gst/rtsp/: Add google RTSP extension, it can only handle udp and responds with unsupported if we do anything else. Fi...

Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtsp.c: (plugin_init):
* gst/rtsp/gstrtspgoogle.c: (gst_rtsp_google_before_send),
(gst_rtsp_google_after_send), (gst_rtsp_google_get_transports),
(_do_init), (gst_rtsp_google_base_init),
(gst_rtsp_google_class_init), (gst_rtsp_google_init),
(gst_rtsp_google_finalize), (gst_rtsp_google_change_state),
(gst_rtsp_google_extension_init):
* gst/rtsp/gstrtspgoogle.h:
Add google RTSP extension, it can only handle udp and responds with
unsupported if we do anything else. Fixes #546465.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_connection_send),
(gst_rtspsrc_connection_receive), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_pause):
Make transport setup code a bit better using GString.
Add some more debug.
Check for closed connections before doing anything on them.

5 files changed, 316 insertions, 27 deletions

diff --git a/gst/rtsp/Makefile.am b/gst/rtsp/Makefile.am
index 0e657159..b2e6b84b 100644
--- a/gst/rtsp/Makefile.am
+++ b/gst/rtsp/Makefile.am
@@ -1,7 +1,7 @@
 plugin_LTLIBRARIES = libgstrtsp.la
 
 libgstrtsp_la_SOURCES = gstrtsp.c gstrtspsrc.c \
-			gstrtpdec.c gstrtspext.c
+			gstrtpdec.c gstrtspext.c gstrtspgoogle.c
 
 libgstrtsp_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS)
 libgstrtsp_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) \
@@ -14,4 +14,5 @@ libgstrtsp_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
 noinst_HEADERS = gstrtspsrc.h     \
 		 gstrtsp.h        \
 		 gstrtpdec.h      \
-		 gstrtspext.h
+		 gstrtspext.h     \
+		 gstrtspgoogle.h
diff --git a/gst/rtsp/gstrtsp.c b/gst/rtsp/gstrtsp.c
index 045c03a6..476ebaa5 100644
--- a/gst/rtsp/gstrtsp.c
+++ b/gst/rtsp/gstrtsp.c
@@ -47,6 +47,7 @@
 
 #include "gstrtpdec.h"
 #include "gstrtspsrc.h"
+#include "gstrtspgoogle.h"
 
 static gboolean
 plugin_init (GstPlugin * plugin)
@@ -57,6 +58,10 @@ plugin_init (GstPlugin * plugin)
   if (!gst_element_register (plugin, "rtpdec", GST_RANK_NONE, GST_TYPE_RTP_DEC))
     return FALSE;
 
+  if (!gst_element_register (plugin, "rtspgoogle", GST_RANK_SECONDARY,
+          GST_TYPE_RTSP_GOOGLE)) {
+    return FALSE;
+  }
   return TRUE;
 }
 
