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gst/rtsp/gstrtspsrc.c: Refactor transport configuration code.

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Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
(gst_rtspsrc_alloc_udp_ports), (gst_rtspsrc_handle_src_event),
(gst_rtspsrc_handle_src_query),
(gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_mcast),
(gst_rtspsrc_stream_configure_udp),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
Refactor transport configuration code.
Create internal pads for TCP transport so that we can implement events
and queries.
Handle events and queries.
Parse range from the SDP.
Fix race in pause handler where the connection could still be flushing.
This commit is contained in:
Wim Taymans 2007-05-03 13:48:54 +00:00
parent 24e51b3c73
commit 17011e9a41
2 changed files with 432 additions and 215 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

21
ChangeLog
View file

@ -1,3 +1,24 @@
2007-05-03 Wim Taymans <wim@fluendo.com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
(gst_rtspsrc_alloc_udp_ports), (gst_rtspsrc_handle_src_event),
(gst_rtspsrc_handle_src_query),
(gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_mcast),
(gst_rtspsrc_stream_configure_udp),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
Refactor transport configuration code.
Create internal pads for TCP transport so that we can implement events
and queries.
Handle events and queries.
Parse range from the SDP.
Fix race in pause handler where the connection could still be flushing.
2007-05-02 Wim Taymans <wim@fluendo.com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),

626
gst/rtsp/gstrtspsrc.c
View file

@ -120,6 +120,12 @@ static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream%d",
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));
/* template used internally */
static GstStaticPadTemplate anytemplate = GST_STATIC_PAD_TEMPLATE ("internal%d",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS_ANY);
enum
{
/* FILL ME */
@ -313,7 +319,6 @@ gst_rtspsrc_finalize (GObject * object)
g_free (rtspsrc->req_location);
g_free (rtspsrc->content_base);
rtsp_url_free (rtspsrc->url);
g_free (rtspsrc->addr);
g_static_rec_mutex_free (rtspsrc->state_rec_lock);
g_free (rtspsrc->state_rec_lock);
@ -939,7 +944,7 @@ again:
g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
/* this should not happen */
/* this should not happen... */
if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
goto port_error;
@ -1005,6 +1010,77 @@ cleanup:
}
}
static gboolean
gst_rtspsrc_handle_src_event (GstPad * pad, GstEvent * event)
{
GstRTSPSrc *src;
gboolean res = TRUE;
src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_QOS:
break;
case GST_EVENT_SEEK:
break;
case GST_EVENT_NAVIGATION:
break;
case GST_EVENT_LATENCY:
break;
default:
break;
}
return res;
}
static gboolean
gst_rtspsrc_handle_src_query (GstPad * pad, GstQuery * query)
{
GstRTSPSrc *src;
gboolean res = TRUE;
src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
break;
}
case GST_QUERY_DURATION:
{
GstFormat format;
gst_query_parse_duration (query, &format, NULL);
switch (format) {
case GST_FORMAT_TIME:
break;
default:
res = FALSE;
break;
}
break;
}
case GST_QUERY_LATENCY:
{
/* we are live with a min latency of 0 and unlimted max latency */
gst_query_set_latency (query, TRUE, 0, -1);
break;
}
default:
break;
}
return res;
}
static void
pad_unblocked (GstPad * pad, gboolean blocked, GstRTSPSrc * src)
{
@ -1135,39 +1211,15 @@ unknown_stream:
}
}
/* sets up all elements needed for streaming over the specified transport.
* Does not yet expose the element pads, this will be done when there is actuall
* dataflow detected, which might never happen when UDP is blocked in a
* firewall, for example.
