"))) {
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if ((ret =
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gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate,
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¶ms))) {
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ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
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if (ret) {
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if (payload != pt) {
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g_warning ("rtpmap of wrong payload type");
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name = NULL;
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@ -511,7 +507,7 @@ gst_rtspsrc_media_to_caps (SDPMedia * media)
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g_warning ("error parsing rtpmap");
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}
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} else {
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g_warning ("rtpmap type not given fot dynamic payload %d", pt);
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g_warning ("rtpmap type not given for dynamic payload %d", pt);
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return NULL;
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}
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}
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@ -576,30 +572,29 @@ gst_rtspsrc_stream_setup_rtp (GstRTSPStream * stream, SDPMedia * media,
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{
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GstStateChangeReturn ret;
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GstRTSPSrc *src;
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GstCaps *caps;
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GstElement *tmp, *rtp, *rtcp;
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GstElement *tmp, *rtpsrc, *rtcpsrc;
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gint tmp_rtp, tmp_rtcp;
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guint count;
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src = stream->parent;
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tmp = NULL;
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rtp = NULL;
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rtcp = NULL;
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rtpsrc = NULL;
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rtcpsrc = NULL;
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count = 0;
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/* try to allocate 2 udp ports, the RTP port should be an even
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/* try to allocate 2 UDP ports, the RTP port should be an even
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* number and the RTCP port should be the next (uneven) port */
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again:
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rtp = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0:0", NULL);
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if (rtp == NULL)
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rtpsrc = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0:0", NULL);
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if (rtpsrc == NULL)
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goto no_udp_rtp_protocol;
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ret = gst_element_set_state (rtp, GST_STATE_PAUSED);
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ret = gst_element_set_state (rtpsrc, GST_STATE_PAUSED);
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if (ret == GST_STATE_CHANGE_FAILURE)
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goto start_rtp_failure;
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g_object_get (G_OBJECT (rtp), "port", &tmp_rtp, NULL);
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g_object_get (G_OBJECT (rtpsrc), "port", &tmp_rtp, NULL);
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GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
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/* check if port is even */
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@ -616,7 +611,7 @@ again:
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gst_element_set_state (tmp, GST_STATE_NULL);
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gst_object_unref (tmp);
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}
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tmp = rtp;
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tmp = rtpsrc;
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GST_DEBUG_OBJECT (src, "retry %d", count);
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goto again;
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}
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@ -628,40 +623,35 @@ again:
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}
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/* allocate port+1 for RTCP now */
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rtcp = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", NULL);
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if (rtcp == NULL)
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rtcpsrc = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", NULL);
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if (rtcpsrc == NULL)
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goto no_udp_rtcp_protocol;
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/* set port */
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tmp_rtcp = tmp_rtp + 1;
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g_object_set (G_OBJECT (rtcp), "port", tmp_rtcp, NULL);
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g_object_set (G_OBJECT (rtcpsrc), "port", tmp_rtcp, NULL);
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GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
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ret = gst_element_set_state (rtcp, GST_STATE_PAUSED);
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ret = gst_element_set_state (rtcpsrc, GST_STATE_PAUSED);
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/* FIXME, this could fail if the next port is not free, we
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* should retry with another port then */
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if (ret == GST_STATE_CHANGE_FAILURE)
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goto start_rtcp_failure;
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/* all fine, do port check */
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g_object_get (G_OBJECT (rtp), "port", rtpport, NULL);
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g_object_get (G_OBJECT (rtcp), "port", rtcpport, NULL);
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g_object_get (G_OBJECT (rtpsrc), "port", rtpport, NULL);
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g_object_get (G_OBJECT (rtcpsrc), "port", rtcpport, NULL);
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/* this should not happen */
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if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
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goto port_error;
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/* we manage these elements */
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stream->rtpsrc = rtp;
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/* we manage these elements, we set the caps in configure_transport */
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stream->rtpsrc = rtpsrc;
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gst_rtspsrc_add_element (src, stream->rtpsrc);
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stream->rtcpsrc = rtcp;
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stream->rtcpsrc = rtcpsrc;
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gst_rtspsrc_add_element (src, stream->rtcpsrc);
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caps = gst_rtspsrc_media_to_caps (media);
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/* set caps */
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g_object_set (G_OBJECT (stream->rtpsrc), "caps", caps, NULL);
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return TRUE;
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/* ERRORS */
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@ -703,13 +693,13 @@ cleanup:
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gst_element_set_state (tmp, GST_STATE_NULL);
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gst_object_unref (tmp);
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}
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if (rtp) {
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gst_element_set_state (rtp, GST_STATE_NULL);
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gst_object_unref (rtp);
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if (rtpsrc) {
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gst_element_set_state (rtpsrc, GST_STATE_NULL);
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gst_object_unref (rtpsrc);
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}
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if (rtcp) {
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gst_element_set_state (rtcp, GST_STATE_NULL);
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gst_object_unref (rtcp);
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if (rtcpsrc) {
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gst_element_set_state (rtcpsrc, GST_STATE_NULL);
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gst_object_unref (rtcpsrc);
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}
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return FALSE;
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}
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@ -734,9 +724,8 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
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/* we manage this element */
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gst_rtspsrc_add_element (src, stream->rtpdec);
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if ((ret =
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gst_element_set_state (stream->rtpdec,
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GST_STATE_PAUSED)) != GST_STATE_CHANGE_SUCCESS)
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ret = gst_element_set_state (stream->rtpdec, GST_STATE_PAUSED);
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if (ret != GST_STATE_CHANGE_SUCCESS)
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goto start_rtpdec_failure;
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stream->rtpdecrtp = gst_element_get_pad (stream->rtpdec, "sinkrtp");
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@ -745,17 +734,55 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
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if (transport->lower_transport == RTSP_LOWER_TRANS_TCP) {
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/* configure for interleaved delivery, nothing needs to be done
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* here, the loop function will call the chain functions of the
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* rtp session manager. */
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* RTP session manager. */
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stream->rtpchannel = transport->interleaved.min;
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stream->rtcpchannel = transport->interleaved.max;
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GST_DEBUG ("stream %p on channels %d-%d", stream,
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stream->rtpchannel, stream->rtcpchannel);
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/* also store the caps in the stream */
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/* also store the caps in the stream, we need this when setting caps on
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* outgoing buffers */
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stream->caps = gst_rtspsrc_media_to_caps (media);
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} else {
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/* configure for UDP delivery, we need to connect the udp pads to
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* the rtp session plugin. */
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/* multicast was selected, create UDP sources and connect to the multicast
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* group. */
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if (transport->multicast) {
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gchar *uri;
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/* creating RTP source */
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uri =
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g_strdup_printf ("udp://%s:%d", transport->destination,
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transport->port.min);
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stream->rtpsrc = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
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g_free (uri);
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if (stream->rtpsrc == NULL)
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goto no_element;
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/* creating RTCP source */
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uri =
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g_strdup_printf ("udp://%s:%d", transport->destination,
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transport->port.max);
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stream->rtcpsrc = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
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g_free (uri);
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if (stream->rtcpsrc == NULL)
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goto no_element;
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/* change state */
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gst_element_set_state (stream->rtpsrc, GST_STATE_PAUSED);
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gst_element_set_state (stream->rtcpsrc, GST_STATE_PAUSED);
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/* we manage these elements */
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gst_rtspsrc_add_element (src, stream->rtpsrc);
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gst_rtspsrc_add_element (src, stream->rtcpsrc);
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}
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/* configure caps on the RTP source element */
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stream->caps = gst_rtspsrc_media_to_caps (media);
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g_object_set (G_OBJECT (stream->rtpsrc), "caps", stream->caps, NULL);
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|
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/* configure for UDP delivery, we need to connect the UDP pads to
|
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* the RTP session plugin. */
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pad = gst_element_get_pad (stream->rtpsrc, "src");
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gst_pad_link (pad, stream->rtpdecrtp);
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gst_object_unref (pad);