diff --git a/gst/rtsp/gstrtspgoogle.c b/gst/rtsp/gstrtspgoogle.c
new file mode 100644
index 00000000..854d6963
--- /dev/null
+++ b/gst/rtsp/gstrtspgoogle.c
@@ -0,0 +1,211 @@
+/* GStreamer
+ * Copyright (C) <2008> Wim Taymans <wim.taymans@google.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+/* Element-Checklist-Version: 5 */
+
+/**
+ * SECTION:element-rtspgoogle
+ *
+ * A Google RTSP extension
+ *
+ * Last reviewed on 2008-08-20 (0.10.10)
+ */
+
+#include <string.h>
+
+#include <gst/rtsp/gstrtspextension.h>
+
+#include "gstrtspgoogle.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtspgoogle_debug);
+#define GST_CAT_DEFAULT (rtspgoogle_debug)
+
+/* elementfactory information */
+static const GstElementDetails rtspgoogle_details =
+GST_ELEMENT_DETAILS ("Google RTSP Extension",
+    "Network/Extension/Protocol",
+    "Extends RTSP so that it can handle Google setup",
+    "Wim Taymans <wim.taymans@gmail.com>");
+
+#define SERVER_PREFIX "Google RTSP"
+
+static GstRTSPResult
+gst_rtsp_google_before_send (GstRTSPExtension * ext, GstRTSPMessage * request)
+{
+  GstRTSPGoogle *ctx = (GstRTSPGoogle *) ext;
+
+  GST_DEBUG_OBJECT (ext, "before send");
+
+  switch (request->type_data.request.method) {
+    case GST_RTSP_OPTIONS:
+    {
+      /* activate ourselves with the first request */
+      ctx->active = TRUE;
+      break;
+    }
+    default:
+      break;
+  }
+  return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+gst_rtsp_google_after_send (GstRTSPExtension * ext, GstRTSPMessage * req,
+    GstRTSPMessage * resp)
+{
+  GstRTSPGoogle *ctx = (GstRTSPGoogle *) ext;
+
+  GST_DEBUG_OBJECT (ext, "after send");
+
+  switch (req->type_data.request.method) {
+    case GST_RTSP_OPTIONS:
+    {
+      gchar *server = NULL;
+
+      gst_rtsp_message_get_header (resp, GST_RTSP_HDR_SERVER, &server, 0);
+      if (server && g_str_has_prefix (server, SERVER_PREFIX))
+        ctx->active = TRUE;
+      else
+        ctx->active = FALSE;
+      break;
+    }
+    default:
+      break;
+  }
+  return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+gst_rtsp_google_get_transports (GstRTSPExtension * ext,
+    GstRTSPLowerTrans protocols, gchar ** transport)
+{
+  GstRTSPResult res;
+  GstRTSPGoogle *ctx = (GstRTSPGoogle *) ext;
+  GString *result;
+
+  if (!ctx->active)
+    return GST_RTSP_OK;
+
+  /* always only suggest UDP */
+  if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
+    result = g_string_new ("");
+
+    GST_DEBUG_OBJECT (ext, "adding UDP unicast");
+
+    g_string_append (result, "RTP/AVP");
+    g_string_append (result, ";unicast;client_port=%%u1-%%u2");
+
+    *transport = g_string_free (result, FALSE);
+
+    res = GST_RTSP_OK;
+  } else {
+    res = GST_RTSP_ERROR;
+  }
+
+  return res;
+}
+
+static void gst_rtsp_google_finalize (GObject * object);
+
+static GstStateChangeReturn gst_rtsp_google_change_state (GstElement * element,
+    GstStateChange transition);
+
+static void gst_rtsp_google_extension_init (gpointer g_iface,
+    gpointer iface_data);
+
+static void
+_do_init (GType rtspgoogle_type)
+{
+  static const GInterfaceInfo rtspextension_info = {
+    gst_rtsp_google_extension_init,
+    NULL,
+    NULL
+  };
+
+  g_type_add_interface_static (rtspgoogle_type, GST_TYPE_RTSP_EXTENSION,
+      &rtspextension_info);
+}
+
+GST_BOILERPLATE_FULL (GstRTSPGoogle, gst_rtsp_google, GstElement,
+    GST_TYPE_ELEMENT, _do_init);
+
+static void
+gst_rtsp_google_base_init (gpointer klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  gst_element_class_set_details (element_class, &rtspgoogle_details);
+}
+
+static void
+gst_rtsp_google_class_init (GstRTSPGoogleClass * g_class)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstRTSPGoogleClass *klass;
+
+  klass = (GstRTSPGoogleClass *) g_class;
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+
+  gobject_class->finalize = gst_rtsp_google_finalize;
+
+  gstelement_class->change_state =
+      GST_DEBUG_FUNCPTR (gst_rtsp_google_change_state);
+
+  GST_DEBUG_CATEGORY_INIT (rtspgoogle_debug, "rtspgoogle", 0,
+      "Google RTSP extension");
+}
+
+static void
+gst_rtsp_google_init (GstRTSPGoogle * rtspgoogle, GstRTSPGoogleClass * klass)
+{
+}
+
+static void
+gst_rtsp_google_finalize (GObject * object)
+{
+  GstRTSPGoogle *rtspgoogle;
+
+  rtspgoogle = GST_RTSP_GOOGLE (object);
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static GstStateChangeReturn
+gst_rtsp_google_change_state (GstElement * element, GstStateChange transition)
+{
+  GstStateChangeReturn ret;
+  GstRTSPGoogle *rtspgoogle;
+
+  rtspgoogle = GST_RTSP_GOOGLE (element);
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+  return ret;
+}
+
+static void
+gst_rtsp_google_extension_init (gpointer g_iface, gpointer iface_data)
+{
+  GstRTSPExtensionInterface *iface = (GstRTSPExtensionInterface *) g_iface;
+
+  iface->before_send = gst_rtsp_google_before_send;
+  iface->after_send = gst_rtsp_google_after_send;
+  iface->get_transports = gst_rtsp_google_get_transports;
+}
diff --git a/gst/rtsp/gstrtspgoogle.h b/gst/rtsp/gstrtspgoogle.h
new file mode 100644
index 00000000..4d5aa50c
--- /dev/null
+++ b/gst/rtsp/gstrtspgoogle.h
@@ -0,0 +1,50 @@
+/* GStreamer
+ * Copyright (C) <2008> Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __GST_RTSP_GOOGLE_H__
+#define __GST_RTSP_GOOGLE_H__
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_GOOGLE  		(gst_rtsp_google_get_type())
+#define GST_IS_RTSP_GOOGLE(obj)  	(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTSP_GOOGLE))
+#define GST_IS_RTSP_GOOGLE_CLASS(klass)	(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTSP_GOOGLE))
+#define GST_RTSP_GOOGLE(obj)  		(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTSP_GOOGLE, GstRTSPGoogle))
+#define GST_RTSP_GOOGLE_CLASS(klass)  	(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTSP_GOOGLE, GstRTSPGoogleClass))
+
+typedef struct _GstRTSPGoogle GstRTSPGoogle;
+typedef struct _GstRTSPGoogleClass GstRTSPGoogleClass;
+
+struct _GstRTSPGoogle {
+  GstElement  element;
+
+  gboolean active;
+};
+
+struct _GstRTSPGoogleClass {
+  GstElementClass parent_class;
+};
+
+GType gst_rtsp_google_get_type(void);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_GOOGLE_H__ */
diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index 048786b5..f393697a 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -1205,7 +1205,10 @@ gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPMessage * message,
   GstRTSPResult ret;
 
