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authorWim Taymans <wim.taymans@gmail.com>2006-02-09 14:20:14 +0000
committerWim Taymans <wim.taymans@gmail.com>2006-02-09 14:20:14 +0000
commit946e1e43637e78c263c82d449ec19dfd84ed8269 (patch)
tree79abf610349beae51daa2c53a10fbb45175cb797
parent509035fcdb2bc6237bf906aa5fb030dbf33852ba (diff)
gst/rtsp/: Resurected rtpdec to make rtspsrc happy again.
Original commit message from CVS: * gst/rtsp/Makefile.am: * gst/rtsp/gstrtpdec.c: (gst_rtpdec_get_type), (gst_rtpdec_class_init), (gst_rtpdec_init), (gst_rtpdec_getcaps), (gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp), (gst_rtpdec_set_property), (gst_rtpdec_get_property), (gst_rtpdec_change_state): * gst/rtsp/gstrtpdec.h: * gst/rtsp/gstrtsp.c: (plugin_init): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport): * gst/rtsp/rtspconnection.c: (read_body), (rtsp_connection_receive): * gst/rtsp/rtspmessage.c: (rtsp_message_dump): Resurected rtpdec to make rtspsrc happy again. Skip attributes from the session id. Don't crash when dumping a message with an empty body.
-rw-r--r--ChangeLog19
-rw-r--r--gst/rtsp/Makefile.am3
-rw-r--r--gst/rtsp/gstrtpdec.c277
-rw-r--r--gst/rtsp/gstrtpdec.h55
-rw-r--r--gst/rtsp/gstrtsp.c3
-rw-r--r--gst/rtsp/gstrtspsrc.c8
-rw-r--r--gst/rtsp/rtspconnection.c14
-rw-r--r--gst/rtsp/rtspmessage.c2
8 files changed, 373 insertions, 8 deletions
diff --git a/ChangeLog b/ChangeLog
index 78634417..2d82fdad 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,5 +1,24 @@
2006-02-09 Wim Taymans <wim@fluendo.com>
+ * gst/rtsp/Makefile.am:
+ * gst/rtsp/gstrtpdec.c: (gst_rtpdec_get_type),
+ (gst_rtpdec_class_init), (gst_rtpdec_init), (gst_rtpdec_getcaps),
+ (gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp),
+ (gst_rtpdec_set_property), (gst_rtpdec_get_property),
+ (gst_rtpdec_change_state):
+ * gst/rtsp/gstrtpdec.h:
+ * gst/rtsp/gstrtsp.c: (plugin_init):
+ * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport):
+ * gst/rtsp/rtspconnection.c: (read_body),
+ (rtsp_connection_receive):
+ * gst/rtsp/rtspmessage.c: (rtsp_message_dump):
+ Resurected rtpdec to make rtspsrc happy again.
+ Skip attributes from the session id.
+ Don't crash when dumping a message with an empty body.
+
+
+2006-02-09 Wim Taymans <wim@fluendo.com>
+
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_chain):
Added more meaningfull warnings when something goes wrong.
Clear F bit on outgoing AMR packets.
diff --git a/gst/rtsp/Makefile.am b/gst/rtsp/Makefile.am
index f2606183..85c2dad4 100644
--- a/gst/rtsp/Makefile.am
+++ b/gst/rtsp/Makefile.am
@@ -1,6 +1,7 @@
plugin_LTLIBRARIES = libgstrtsp.la
libgstrtsp_la_SOURCES = gstrtsp.c gstrtspsrc.c \
+ gstrtpdec.c \
rtspconnection.c \
rtspdefs.c \
rtspmessage.c \
@@ -18,4 +19,4 @@ test_SOURCES = test.c rtspdefs.c rtspurl.c rtspconnection.c rtspmessage.c rtsptr
test_CFLAGS = $(GST_CFLAGS)
test_LDFLAGS = $(GST_LIBS)
-noinst_HEADERS = gstrtspsrc.h gstrtsp.h rtsptransport.h rtsp.h rtspurl.h rtspconnection.h rtspdefs.h rtspmessage.h sdp.h sdpmessage.h
+noinst_HEADERS = gstrtspsrc.h gstrtsp.h gstrtpdec.h rtsptransport.h rtsp.h rtspurl.h rtspconnection.h rtspdefs.h rtspmessage.h sdp.h sdpmessage.h
diff --git a/gst/rtsp/gstrtpdec.c b/gst/rtsp/gstrtpdec.c
new file mode 100644
index 00000000..7484ad33
--- /dev/null
+++ b/gst/rtsp/gstrtpdec.c
@@ -0,0 +1,277 @@
+/* GStreamer
+ * Copyright (C) <2005> Wim Taymans <wim@fluendo.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 */
