+2008-08-20 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * 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.
+
2008-08-20 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* ext/pulse/pulsesrc.c: (gst_pulsesrc_class_init),
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) \
noinst_HEADERS = gstrtspsrc.h \
gstrtsp.h \
gstrtpdec.h \
- gstrtspext.h
+ gstrtspext.h \
+ gstrtspgoogle.h
#include "gstrtpdec.h"
#include "gstrtspsrc.h"
+#include "gstrtspgoogle.h"
static gboolean
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;
}
--- /dev/null
+/* 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;
+}
--- /dev/null
+/* 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__ */
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;
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;
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);
}
gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
GstRTSPLowerTrans protocols, gchar ** transports)
{
- gchar *result;
GstRTSPResult res;
+ GString *result;
+ gboolean add_udp_str;
*transports = NULL;
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;
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,
cleanup_error:
{
if (src->connection) {
+ GST_DEBUG_OBJECT (src, "free connection");
gst_rtsp_connection_free (src->connection);
src->connection = NULL;
}
src->task = NULL;
}
+ if (!src->connection)
+ goto done;
+
GST_DEBUG_OBJECT (src, "stop connection flush");
gst_rtsp_connection_flush (src->connection, FALSE);
gst_rtsp_connection_free (src->connection);
src->connection = NULL;
+done:
/* cleanup */
gst_rtspsrc_cleanup (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);
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
+no_connection:
src->state = GST_RTSP_STATE_READY;
done:
projects / platform / upstream / gstreamer.git / commitdiff