im not being able to upload the patch, it keeps telling me that the file does
not exist or that i dont have permission to read it....but it does exists and i
can read it. So im pasting here the patch content:

From 22f24ed301f7980fac9a3832dc093b534c54fc98 Mon Sep 17 00:00:00 2001
From: Tiago Katcipis <katcipis@katcipis-laptop.(none)>
Date: Thu, 8 Oct 2009 12:16:52 -0300
Subject: [PATCH] adding rtpg723pay plugin

---
 configure.ac                   |    2 +
 gst/rtpg723pay/Makefile.am     |   11 ++
 gst/rtpg723pay/gstrtpg723pay.c |  310 ++++++++++++++++++++++++++++++++++++++++
 gst/rtpg723pay/gstrtpg723pay.h |   59 ++++++++
 4 files changed, 382 insertions(+), 0 deletions(-)
 create mode 100644 gst/rtpg723pay/Makefile.am
 create mode 100644 gst/rtpg723pay/gstrtpg723pay.c
 create mode 100644 gst/rtpg723pay/gstrtpg723pay.h

diff --git a/configure.ac b/configure.ac
index 7c32666..510efe5 100644
--- a/configure.ac
+++ b/configure.ac
@@ -285,6 +285,7 @@ AG_GST_CHECK_PLUGIN(pnm)
 AG_GST_CHECK_PLUGIN(qtmux)
 AG_GST_CHECK_PLUGIN(rawparse)
 AG_GST_CHECK_PLUGIN(real)
+AG_GST_CHECK_PLUGIN(rtpg723pay)
 AG_GST_CHECK_PLUGIN(rtpmux)
 AG_GST_CHECK_PLUGIN(scaletempo)
 AG_GST_CHECK_PLUGIN(sdp)
@@ -1767,6 +1768,7 @@ gst/pnm/Makefile
 gst/qtmux/Makefile
 gst/rawparse/Makefile
 gst/real/Makefile
+gst/rtpg723pay/Makefile
 gst/rtpmux/Makefile
 gst/scaletempo/Makefile
 gst/sdp/Makefile
diff --git a/gst/rtpg723pay/Makefile.am b/gst/rtpg723pay/Makefile.am
new file mode 100644
index 0000000..75d13f3
--- /dev/null
+++ b/gst/rtpg723pay/Makefile.am
@@ -0,0 +1,11 @@
+
+plugin_LTLIBRARIES = libgstrtpg723pay.la
+
+libgstrtpg723pay_la_SOURCES = gstrtpg723pay.c 
+libgstrtpg723pay_la_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
+libgstrtpg723pay_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS)
+libgstrtpg723pay_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+libgstrtpg723pay_la_LIBTOOLFLAGS = --tag=disable-static
+
+noinst_HEADERS = \
+	gstrtpg723pay.h
diff --git a/gst/rtpg723pay/gstrtpg723pay.c b/gst/rtpg723pay/gstrtpg723pay.c
new file mode 100644
index 0000000..86f86f7
--- /dev/null
+++ b/gst/rtpg723pay/gstrtpg723pay.c
@@ -0,0 +1,310 @@
+/* GStreamer
+ * Copyright (C) <2007> Nokia Corporation
+ * Copyright (C) <2007> Collabora Ltd
+ *  @author: Olivier Crete <olivier.crete@collabora.co.uk>
+ *
+ * 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.
+ */
+
+/*
+ * This payloader assumes that the data will ALWAYS come as zero or more
+ * 10 bytes frame of audio followed by 0 or 1 2 byte frame of silence.
+ * Any other buffer format won't work
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <string.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/base/gstadapter.h>
+
+#include "gstrtpg723pay.h"
+
+#define GST_RTP_PAYLOAD_G723 4
+#define GST_RTP_PAYLOAD_G723_STRING "4"
+
+/* According to RFC 3551, works only with G723 encoded with 6.3 kb/s high-rate */
+#define G723_FRAME_SIZE 24
+#define G723B_SID_FRAME_SIZE 4
+#define G723_FRAME_DURATION (30 * GST_MSECOND)
+#define G723_FRAME_DURATION_MS (30)
+
+static gboolean
+gst_rtp_g723_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps);
+static GstFlowReturn
+gst_rtp_g723_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf);
+
+