*/
/* try to get and configure a manager */
static gboolean
gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
RTSPTransport * transport)
{
GstRTSPSrc *src;
GstPad *outpad = NULL;
GstPadTemplate *template;
GstStateChangeReturn ret;
const gchar *manager;
gchar *name;
GstStructure *s;
const gchar *mime, *manager;
RTSPResult res;
src = stream->parent;
GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
s = gst_caps_get_structure (stream->caps, 0);
/* get the proper mime type for this stream now */
if ((res = rtsp_transport_get_mime (transport->trans, &mime)) < 0)
goto no_mime;
if (!mime)
goto no_mime;
/* configure the final mime type */
GST_DEBUG_OBJECT (src, "setting mime to %s", mime);
gst_structure_set_name (s, mime);
GstStateChangeReturn ret;
/* find a manager */
if ((res = rtsp_transport_get_manager (transport->trans, &manager, 0)) < 0)
@ -1199,6 +1251,15 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
goto start_session_failure;
g_object_set (src->session, "latency", src->latency, NULL);
/* connect to signals if we did not already do so */
GST_DEBUG_OBJECT (src, "connect to signals on session manager");
src->session_sig_id =
g_signal_connect (src->session, "pad-added",
(GCallback) new_session_pad, src);
src->session_ptmap_id =
g_signal_connect (src->session, "request-pt-map",
(GCallback) request_pt_map, src);
}
/* we stream directly to the manager, get some pads. Each RTSP stream goes
@ -1210,197 +1271,341 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
stream->channelpad[1] = gst_element_get_request_pad (src->session, name);
g_free (name);
}
use_no_manager:
return TRUE;
if (transport->lower_transport == RTSP_LOWER_TRANS_TCP) {
gint i;
/* ERRORS */
no_manager:
{
GST_DEBUG_OBJECT (src, "cannot get a session manager");
return FALSE;
}
manager_failed:
{
GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
return FALSE;
}
start_session_failure:
{
GST_DEBUG_OBJECT (src, "could not start session");
return FALSE;
}
}
/* configure for interleaved delivery, nothing needs to be done
* here, the loop function will call the chain functions of the
* session manager. */
stream->channel[0] = transport->interleaved.min;
stream->channel[1] = transport->interleaved.max;
GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
stream->channel[0], stream->channel[1]);
/* free the UDP sources allocated when negotiating a transport.
* This function is called when the server negotiated to a transport where the
* UDP sources are not needed anymore, such as TCP or multicast. */
static void
gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
{
gint i;
/* we can remove the allocated UDP ports now */
for (i = 0; i < 2; i++) {
if (stream->udpsrc[i]) {
gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
gst_object_unref (stream->udpsrc[i]);
stream->udpsrc[i] = NULL;
}
for (i = 0; i < 2; i++) {
if (stream->udpsrc[i]) {
gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
gst_object_unref (stream->udpsrc[i]);
stream->udpsrc[i] = NULL;
}
}
}
/* no session manager, send data to srcpad directly */
if (!stream->channelpad[0]) {
GST_DEBUG_OBJECT (src, "no manager, creating pad");
/* for TCP, create pads to send and receive data to and from the manager and to
* intercept various events and queries
*/
static gboolean
gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
RTSPTransport * transport, GstPad ** outpad)
{
gchar *name;
GstPadTemplate *template;
GstPad *pad0, *pad1;
/* create a new pad we will use to stream to */
name = g_strdup_printf ("stream%d", stream->id);
template = gst_static_pad_template_get (&rtptemplate);
stream->channelpad[0] = gst_pad_new_from_template (template, name);
gst_object_unref (template);
g_free (name);
/* configure for interleaved delivery, nothing needs to be done
* here, the loop function will call the chain functions of the
* session manager. */
stream->channel[0] = transport->interleaved.min;
stream->channel[1] = transport->interleaved.max;
GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
stream->channel[0], stream->channel[1]);
/* set caps and activate */
gst_pad_use_fixed_caps (stream->channelpad[0]);
gst_pad_set_active (stream->channelpad[0], TRUE);
/* we can remove the allocated UDP ports now */
gst_rtspsrc_stream_free_udp (stream);
outpad = gst_object_ref (stream->channelpad[0]);
} else {
GST_DEBUG_OBJECT (src, "using manager source pad");
/* we connected to pad-added signal to get pads from the manager */
}
/* no session manager, send data to srcpad directly */
if (!stream->channelpad[0]) {
GST_DEBUG_OBJECT (src, "no manager, creating pad");
/* create a new pad we will use to stream to */
name = g_strdup_printf ("stream%d", stream->id);
template = gst_static_pad_template_get (&rtptemplate);
stream->channelpad[0] = gst_pad_new_from_template (template, name);
gst_object_unref (template);
g_free (name);
/* set caps and activate */
gst_pad_use_fixed_caps (stream->channelpad[0]);
gst_pad_set_active (stream->channelpad[0], TRUE);
*outpad = gst_object_ref (stream->channelpad[0]);
} else {
/* multicast was selected, create UDP sources and join the multicast
* group. */
if (transport->lower_transport == RTSP_LOWER_TRANS_UDP_MCAST) {
gchar *uri;
GstPadTemplate *template;
GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
GST_DEBUG_OBJECT (src, "using manager source pad");
/* creating UDP source */
if (transport->port.min != -1) {
uri = g_strdup_printf ("udp://%s:%d", transport->destination,
transport->port.min);
stream->udpsrc[0] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
g_free (uri);
if (stream->udpsrc[0] == NULL)
goto no_element;
template = gst_static_pad_template_get (&anytemplate);
/* take ownership */
gst_object_ref (stream->udpsrc[0]);
gst_object_sink (stream->udpsrc[0]);
/* allocate pads for sending the channel data into the manager */
pad0 = gst_pad_new_from_template (template, "internal0");
gst_pad_set_event_function (pad0, gst_rtspsrc_handle_src_event);
gst_pad_set_query_function (pad0, gst_rtspsrc_handle_src_query);
gst_pad_link (pad0, stream->channelpad[0]);
stream->channelpad[0] = pad0;
gst_pad_set_element_private (pad0, src);
/* change state */
gst_element_set_state (stream->udpsrc[0], GST_STATE_READY);
}
/* creating another UDP source */
if (transport->port.max != -1) {
uri = g_strdup_printf ("udp://%s:%d", transport->destination,
transport->port.max);
stream->udpsrc[1] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
g_free (uri);
if (stream->udpsrc[1] == NULL)
goto no_element;
/* take ownership */
gst_object_ref (stream->udpsrc[1]);
gst_object_sink (stream->udpsrc[1]);
gst_element_set_state (stream->udpsrc[1], GST_STATE_READY);
}
if (stream->channelpad[1]) {
/* if we have a sinkpad for the other channel, create a pad and link to the
* manager. */
pad1 = gst_pad_new_from_template (template, "internal1");
gst_pad_link (pad1, stream->channelpad[1]);
stream->channelpad[1] = pad1;
}
gst_object_unref (template);
}
return TRUE;
}
/* we manage the UDP elements now. For unicast, the UDP sources where
* allocated in the stream when we suggested a transport. */
if (stream->udpsrc[0]) {
gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
/* For multicast create UDP sources and join the multicast group. */
static gboolean
gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
RTSPTransport * transport, GstPad ** outpad)
{
gchar *uri;
GST_DEBUG_OBJECT (src, "setting up UDP source");
GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
/* configure a timeout on the UDP port. When the timeout message is
* posted, we assume UDP transport is not possible. We reconnect using TCP
* if we can. */
g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", src->timeout,
NULL);
/* we can remove the allocated UDP ports now */
gst_rtspsrc_stream_free_udp (stream);
/* get output pad of the UDP source. */
outpad = gst_element_get_pad (stream->udpsrc[0], "src");
/* creating UDP source */
if (transport->port.min != -1) {
uri = g_strdup_printf ("udp://%s:%d", transport->destination,
transport->port.min);
stream->udpsrc[0] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
g_free (uri);
if (stream->udpsrc[0] == NULL)
goto no_element;
/* save it so we can unblock */
stream->blockedpad = outpad;
/* take ownership */
gst_object_ref (stream->udpsrc[0]);
gst_object_sink (stream->udpsrc[0]);
/* configure pad block on the pad. As soon as there is dataflow on the
* UDP source, we know that UDP is not blocked by a firewall and we can
* configure all the streams to let the application autoplug decoders. */
gst_pad_set_blocked_async (outpad, TRUE,
(GstPadBlockCallback) pad_blocked, src);
/* change state */
gst_element_set_state (stream->udpsrc[0], GST_STATE_READY);
}
if (stream->channelpad[0]) {
GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
/* configure for UDP delivery, we need to connect the UDP pads to
* the session plugin. */
gst_pad_link (outpad, stream->channelpad[0]);
gst_object_unref (outpad);
outpad = NULL;
/* we connected to pad-added signal to get pads from the manager */
} else {
GST_DEBUG_OBJECT (src, "using UDP src pad as output");
}
}
/* creating another UDP source */
if (transport->port.max != -1) {
uri = g_strdup_printf ("udp://%s:%d", transport->destination,
transport->port.max);
stream->udpsrc[1] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
g_free (uri);
if (stream->udpsrc[1] == NULL)
goto no_element;
if (stream->udpsrc[1]) {
gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
/* take ownership */
gst_object_ref (stream->udpsrc[1]);
gst_object_sink (stream->udpsrc[1]);
if (stream->channelpad[1]) {
GstPad *pad;
gst_element_set_state (stream->udpsrc[1], GST_STATE_READY);
}
return TRUE;
GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
/* ERRORS */
no_element:
{
GST_DEBUG_OBJECT (src, "no UDP source element found");
return FALSE;
}
}
pad = gst_element_get_pad (stream->udpsrc[1], "src");
gst_pad_link (pad, stream->channelpad[1]);
gst_object_unref (pad);
}
}
/* configure udpsink back to the server for RTCP messages. */
{
GstPad *pad;
gint port;
gchar *destination, *uri;
/* configure the remainder of the UDP ports */
static gboolean
gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
RTSPTransport * transport, GstPad ** outpad)
{
/* we manage the UDP elements now. For unicast, the UDP sources where
* allocated in the stream when we suggested a transport. */
if (stream->udpsrc[0]) {
gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
/* get host and port */
if (transport->lower_transport == RTSP_LOWER_TRANS_UDP_MCAST)
port = transport->port.max;
else
port = transport->server_port.max;
GST_DEBUG_OBJECT (src, "setting up UDP source");
/* first take the source, then the endpoint to figure out where to send
* the RTCP. */
destination = transport->source;
if (destination == NULL)
destination = src->connection->ip;
/* configure a timeout on the UDP port. When the timeout message is
* posted, we assume UDP transport is not possible. We reconnect using TCP
* if we can. */
g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", src->timeout, NULL);
GST_DEBUG_OBJECT (src, "configure UDP sink for %s:%d", destination, port);
/* get output pad of the UDP source. */
*outpad = gst_element_get_pad (stream->udpsrc[0], "src");
uri = g_strdup_printf ("udp://%s:%d", destination, port);
stream->udpsink = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
g_free (uri);
if (stream->udpsink == NULL)
goto no_sink_element;
/* save it so we can unblock */
stream->blockedpad = *outpad;
/* we keep this playing always */
gst_element_set_locked_state (stream->udpsink, TRUE);
gst_element_set_state (stream->udpsink, GST_STATE_PLAYING);
/* configure pad block on the pad. As soon as there is dataflow on the
* UDP source, we know that UDP is not blocked by a firewall and we can
* configure all the streams to let the application autoplug decoders. */
gst_pad_set_blocked_async (stream->blockedpad, TRUE,
(GstPadBlockCallback) pad_blocked, src);
/* no sync needed */
g_object_set (G_OBJECT (stream->udpsink), "sync", FALSE, NULL);
gst_object_ref (stream->udpsink);
gst_bin_add (GST_BIN_CAST (src), stream->udpsink);
stream->rtcppad = gst_element_get_pad (stream->udpsink, "sink");
/* get session RTCP pad */
name = g_strdup_printf ("send_rtcp_src_%d", stream->id);
pad = gst_element_get_request_pad (src->session, name);
g_free (name);
/* and link */
gst_pad_link (pad, stream->rtcppad);
if (stream->channelpad[0]) {
GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
/* configure for UDP delivery, we need to connect the UDP pads to
* the session plugin. */
gst_pad_link (*outpad, stream->channelpad[0]);
gst_object_unref (*outpad);
*outpad = NULL;
/* we connected to pad-added signal to get pads from the manager */
} else {
GST_DEBUG_OBJECT (src, "using UDP src pad as output");
}
}
if (src->session && !src->session_sig_id) {
GST_DEBUG_OBJECT (src, "connect to signals on session manager");
src->session_sig_id =
g_signal_connect (src->session, "pad-added",
(GCallback) new_session_pad, src);
src->session_ptmap_id =
g_signal_connect (src->session, "request-pt-map",
(GCallback) request_pt_map, src);
/* RTCP port */
if (stream->udpsrc[1]) {
gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
if (stream->channelpad[1]) {
GstPad *pad;
GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
pad = gst_element_get_pad (stream->udpsrc[1], "src");
gst_pad_link (pad, stream->channelpad[1]);
gst_object_unref (pad);
} else {
/* leave unlinked */
}
}
return TRUE;
}
/* configure the UDP sink back to the server for status reports */
static gboolean
gst_rtspsrc_stream_configure_udp_sink (GstRTSPSrc * src, GstRTSPStream * stream,
RTSPTransport * transport)
{
GstPad *pad;
gint port;
gchar *destination, *uri, *name;
/* get host and port */
if (transport->lower_transport == RTSP_LOWER_TRANS_UDP_MCAST)
port = transport->port.max;
else
port = transport->server_port.max;
/* first take the source, then the endpoint to figure out where to send
* the RTCP. */
destination = transport->source;
if (destination == NULL)
destination = src->connection->ip;
GST_DEBUG_OBJECT (src, "configure UDP sink for %s:%d", destination, port);
uri = g_strdup_printf ("udp://%s:%d", destination, port);
stream->udpsink = gst_element_make_from_uri (GST_URI_SINK, uri, NULL);
g_free (uri);
if (stream->udpsink == NULL)
goto no_sink_element;
/* we keep this playing always */
gst_element_set_locked_state (stream->udpsink, TRUE);
gst_element_set_state (stream->udpsink, GST_STATE_PLAYING);
/* no sync needed */
g_object_set (G_OBJECT (stream->udpsink), "sync", FALSE, NULL);
gst_object_ref (stream->udpsink);
gst_bin_add (GST_BIN_CAST (src), stream->udpsink);
stream->rtcppad = gst_element_get_pad (stream->udpsink, "sink");
/* get session RTCP pad */
name = g_strdup_printf ("send_rtcp_src_%d", stream->id);
pad = gst_element_get_request_pad (src->session, name);
g_free (name);
/* and link */
gst_pad_link (pad, stream->rtcppad);
return TRUE;
/* ERRORS */
no_sink_element:
{
GST_DEBUG_OBJECT (src, "no UDP sink element found");
return FALSE;
}
}
/* sets up all elements needed for streaming over the specified transport.
* Does not yet expose the element pads, this will be done when there is actuall
* dataflow detected, which might never happen when UDP is blocked in a
* firewall, for example.
*/
static gboolean
gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
RTSPTransport * transport)
{
GstRTSPSrc *src;
GstPad *outpad = NULL;
GstPadTemplate *template;
gchar *name;
GstStructure *s;
const gchar *mime;
RTSPResult res;
src = stream->parent;
GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
s = gst_caps_get_structure (stream->caps, 0);
/* get the proper mime type for this stream now */
if ((res = rtsp_transport_get_mime (transport->trans, &mime)) < 0)
goto unknown_transport;
if (!mime)
goto unknown_transport;
/* configure the final mime type */
GST_DEBUG_OBJECT (src, "setting mime to %s", mime);
gst_structure_set_name (s, mime);
/* try to get and configure a manager, channelpad[0-1] will be configured with
* the pads for the manager, or NULL when no manager is needed. */
if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
goto no_manager;
switch (transport->lower_transport) {
case RTSP_LOWER_TRANS_TCP:
if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
goto transport_failed;
break;
case RTSP_LOWER_TRANS_UDP_MCAST:
if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
goto transport_failed;
/* fallthrough, the rest is the same for UDP and MCAST */
case RTSP_LOWER_TRANS_UDP:
if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
goto transport_failed;
/* configure udpsink back to the server for RTCP messages. */
if (!gst_rtspsrc_stream_configure_udp_sink (src, stream, transport))
goto transport_failed;
break;
default:
goto unknown_transport;
}
if (outpad) {
@ -1424,7 +1629,12 @@ use_no_manager:
return TRUE;
/* ERRORS */
no_mime:
transport_failed:
{
GST_DEBUG_OBJECT (src, "failed to configure transport");
return FALSE;
}
unknown_transport:
{
GST_DEBUG_OBJECT (src, "unknown transport");
return FALSE;
@ -1434,26 +1644,6 @@ no_manager:
GST_DEBUG_OBJECT (src, "cannot get a session manager");
return FALSE;
}
manager_failed:
{
GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
return FALSE;
}
no_element:
{
GST_DEBUG_OBJECT (src, "no UDP source element found");
return FALSE;
}
no_sink_element:
{
GST_DEBUG_OBJECT (src, "no UDP sink element found");
return FALSE;
}
start_session_failure:
{
GST_DEBUG_OBJECT (src, "could not start session");
return FALSE;
}
}
/* Adds the source pads of all configured streams to the element.
@ -1543,7 +1733,7 @@ gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
for (streams = src->streams; streams; streams = g_list_next (streams)) {
GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
/* only pads that have a connection to the outside world */
/* only streams that have a connection to the outside world */
if (ostream->srcpad == NULL)
continue;
@ -1867,7 +2057,7 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
rtsp_connection_flush (src->connection, FALSE);
goto interrupt;
case RTSP_ETIMEOUT:
/* ignore result, a warning was posted */
/* send keep-alive, ignore the result, a warning will be posted. */
GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
res = gst_rtspsrc_send_keep_alive (src);
continue;
@ -2814,14 +3004,13 @@ gst_rtspsrc_open (GstRTSPSrc * src)
if (src->extension && src->extension->parse_sdp)
src->extension->parse_sdp (src->extension, &sdp);
/* parse address */
/* parse range */
{
SDPOrigin *origin;
gchar *range;
origin = sdp_message_get_origin (&sdp);
range = sdp_message_get_attribute_val (&sdp, "range");
g_free (src->addr);
src->addr = g_strdup (origin->addr);
GST_DEBUG_OBJECT (src, "got range: %s", GST_STR_NULL (range));
}
/* create streams */
@ -3070,6 +3259,7 @@ gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
timebase = atol (fields[j] + 8);
}
}
g_strfreev (fields);
/* now we need to store the values in the caps of the stream and make sure
* that the UDP elements have the same caps property set before they receive
* the first buffer. */
@ -3132,7 +3322,7 @@ gst_rtspsrc_play (GstRTSPSrc * src)
/* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
* for the RTP packets. If this is not present, we assume all starts from 0...
* FIXME, this is info for the RTP session manager ideally. */
* This is info for the RTP session manager that we pass to it in caps. */
rtsp_message_get_header (&response, RTSP_HDR_RTP_INFO, &rtpinfo);
if (rtpinfo)
gst_rtspsrc_parse_rtpinfo (src, rtpinfo);
@ -3201,6 +3391,12 @@ gst_rtspsrc_pause (GstRTSPSrc * src)
if (src->state == RTSP_STATE_READY)
goto was_paused;
/* wait for streaming to finish */
GST_RTSP_STREAM_LOCK (src);
GST_RTSP_STREAM_UNLOCK (src);
rtsp_connection_flush (src->connection, FALSE);
/* do pause */
res = rtsp_message_init_request (&request, RTSP_PAUSE, src->req_location);
if (res < 0)