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-rw-r--r--ChangeLog21
-rw-r--r--gst/rtsp/URLS1
-rw-r--r--gst/rtsp/gstrtspsrc.c118
-rw-r--r--gst/rtsp/gstrtspsrc.h6
-rw-r--r--gst/rtsp/rtspdefs.c9
-rw-r--r--gst/rtsp/rtspdefs.h10
-rw-r--r--gst/rtsp/rtspurl.c5
7 files changed, 156 insertions, 14 deletions
diff --git a/ChangeLog b/ChangeLog
index 1c6902c4..95d2b2df 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,24 @@
+2006-10-11 Wim Taymans <wim@fluendo.com>
+
+ * 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.
+
2006-10-11 Tim-Philipp Müller <tim at centricular dot net>
* gst/apetag/gsttagdemux.c: (gst_tag_demux_add_srcpad):
diff --git a/gst/rtsp/URLS b/gst/rtsp/URLS
index 16d5d5a4..4c08f804 100644
--- a/gst/rtsp/URLS
+++ b/gst/rtsp/URLS
@@ -16,3 +16,4 @@ MP4V-ES/mpeg4-generic(ACC):
REAL:
rtsp://213.254.239.61/farm/*/encoder/tagesschau/live1high.rm
+ 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
diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index f67ac46b..47917535 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -95,6 +95,10 @@
#include "gstrtspsrc.h"
#include "sdp.h"
+#if 0
+#define WITH_EXT_REAL
+#endif
+
#include "rtspextwms.h"
#ifdef WITH_EXT_REAL
#include "rtspextreal.h"
@@ -173,6 +177,11 @@ static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
+static gboolean gst_rtspsrc_open (GstRTSPSrc * src);
+static gboolean gst_rtspsrc_play (GstRTSPSrc * src);
+static gboolean gst_rtspsrc_pause (GstRTSPSrc * src);
+static gboolean gst_rtspsrc_close (GstRTSPSrc * src);
+
static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
const gchar * uri);
@@ -1207,9 +1216,12 @@ need_pause:
static void
gst_rtspsrc_loop_udp (GstRTSPSrc * src)
{
+ gboolean restart = FALSE;
+
GST_OBJECT_LOCK (src);
if (src->loop_cmd == CMD_STOP)
goto stopping;
+
while (src->loop_cmd == CMD_WAIT) {
GST_DEBUG_OBJECT (src, "waiting");
GST_RTSP_LOOP_WAIT (src);
@@ -1220,9 +1232,42 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
if (src->loop_cmd == CMD_RECONNECT) {
/* FIXME, when we get here we have to reconnect using tcp */
src->loop_cmd = CMD_WAIT;
+ restart = TRUE;
}
GST_OBJECT_UNLOCK (src);
+ if (restart) {
+ gst_rtspsrc_pause (src);
+
+ if (src->task) {
+ /* stop task, we cannot join as this would deadlock */
+ gst_task_stop (src->task);
+ /* and free the task so that _close will not stop/join it again. */
+ gst_object_unref (GST_OBJECT (src->task));
+ src->task = NULL;
+ }
+ gst_rtspsrc_close (src);
+
+ /* see if we have TCP left to try */
+ if (src->cur_protocols & RTSP_LOWER_TRANS_TCP) {
+ /* We post a warning message now to inform the user
+ * that nothing happened. It's most likely a firewall thing. */
+ GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
+ ("Could not receive any UDP packets for %.4f seconds, maybe your "
+ "firewall is blocking it. Retrying using a TCP connection.",
+ (gdouble) src->timeout / 1000000));
+ /* we can try only TCP now */
+ src->cur_protocols = RTSP_LOWER_TRANS_TCP;
+ gst_rtspsrc_open (src);
+ gst_rtspsrc_play (src);
+ } else {
+ src->cur_protocols = 0;
+ /* no transport possible, post an error */
+ GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
