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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.
This commit is contained in:
Wim Taymans
parent
9a5c8cd25e
commit
c831aef495
6 changed files with 160 additions and 29 deletions
16
ChangeLog
16
ChangeLog
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@ -1,3 +1,19 @@
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2005-08-18 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send), (gst_rtspsrc_open),
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(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
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(gst_rtspsrc_change_state):
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* gst/rtsp/gstrtspsrc.h:
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* gst/rtsp/rtspdefs.c: (rtsp_method_as_text), (rtsp_find_method):
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* gst/rtsp/rtspdefs.h:
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* gst/rtsp/rtsptransport.c: (rtsp_transport_parse):
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Handle RTSP defaults better.
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Issue OPTIONS request to figure out what we are allowed to do.
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Make the methods a bitfield so we can easily collect supported
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options.
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Fix rtsp_find_method.
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Do proper RTSP connection shutdown.
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2005-08-18 Wim Taymans <wim@fluendo.com>
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* gst/rtp/Makefile.am:
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@ -495,7 +495,7 @@ need_pause:
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static gboolean
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gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
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RTSPMessage * response)
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RTSPMessage * response, RTSPStatusCode * code)
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{
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RTSPResult res;
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@ -507,6 +507,11 @@ gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
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if ((res = rtsp_connection_receive (src->connection, response)) < 0)
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goto receive_error;
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if (code) {
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*code = response->type_data.response.code;
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}
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if (response->type_data.response.code != RTSP_STS_OK)
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goto error_response;
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@ -559,6 +564,48 @@ gst_rtspsrc_open (GstRTSPSrc * src)
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if ((res = rtsp_connection_open (url, &src->connection)) < 0)
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goto could_not_open;
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/* create OPTIONS */
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GST_DEBUG ("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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goto create_request_failed;
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/* send OPTIONS */
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GST_DEBUG ("send options...");
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if (!gst_rtspsrc_send (src, &request, &response, NULL))
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goto send_error;
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{
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gchar *respoptions = NULL;
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gchar **options;
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gint i;
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rtsp_message_get_header (&response, RTSP_HDR_ALLOW, &respoptions);
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if (!respoptions)
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goto no_options;
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/* parse options */
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options = g_strsplit (respoptions, ",", 0);
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i = 0;
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while (options[i]) {
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gint method = rtsp_find_method (options[i]);
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/* keep bitfield of supported methods */
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if (method != -1)
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src->options |= method;
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i++;
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}
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g_strfreev (options);
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/* we need describe and setup */
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if (!(src->options & RTSP_DESCRIBE))
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goto no_describe;
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if (!(src->options & RTSP_SETUP))
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goto no_setup;
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}
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/* create DESCRIBE */
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GST_DEBUG ("create describe...");
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if ((res =
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@ -570,9 +617,22 @@ gst_rtspsrc_open (GstRTSPSrc * src)
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/* send DESCRIBE */
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GST_DEBUG ("send describe...");
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if (!gst_rtspsrc_send (src, &request, &response))
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if (!gst_rtspsrc_send (src, &request, &response, NULL))
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goto send_error;
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/* check if reply is SDP */
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{
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gchar *respcont = NULL;
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rtsp_message_get_header (&response, RTSP_HDR_CONTENT_TYPE, &respcont);
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/* could not be set but since the request returned OK, we assume it
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* was SDP, else check it. */
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if (respcont) {
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if (!g_ascii_strcasecmp (respcont, "application/sdp") == 0)
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goto wrong_content_type;
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}
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}
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/* parse SDP */
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rtsp_message_get_body (&response, &data, &size);
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@ -622,7 +682,6 @@ gst_rtspsrc_open (GstRTSPSrc * src)
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}
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g_free (setup_url);
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transports = g_strdup ("");
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if (protocols & GST_RTSP_PROTO_UDP_UNICAST) {
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gchar *new;
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@ -662,15 +721,17 @@ 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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if (!gst_rtspsrc_send (src, &request, &response))
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if (!gst_rtspsrc_send (src, &request, &response, NULL))
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goto send_error;
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/* parse response transport */
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{
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gchar *resptrans;
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gchar *resptrans = NULL;
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RTSPTransport transport = { 0 };
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rtsp_message_get_header (&response, RTSP_HDR_TRANSPORT, &resptrans);
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if (!resptrans)
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goto no_transport;
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/* parse transport */
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rtsp_transport_parse (resptrans, &transport);
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@ -723,14 +784,44 @@ send_error:
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("Could not send message."), (NULL));
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return FALSE;
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}
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no_options:
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{
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GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
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("Invalid OPTIONS response."), (NULL));
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return FALSE;
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}
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no_describe:
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{
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GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
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("Server does not support DESCRIBE."), (NULL));
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return FALSE;
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}
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no_setup:
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{
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GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
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("Server does not support SETUP."), (NULL));
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return FALSE;
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}
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wrong_content_type:
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{
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GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
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("Server does not support SDP."), (NULL));
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return FALSE;