+static const GstElementDetails gst_rtp_g723_pay_details =
+GST_ELEMENT_DETAILS ("RTP G.723 payloader",
+    "Codec/Payloader/Network",
+    "Packetize 6.3kb/s G.723 audio into RTP packets",
+    "Tiago Katcipis <tiago.katcipis@digitro.com.br>");
+
+static GstStaticPadTemplate gst_rtp_g723_pay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/G723, "     /* according to RFC 3551 */
+        "channels = (int) 1, " "rate = (int) 8000")
+    );
+
+static GstStaticPadTemplate gst_rtp_g723_pay_src_template =
+    GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, "
+        "media = (string) \"audio\", "
+        "payload = (int) " GST_RTP_PAYLOAD_G723_STRING ", "
+        "clock-rate = (int) 8000, "
+        "encoding-name = (string) \"G723\"; "
+        "application/x-rtp, "
+        "media = (string) \"audio\", "
+        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+        "clock-rate = (int) 8000, " "encoding-name = (string) \"G723\"")
+    );
+
+static void
+gst_rtp_g723_pay_init (GstRTPG723Pay * pay, GstRTPG723PayClass * klass);
+
+GST_BOILERPLATE (GstRTPG723Pay, gst_rtp_g723_pay, GstBaseRTPAudioPayload,
+    GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
+
+static void
+gst_rtp_g723_pay_base_init (gpointer klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtp_g723_pay_sink_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtp_g723_pay_src_template));
+  gst_element_class_set_details (element_class, &gst_rtp_g723_pay_details);
+}
+
+static void
+gst_rtp_g723_pay_class_init (GstRTPG723PayClass * klass)
+{
+  GstBaseRTPPayloadClass *payload_class = GST_BASE_RTP_PAYLOAD_CLASS (klass);
+
+  payload_class->set_caps = gst_rtp_g723_pay_set_caps;
+  payload_class->handle_buffer = gst_rtp_g723_pay_handle_buffer;
+}
+
+static void
+gst_rtp_g723_pay_init (GstRTPG723Pay * pay, GstRTPG723PayClass * klass)
+{
+  GstBaseRTPPayload *payload = GST_BASE_RTP_PAYLOAD (pay);
+  GstBaseRTPAudioPayload *audiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (pay);
+
+  payload->pt = GST_RTP_PAYLOAD_G723;
+  gst_basertppayload_set_options (payload, "audio", FALSE, "G723", 8000);
+
+  gst_base_rtp_audio_payload_set_frame_based (audiopayload);
+  gst_base_rtp_audio_payload_set_frame_options (audiopayload,
+      G723_FRAME_DURATION_MS, G723_FRAME_SIZE);
+
+}
+
+static gboolean
+gst_rtp_g723_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps)
+{
+  gboolean res;
+  GstStructure *structure;
+  gint pt;
+
+  structure = gst_caps_get_structure (caps, 0);
+  if (!gst_structure_get_int (structure, "payload", &pt))
+    pt = GST_RTP_PAYLOAD_G723;
+
+  payload->pt = pt;
+  payload->dynamic = pt != GST_RTP_PAYLOAD_G723;
+
+  res = gst_basertppayload_set_outcaps (payload, NULL);
+
+  return res;
+}
+
+static GstFlowReturn
+gst_rtp_g723_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
+{
+  GstFlowReturn ret = GST_FLOW_OK;
+  GstBaseRTPAudioPayload *basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (payload);
+  GstAdapter *adapter = NULL;
+  guint payload_len;
+  const guint8 *data = NULL;
+  guint available;
+  guint maxptime_octets = G_MAXUINT;
+  guint minptime_octets = 0;
+  guint min_payload_len;
+  guint max_payload_len;
+  gboolean use_adapter = FALSE;
+
+  available = GST_BUFFER_SIZE (buf);
+
+  if (available % G723_FRAME_SIZE != 0 &&
+      available % G723_FRAME_SIZE != G723B_SID_FRAME_SIZE)
+    goto invalid_size;
+
+  /* max number of bytes based on given ptime, has to be multiple of
+   * frame_duration */
+  if (payload->max_ptime != -1) {
+    guint ptime_ms = payload->max_ptime / 1000000;
+
+    maxptime_octets = G723_FRAME_SIZE *
+        (int) (ptime_ms / G723_FRAME_DURATION_MS);
+
+    if (maxptime_octets < G723_FRAME_SIZE) {
+      GST_WARNING_OBJECT (basertpaudiopayload, "Given ptime %" G_GINT64_FORMAT
+          " is smaller than minimum %d ns, overwriting to minimum",
+          payload->max_ptime, G723_FRAME_DURATION_MS);
+      maxptime_octets = G723_FRAME_SIZE;
+    }
+  }
+
+  max_payload_len = MIN (
+      /* MTU max */
+      (int) (gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
+              (basertpaudiopayload), 0, 0) / G723_FRAME_SIZE) * G723_FRAME_SIZE,
+      /* ptime max */
+      maxptime_octets);
+
+  /* min number of bytes based on a given ptime, has to be a multiple
+     of frame duration */
+  {
+    guint64 min_ptime;
+
+    g_object_get (G_OBJECT (payload), "min-ptime", &min_ptime, NULL);
+
+    min_ptime = min_ptime / 1000000;
+    minptime_octets = G723_FRAME_SIZE *
+        (int) (min_ptime / G723_FRAME_DURATION_MS);
+  }
+
+  min_payload_len = MAX (minptime_octets, G723_FRAME_SIZE);
+
+  if (min_payload_len > max_payload_len) {
+    min_payload_len = max_payload_len;
+  }
+
+  GST_DEBUG_OBJECT (basertpaudiopayload,
+      "Calculated min_payload_len %u and max_payload_len %u",
+      min_payload_len, max_payload_len);
+
+  adapter = gst_base_rtp_audio_payload_get_adapter (basertpaudiopayload);
+
+  if (adapter && gst_adapter_available (adapter)) {
+    /* If there is always data in the adapter, we have to use it */
+    gst_adapter_push (adapter, buf);
+    available = gst_adapter_available (adapter);
+    use_adapter = TRUE;
+  } else {
+    /* let's set the base timestamp */
+    basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buf);
+
+    /* If buffer fits on an RTP packet, let's just push it through */
+    /* this will check against max_ptime and max_mtu */
+    if (GST_BUFFER_SIZE (buf) >= min_payload_len &&
+        GST_BUFFER_SIZE (buf) <= max_payload_len) {
+      ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
+          GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
+          GST_BUFFER_TIMESTAMP (buf));
+      gst_buffer_unref (buf);
+      return ret;
+    }
+
+    available = GST_BUFFER_SIZE (buf);
+    data = (guint8 *) GST_BUFFER_DATA (buf);
+  }
+
+  /* as long as we have full frames */
+  /* this loop will push all available buffers till the last frame */
+  while (available >= min_payload_len ||
+      available % G723_FRAME_SIZE == G723B_SID_FRAME_SIZE) {
+    guint num;
+
+    /* We send as much as we can */
+    if (available <= max_payload_len) {
+      payload_len = available;
+    } else {
+      payload_len = MIN (max_payload_len,
+          (available / G723_FRAME_SIZE) * G723_FRAME_SIZE);
+    }
+
+    if (use_adapter) {
+      data = gst_adapter_peek (adapter, payload_len);
+    }
+
+    ret = gst_base_rtp_audio_payload_push (basertpaudiopayload, data,
+        payload_len, basertpaudiopayload->base_ts);
+
+    num = payload_len / G723_FRAME_SIZE;
+    basertpaudiopayload->base_ts += G723_FRAME_DURATION * num;
+
+    if (use_adapter) {
+      gst_adapter_flush (adapter, payload_len);
+      available = gst_adapter_available (adapter);
+    } else {
+      available -= payload_len;
+      data += payload_len;
+    }
+  }
+
+  if (!use_adapter) {
+    if (available != 0 && adapter) {
+      GstBuffer *buf2;
+      buf2 = gst_buffer_create_sub (buf,
+          GST_BUFFER_SIZE (buf) - available, available);
+      gst_adapter_push (adapter, buf2);
+    } else {
+      gst_buffer_unref (buf);
+    }
+  }
+
+  if (adapter) {
+    g_object_unref (adapter);
+  }
+
+  return ret;
+
+  /* ERRORS */
+invalid_size:
+  {
+    GST_ELEMENT_ERROR (payload, STREAM, WRONG_TYPE,
+        ("Invalid input buffer size"),
+        ("Invalid buffer size, should be a multiple of"
+            " G723_FRAME_SIZE(24) with an optional G723B_SID_FRAME_SIZE(4)"
+            " added to it, but it is %u", available));
+    gst_buffer_unref (buf);
+    return GST_FLOW_ERROR;
+  }
+}
+
+/*Plugin init functions*/
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpg723pay", GST_RANK_NONE, gst_rtp_g723_pay_get_type());
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "rtpg723pay",
+    "Packetize 6.3kb/s G.723 audio into RTP packets",
+    plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
+
diff --git a/gst/rtpg723pay/gstrtpg723pay.h b/gst/rtpg723pay/gstrtpg723pay.h
new file mode 100644
index 0000000..b027e22
--- /dev/null
+++ b/gst/rtpg723pay/gstrtpg723pay.h
@@ -0,0 +1,59 @@
+/* GStreamer
+ * Copyright (C) <2007> Nokia Corporation
+ * Copyright (C) <2007> Collabora Ltd
+ *
+ * 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_RTP_G723_PAY_H__
+#define __GST_RTP_G723_PAY_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertpaudiopayload.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTP_G723_PAY \
+  (gst_rtp_g723_pay_get_type())
+#define GST_RTP_G723_PAY(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_G723_PAY,GstRTPG723Pay))
+#define GST_RTP_G723_PAY_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_G723_PAY,GstRTPG723PayClass))
+#define GST_IS_RTP_G723_PAY(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_G723_PAY))
+#define GST_IS_RTP_G723_PAY_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_G723_PAY))
+
+typedef struct _GstRTPG723Pay GstRTPG723Pay;
+typedef struct _GstRTPG723PayClass GstRTPG723PayClass;
+
+struct _GstRTPG723Pay
+{
+  GstBaseRTPAudioPayload audiopayload;
+};
+
+struct _GstRTPG723PayClass
+{
+  GstBaseRTPAudioPayloadClass parent_class;
+};
+
+gboolean gst_rtp_g723_pay_plugin_init (GstPlugin * plugin);
+
+GType gst_rtp_g723_pay_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_G723_PAY_H__ */
-- 
1.6.0.4

Created attachment 145079 [details] [review]
rtpg723pay patch

now at home the patches are going :-).

Have my patch been accepted? if don't...what i have to do so it can be accepted?

commit 908a9ee63b9747111a4004c22e55905b5c1e430b
Author: Tiago Katcipis <tiago.katcipis@digitro.com.br>
Date:   Thu Dec 17 17:27:42 2009 +0100

    rtp: add G723 payloader
    
    Fixes #597823