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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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gst/rtsp/gstrtspsrc.*: Setup UDP sources correctly, receives raw data from RTSP compliant servers now.
Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type), (gst_rtspsrc_class_init), (gst_rtspsrc_init), (gst_rtspsrc_create_stream), (gst_rtspsrc_add_element), (gst_rtspsrc_set_state), (gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_stream_configure_transport), (find_stream), (gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Setup UDP sources correctly, receives raw data from RTSP compliant servers now.
This commit is contained in:
Wim Taymans
parent
6f0ea35883
commit
91ce2b294e
3 changed files with 120 additions and 80 deletions
13
ChangeLog
13
ChangeLog
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@ -1,3 +1,16 @@
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2005-05-11 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
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(gst_rtspsrc_class_init), (gst_rtspsrc_init),
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(gst_rtspsrc_create_stream), (gst_rtspsrc_add_element),
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(gst_rtspsrc_set_state), (gst_rtspsrc_stream_setup_rtp),
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(gst_rtspsrc_stream_configure_transport), (find_stream),
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(gst_rtspsrc_loop), (gst_rtspsrc_open), (gst_rtspsrc_close),
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(gst_rtspsrc_play), (gst_rtspsrc_change_state):
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* gst/rtsp/gstrtspsrc.h:
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Setup UDP sources correctly, receives raw data from RTSP
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compliant servers now.
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2005-05-11 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/.cvsignore:
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@ -89,7 +89,6 @@ static void gst_rtspsrc_class_init (GstRTSPSrc * klass);
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static void gst_rtspsrc_init (GstRTSPSrc * rtspsrc);
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static GstElementStateReturn gst_rtspsrc_change_state (GstElement * element);
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static gboolean gst_rtspsrc_activate (GstPad * pad, GstActivateMode mode);
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static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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@ -167,7 +166,7 @@ gst_rtspsrc_class_init (GstRTSPSrc * klass)
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_DEBUG,
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g_param_spec_boolean ("debug", "Debug",
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"Dump request qnd response messages to stdout",
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"Dump request and response messages to stdout",
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DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
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gstelement_class->change_state = gst_rtspsrc_change_state;
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@ -176,14 +175,6 @@ gst_rtspsrc_class_init (GstRTSPSrc * klass)
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static void
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gst_rtspsrc_init (GstRTSPSrc * src)
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{
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/*
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src->srcpad =
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gst_pad_new_from_template (gst_static_pad_template_get (&srctemplate),
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"src");
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gst_pad_set_loop_function (src->srcpad, gst_rtspsrc_loop);
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gst_pad_set_activate_function (src->srcpad, gst_rtspsrc_activate);
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gst_element_add_pad (GST_ELEMENT (src), src->srcpad);
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*/
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}
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static void
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@ -242,6 +233,7 @@ gst_rtspsrc_create_stream (GstRTSPSrc * src)
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s = g_new0 (GstRTSPStream, 1);
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s->parent = src;
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s->id = src->numstreams++;
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src->streams = g_list_append (src->streams, s);
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@ -249,7 +241,7 @@ gst_rtspsrc_create_stream (GstRTSPSrc * src)
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}
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static gboolean
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rtspsrc_add_element (GstRTSPSrc * src, GstElement * element)
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gst_rtspsrc_add_element (GstRTSPSrc * src, GstElement * element)
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{
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gst_object_set_parent (GST_OBJECT (element), GST_OBJECT (src));
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gst_element_set_manager (element, GST_ELEMENT_MANAGER (src));
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@ -258,6 +250,42 @@ rtspsrc_add_element (GstRTSPSrc * src, GstElement * element)
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return TRUE;
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}
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static GstElementStateReturn
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gst_rtspsrc_set_state (GstRTSPSrc * src, GstElementState state)
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{
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GstElementStateReturn ret;
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GList *streams;
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/* for all streams */
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for (streams = src->streams; streams; streams = g_list_next (streams)) {
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GstRTSPStream *stream;
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stream = (GstRTSPStream *) streams->data;
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/* first our rtp session manager */
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if ((ret =
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gst_element_set_state (stream->rtpdec, state)) != GST_STATE_SUCCESS)
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goto done;
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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_SUCCESS)
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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_SUCCESS)
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goto done;
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}
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}
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done:
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return ret;
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}
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static gboolean
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gst_rtspsrc_stream_setup_rtp (GstRTSPStream * stream, gint * rtpport,