+ ("Could not receive any UDP packets for %.4f seconds, maybe your "
+ "firewall is blocking it.", (gdouble) src->timeout / 1000000));
+ }
+ }
return;
/* ERRORS */
@@ -1253,6 +1298,36 @@ gst_rtspsrc_loop (GstRTSPSrc * src)
gst_rtspsrc_loop_udp (src);
}
+static RTSPResult
+gst_rtspsrc_handle_request (GstRTSPSrc * src, RTSPMessage * request)
+{
+ RTSPMessage response = { 0 };
+ RTSPResult res;
+
+ res = rtsp_message_init_response (&response, RTSP_STS_OK, "OK", request);
+ if (res < 0)
+ goto send_error;
+
+ if (src->debug)
+ rtsp_message_dump (&response);
+
+ if ((res = rtsp_connection_send (src->connection, &response)) < 0)
+ goto send_error;
+
+ return RTSP_OK;
+
+ /* ERRORS */
+send_error:
+ {
+ gchar *str = rtsp_strresult (res);
+
+ GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
+ ("Could not send message. (%s)", str));
+ g_free (str);
+ return res;
+ }
+}
+
/**
* gst_rtspsrc_send:
* @src: the rtsp source
@@ -1287,12 +1362,28 @@ gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
if ((res = rtsp_connection_send (src->connection, request)) < 0)
goto send_error;
+next:
if ((res = rtsp_connection_receive (src->connection, response)) < 0)
goto receive_error;
if (src->debug)
rtsp_message_dump (response);
+ switch (response->type) {
+ case RTSP_MESSAGE_REQUEST:
+ /* FIXME, handle server request, reply with OK, for now */
+ if ((res = gst_rtspsrc_handle_request (src, response)) < 0)
+ goto handle_request_failed;
+ goto next;
+ case RTSP_MESSAGE_RESPONSE:
+ /* ok, a response is good */
+ break;
+ default:
+ case RTSP_MESSAGE_DATA:
+ /* get next response */
+ goto next;
+ }
+
thecode = response->type_data.response.code;
/* if the caller wanted the result code, we store it. Else we check if it's
* OK. */
@@ -1332,6 +1423,11 @@ receive_error:
g_free (str);
return FALSE;
}
+handle_request_failed:
+ {
+ /* ERROR was posted */
+ return FALSE;
+ }
error_response:
{
switch (response->type_data.response.code) {
@@ -1566,6 +1662,11 @@ gst_rtspsrc_open (GstRTSPSrc * src)
GstRTSPStream *stream = NULL;
gchar *respcont = NULL;
+ /* reset our state */
+ src->free_channel = 0;
+ src->interleaved = FALSE;
+ gst_segment_init (&src->segment, GST_FORMAT_TIME);
+
/* can't continue without a valid url */
if (G_UNLIKELY (src->url == NULL))
goto no_url;
@@ -1639,7 +1740,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
/* we initially allow all configured lower transports. based on the
* replies from the server we narrow them down. */
- protocols = src->protocols;
+ protocols = src->cur_protocols;
/* setup streams */
n_streams = sdp_message_medias_len (&sdp);
@@ -1952,7 +2053,7 @@ gst_rtspsrc_play (GstRTSPSrc * src)
if (res < 0)
goto create_request_failed;
- rtsp_message_add_header (&request, RTSP_HDR_RANGE, "npt=0.000-");
+ rtsp_message_add_header (&request, RTSP_HDR_RANGE, "npt=0-");
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
@@ -2051,13 +2152,6 @@ gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
GST_DEBUG_OBJECT (bin, "timeout on UDP port");
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT);
- /* FIXME, we post an error message now to inform the user
- * that nothing happened. It's most likely a firewall thing. Ideally we
- * notify the thread and redo the setup with only TCP. */
- GST_ELEMENT_ERROR (rtspsrc, RESOURCE, READ, (NULL),
- ("Could not receive any UDP packets for %.4f seconds, maybe your "
- "firewall is blocking it.",
- (gdouble) rtspsrc->timeout / 1000000));
return;
}
}
@@ -2080,15 +2174,13 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
- /* the first free channel for interleaved mode */
- rtspsrc->free_channel = 0;
- rtspsrc->interleaved = FALSE;
- gst_segment_init (&rtspsrc->segment, GST_FORMAT_TIME);
+ rtspsrc->cur_protocols = rtspsrc->protocols;
if (!gst_rtspsrc_open (rtspsrc))
goto open_failed;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
rtsp_connection_flush (rtspsrc->connection, FALSE);
+ /* copy configuerd protocols */
gst_rtspsrc_play (rtspsrc);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
diff --git a/gst/rtsp/gstrtspsrc.h b/gst/rtsp/gstrtspsrc.h
index 94b9ed3d..d7ea7e59 100644
--- a/gst/rtsp/gstrtspsrc.h
+++ b/gst/rtsp/gstrtspsrc.h
@@ -124,14 +124,18 @@ struct _GstRTSPSrc {
GList *streams;
GstStructure *props;
+ /* properties */
gchar *location;
RTSPUrl *url;
- gchar *content_base;
RTSPLowerTrans protocols;
gboolean debug;
guint retry;
guint64 timeout;
+ /* state */
+ gchar *content_base;
+ RTSPLowerTrans cur_protocols;
+
/* supported methods */
gint methods;
diff --git a/gst/rtsp/rtspdefs.c b/gst/rtsp/rtspdefs.c
index 0138d792..63b68a3f 100644
--- a/gst/rtsp/rtspdefs.c
+++ b/gst/rtsp/rtspdefs.c
@@ -128,7 +128,16 @@ static const gchar *rtsp_headers[] = {
"ClientChallenge", /* ClientChallenge */
"RealChallenge1", /* RealChallenge1 */
"RealChallenge2", /* RealChallenge2 */
+ "RealChallenge3", /* RealChallenge3 */
"Subscribe", /* Subscribe */
+ "Alert", /* Alert */
+ "ClientID", /* ClientID */
+ "CompanyID", /* CompanyID */
+ "GUID", /* GUID */
+ "RegionData", /* RegionData */
+ "SupportsMaximumASMBandwidth", /* SupportsMaximumASMBandwidth */
+ "Language", /* Language */
+ "PlayerStarttime", /* PlayerStarttime */
NULL
};
diff --git a/gst/rtsp/rtspdefs.h b/gst/rtsp/rtspdefs.h
index 229fc42a..7afece94 100644
--- a/gst/rtsp/rtspdefs.h
+++ b/gst/rtsp/rtspdefs.h
@@ -155,7 +155,17 @@ typedef enum {
RTSP_HDR_CLIENT_CHALLENGE, /* ClientChallenge */
RTSP_HDR_REAL_CHALLENGE1, /* RealChallenge1 */
RTSP_HDR_REAL_CHALLENGE2, /* RealChallenge2 */
+ RTSP_HDR_REAL_CHALLENGE3, /* RealChallenge3 */
RTSP_HDR_SUBSCRIBE, /* Subscribe */
+ RTSP_HDR_ALERT, /* Alert */
+ RTSP_HDR_CLIENT_ID, /* ClientID */
+ RTSP_HDR_COMPANY_ID, /* CompanyID */
+ RTSP_HDR_GUID, /* GUID */
+ RTSP_HDR_REGION_DATA, /* RegionData */
+ RTSP_HDR_MAX_ASM_WIDTH, /* SupportsMaximumASMBandwidth */
+ RTSP_HDR_LANGUAGE, /* Language */
+ RTSP_HDR_PLAYER_START_TIME, /* PlayerStarttime */
+
} RTSPHeaderField;
diff --git a/gst/rtsp/rtspurl.c b/gst/rtsp/rtspurl.c
index 04b455ac..cc11e91f 100644
--- a/gst/rtsp/rtspurl.c
+++ b/gst/rtsp/rtspurl.c
@@ -95,6 +95,11 @@ rtsp_url_parse (const gchar * urlstr, RTSPUrl ** url)
}
col = strstr (p, ":");
+ /* we have a ':' and a slash but the ':' is after the slash, it's not really
+ * part of the hostname */
+ if (col && slash && col >= slash)
+ col = NULL;
+
if (col) {
res->host = g_strndup (p, col - p);
p = col + 1;
generated by cgit (git 2.34.1) at 2025-06-19 00:44:56 +0000