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gst/rtsp/gstrtspsrc.*: Fix race when multiple udp sources post timeouts, just act on the first received timeout.
Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init), (gst_rtspsrc_finalize), (new_session_pad), (request_pt_map), (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send), (gst_rtspsrc_send), (gst_rtspsrc_async_open), (gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Fix race when multiple udp sources post timeouts, just act on the first received timeout. Protect stream list with a recursive lock to fix some races. Flush connection when we need to do a reconnect or stop. Make state lock recursive. * gst/rtsp/rtspconnection.c: (rtsp_connection_connect), (rtsp_connection_close): Some small cleanups.
This commit is contained in:
parent
13ae0cde51
commit
24e51b3c73
4 changed files with 182 additions and 102 deletions
19
ChangeLog
19
ChangeLog
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@ -1,3 +1,22 @@
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2007-05-02 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
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(gst_rtspsrc_finalize), (new_session_pad), (request_pt_map),
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(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
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(gst_rtspsrc_send), (gst_rtspsrc_async_open), (gst_rtspsrc_close),
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(gst_rtspsrc_play), (gst_rtspsrc_handle_message),
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(gst_rtspsrc_change_state):
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* gst/rtsp/gstrtspsrc.h:
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Fix race when multiple udp sources post timeouts, just act on the first
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received timeout.
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Protect stream list with a recursive lock to fix some races.
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Flush connection when we need to do a reconnect or stop.
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Make state lock recursive.
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* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
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(rtsp_connection_close):
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Some small cleanups.
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2007-05-02 Wim Taymans <wim@fluendo.com>
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* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
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@ -179,12 +179,6 @@ static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
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GstStateChange transition);
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static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
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static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
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RTSPMessage * response);
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static gboolean gst_rtspsrc_try_send (GstRTSPSrc * src, RTSPMessage * request,
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RTSPMessage * response, RTSPStatusCode * code);
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static gboolean gst_rtspsrc_open (GstRTSPSrc * src);
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static gboolean gst_rtspsrc_play (GstRTSPSrc * src);
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static gboolean gst_rtspsrc_pause (GstRTSPSrc * src);
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@ -301,7 +295,8 @@ gst_rtspsrc_init (GstRTSPSrc * src, GstRTSPSrcClass * g_class)
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#endif
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src->extension->src = (gpointer) src;
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src->state_lock = g_mutex_new ();
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src->state_rec_lock = g_new (GStaticRecMutex, 1);
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g_static_rec_mutex_init (src->state_rec_lock);
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src->state = RTSP_STATE_INVALID;
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}
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@ -319,7 +314,8 @@ gst_rtspsrc_finalize (GObject * object)
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g_free (rtspsrc->content_base);
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rtsp_url_free (rtspsrc->url);
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g_free (rtspsrc->addr);
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g_mutex_free (rtspsrc->state_lock);
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g_static_rec_mutex_free (rtspsrc->state_rec_lock);
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g_free (rtspsrc->state_rec_lock);
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if (rtspsrc->extension) {
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#ifdef WITH_EXT_REAL
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@ -1050,9 +1046,11 @@ new_session_pad (GstElement * session, GstPad * pad, GstRTSPSrc * src)
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gint id, ssrc, pt;
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GList *lstream;
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GstRTSPStream *stream;
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gboolean all_added;
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GST_DEBUG_OBJECT (src, "got new session pad %" GST_PTR_FORMAT, pad);
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GST_RTSP_STATE_LOCK (src);
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/* find stream */
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name = gst_object_get_name (GST_OBJECT_CAST (pad));
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if (sscanf (name, "recv_rtp_src_%d_%d_%d", &id, &ssrc, &pt) != 3)
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@ -1079,23 +1077,30 @@ new_session_pad (GstElement * session, GstPad * pad, GstRTSPSrc * src)
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gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
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/* check if we added all streams */
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all_added = TRUE;
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for (lstream = src->streams; lstream; lstream = g_list_next (lstream)) {
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stream = (GstRTSPStream *) lstream->data;
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if (!stream->added)
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goto done;
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if (!stream->added) {
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all_added = FALSE;
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break;
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}
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}
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GST_RTSP_STATE_UNLOCK (src);
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if (all_added) {
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GST_DEBUG_OBJECT (src, "We added all streams");
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/* when we get here, all stream are added and we can fire the no-more-pads
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* signal. */
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gst_element_no_more_pads (GST_ELEMENT_CAST (src));
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}
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done:
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return;
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/* ERRORS */
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unknown_stream:
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{
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GST_DEBUG_OBJECT (src, "ignoring unknown stream");
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GST_RTSP_STATE_UNLOCK (src);
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g_free (name);
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return;
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}
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@ -1106,21 +1111,26 @@ request_pt_map (GstElement * sess, guint session, guint pt, GstRTSPSrc * src)
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{
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GstRTSPStream *stream;
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GList *lstream;
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GstCaps *caps;
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GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
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GST_RTSP_STATE_LOCK (src);
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lstream = g_list_find_custom (src->streams, GINT_TO_POINTER (session),
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(GCompareFunc) find_stream_by_id);
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if (!lstream)
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goto unknown_stream;
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stream = (GstRTSPStream *) lstream->data;
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caps = stream->caps;
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GST_RTSP_STATE_UNLOCK (src);
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return stream->caps;
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return caps;
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unknown_stream:
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{
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GST_DEBUG_OBJECT (src, "unknown stream %d", session);
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GST_RTSP_STATE_UNLOCK (src);
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return NULL;
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}
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}
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@ -1852,7 +1862,7 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
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break;
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case RTSP_EINTR:
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/* we got interrupted, see what we have to do */
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GST_DEBUG_OBJECT (src, "we got interrupted");
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GST_DEBUG_OBJECT (src, "we got interrupted, unset flushing");
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/* unset flushing so we can do something else */
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rtsp_connection_flush (src->connection, FALSE);
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goto interrupt;
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@ -1992,12 +2002,14 @@ play_failed:
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}
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static void
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gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd)
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gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gboolean flush)
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{
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GST_OBJECT_LOCK (src);
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src->loop_cmd = cmd;
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if (cmd != CMD_WAIT)
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if (flush) {
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GST_DEBUG_OBJECT (src, "start flush");
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rtsp_connection_flush (src->connection, TRUE);
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}
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GST_OBJECT_UNLOCK (src);
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}
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@ -2153,76 +2165,6 @@ no_user_pass:
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}
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}
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/**
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* gst_rtspsrc_send:
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* @src: the rtsp source
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* @request: must point to a valid request
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* @response: must point to an empty #RTSPMessage
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*
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* send @request and retrieve the response in @response. optionally @code can be
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* non-NULL in which case it will contain the status code of the response.
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*
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* If This function returns TRUE, @response will contain a valid response
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* message that should be cleaned with rtsp_message_unset() after usage.
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*
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* If @code is NULL, this function will return FALSE (with an invalid @response
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* message) if the response code was not 200 (OK).
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*
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* If the attempt results in an authentication failure, then this will attempt
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* to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
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* the request.
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*
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* Returns: TRUE if the processing was successful.
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*/
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gboolean
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gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
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RTSPMessage * response, RTSPStatusCode * code)
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{
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RTSPStatusCode int_code = RTSP_STS_OK;
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gboolean res;
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gboolean retry;
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do {
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retry = FALSE;
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res = gst_rtspsrc_try_send (src, request, response, &int_code);
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if (int_code == RTSP_STS_UNAUTHORIZED) {
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if (gst_rtspsrc_setup_auth (src, response)) {
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/* Try the request/response again after configuring the auth info
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* and loop again */
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retry = TRUE;
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}
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}
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} while (retry == TRUE);
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/* If the user requested the code, let them handle errors, otherwise
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* post an error below */
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if (code != NULL)
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*code = int_code;
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else if (int_code != RTSP_STS_OK)
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goto error_response;
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return res;
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error_response:
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{
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switch (response->type_data.response.code) {
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case RTSP_STS_NOT_FOUND:
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GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
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response->type_data.response.reason));
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break;
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default:
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GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
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("Got error response: %d (%s).", response->type_data.response.code,
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response->type_data.response.reason));
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break;
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}
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/* we return FALSE so we should unset the response ourselves */
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rtsp_message_unset (response);
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return FALSE;
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}
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}
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static gboolean
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gst_rtspsrc_try_send (GstRTSPSrc * src, RTSPMessage * request,
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RTSPMessage * response, RTSPStatusCode * code)
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@ -2234,6 +2176,8 @@ gst_rtspsrc_try_send (GstRTSPSrc * src, RTSPMessage * request,
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if (src->extension && src->extension->before_send)
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src->extension->before_send (src->extension, request);
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GST_DEBUG_OBJECT (src, "sending message");
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if (src->debug)
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rtsp_message_dump (request);
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@ -2309,6 +2253,76 @@ handle_request_failed:
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}
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}
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/**
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* gst_rtspsrc_send:
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* @src: the rtsp source
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* @request: must point to a valid request
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* @response: must point to an empty #RTSPMessage
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*
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* send @request and retrieve the response in @response. optionally @code can be
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* non-NULL in which case it will contain the status code of the response.
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*
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* If This function returns TRUE, @response will contain a valid response
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* message that should be cleaned with rtsp_message_unset() after usage.
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*
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* If @code is NULL, this function will return FALSE (with an invalid @response
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* message) if the response code was not 200 (OK).
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*
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* If the attempt results in an authentication failure, then this will attempt
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* to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
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* the request.
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*
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* Returns: TRUE if the processing was successful.
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*/
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gboolean
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gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
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RTSPMessage * response, RTSPStatusCode * code)
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{
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RTSPStatusCode int_code = RTSP_STS_OK;
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gboolean res;
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gboolean retry;
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do {
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retry = FALSE;
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res = gst_rtspsrc_try_send (src, request, response, &int_code);
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if (int_code == RTSP_STS_UNAUTHORIZED) {
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if (gst_rtspsrc_setup_auth (src, response)) {
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/* Try the request/response again after configuring the auth info
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* and loop again */
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retry = TRUE;
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}
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}
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} while (retry == TRUE);
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/* If the user requested the code, let them handle errors, otherwise
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* post an error below */
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if (code != NULL)
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*code = int_code;
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else if (int_code != RTSP_STS_OK)
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goto error_response;
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return res;
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error_response:
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{
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switch (response->type_data.response.code) {
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case RTSP_STS_NOT_FOUND:
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GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
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response->type_data.response.reason));
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break;
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default:
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GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
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("Got error response: %d (%s).", response->type_data.response.code,
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response->type_data.response.reason));
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break;
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}
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/* we return FALSE so we should unset the response ourselves */
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rtsp_message_unset (response);
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return FALSE;
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}
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}
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/* parse the response and collect all the supported methods. We need this
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* information so that we don't try to send an unsupported request to the
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* server.
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@ -2896,6 +2910,23 @@ cleanup_error:
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}
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}
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#if 0
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static gboolean
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gst_rtspsrc_async_open (GstRTSPSrc * src)
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{
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GError *error = NULL;
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gboolean res = TRUE;
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src->thread =
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g_thread_create ((GThreadFunc) gst_rtspsrc_open, src, TRUE, &error);
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if (error != NULL) {
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GST_ELEMENT_ERROR (src, RESOURCE, INIT, (NULL),
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("Could not start async thread (%s).", error->message));
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}
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return res;
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}
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#endif
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static gboolean
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gst_rtspsrc_close (GstRTSPSrc * src)
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{
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@ -2907,15 +2938,15 @@ gst_rtspsrc_close (GstRTSPSrc * src)
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GST_RTSP_STATE_LOCK (src);
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gst_rtspsrc_loop_send_cmd (src, CMD_STOP);
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gst_rtspsrc_loop_send_cmd (src, CMD_STOP, TRUE);
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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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/* make sure it is not running */
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g_static_rec_mutex_lock (src->stream_rec_lock);
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g_static_rec_mutex_unlock (src->stream_rec_lock);
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GST_RTSP_STREAM_LOCK (src);
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GST_RTSP_STREAM_UNLOCK (src);
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/* no wait for the task to finish */
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gst_task_join (src->task);
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@ -2925,6 +2956,9 @@ gst_rtspsrc_close (GstRTSPSrc * src)
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src->task = NULL;
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}
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GST_DEBUG_OBJECT (src, "stop flush");
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rtsp_connection_flush (src->connection, FALSE);
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if (src->methods & RTSP_PLAY) {
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/* do TEARDOWN */
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res =
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@ -3096,7 +3130,6 @@ gst_rtspsrc_play (GstRTSPSrc * src)
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* Play Time) and should be put in the NEWSEGMENT position field. */
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rtsp_message_get_header (&response, RTSP_HDR_RANGE, &range);
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/* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
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* for the RTP packets. If this is not present, we assume all starts from 0...
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* FIXME, this is info for the RTP session manager ideally. */
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@ -3111,11 +3144,11 @@ gst_rtspsrc_play (GstRTSPSrc * src)
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* For UDP we start the task as well to look for server info and UDP timeouts. */
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if (src->task == NULL) {
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src->task = gst_task_create ((GstTaskFunction) gst_rtspsrc_loop, src);
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gst_task_set_lock (src->task, src->stream_rec_lock);
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gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
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}
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src->running = TRUE;
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src->state = RTSP_STATE_PLAYING;
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gst_rtspsrc_loop_send_cmd (src, CMD_WAIT);
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gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, FALSE);
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gst_task_start (src->task);
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done:
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@ -3227,8 +3260,21 @@ gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
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const GstStructure *s = gst_message_get_structure (message);
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if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
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gboolean ignore_timeout;
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GST_DEBUG_OBJECT (bin, "timeout on UDP port");
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gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT);
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GST_OBJECT_LOCK (rtspsrc);
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ignore_timeout = rtspsrc->ignore_timeout;
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rtspsrc->ignore_timeout = TRUE;
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GST_OBJECT_UNLOCK (rtspsrc);
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/* we only act on the first udp timeout message, others are irrelevant
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* and can be ignored. */
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if (ignore_timeout)
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gst_message_unref (message);
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else
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gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, TRUE);
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return;
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}
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GST_BIN_CLASS (parent_class)->handle_message (bin, message);
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||||
|
@ -3300,10 +3346,13 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
|
|||
break;
|
||||
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
||||
rtspsrc->cur_protocols = rtspsrc->protocols;
|
||||
/* first attempt, don't ignore timeouts */
|
||||
rtspsrc->ignore_timeout = FALSE;
|
||||
if (!gst_rtspsrc_open (rtspsrc))
|
||||
goto open_failed;
|
||||
break;
|
||||
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
||||
GST_DEBUG_OBJECT (rtspsrc, "stop flush");
|
||||
rtsp_connection_flush (rtspsrc->connection, FALSE);
|
||||
/* FIXME, the server might send UDP packets before we activate the UDP
|
||||
* ports */
|
||||
|
@ -3311,6 +3360,7 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
|
|||
break;
|
||||
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
||||
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
||||
GST_DEBUG_OBJECT (rtspsrc, "start flush");
|
||||
rtsp_connection_flush (rtspsrc->connection, TRUE);
|
||||
break;
|
||||
default:
|
||||
|
|
|
@ -67,9 +67,13 @@ G_BEGIN_DECLS
|
|||
typedef struct _GstRTSPSrc GstRTSPSrc;
|
||||
typedef struct _GstRTSPSrcClass GstRTSPSrcClass;
|
||||
|
||||
#define GST_RTSP_STATE_GET_LOCK(rtsp) (GST_RTSPSRC_CAST(rtsp)->state_lock)
|
||||
#define GST_RTSP_STATE_LOCK(rtsp) (g_mutex_lock (GST_RTSP_STATE_GET_LOCK(rtsp)))
|
||||
#define GST_RTSP_STATE_UNLOCK(rtsp) (g_mutex_unlock (GST_RTSP_STATE_GET_LOCK(rtsp)))
|
||||
#define GST_RTSP_STATE_GET_LOCK(rtsp) (GST_RTSPSRC_CAST(rtsp)->state_rec_lock)
|
||||
#define GST_RTSP_STATE_LOCK(rtsp) (g_static_rec_mutex_lock (GST_RTSP_STATE_GET_LOCK(rtsp)))
|
||||
#define GST_RTSP_STATE_UNLOCK(rtsp) (g_static_rec_mutex_unlock (GST_RTSP_STATE_GET_LOCK(rtsp)))
|
||||
|
||||
#define GST_RTSP_STREAM_GET_LOCK(rtsp) (GST_RTSPSRC_CAST(rtsp)->stream_rec_lock)
|
||||
#define GST_RTSP_STREAM_LOCK(rtsp) (g_static_rec_mutex_lock (GST_RTSP_STREAM_GET_LOCK(rtsp)))
|
||||
#define GST_RTSP_STREAM_UNLOCK(rtsp) (g_static_rec_mutex_unlock (GST_RTSP_STREAM_GET_LOCK(rtsp)))
|
||||
|
||||
typedef struct _GstRTSPStream GstRTSPStream;
|
||||
|
||||
|
@ -121,9 +125,12 @@ struct _GstRTSPSrc {
|
|||
gboolean running;
|
||||
gint free_channel;
|
||||
|
||||
/* cond to signal loop */
|
||||
/* UDP mode loop */
|
||||
gint loop_cmd;
|
||||
GMutex *state_lock;
|
||||
gboolean ignore_timeout;
|
||||
|
||||
/* mutex for protecting state changes */
|
||||
GStaticRecMutex *state_rec_lock;
|
||||
|
||||
gint numstreams;
|
||||
GList *streams;
|
||||
|
|
|
@ -205,6 +205,9 @@ rtsp_connection_connect (RTSPConnection * conn, GTimeVal * timeout)
|
|||
if (fd == -1)
|
||||
goto sys_error;
|
||||
|
||||
/* set to non-blocking mode so that we can cancel the connect */
|
||||
//fcntl (fd, F_SETFL, O_NONBLOCK);
|
||||
|
||||
ret = connect (fd, (struct sockaddr *) &sin, sizeof (sin));
|
||||
if (ret != 0)
|
||||
goto sys_error;
|
||||
|
@ -216,6 +219,8 @@ rtsp_connection_connect (RTSPConnection * conn, GTimeVal * timeout)
|
|||
|
||||
sys_error:
|
||||
{
|
||||
if (fd != -1)
|
||||
CLOSE_SOCKET (fd);
|
||||
return RTSP_ESYS;
|
||||
}
|
||||
not_resolved:
|
||||
|
@ -828,7 +833,6 @@ rtsp_connection_close (RTSPConnection * conn)
|
|||
gint res;
|
||||
|
||||
g_return_val_if_fail (conn != NULL, RTSP_EINVAL);
|
||||
g_return_val_if_fail (conn->fd >= 0, RTSP_EINVAL);
|
||||
|
||||
if (conn->fd != -1) {
|
||||
res = CLOSE_SOCKET (conn->fd);
|
||||
|
|
Loading…
Reference in a new issue