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gint * rtcpport)
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@ -273,7 +301,7 @@ gst_rtspsrc_stream_setup_rtp (GstRTSPStream * stream, gint * rtpport,
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goto no_udp_rtp_protocol;
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/* we manage this element */
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rtspsrc_add_element (src, stream->rtpsrc);
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gst_rtspsrc_add_element (src, stream->rtpsrc);
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if ((ret =
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gst_element_set_state (stream->rtpsrc,
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@ -285,7 +313,7 @@ gst_rtspsrc_stream_setup_rtp (GstRTSPStream * stream, gint * rtpport,
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goto no_udp_rtcp_protocol;
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/* we manage this element */
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rtspsrc_add_element (src, stream->rtcpsrc);
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gst_rtspsrc_add_element (src, stream->rtcpsrc);
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if ((ret =
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gst_element_set_state (stream->rtcpsrc,
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@ -325,27 +353,47 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
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RTSPTransport * transport)
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{
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GstRTSPSrc *src;
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GstPad *pad;
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GstElementStateReturn ret;
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gchar *name;
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src = stream->parent;
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if (transport->lower_transport == RTSP_LOWER_TRANS_TCP) {
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GstPad *pad;
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/* configure for interleaved delivery */
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if (!(stream->rtpdec = gst_element_factory_make ("rtpdec", NULL)))
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goto no_element;
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/* we manage this element */
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rtspsrc_add_element (src, stream->rtpdec);
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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_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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stream->rtpdecrtcp = gst_element_get_pad (stream->rtpdec, "sinkrtcp");
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/* FIXME, make sure it outputs the caps */
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pad = gst_element_get_pad (stream->rtpdec, "srcrtp");
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gst_element_add_ghost_pad (GST_ELEMENT (src), pad, "srcrtp");
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name = g_strdup_printf ("rtp_stream%d", stream->id);
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gst_element_add_ghost_pad (GST_ELEMENT (src), pad, name);
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g_free (name);
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gst_object_unref (GST_OBJECT (pad));
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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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} else {
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/* configure for UDP delivery, FIXME */
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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 (GST_OBJECT (pad));
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pad = gst_element_get_pad (stream->rtcpsrc, "src");
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gst_pad_link (pad, stream->rtpdecrtcp);
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gst_object_unref (GST_OBJECT (pad));
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}
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return TRUE;
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@ -354,6 +402,11 @@ no_element:
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GST_DEBUG ("no rtpdec element found");
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return FALSE;
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}
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start_rtpdec_failure:
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{
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GST_DEBUG ("could not start RTP session");
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return FALSE;
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}
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}
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static gint
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@ -602,8 +655,6 @@ gst_rtspsrc_open (GstRTSPSrc * src)
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rtsp_message_add_header (&request, RTSP_HDR_TRANSPORT, transports);
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g_free (transports);
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rtsp_message_dump (&request);
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if (!gst_rtspsrc_send (src, &request, &response))
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goto send_error;
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@ -614,8 +665,9 @@ gst_rtspsrc_open (GstRTSPSrc * src)
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rtsp_message_get_header (&response, RTSP_HDR_TRANSPORT, &resptrans);
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/* update allowed transports for other streams */
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/* parse transport */
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rtsp_transport_parse (resptrans, &transport);
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/* update allowed transports for other streams */
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if (transport.lower_transport == RTSP_LOWER_TRANS_TCP) {
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protocols = GST_RTSP_PROTO_TCP;
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src->interleaved = TRUE;
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@ -628,7 +680,11 @@ gst_rtspsrc_open (GstRTSPSrc * src)
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protocols = GST_RTSP_PROTO_UDP_UNICAST;
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}
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}
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gst_rtspsrc_stream_configure_transport (stream, &transport);
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/* now configure the stream with the transport */
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if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
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GST_DEBUG ("could not configure stream transport, skipping stream");
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}
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/* clean up our transport struct */
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rtsp_transport_init (&transport);
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}
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}
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@ -670,6 +726,14 @@ gst_rtspsrc_close (GstRTSPSrc * src)
