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gst/rtsp/URLS: Added some other URL.
Original commit message from CVS: * gst/rtsp/URLS: Added some other URL. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp), (gst_rtspsrc_handle_request), (gst_rtspsrc_send), (gst_rtspsrc_open), (gst_rtspsrc_play), (gst_rtspsrc_handle_message), (gst_rtspsrc_change_state): * gst/rtsp/gstrtspsrc.h: Work on fallback to TCP connection when the UDP socket times out. Handler server requests, just reply with OK for now. * gst/rtsp/rtspdefs.c: (rtsp_strresult): * gst/rtsp/rtspdefs.h: Added some more Real extension headers. * gst/rtsp/rtspurl.c: (rtsp_url_parse): Fix parsing of urls with a ':' that is not part of the hostname:port part of the url.
This commit is contained in:
Wim Taymans
parent
7cb2c5f8a0
commit
7accf76da8
7 changed files with 156 additions and 14 deletions
21
ChangeLog
21
ChangeLog
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@ -1,3 +1,24 @@
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2006-10-11 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/URLS:
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Added some other URL.
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp),
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(gst_rtspsrc_handle_request), (gst_rtspsrc_send),
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(gst_rtspsrc_open), (gst_rtspsrc_play),
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(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
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* gst/rtsp/gstrtspsrc.h:
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Work on fallback to TCP connection when the UDP socket times out.
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Handler server requests, just reply with OK for now.
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* gst/rtsp/rtspdefs.c: (rtsp_strresult):
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* gst/rtsp/rtspdefs.h:
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Added some more Real extension headers.
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* gst/rtsp/rtspurl.c: (rtsp_url_parse):
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Fix parsing of urls with a ':' that is not part of the hostname:port
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part of the url.
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2006-10-11 Tim-Philipp Müller <tim at centricular dot net>
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* gst/apetag/gsttagdemux.c: (gst_tag_demux_add_srcpad):
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@ -16,3 +16,4 @@ MP4V-ES/mpeg4-generic(ACC):
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REAL:
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rtsp://213.254.239.61/farm/*/encoder/tagesschau/live1high.rm
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rtsp://64.192.137.105:554/real.amazon-de.eu2/phononet/B/0/0/0/H/W/Y/4/K/S/01.01.rm?cloakport=80,554,7070
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@ -95,6 +95,10 @@
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#include "gstrtspsrc.h"
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#include "sdp.h"
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#if 0
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#define WITH_EXT_REAL
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#endif
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#include "rtspextwms.h"
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#ifdef WITH_EXT_REAL
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#include "rtspextreal.h"
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@ -173,6 +177,11 @@ static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
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static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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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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static gboolean gst_rtspsrc_close (GstRTSPSrc * src);
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static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
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const gchar * uri);
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@ -1207,9 +1216,12 @@ need_pause:
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static void
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gst_rtspsrc_loop_udp (GstRTSPSrc * src)
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{
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gboolean restart = FALSE;
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GST_OBJECT_LOCK (src);
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if (src->loop_cmd == CMD_STOP)
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goto stopping;
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while (src->loop_cmd == CMD_WAIT) {
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GST_DEBUG_OBJECT (src, "waiting");
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GST_RTSP_LOOP_WAIT (src);
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@ -1220,9 +1232,42 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
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if (src->loop_cmd == CMD_RECONNECT) {
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/* FIXME, when we get here we have to reconnect using tcp */
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src->loop_cmd = CMD_WAIT;
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restart = TRUE;
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}
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GST_OBJECT_UNLOCK (src);
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if (restart) {
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gst_rtspsrc_pause (src);
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if (src->task) {
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/* stop task, we cannot join as this would deadlock */
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gst_task_stop (src->task);
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/* and free the task so that _close will not stop/join it again. */
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gst_object_unref (GST_OBJECT (src->task));
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src->task = NULL;
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}
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gst_rtspsrc_close (src);
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/* see if we have TCP left to try */
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if (src->cur_protocols & RTSP_LOWER_TRANS_TCP) {
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/* We post a warning message now to inform the user
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* that nothing happened. It's most likely a firewall thing. */
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GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
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("Could not receive any UDP packets for %.4f seconds, maybe your "
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"firewall is blocking it. Retrying using a TCP connection.",
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(gdouble) src->timeout / 1000000));
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/* we can try only TCP now */
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src->cur_protocols = RTSP_LOWER_TRANS_TCP;
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gst_rtspsrc_open (src);
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gst_rtspsrc_play (src);
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} else {
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src->cur_protocols = 0;
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/* no transport possible, post an error */
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GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
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("Could not receive any UDP packets for %.4f seconds, maybe your "
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"firewall is blocking it.", (gdouble) src->timeout / 1000000));
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}
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}
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return;
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/* ERRORS */
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@ -1253,6 +1298,36 @@ gst_rtspsrc_loop (GstRTSPSrc * src)
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gst_rtspsrc_loop_udp (src);
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}
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static RTSPResult
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gst_rtspsrc_handle_request (GstRTSPSrc * src, RTSPMessage * request)
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{
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RTSPMessage response = { 0 };
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RTSPResult res;
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res = rtsp_message_init_response (&response, RTSP_STS_OK, "OK", request);
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if (res < 0)
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goto send_error;
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if (src->debug)
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rtsp_message_dump (&response);
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if ((res = rtsp_connection_send (src->connection, &response)) < 0)
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goto send_error;
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return RTSP_OK;
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/* ERRORS */
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send_error:
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{
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gchar *str = rtsp_strresult (res);
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GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
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("Could not send message. (%s)", str));
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g_free (str);
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return res;
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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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@ -1287,12 +1362,28 @@ gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
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if ((res = rtsp_connection_send (src->connection, request)) < 0)
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goto send_error;
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next:
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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 (src->debug)
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rtsp_message_dump (response);
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switch (response->type) {
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case RTSP_MESSAGE_REQUEST:
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/* FIXME, handle server request, reply with OK, for now */
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if ((res = gst_rtspsrc_handle_request (src, response)) < 0)
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goto handle_request_failed;
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goto next;
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case RTSP_MESSAGE_RESPONSE:
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/* ok, a response is good */
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break;
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default:
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case RTSP_MESSAGE_DATA:
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/* get next response */
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goto next;
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}
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thecode = response->type_data.response.code;
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/* if the caller wanted the result code, we store it. Else we check if it's
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* OK. */
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@ -1332,6 +1423,11 @@ receive_error:
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g_free (str);
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return FALSE;
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}
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handle_request_failed:
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{
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/* ERROR was posted */
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return FALSE;
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}
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error_response:
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{
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switch (response->type_data.response.code) {
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@ -1566,6 +1662,11 @@ gst_rtspsrc_open (GstRTSPSrc * src)
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GstRTSPStream *stream = NULL;
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gchar *respcont = NULL;
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/* reset our state */
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src->free_channel = 0;
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src->interleaved = FALSE;
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gst_segment_init (&src->segment, GST_FORMAT_TIME);
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/* can't continue without a valid url */
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if (G_UNLIKELY (src->url == NULL))
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goto no_url;
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@ -1639,7 +1740,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
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/* we initially allow all configured lower transports. based on the
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* replies from the server we narrow them down. */
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protocols = src->protocols;
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protocols = src->cur_protocols;
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/* setup streams */
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n_streams = sdp_message_medias_len (&sdp);
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@ -1952,7 +2053,7 @@ gst_rtspsrc_play (GstRTSPSrc * src)
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if (res < 0)
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goto create_request_failed;
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rtsp_message_add_header (&request, RTSP_HDR_RANGE, "npt=0.000-");