   GST_RTSP_CONN_LOCK (src);
-  ret = gst_rtsp_connection_send (src->connection, message, timeout);
+  if (src->connection)
+    ret = gst_rtsp_connection_send (src->connection, message, timeout);
+  else
+    ret = GST_RTSP_ERROR;
   GST_RTSP_CONN_UNLOCK (src);
 
   return ret;
@@ -1218,7 +1221,10 @@ gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPMessage * message,
   GstRTSPResult ret;
 
   GST_RTSP_CONN_LOCK (src);
-  ret = gst_rtsp_connection_receive (src->connection, message, timeout);
+  if (src->connection)
+    ret = gst_rtsp_connection_receive (src->connection, message, timeout);
+  else
+    ret = GST_RTSP_ERROR;
   GST_RTSP_CONN_UNLOCK (src);
 
   return ret;
@@ -2919,10 +2925,12 @@ gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gboolean flush)
   src->loop_cmd = cmd;
   if (flush) {
     GST_DEBUG_OBJECT (src, "start connection flush");
-    gst_rtsp_connection_flush (src->connection, TRUE);
+    if (src->connection)
+      gst_rtsp_connection_flush (src->connection, TRUE);
   } else {
     GST_DEBUG_OBJECT (src, "stop connection flush");
-    gst_rtsp_connection_flush (src->connection, FALSE);
+    if (src->connection)
+      gst_rtsp_connection_flush (src->connection, FALSE);
   }
   GST_OBJECT_UNLOCK (src);
 }
@@ -3577,8 +3585,9 @@ static GstRTSPResult
 gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
     GstRTSPLowerTrans protocols, gchar ** transports)
 {
-  gchar *result;
   GstRTSPResult res;
+  GString *result;
+  gboolean add_udp_str;
 
   *transports = NULL;
 
@@ -3594,42 +3603,42 @@ gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
   if (*transports != NULL)
     return GST_RTSP_OK;
 
+  /* it's the default but some servers need it */
+  add_udp_str = TRUE;
+
   /* the default RTSP transports */
-  result = g_strdup ("");
+  result = g_string_new ("");
   if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
-    gchar *new;
-
     GST_DEBUG_OBJECT (src, "adding UDP unicast");
 
-    new =
-        g_strconcat (result, "RTP/AVP/UDP;unicast;client_port=%%u1-%%u2", NULL);
-    g_free (result);
-    result = new;
+    g_string_append (result, "RTP/AVP");
+    if (add_udp_str)
+      g_string_append (result, "/UDP");
+    g_string_append (result, ";unicast;client_port=%%u1-%%u2");
   }
   if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
-    gchar *new;
-
     GST_DEBUG_OBJECT (src, "adding UDP multicast");
 
     /* we don't have to allocate any UDP ports yet, if the selected transport
      * turns out to be multicast we can create them and join the multicast
      * group indicated in the transport reply */
-    new = g_strconcat (result, result[0] ? "," : "",
-        "RTP/AVP/UDP;multicast", NULL);
-    g_free (result);
-    result = new;
+    if (result->len > 0)
+      g_string_append (result, ",");
+    g_string_append (result, "RTP/AVP");
+    if (add_udp_str)
+      g_string_append (result, "/UDP");
+    g_string_append (result, ";multicast");
   }
   if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
-    gchar *new;
-
     GST_DEBUG_OBJECT (src, "adding TCP");
 
-    new = g_strconcat (result, result[0] ? "," : "",
-        "RTP/AVP/TCP;unicast;interleaved=%%i1-%%i2", NULL);
-    g_free (result);
-    result = new;
+    if (result->len > 0)
+      g_string_append (result, ",");
+    g_string_append (result, "RTP/AVP/TCP;unicast;interleaved=%%i1-%%i2");
   }
-  *transports = result;
+  *transports = g_string_free (result, FALSE);
+
+  GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports));
 
   return GST_RTSP_OK;
 
@@ -3791,12 +3800,16 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src)
     if (res < 0)
       goto setup_transport_failed;
 
+    GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports));
+
     /* replace placeholders with real values, this function will optionally
      * allocate UDP ports and other info needed to execute the setup request */
     res = gst_rtspsrc_prepare_transports (stream, &transports);
     if (res < 0)
       goto setup_transport_failed;
 
+    GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
+
     /* create SETUP request */
     res =
         gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
@@ -4209,6 +4222,7 @@ setup_failed:
 cleanup_error:
   {
     if (src->connection) {
+      GST_DEBUG_OBJECT (src, "free connection");
       gst_rtsp_connection_free (src->connection);
       src->connection = NULL;
     }
@@ -4266,6 +4280,9 @@ gst_rtspsrc_close (GstRTSPSrc * src)
     src->task = NULL;
   }
 
+  if (!src->connection)
+    goto done;
+
   GST_DEBUG_OBJECT (src, "stop connection flush");
   gst_rtsp_connection_flush (src->connection, FALSE);
 
@@ -4297,6 +4314,7 @@ gst_rtspsrc_close (GstRTSPSrc * src)
   gst_rtsp_connection_free (src->connection);
   src->connection = NULL;
 
+done:
   /* cleanup */
   gst_rtspsrc_cleanup (src);
 
@@ -4594,6 +4612,9 @@ gst_rtspsrc_pause (GstRTSPSrc * src)
   GST_DEBUG_OBJECT (src, "connection is idle now");
   GST_RTSP_CONN_UNLOCK (src);
 
+  if (!src->connection)
+    goto no_connection;
+
   GST_DEBUG_OBJECT (src, "stop connection flush");
   gst_rtsp_connection_flush (src->connection, FALSE);
 
@@ -4610,6 +4631,7 @@ gst_rtspsrc_pause (GstRTSPSrc * src)
   gst_rtsp_message_unset (&request);
   gst_rtsp_message_unset (&response);
 
+no_connection:
   src->state = GST_RTSP_STATE_READY;
 
 done: