Commit c831aef4 authored by Wim Taymans's avatar Wim Taymans
Browse files

gst/rtsp/: Handle RTSP defaults better.

Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.c: (rtsp_method_as_text), (rtsp_find_method):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_parse):
Handle RTSP defaults better.
Issue OPTIONS request to figure out what we are allowed to do.
Make the methods a bitfield so we can easily collect supported
options.
Fix rtsp_find_method.
Do proper RTSP connection shutdown.
parent 9a5c8cd2
2005-08-18 Wim Taymans <wim@fluendo.com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.c: (rtsp_method_as_text), (rtsp_find_method):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_parse):
Handle RTSP defaults better.
Issue OPTIONS request to figure out what we are allowed to do.
Make the methods a bitfield so we can easily collect supported
options.
Fix rtsp_find_method.
Do proper RTSP connection shutdown.
2005-08-18 Wim Taymans <wim@fluendo.com>
 
* gst/rtp/Makefile.am:
......
......@@ -495,7 +495,7 @@ need_pause:
static gboolean
gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
RTSPMessage * response)
RTSPMessage * response, RTSPStatusCode * code)
{
RTSPResult res;
......@@ -507,6 +507,11 @@ gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
if ((res = rtsp_connection_receive (src->connection, response)) < 0)
goto receive_error;
if (code) {
*code = response->type_data.response.code;
}
if (response->type_data.response.code != RTSP_STS_OK)
goto error_response;
......@@ -559,6 +564,48 @@ gst_rtspsrc_open (GstRTSPSrc * src)
if ((res = rtsp_connection_open (url, &src->connection)) < 0)
goto could_not_open;
/* create OPTIONS */
GST_DEBUG ("create options...");
if ((res =
rtsp_message_init_request (RTSP_OPTIONS, src->location,
&request)) < 0)
goto create_request_failed;
/* send OPTIONS */
GST_DEBUG ("send options...");
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
{
gchar *respoptions = NULL;
gchar **options;
gint i;
rtsp_message_get_header (&response, RTSP_HDR_ALLOW, &respoptions);
if (!respoptions)
goto no_options;
/* parse options */
options = g_strsplit (respoptions, ",", 0);
i = 0;
while (options[i]) {
gint method = rtsp_find_method (options[i]);
/* keep bitfield of supported methods */
if (method != -1)
src->options |= method;
i++;
}
g_strfreev (options);
/* we need describe and setup */
if (!(src->options & RTSP_DESCRIBE))
goto no_describe;
if (!(src->options & RTSP_SETUP))
goto no_setup;
}
/* create DESCRIBE */
GST_DEBUG ("create describe...");
if ((res =
......@@ -570,9 +617,22 @@ gst_rtspsrc_open (GstRTSPSrc * src)
/* send DESCRIBE */
GST_DEBUG ("send describe...");
if (!gst_rtspsrc_send (src, &request, &response))
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
/* check if reply is SDP */
{
gchar *respcont = NULL;
rtsp_message_get_header (&response, RTSP_HDR_CONTENT_TYPE, &respcont);
/* could not be set but since the request returned OK, we assume it
* was SDP, else check it. */
if (respcont) {
if (!g_ascii_strcasecmp (respcont, "application/sdp") == 0)
goto wrong_content_type;
}
}
/* parse SDP */
rtsp_message_get_body (&response, &data, &size);
......@@ -622,7 +682,6 @@ gst_rtspsrc_open (GstRTSPSrc * src)
}
g_free (setup_url);
transports = g_strdup ("");
if (protocols & GST_RTSP_PROTO_UDP_UNICAST) {
gchar *new;
......@@ -662,15 +721,17 @@ gst_rtspsrc_open (GstRTSPSrc * src)
rtsp_message_add_header (&request, RTSP_HDR_TRANSPORT, transports);
g_free (transports);
if (!gst_rtspsrc_send (src, &request, &response))
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
/* parse response transport */
{
gchar *resptrans;
gchar *resptrans = NULL;
RTSPTransport transport = { 0 };
rtsp_message_get_header (&response, RTSP_HDR_TRANSPORT, &resptrans);
if (!resptrans)
goto no_transport;
/* parse transport */
rtsp_transport_parse (resptrans, &transport);
......@@ -723,14 +784,44 @@ send_error:
("Could not send message."), (NULL));
return FALSE;
}
no_options:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Invalid OPTIONS response."), (NULL));
return FALSE;
}
no_describe:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Server does not support DESCRIBE."), (NULL));
return FALSE;
}
no_setup:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Server does not support SETUP."), (NULL));