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@ -1008,9 +1035,8 @@ gst_rtspsrc_open (GstRTSPSrc * src)
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/* create OPTIONS */
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GST_DEBUG_OBJECT (src, "create options...");
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if ((res =
|
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rtsp_message_init_request (RTSP_OPTIONS, src->location,
|
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&request)) < 0)
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res = rtsp_message_init_request (RTSP_OPTIONS, src->location, &request);
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if (res < 0)
|
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goto create_request_failed;
|
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|
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/* send OPTIONS */
|
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|
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@ -1067,11 +1093,11 @@ gst_rtspsrc_open (GstRTSPSrc * src)
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/* create DESCRIBE */
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GST_DEBUG_OBJECT (src, "create describe...");
|
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if ((res =
|
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rtsp_message_init_request (RTSP_DESCRIBE, src->location,
|
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&request)) < 0)
|
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res = rtsp_message_init_request (RTSP_DESCRIBE, src->location, &request);
|
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if (res < 0)
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goto create_request_failed;
|
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/* we accept SDP for now */
|
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|
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/* we only accept SDP for now */
|
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rtsp_message_add_header (&request, RTSP_HDR_ACCEPT, "application/sdp");
|
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|
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/* send DESCRIBE */
|
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|
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@ -1092,7 +1118,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
|
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}
|
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}
|
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|
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/* parse SDP */
|
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/* get message body and parse as SDP */
|
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rtsp_message_get_body (&response, &data, &size);
|
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|
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GST_DEBUG_OBJECT (src, "parse sdp...");
|
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|
|
@ -1102,8 +1128,10 @@ gst_rtspsrc_open (GstRTSPSrc * src)
|
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if (src->debug)
|
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sdp_message_dump (&sdp);
|
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|
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/* we allow all configured protocols */
|
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/* we initially allow all configured protocols. based on the replies from the
|
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* server we narrow them down. */
|
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protocols = src->protocols;
|
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|
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/* setup streams */
|
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{
|
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gint i;
|
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|
|
@ -1135,14 +1163,12 @@ gst_rtspsrc_open (GstRTSPSrc * src)
|
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}
|
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|
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GST_DEBUG_OBJECT (src, "setup %s", setup_url);
|
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|
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/* create SETUP request */
|
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if ((res =
|
||||
rtsp_message_init_request (RTSP_SETUP, setup_url,
|
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&request)) < 0) {
|
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g_free (setup_url);
|
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goto create_request_failed;
|
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}
|
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res = rtsp_message_init_request (RTSP_SETUP, setup_url, &request);
|
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g_free (setup_url);
|
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if (res < 0)
|
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goto create_request_failed;
|
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|
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transports = g_strdup ("");
|
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if (protocols & GST_RTSP_PROTO_UDP_UNICAST) {
|
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|
|
@ -1150,7 +1176,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
|
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gint rtpport, rtcpport;
|
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gchar *trxparams;
|
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|
||||
/* allocate two udp ports */
|
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/* allocate two UDP ports */
|
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if (!gst_rtspsrc_stream_setup_rtp (stream, media, &rtpport, &rtcpport))
|
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goto setup_rtp_failed;
|
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|
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|
|
@ -1167,6 +1193,9 @@ gst_rtspsrc_open (GstRTSPSrc * src)
|
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|
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GST_DEBUG_OBJECT (src, "setting up MULTICAST");
|
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|
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/* 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 =
|
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g_strconcat (transports, transports[0] ? "," : "",
|
||||
"RTP/AVP/UDP;multicast", NULL);
|
||||
|
|
@ -1203,18 +1232,21 @@ gst_rtspsrc_open (GstRTSPSrc * src)
|
|||
|
||||
/* 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;
|
||||
}
|
||||
|
|
@ -1314,9 +1346,8 @@ gst_rtspsrc_close (GstRTSPSrc * src)
|
|||
|
||||
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))
|
||||
|
|
@ -1363,8 +1394,8 @@ gst_rtspsrc_play (GstRTSPSrc * src)
|
|||
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))
|
||||
|
|
@ -1406,8 +1437,8 @@ gst_rtspsrc_pause (GstRTSPSrc * src)
|
|||
|
||||
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))
|
||||
|
|
|
|||
|
|
@ -45,12 +45,12 @@
|
|||
#endif
|
||||
|
||||
#ifdef G_OS_WIN32
|
||||
/* note that inet_aton is deprecated on unix because
|
||||
* inet_addr returns -1 (INADDR_NONE) for the valid 255.255.255.255
|
||||
* address. */
|
||||
static int
|
||||
inet_aton (const char *c, struct in_addr *paddr)
|
||||
{
|
||||
/* note that inet_addr is deprecated on unix because
|
||||
* inet_addr returns -1 (INADDR_NONE) for the valid 255.255.255.255
|
||||
* address. */
|
||||
paddr->s_addr = inet_addr (c);
|
||||
|
||||
if (paddr->s_addr == INADDR_NONE)
|
||||
|
|
|
|||
Loading…
Reference in a new issue