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rtsp_message_add_header (&request, RTSP_HDR_RANGE, "npt=0-");
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if (!gst_rtspsrc_send (src, &request, &response, NULL))
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goto send_error;
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@ -2051,13 +2152,6 @@ gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
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if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
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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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/* FIXME, we post an error message now to inform the user
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* that nothing happened. It's most likely a firewall thing. Ideally we
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* notify the thread and redo the setup with only TCP. */
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GST_ELEMENT_ERROR (rtspsrc, RESOURCE, READ, (NULL),
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("Could not receive any UDP packets for %.4f seconds, maybe your "
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"firewall is blocking it.",
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(gdouble) rtspsrc->timeout / 1000000));
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return;
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}
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}
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@ -2080,15 +2174,13 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
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case GST_STATE_CHANGE_NULL_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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/* the first free channel for interleaved mode */
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rtspsrc->free_channel = 0;
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rtspsrc->interleaved = FALSE;
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gst_segment_init (&rtspsrc->segment, GST_FORMAT_TIME);
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rtspsrc->cur_protocols = rtspsrc->protocols;
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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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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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rtsp_connection_flush (rtspsrc->connection, FALSE);
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/* copy configuerd protocols */
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gst_rtspsrc_play (rtspsrc);
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break;
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case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
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@ -124,14 +124,18 @@ struct _GstRTSPSrc {
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GList *streams;
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GstStructure *props;
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/* properties */
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gchar *location;
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RTSPUrl *url;
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gchar *content_base;
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RTSPLowerTrans protocols;
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gboolean debug;
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guint retry;
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guint64 timeout;
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/* state */
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gchar *content_base;
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RTSPLowerTrans cur_protocols;
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/* supported methods */
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gint methods;
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@ -128,7 +128,16 @@ static const gchar *rtsp_headers[] = {
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"ClientChallenge", /* ClientChallenge */
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"RealChallenge1", /* RealChallenge1 */
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"RealChallenge2", /* RealChallenge2 */
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"RealChallenge3", /* RealChallenge3 */
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"Subscribe", /* Subscribe */
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"Alert", /* Alert */
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"ClientID", /* ClientID */
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"CompanyID", /* CompanyID */
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"GUID", /* GUID */
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"RegionData", /* RegionData */
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"SupportsMaximumASMBandwidth", /* SupportsMaximumASMBandwidth */
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"Language", /* Language */
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"PlayerStarttime", /* PlayerStarttime */
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NULL
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};
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@ -155,7 +155,17 @@ typedef enum {
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RTSP_HDR_CLIENT_CHALLENGE, /* ClientChallenge */
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RTSP_HDR_REAL_CHALLENGE1, /* RealChallenge1 */
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RTSP_HDR_REAL_CHALLENGE2, /* RealChallenge2 */
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RTSP_HDR_REAL_CHALLENGE3, /* RealChallenge3 */
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RTSP_HDR_SUBSCRIBE, /* Subscribe */
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RTSP_HDR_ALERT, /* Alert */
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RTSP_HDR_CLIENT_ID, /* ClientID */
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RTSP_HDR_COMPANY_ID, /* CompanyID */
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RTSP_HDR_GUID, /* GUID */
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RTSP_HDR_REGION_DATA, /* RegionData */
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RTSP_HDR_MAX_ASM_WIDTH, /* SupportsMaximumASMBandwidth */
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RTSP_HDR_LANGUAGE, /* Language */
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RTSP_HDR_PLAYER_START_TIME, /* PlayerStarttime */
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} RTSPHeaderField;
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@ -95,6 +95,11 @@ rtsp_url_parse (const gchar * urlstr, RTSPUrl ** url)
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}
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col = strstr (p, ":");
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/* we have a ':' and a slash but the ':' is after the slash, it's not really
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* part of the hostname */
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if (col && slash && col >= slash)
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col = NULL;
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if (col) {
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res->host = g_strndup (p, col - p);
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p = col + 1;
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