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RTSPResult res;
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GST_DEBUG ("TEARDOWN...");
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/* stop task if any */
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if (src->task) {
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gst_task_stop (src->task);
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gst_object_unref (GST_OBJECT (src->task));
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src->task = NULL;
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}
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/* do TEARDOWN */
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if ((res =
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rtsp_message_init_request (RTSP_TEARDOWN, src->location,
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@ -713,6 +777,7 @@ gst_rtspsrc_play (GstRTSPSrc * src)
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RTSPResult res;
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GST_DEBUG ("PLAY...");
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/* do play */
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if ((res =
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rtsp_message_init_request (RTSP_PLAY, src->location, &request)) < 0)
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@ -777,52 +842,6 @@ send_error:
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}
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}
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static gboolean
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gst_rtspsrc_activate (GstPad * pad, GstActivateMode mode)
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{
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gboolean result;
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GstRTSPSrc *rtspsrc;
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rtspsrc = GST_RTSPSRC (GST_OBJECT_PARENT (pad));
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switch (mode) {
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case GST_ACTIVATE_PUSH:
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/* if we have a scheduler we can start the task */
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if (GST_ELEMENT_SCHEDULER (rtspsrc) && rtspsrc->interleaved) {
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GST_STREAM_LOCK (pad);
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GST_RPAD_TASK (pad) =
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gst_scheduler_create_task (GST_ELEMENT_SCHEDULER (rtspsrc),
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(GstTaskFunction) gst_rtspsrc_loop, pad);
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gst_task_start (GST_RPAD_TASK (pad));
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GST_STREAM_UNLOCK (pad);
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result = TRUE;
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}
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break;
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case GST_ACTIVATE_PULL:
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result = FALSE;
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break;
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case GST_ACTIVATE_NONE:
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/* step 1, unblock clock sync (if any) */
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/* step 2, make sure streaming finishes */
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GST_STREAM_LOCK (pad);
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gst_rtspsrc_close (rtspsrc);
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/* step 3, stop the task */
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if (GST_RPAD_TASK (pad)) {
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gst_task_stop (GST_RPAD_TASK (pad));
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gst_object_unref (GST_OBJECT (GST_RPAD_TASK (pad)));
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GST_RPAD_TASK (pad) = NULL;
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}
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GST_STREAM_UNLOCK (pad);
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result = TRUE;
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break;
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}
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return result;
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}
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static GstElementStateReturn
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gst_rtspsrc_change_state (GstElement * element)
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{
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@ -845,18 +864,22 @@ gst_rtspsrc_change_state (GstElement * element)
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gst_rtspsrc_play (rtspsrc);
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break;
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case GST_STATE_PAUSED_TO_PLAYING:
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gst_rtspsrc_play (rtspsrc);
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break;
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default:
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break;
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}
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ret = gst_rtspsrc_set_state (rtspsrc, GST_STATE_PENDING (rtspsrc));
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if (ret != GST_STATE_SUCCESS)
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goto error;
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element);
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switch (transition) {
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case GST_STATE_PLAYING_TO_PAUSED:
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gst_rtspsrc_pause (rtspsrc);
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break;
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case GST_STATE_PAUSED_TO_READY:
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gst_rtspsrc_pause (rtspsrc);
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break;
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case GST_STATE_READY_TO_NULL:
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break;
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@ -864,5 +887,6 @@ gst_rtspsrc_change_state (GstElement * element)
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break;
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}
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error:
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return ret;
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}
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@ -55,6 +55,8 @@ typedef enum
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typedef struct _GstRTSPStream GstRTSPStream;
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struct _GstRTSPStream {
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gint id;
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gint rtpchannel;
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gint rtcpchannel;
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@ -79,6 +81,7 @@ struct _GstRTSPSrc {
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gboolean interleaved;
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GstTask *task;
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gint numstreams;
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GList *streams;
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gchar *location;
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