return FALSE;
}
wrong_content_type:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Server does not support SDP."), (NULL));
return FALSE;
}
setup_rtp_failed:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, ("Could not setup rtp."), (NULL));
return FALSE;
}
no_transport:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Server did not select transport."), (NULL));
return FALSE;
}
}
G_GNUC_UNUSED static gboolean
static gboolean
gst_rtspsrc_close (GstRTSPSrc * src)
{
RTSPMessage request = { 0 };
......@@ -746,14 +837,16 @@ gst_rtspsrc_close (GstRTSPSrc * src)
src->task = NULL;
}
/* do TEARDOWN */
if ((res =
rtsp_message_init_request (RTSP_TEARDOWN, src->location,
&request)) < 0)
goto create_request_failed;
if (src->options & RTSP_PLAY) {
/* do TEARDOWN */
if ((res =
rtsp_message_init_request (RTSP_TEARDOWN, src->location,
&request)) < 0)
goto create_request_failed;
if (!gst_rtspsrc_send (src, &request, &response))
goto send_error;
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
}
/* close connection */
GST_DEBUG ("closing connection...");
......@@ -788,6 +881,9 @@ gst_rtspsrc_play (GstRTSPSrc * src)
RTSPMessage response = { 0 };
RTSPResult res;
if (!(src->options & RTSP_PLAY))
return TRUE;
GST_DEBUG ("PLAY...");
/* do play */
......@@ -795,7 +891,7 @@ gst_rtspsrc_play (GstRTSPSrc * src)
rtsp_message_init_request (RTSP_PLAY, src->location, &request)) < 0)
goto create_request_failed;
if (!gst_rtspsrc_send (src, &request, &response))
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
if (src->interleaved) {
......@@ -827,13 +923,16 @@ gst_rtspsrc_pause (GstRTSPSrc * src)
RTSPMessage response = { 0 };
RTSPResult res;
if (!(src->options & RTSP_PAUSE))
return TRUE;
GST_DEBUG ("PAUSE...");
/* do pause */
if ((res =
rtsp_message_init_request (RTSP_PAUSE, src->location, &request)) < 0)
goto create_request_failed;
if (!gst_rtspsrc_send (src, &request, &response))
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
return TRUE;
......@@ -868,6 +967,7 @@ gst_rtspsrc_change_state (GstElement * element)
break;
case GST_STATE_READY_TO_PAUSED:
rtspsrc->interleaved = FALSE;
rtspsrc->options = 0;
if (!gst_rtspsrc_open (rtspsrc))
goto open_failed;
break;
......@@ -891,6 +991,7 @@ gst_rtspsrc_change_state (GstElement * element)
gst_rtspsrc_pause (rtspsrc);
break;
case GST_STATE_PAUSED_TO_READY:
gst_rtspsrc_close (rtspsrc);
break;
case GST_STATE_READY_TO_NULL:
break;
......
......@@ -88,6 +88,8 @@ struct _GstRTSPSrc {
gboolean debug;
GstRTSPProto protocols;
/* supported options */
gint options;
RTSPConnection *connection;
RTSPMessage *request;
......
......@@ -136,7 +136,17 @@ rtsp_init_status (void)
const gchar *
rtsp_method_as_text (RTSPMethod method)
{
return rtsp_methods[method];
gint i;
if (method == 0)
return NULL;
i = 0;
while ((method & 1) == 0) {
i++;
method >>= 1;
}
return rtsp_methods[i];
}
const gchar *
......@@ -176,8 +186,8 @@ rtsp_find_method (gchar * method)
gint idx;
for (idx = 0; rtsp_methods[idx]; idx++) {
if (g_ascii_strcasecmp (rtsp_headers[idx], method) == 0) {
return idx;
if (g_ascii_strcasecmp (rtsp_methods[idx], method) == 0) {
return (1 << idx);
}
}
return -1;
......
......@@ -54,17 +54,17 @@ typedef enum {
} RTSPState;
typedef enum {
RTSP_DESCRIBE,
RTSP_ANNOUNCE,
RTSP_GET_PARAMETER,
RTSP_OPTIONS,
RTSP_PAUSE,
RTSP_PLAY,
RTSP_RECORD,
RTSP_REDIRECT,
RTSP_SETUP,
RTSP_SET_PARAMETER,
RTSP_TEARDOWN,
RTSP_DESCRIBE = (1 << 0),
RTSP_ANNOUNCE = (1 << 1),
RTSP_GET_PARAMETER = (1 << 2),
RTSP_OPTIONS = (1 << 3),
RTSP_PAUSE = (1 << 4),
RTSP_PLAY = (1 << 5),
RTSP_RECORD = (1 << 6),
RTSP_REDIRECT = (1 << 7),
RTSP_SETUP = (1 << 8),
RTSP_SET_PARAMETER = (1 << 9),
RTSP_TEARDOWN = (1 << 10),
} RTSPMethod;
typedef enum {
......
......@@ -95,6 +95,8 @@ rtsp_transport_parse (gchar * str, RTSPTransport * transport)
transport->lower_transport = RTSP_LOWER_TRANS_UDP;
} else if (g_str_has_prefix (split[i], "RTP/AVP/TCP")) {
transport->lower_transport = RTSP_LOWER_TRANS_TCP;
} else if (g_str_has_prefix (split[i], "RTP/AVP")) {
transport->lower_transport = RTSP_LOWER_TRANS_UDP;
} else if (g_str_has_prefix (split[i], "multicast")) {
transport->multicast = TRUE;
} else if (g_str_has_prefix (split[i], "unicast")) {
......
Markdown is supported
0% or .
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment