From 485d25aef1318f18e41c6b6f224deb37149d9632 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Wed, 21 Sep 2005 17:53:26 +0000 Subject: gst/rtsp/gstrtspsrc.c: More SDP parsing and caps setting. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send), (gst_rtspsrc_change_state): More SDP parsing and caps setting. Do NO_PREROLL differently. add pads only after negotiated. * gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_getcaps): Implement the getcaps function. --- gst/rtsp/gstrtspsrc.c | 31 +++++++++++++++++++------------ 1 file changed, 19 insertions(+), 12 deletions(-) (limited to 'gst/rtsp') diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index f0c5d06a..b36813b7 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -448,8 +448,15 @@ gst_rtspsrc_media_to_caps (SDPMedia * media) gchar **keyval; keyval = g_strsplit (pairs[i], "=", 0); - if (keyval[0] && keyval[1]) { - gst_structure_set (s, keyval[0], G_TYPE_STRING, keyval[1], NULL); + if (keyval[0]) { + gchar *val; + + if (keyval[1]) + val = keyval[1]; + else + val = "1"; + + gst_structure_set (s, keyval[0], G_TYPE_STRING, val, NULL); } g_strfreev (keyval); } @@ -499,9 +506,6 @@ gst_rtspsrc_stream_setup_rtp (GstRTSPStream * stream, SDPMedia * media, g_object_get (G_OBJECT (stream->rtpsrc), "port", rtpport, NULL); g_object_get (G_OBJECT (stream->rtcpsrc), "port", rtcpport, NULL); - gst_element_set_state (stream->rtpsrc, GST_STATE_READY); - gst_element_set_state (stream->rtcpsrc, GST_STATE_READY); - return TRUE; /* ERRORS, FIXME, cleanup */ @@ -552,13 +556,6 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream, 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_pad (GST_ELEMENT (src), gst_ghost_pad_new (name, pad)); - 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 @@ -574,6 +571,13 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream, gst_pad_link (pad, stream->rtpdecrtcp); gst_object_unref (GST_OBJECT (pad)); } + + pad = gst_element_get_pad (stream->rtpdec, "srcrtp"); + name = g_strdup_printf ("rtp_stream%d", stream->id); + gst_element_add_pad (GST_ELEMENT (src), gst_ghost_pad_new (name, pad)); + g_free (name); + gst_object_unref (GST_OBJECT (pad)); + return TRUE; no_element: @@ -1181,6 +1185,9 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition) goto done; switch (transition) { + case GST_STATE_CHANGE_READY_TO_PAUSED: + ret = GST_STATE_CHANGE_NO_PREROLL; + break; case GST_STATE_CHANGE_PLAYING_TO_PAUSED: gst_rtspsrc_pause (rtspsrc); break; -- cgit