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}
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setup_rtp_failed:
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{
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GST_ELEMENT_ERROR (src, RESOURCE, WRITE, ("Could not setup rtp."), (NULL));
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return FALSE;
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}
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no_transport:
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{
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GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
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("Server did not select transport."), (NULL));
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return FALSE;
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}
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}
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G_GNUC_UNUSED static gboolean
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static gboolean
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gst_rtspsrc_close (GstRTSPSrc * src)
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{
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RTSPMessage request = { 0 };
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@ -746,14 +837,16 @@ gst_rtspsrc_close (GstRTSPSrc * src)
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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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&request)) < 0)
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goto create_request_failed;
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if (src->options & RTSP_PLAY) {
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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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&request)) < 0)
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goto create_request_failed;
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if (!gst_rtspsrc_send (src, &request, &response))
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goto send_error;
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if (!gst_rtspsrc_send (src, &request, &response, NULL))
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goto send_error;
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}
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/* close connection */
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GST_DEBUG ("closing connection...");
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@ -788,6 +881,9 @@ gst_rtspsrc_play (GstRTSPSrc * src)
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RTSPMessage response = { 0 };
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RTSPResult res;
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if (!(src->options & RTSP_PLAY))
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return TRUE;
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GST_DEBUG ("PLAY...");
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/* do play */
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@ -795,7 +891,7 @@ gst_rtspsrc_play (GstRTSPSrc * src)
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rtsp_message_init_request (RTSP_PLAY, src->location, &request)) < 0)
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goto create_request_failed;
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if (!gst_rtspsrc_send (src, &request, &response))
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if (!gst_rtspsrc_send (src, &request, &response, NULL))
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goto send_error;
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if (src->interleaved) {
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@ -827,13 +923,16 @@ gst_rtspsrc_pause (GstRTSPSrc * src)
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RTSPMessage response = { 0 };
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RTSPResult res;
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if (!(src->options & RTSP_PAUSE))
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return TRUE;
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GST_DEBUG ("PAUSE...");
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/* do pause */
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if ((res =
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rtsp_message_init_request (RTSP_PAUSE, src->location, &request)) < 0)
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goto create_request_failed;
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if (!gst_rtspsrc_send (src, &request, &response))
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if (!gst_rtspsrc_send (src, &request, &response, NULL))
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goto send_error;
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return TRUE;
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@ -868,6 +967,7 @@ gst_rtspsrc_change_state (GstElement * element)
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break;
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case GST_STATE_READY_TO_PAUSED:
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rtspsrc->interleaved = FALSE;
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rtspsrc->options = 0;
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if (!gst_rtspsrc_open (rtspsrc))
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goto open_failed;
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break;
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@ -891,6 +991,7 @@ gst_rtspsrc_change_state (GstElement * element)
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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_close (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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@ -88,6 +88,8 @@ struct _GstRTSPSrc {
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gboolean debug;
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GstRTSPProto protocols;
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/* supported options */
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gint options;
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RTSPConnection *connection;
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RTSPMessage *request;
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@ -136,7 +136,17 @@ rtsp_init_status (void)
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const gchar *
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rtsp_method_as_text (RTSPMethod method)
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{
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return rtsp_methods[method];
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gint i;
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if (method == 0)
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return NULL;
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i = 0;
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while ((method & 1) == 0) {
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i++;
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method >>= 1;
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}
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return rtsp_methods[i];
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}
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const gchar *
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@ -176,8 +186,8 @@ rtsp_find_method (gchar * method)
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gint idx;
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for (idx = 0; rtsp_methods[idx]; idx++) {
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if (g_ascii_strcasecmp (rtsp_headers[idx], method) == 0) {
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return idx;
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if (g_ascii_strcasecmp (rtsp_methods[idx], method) == 0) {
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return (1 << idx);
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}
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}
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return -1;
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@ -54,17 +54,17 @@ typedef enum {
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} RTSPState;
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typedef enum {
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RTSP_DESCRIBE,
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RTSP_ANNOUNCE,
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RTSP_GET_PARAMETER,
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RTSP_OPTIONS,
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RTSP_PAUSE,
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RTSP_PLAY,
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RTSP_RECORD,
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RTSP_REDIRECT,
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RTSP_SETUP,
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RTSP_SET_PARAMETER,
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RTSP_TEARDOWN,
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RTSP_DESCRIBE = (1 << 0),
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RTSP_ANNOUNCE = (1 << 1),
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RTSP_GET_PARAMETER = (1 << 2),
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RTSP_OPTIONS = (1 << 3),
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RTSP_PAUSE = (1 << 4),
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RTSP_PLAY = (1 << 5),
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RTSP_RECORD = (1 << 6),
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RTSP_REDIRECT = (1 << 7),
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RTSP_SETUP = (1 << 8),
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RTSP_SET_PARAMETER = (1 << 9),
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RTSP_TEARDOWN = (1 << 10),
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} RTSPMethod;
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typedef enum {
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@ -95,6 +95,8 @@ rtsp_transport_parse (gchar * str, RTSPTransport * transport)
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transport->lower_transport = RTSP_LOWER_TRANS_UDP;
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} else if (g_str_has_prefix (split[i], "RTP/AVP/TCP")) {
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transport->lower_transport = RTSP_LOWER_TRANS_TCP;
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} else if (g_str_has_prefix (split[i], "RTP/AVP")) {
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transport->lower_transport = RTSP_LOWER_TRANS_UDP;
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} else if (g_str_has_prefix (split[i], "multicast")) {
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transport->multicast = TRUE;
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} else if (g_str_has_prefix (split[i], "unicast")) {
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