+
+#include "gstrtpdec.h"
+
+GST_DEBUG_CATEGORY (rtpdec_debug);
+#define GST_CAT_DEFAULT (rtpdec_debug)
+
+/* elementfactory information */
+static GstElementDetails rtpdec_details = GST_ELEMENT_DETAILS ("RTP Decoder",
+ "Codec/Parser/Network",
+ "Accepts raw RTP and RTCP packets and sends them forward",
+ "Wim Taymans <wim@fluendo.com>");
+
+/* GstRTPDec signals and args */
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+enum
+{
+ ARG_0,
+ ARG_SKIP
+ /* FILL ME */
+};
+
+static GstStaticPadTemplate gst_rtpdec_src_rtp_template =
+GST_STATIC_PAD_TEMPLATE ("srcrtp",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtp")
+ );
+
+static GstStaticPadTemplate gst_rtpdec_src_rtcp_template =
+GST_STATIC_PAD_TEMPLATE ("srcrtcp",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtcp")
+ );
+
+static GstStaticPadTemplate gst_rtpdec_sink_rtp_template =
+GST_STATIC_PAD_TEMPLATE ("sinkrtp",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtp")
+ );
+
+static GstStaticPadTemplate gst_rtpdec_sink_rtcp_template =
+GST_STATIC_PAD_TEMPLATE ("sinkrtcp",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtcp")
+ );
+
+static void gst_rtpdec_class_init (gpointer g_class);
+static void gst_rtpdec_init (GstRTPDec * rtpdec);
+
+static GstCaps *gst_rtpdec_getcaps (GstPad * pad);
+static GstFlowReturn gst_rtpdec_chain_rtp (GstPad * pad, GstBuffer * buffer);
+static GstFlowReturn gst_rtpdec_chain_rtcp (GstPad * pad, GstBuffer * buffer);
+
+static void gst_rtpdec_set_property (GObject * object,
+ guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_rtpdec_get_property (GObject * object,
+ guint prop_id, GValue * value, GParamSpec * pspec);
+
+static GstStateChangeReturn gst_rtpdec_change_state (GstElement * element,
+ GstStateChange transition);
+
+static GstElementClass *parent_class = NULL;
+
+/*static guint gst_rtpdec_signals[LAST_SIGNAL] = { 0 };*/
+
+GType
+gst_rtpdec_get_type (void)
+{
+ static GType rtpdec_type = 0;
+
+ if (!rtpdec_type) {
+ static const GTypeInfo rtpdec_info = {
+ sizeof (GstRTPDecClass), NULL,
+ NULL,
+ (GClassInitFunc) gst_rtpdec_class_init,
+ NULL,
+ NULL,
+ sizeof (GstRTPDec),
+ 0,
+ (GInstanceInitFunc) gst_rtpdec_init,
+ };
+
+ rtpdec_type =
+ g_type_register_static (GST_TYPE_ELEMENT, "GstRTPDec", &rtpdec_info, 0);
+ }
+ return rtpdec_type;
+}
+
+static void
+gst_rtpdec_class_init (gpointer g_class)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstRTPDecClass *klass;
+
+ klass = (GstRTPDecClass *) g_class;
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_rtpdec_src_rtp_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_rtpdec_src_rtcp_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_rtpdec_sink_rtp_template));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&gst_rtpdec_sink_rtcp_template));
+ gst_element_class_set_details (gstelement_class, &rtpdec_details);
+
+ gobject_class->set_property = gst_rtpdec_set_property;
+ gobject_class->get_property = gst_rtpdec_get_property;
+
+ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SKIP, g_param_spec_int ("skip", "skip", "skip", G_MININT, G_MAXINT, 0, G_PARAM_READWRITE)); /* CHECKME */
+
+ parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+
+ gstelement_class->change_state = gst_rtpdec_change_state;
+
+ GST_DEBUG_CATEGORY_INIT (rtpdec_debug, "rtpdec", 0, "RTP decoder");
+}
+
+static void
+gst_rtpdec_init (GstRTPDec * rtpdec)
+{
+ /* the input rtp pad */
+ rtpdec->sink_rtp =
+ gst_pad_new_from_template (gst_static_pad_template_get
+ (&gst_rtpdec_sink_rtp_template), "sinkrtp");
+ gst_element_add_pad (GST_ELEMENT (rtpdec), rtpdec->sink_rtp);
+ gst_pad_set_getcaps_function (rtpdec->sink_rtp, gst_rtpdec_getcaps);
+ gst_pad_set_chain_function (rtpdec->sink_rtp, gst_rtpdec_chain_rtp);
+
+ /* the input rtcp pad */
+ rtpdec->sink_rtcp =
+ gst_pad_new_from_template (gst_static_pad_template_get
+ (&gst_rtpdec_sink_rtcp_template), "sinkrtcp");
+ gst_element_add_pad (GST_ELEMENT (rtpdec), rtpdec->sink_rtcp);
+ gst_pad_set_chain_function (rtpdec->sink_rtcp, gst_rtpdec_chain_rtcp);
+
+ /* the output rtp pad */
+ rtpdec->src_rtp =
+ gst_pad_new_from_template (gst_static_pad_template_get
+ (&gst_rtpdec_src_rtp_template), "srcrtp");
+ gst_pad_set_getcaps_function (rtpdec->src_rtp, gst_rtpdec_getcaps);
+ gst_element_add_pad (GST_ELEMENT (rtpdec), rtpdec->src_rtp);
+
+ /* the output rtcp pad */
+ rtpdec->src_rtcp =
+ gst_pad_new_from_template (gst_static_pad_template_get
+ (&gst_rtpdec_src_rtcp_template), "srcrtcp");
+ gst_element_add_pad (GST_ELEMENT (rtpdec), rtpdec->src_rtcp);
+}
+
+static GstCaps *
+gst_rtpdec_getcaps (GstPad * pad)
+{
+ GstRTPDec *src;
+ GstPad *other;
+ GstCaps *caps;
+
+ src = GST_RTPDEC (GST_PAD_PARENT (pad));
+
+ other = pad == src->src_rtp ? src->sink_rtp : src->src_rtp;
+
+ caps = gst_pad_peer_get_caps (other);
+
+ if (caps == NULL)
+ caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
+
+ return caps;
+}
+
+static GstFlowReturn
+gst_rtpdec_chain_rtp (GstPad * pad, GstBuffer * buffer)
+{
+ GstRTPDec *src;
+
+ src = GST_RTPDEC (GST_PAD_PARENT (pad));
+
+ GST_DEBUG ("got rtp packet");
+ return gst_pad_push (src->src_rtp, buffer);
+}
+
+static GstFlowReturn
+gst_rtpdec_chain_rtcp (GstPad * pad, GstBuffer * buffer)
+{
+ GST_DEBUG ("got rtcp packet");
+
+ gst_buffer_unref (buffer);
+ return GST_FLOW_OK;
+}
+
+static void
+gst_rtpdec_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRTPDec *src;
+
+ src = GST_RTPDEC (object);
+
+ switch (prop_id) {
+ case ARG_SKIP:
+ break;
+ default:
+ break;
+ }
+}
+
+static void
+gst_rtpdec_get_property (GObject * object, guint prop_id, GValue * value,
+ GParamSpec * pspec)
+{
+ GstRTPDec *src;
+
+ src = GST_RTPDEC (object);
+
+ switch (prop_id) {
+ case ARG_SKIP:
+ break;
+ default:
+ break;
+ }
+}
+
+static GstStateChangeReturn
+gst_rtpdec_change_state (GstElement * element, GstStateChange transition)
+{
+ GstStateChangeReturn ret;
+ GstRTPDec *rtpdec;
+
+ rtpdec = GST_RTPDEC (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ break;
+ default:
+ break;
+ }
+
+ ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ break;
+ default:
+ break;
+ }
+
+ return ret;
+}
diff --git a/gst/rtsp/gstrtpdec.h b/gst/rtsp/gstrtpdec.h
new file mode 100644
index 00000000..5d75e466
--- /dev/null
+++ b/gst/rtsp/gstrtpdec.h
@@ -0,0 +1,55 @@
+/* GStreamer
+ * Copyright (C) <2005> Wim Taymans <wim@fluendo.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_RTPDEC_H__
+#define __GST_RTPDEC_H__
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTPDEC (gst_rtpdec_get_type())
+#define GST_IS_RTPDEC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTPDEC))
+#define GST_IS_RTPDEC_CLASS(obj) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTPDEC))
+#define GST_RTPDEC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTPDEC, GstRTPDec))
+#define GST_RTPDEC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTPDEC, GstRTPDecClass))
+
+typedef struct _GstRTPDec GstRTPDec;
+typedef struct _GstRTPDecClass GstRTPDecClass;
+
+struct _GstRTPDec {
+ GstElement element;
+
+ GstPad *sink_rtp;
+ GstPad *sink_rtcp;
+ GstPad *src_rtp;
+ GstPad *src_rtcp;
+};
+
+struct _GstRTPDecClass {
+ GstElementClass parent_class;
+};
+
+gboolean gst_rtpdec_plugin_init (GstPlugin * plugin);
+
+GType gst_rtpdec_get_type(void);
+
+G_END_DECLS
+
+#endif /* __GST_RTPDEC_H__ */
diff --git a/gst/rtsp/gstrtsp.c b/gst/rtsp/gstrtsp.c
index 8fe43ac9..1990622e 100644
--- a/gst/rtsp/gstrtsp.c
+++ b/gst/rtsp/gstrtsp.c
@@ -21,6 +21,7 @@
#include "config.h"
#endif
+#include "gstrtpdec.h"
#include "gstrtspsrc.h"
static gboolean
@@ -29,6 +30,8 @@ plugin_init (GstPlugin * plugin)
if (!gst_element_register (plugin, "rtspsrc", GST_RANK_NONE,
GST_TYPE_RTSPSRC))
return FALSE;
+ if (!gst_element_register (plugin, "rtpdec", GST_RANK_NONE, GST_TYPE_RTPDEC))
+ return FALSE;
return TRUE;
}
diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index 70110501..bc039281 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -567,18 +567,18 @@ gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
* the rtp session plugin. */
pad = gst_element_get_pad (stream->rtpsrc, "src");
gst_pad_link (pad, stream->rtpdecrtp);
- gst_object_unref (GST_OBJECT (pad));
+ gst_object_unref (pad);
pad = gst_element_get_pad (stream->rtcpsrc, "src");
gst_pad_link (pad, stream->rtpdecrtcp);
- gst_object_unref (GST_OBJECT (pad));
+ gst_object_unref (pad);
}
pad = gst_element_get_pad (stream->rtpdec, "srcrtp");
name = g_strdup_printf ("rtp_stream%d", stream->id);
- gst_element_add_pad (GST_ELEMENT (src), gst_ghost_pad_new (name, pad));
+ gst_element_add_pad (GST_ELEMENT_CAST (src), gst_ghost_pad_new (name, pad));
g_free (name);
- gst_object_unref (GST_OBJECT (pad));
+ gst_object_unref (pad);
return TRUE;
diff --git a/gst/rtsp/rtspconnection.c b/gst/rtsp/rtspconnection.c
index ae39df44..221369d9 100644
--- a/gst/rtsp/rtspconnection.c
+++ b/gst/rtsp/rtspconnection.c
@@ -376,9 +376,10 @@ read_body (gint fd, glong content_length, RTSPMessage * msg)
bodyptr += r;
}
}
+ content_length += 1;
done:
- rtsp_message_set_body (msg, (guint8 *) body, content_length + 1);
+ rtsp_message_set_body (msg, (guint8 *) body, content_length);
return RTSP_OK;
@@ -499,7 +500,16 @@ rtsp_connection_receive (RTSPConnection * conn, RTSPMessage * msg)
if (rtsp_message_get_header (msg, RTSP_HDR_SESSION,
&session_id) == RTSP_OK) {
- gint maxlen = sizeof (conn->session_id) - 1;
+ gint sesslen, maxlen, i;
+
+ sesslen = strlen (session_id);
+ maxlen = sizeof (conn->session_id) - 1;
+ /* the sessionid can have attributes marked with ;
+ * Make sure we strip them */
+ for (i = 0; i < sesslen; i++) {
+ if (session_id[i] == ';')
+ maxlen = i;
+ }
/* make sure to not overflow */
strncpy (conn->session_id, session_id, maxlen);
diff --git a/gst/rtsp/rtspmessage.c b/gst/rtsp/rtspmessage.c
index 1fd7ed3f..4c9571f9 100644
--- a/gst/rtsp/rtspmessage.c
+++ b/gst/rtsp/rtspmessage.c
@@ -263,8 +263,8 @@ rtsp_message_dump (RTSPMessage * msg)
g_print (" reason: '%s'\n", msg->type_data.response.reason);
g_print (" headers:\n");
g_hash_table_foreach (msg->hdr_fields, dump_key_value, NULL);
- g_print (" body:\n");
rtsp_message_get_body (msg, &data, &size);
+ g_print (" body: length %d\n", size);
dump_mem (data, size);
} else if (msg->type == RTSP_MESSAGE_DATA) {
g_print ("data message %p\n", msg);