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

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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.
This commit is contained in:
Wim Taymans 2008-08-20 17:30:19 +00:00
parent 4ab89b397e
commit dd54e000ea
6 changed files with 340 additions and 28 deletions
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23
ChangeLog
View file

@ -1,3 +1,26 @@
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),

5
gst/rtsp/Makefile.am
View file

@ -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

5
gst/rtsp/gstrtsp.c
View file

@ -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;
}

211
gst/rtsp/gstrtspgoogle.c Normal file
View file

@ -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;
}

50
gst/rtsp/gstrtspgoogle.h Normal file
View file

@ -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__ */

66
gst/rtsp/gstrtspsrc.c
View file

@ -1205,7 +1205,10 @@ gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPMessage * message,
GstRTSPResult ret;
GST_RTSP_CONN_LOCK (src);
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);
if (src->connection)
ret = gst_rtsp_connection_receive (src->connection, message, timeout);
else
ret = GST_RTSP_ERROR;
GST_RTSP_CONN_UNLOCK (src);
return ret;
@ -2919,9 +2925,11 @@ gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gboolean flush)
src->loop_cmd = cmd;
if (flush) {
GST_DEBUG_OBJECT (src, "start connection flush");
if (src->connection)
gst_rtsp_connection_flush (src->connection, TRUE);
} else {
GST_DEBUG_OBJECT (src, "stop connection flush");
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;
/* the default RTSP transports */
result = g_strdup ("");
if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
gchar *new;
/* it's the default but some servers need it */
add_udp_str = TRUE;
/* the default RTSP transports */
result = g_string_new ("");
if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
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: