[MOVED FROM GST-P-FARSIGHT] Add RTP DTMF event packet generator element and test app for it

20070214131352-65035-3c14a1047c3cd6696f3a716a62b21d3f4b9da62b.gz

4 files changed, 605 insertions, 0 deletions

diff --git a/gst/rtpdtmf/.git-darcs-dir b/gst/rtpdtmf/.git-darcs-dir
new file mode 100644
index 00000000..e69de29b
--- /dev/null
+++ b/gst/rtpdtmf/.git-darcs-dir
diff --git a/gst/rtpdtmf/Makefile.am b/gst/rtpdtmf/Makefile.am
new file mode 100644
index 00000000..603b32c2
--- /dev/null
+++ b/gst/rtpdtmf/Makefile.am
@@ -0,0 +1,9 @@
+plugin_LTLIBRARIES = libgstrtpdtmf.la
+
+libgstrtpdtmf_la_SOURCES = gstrtpdtmfsrc.c
+
+libgstrtpdtmf_la_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) $(ERROR_CFLAGS) -DEXTERN_BUF -DRTP_SUPPORT
+libgstrtpdtmf_la_LIBADD = $(GST_LIBS_LIBS)
+libgstrtpdtmf_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstrtp-@GST_MAJORMINOR@
+
+noinst_HEADERS = gstrtpdtmfsrc.h
diff --git a/gst/rtpdtmf/gstrtpdtmfsrc.c b/gst/rtpdtmf/gstrtpdtmfsrc.c
new file mode 100644
index 00000000..0d036bdf
--- /dev/null
+++ b/gst/rtpdtmf/gstrtpdtmfsrc.c
@@ -0,0 +1,511 @@
+/* GStreamer RTP DTMF source
+ * Copyright (C) 2007 Zeeshan Ali <zeenix@gstreamer.net>
+ *
+ * gstrtpdtmfsrc.c:
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#  include "config.h"
+#endif
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "gstrtpdtmfsrc.h"
+
+#define GST_RTP_DTMF_TYPE_EVENT 1
+#define DEFAULT_PACKET_INTERVAL ((guint16) 50) /* ms */
+#define DEFAULT_SSRC            -1
+#define DEFAULT_PT              96
+#define DEFAULT_CLOCK_RATE      8000
+#define MIN_EVENT               0
+#define MAX_EVENT               16
+#define MIN_EVENT_STRING        "0"
+#define MAX_EVENT_STRING        "16"
+#define MIN_VOLUME              0
+#define MAX_VOLUME              36
+
+/* elementfactory information */
+static const GstElementDetails gst_rtp_dtmf_src_details =
+GST_ELEMENT_DETAILS ("RTP DTMF packet generator",
+    "Source/Network",
+    "Generates RTP DTMF packets",
+    "Zeeshan Ali <zeeshan.ali@nokia.com>");
+
+GST_DEBUG_CATEGORY_STATIC (gst_rtp_dtmf_src_debug);
+#define GST_CAT_DEFAULT gst_rtp_dtmf_src_debug
+
+/* signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum
+{
+  PROP_0,
+  PROP_SSRC,
+  PROP_PT,
+  PROP_CLOCK_RATE,
+};
+
+static GstStaticPadTemplate gst_rtp_dtmf_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("application/x-rtp, "
+        "media = (string) \"audio\", "
+        "payload = (int) [ 96, 127 ], "
+        "clock-rate = (int) [ 0, MAX ], " 
+        "ssrc = (int) [ 0, MAX ], " 
+        "event = (int) [ " MIN_EVENT_STRING ", " MAX_EVENT_STRING " ], "
+        "encoding-name = (string) \"telephone-event\"")
+    );
+
+static GstElementClass *parent_class = NULL;
+
+static void gst_rtp_dtmf_src_base_init (gpointer g_class);
+static void gst_rtp_dtmf_src_class_init (GstRTPDTMFSrcClass * klass);
+static void gst_rtp_dtmf_src_init (GstRTPDTMFSrc * dtmfsrc, gpointer g_class);
+static void gst_rtp_dtmf_src_finalize (GObject * object);
+
+GType
+gst_rtp_dtmf_src_get_type (void)
+{
+  static GType base_src_type = 0;
+
+  if (G_UNLIKELY (base_src_type == 0)) {
+    static const GTypeInfo base_src_info = {
+      sizeof (GstRTPDTMFSrcClass),
+      (GBaseInitFunc) gst_rtp_dtmf_src_base_init,
+      NULL,
+      (GClassInitFunc) gst_rtp_dtmf_src_class_init,
+      NULL,
+      NULL,
+      sizeof (GstRTPDTMFSrc),
+      0,
+      (GInstanceInitFunc) gst_rtp_dtmf_src_init,
+    };
+
+    base_src_type = g_type_register_static (GST_TYPE_ELEMENT,
+        "GstRTPDTMFSrc", &base_src_info, 0);
+  }
+  return base_src_type;
+}
+
+static void gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+static gboolean gst_rtp_dtmf_src_handle_event (GstPad * pad, GstEvent * event);
+static GstStateChangeReturn gst_rtp_dtmf_src_change_state (GstElement * element,
+    GstStateChange transition);
+static void gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc);
+static void gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc, gint event_number,
+    gint event_volume);
+static void gst_rtp_dtmf_src_stop (GstRTPDTMFSrc *dtmfsrc);
+static void gst_rtp_dtmf_src_set_caps (GstRTPDTMFSrc *dtmfsrc);
+
+static void
+gst_rtp_dtmf_src_base_init (gpointer g_class)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+  
+  GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_src_debug,
+          "dtmfsrc", 0, "dtmfsrc element");
+  
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&gst_rtp_dtmf_src_template));
+  
+  gst_element_class_set_details (element_class, &gst_rtp_dtmf_src_details);
+}
+
+static void
+gst_rtp_dtmf_src_class_init (GstRTPDTMFSrcClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+
+  gobject_class = G_OBJECT_CLASS (klass);
+  gstelement_class = GST_ELEMENT_CLASS (klass);
+
+  parent_class = g_type_class_peek_parent (klass);
+
+  gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_finalize);
+  gobject_class->set_property = 
+      GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_set_property);
+  gobject_class->get_property =
+      GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_get_property);
+
+  g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_CLOCK_RATE,
+      g_param_spec_uint ("clock-rate", "clockrate",
+          "The clock-rate at which to generate the dtmf packets",
+          0, G_MAXUINT, DEFAULT_CLOCK_RATE, G_PARAM_READWRITE));
+  g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SSRC,
+      g_param_spec_uint ("ssrc", "SSRC",
+          "The SSRC of the packets (-1 == random)",
+          0, G_MAXUINT, DEFAULT_SSRC, G_PARAM_READWRITE));
+  g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PT,
+      g_param_spec_uint ("pt", "payload type",
+          "The payload type of the packets",
+          0, 0x80, DEFAULT_PT, G_PARAM_READWRITE));
+
+  gstelement_class->change_state =
+      GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_change_state);
+}
+
+static void
+gst_rtp_dtmf_src_init (GstRTPDTMFSrc * dtmfsrc, gpointer g_class)
+{
+  dtmfsrc->srcpad =
+      gst_pad_new_from_static_template (&gst_rtp_dtmf_src_template, "src");
+  GST_DEBUG_OBJECT (dtmfsrc, "adding src pad");
+  gst_element_add_pad (GST_ELEMENT (dtmfsrc), dtmfsrc->srcpad);
+
+  gst_pad_set_event_function (dtmfsrc->srcpad, gst_rtp_dtmf_src_handle_event);
+  
+  dtmfsrc->ssrc = DEFAULT_SSRC;
+  dtmfsrc->pt = DEFAULT_PT;
+  dtmfsrc->clock_rate = DEFAULT_CLOCK_RATE;
+  
+  gst_rtp_dtmf_src_set_caps (dtmfsrc);
+  
+  GST_DEBUG_OBJECT (dtmfsrc, "init done");
+}
+
+static void
+gst_rtp_dtmf_src_finalize (GObject * object)
+{
+  GstRTPDTMFSrc *dtmfsrc;
+
+  dtmfsrc = GST_RTP_DTMF_SRC (object);
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static gboolean
+gst_rtp_dtmf_src_handle_event (GstPad * pad, GstEvent * event)
+{
+  GstRTPDTMFSrc *dtmfsrc;
+  gboolean result = FALSE;
+
+  dtmfsrc = GST_RTP_DTMF_SRC (GST_PAD_PARENT (pad));
+
+  GST_DEBUG_OBJECT (dtmfsrc, "Received an event on the src pad");
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_CUSTOM_UPSTREAM:
+    {
+      const GstStructure *structure;
+
+      if (GST_STATE (dtmfsrc) != GST_STATE_PLAYING) {
+        GST_DEBUG_OBJECT (dtmfsrc, "Received event while not in PLAYING state");
+        break;
+      }
+
+      GST_DEBUG_OBJECT (dtmfsrc, "Received event is of our interest");
+      structure = gst_event_get_structure (event);
+      if (structure && gst_structure_has_name (structure, "dtmf-event")) {
+        gint event_type;
+        gboolean start;
+
+        if (!gst_structure_get_int (structure, "type", &event_type) ||
+            !gst_structure_get_boolean (structure, "start", &start) ||
+            event_type != GST_RTP_DTMF_TYPE_EVENT)
+          break;
+
+        if (start) {
+          gint event_number;
+          gint event_volume;
+
+          if (!gst_structure_get_int (structure, "number", &event_number) ||
+              !gst_structure_get_int (structure, "volume", &event_volume)) 
+            break;
+
+          gst_rtp_dtmf_src_start (dtmfsrc, event_number, event_volume);
+        }
+
+        else {
+          gst_rtp_dtmf_src_stop (dtmfsrc);
+        }
+      }
+      break;
+    }
+    default:
+      break;
+  }
+  gst_event_unref (event);
+  return result;
+}
+
+static void
+gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRTPDTMFSrc *dtmfsrc;
+
+  dtmfsrc = GST_RTP_DTMF_SRC (object);
+
+  switch (prop_id) {
+    case PROP_CLOCK_RATE:
+      dtmfsrc->clock_rate = g_value_get_uint (value);
+      gst_rtp_dtmf_src_set_caps (dtmfsrc);
+      break;
+    case PROP_SSRC:
+      dtmfsrc->ssrc = g_value_get_uint (value);
+      break;
+    case PROP_PT:
+      dtmfsrc->pt = g_value_get_uint (value);
+      gst_rtp_dtmf_src_set_caps (dtmfsrc);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id, GValue * value,
+    GParamSpec * pspec)
+{
+  GstRTPDTMFSrc *dtmfsrc;
+
+  dtmfsrc = GST_RTP_DTMF_SRC (object);
+
+  switch (prop_id) {
+    case PROP_CLOCK_RATE:
+      g_value_set_uint (value, dtmfsrc->clock_rate);
+      break;
+    case PROP_SSRC:
+      g_value_set_uint (value, dtmfsrc->ssrc);
+      break;
+    case PROP_PT:
+      g_value_set_uint (value, dtmfsrc->pt);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc,
+        gint event_number, gint event_volume)
+{
+  g_return_if_fail (dtmfsrc->payload == NULL);
+
+  dtmfsrc->payload = g_new0 (GstRTPDTMFPayload, 1);
+  dtmfsrc->payload->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
+  dtmfsrc->payload->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
+
+  dtmfsrc->next_ts = gst_clock_get_time (GST_ELEMENT_CLOCK (dtmfsrc));
+  dtmfsrc->first_packet = TRUE;
+
+  if (dtmfsrc->ssrc == -1)
+    dtmfsrc->current_ssrc = g_random_int ();
+  else
+    dtmfsrc->current_ssrc = dtmfsrc->ssrc;
+
+  if (!gst_pad_start_task (dtmfsrc->srcpad,
+      (GstTaskFunction) gst_rtp_dtmf_src_push_next_rtp_packet, dtmfsrc)) {
+    GST_ERROR_OBJECT (dtmfsrc, "Failed to start task on src pad");
+  }
+}
+
+static void
+gst_rtp_dtmf_src_stop (GstRTPDTMFSrc *dtmfsrc)
+{
+  g_return_if_fail (dtmfsrc->payload != NULL);
+
+  if (!gst_pad_stop_task (dtmfsrc->srcpad)) {
+    GST_ERROR_OBJECT (dtmfsrc, "Failed to stop task on src pad");
+    return;
+  }
+
+  /* Push the last packet with e-bit set */
+  dtmfsrc->payload->e = 1;
+  gst_rtp_dtmf_src_push_next_rtp_packet (dtmfsrc);
+
+  g_free (dtmfsrc->payload);
+  dtmfsrc->payload = NULL;
+}
+
+static void
+gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
+{
+  GstBuffer *buf = NULL;
+  GstFlowReturn ret;
+  GstRTPDTMFPayload *payload;
+  GstClock * clock;
+  guint32 rtp_ts;
+
+  /* create buffer to hold the payload */
+  buf = gst_rtp_buffer_new_allocate (sizeof (GstRTPDTMFPayload), 0, 0);
+
+  gst_rtp_buffer_set_ssrc (buf, dtmfsrc->current_ssrc);
+  gst_rtp_buffer_set_payload_type (buf, dtmfsrc->pt);
+  if (dtmfsrc->first_packet) {
+    gst_rtp_buffer_set_marker (buf, TRUE);
+    dtmfsrc->first_packet = FALSE;
+  }
+
+  /* timestamp and duration of GstBuffer */ 
+  GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->next_ts;
+  GST_BUFFER_DURATION (buf) = DEFAULT_PACKET_INTERVAL * GST_MSECOND;
+  
+  /* duration of DTMF payload */
+  dtmfsrc->payload->duration +=
+      DEFAULT_PACKET_INTERVAL * dtmfsrc->clock_rate / 1000;
+
+  payload = (GstRTPDTMFPayload *) gst_rtp_buffer_get_payload (buf);
+  /* timestamp of RTP header */
+  rtp_ts = dtmfsrc->next_ts * dtmfsrc->clock_rate / GST_SECOND;
+  gst_rtp_buffer_set_timestamp (buf, rtp_ts);
+  
+  /* copy payload and convert to network-byte order */
+  g_memmove (payload, dtmfsrc->payload, sizeof (GstRTPDTMFPayload));
+  payload->duration = g_htons (payload->duration);
+
+  /* FIXME: Should we sync to clock ourselves or leave it to sink */
+  clock = GST_ELEMENT_CLOCK (dtmfsrc);
+  if (clock != NULL) {
+    GstClockID clock_id;
+    GstClockReturn clock_ret;
+
+    clock_id = gst_clock_new_single_shot_id (clock, dtmfsrc->next_ts);
+    clock_ret = gst_clock_id_wait (clock_id, NULL);
+    if (clock_ret != GST_CLOCK_OK && clock_ret != GST_CLOCK_EARLY) {
+      GST_ERROR_OBJECT (dtmfsrc, "Failed to wait on clock %s",
+              GST_ELEMENT_NAME (clock));
+    }
+    gst_clock_id_unref (clock_id);
+  }
+
+  else {
+    GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s", GST_ELEMENT_NAME (dtmfsrc));
+  }
+
+  dtmfsrc->next_ts += GST_BUFFER_DURATION (buf);
+
+  GST_DEBUG_OBJECT (dtmfsrc,
+          "pushing buffer on src pad of size %d", GST_BUFFER_SIZE (buf));
+  ret = gst_pad_push (dtmfsrc->srcpad, buf);
+  if (ret != GST_FLOW_OK)
+    GST_ERROR_OBJECT (dtmfsrc,
+            "Failed to push buffer on src pad", GST_BUFFER_SIZE (buf));
+    
+  GST_DEBUG_OBJECT (dtmfsrc,
+          "pushed DTMF event '%d' on src pad", dtmfsrc->payload->event);
+}
+
+static void
+gst_rtp_dtmf_src_set_caps (GstRTPDTMFSrc *dtmfsrc)
+{
+  GstCaps *caps;
+
+  caps = gst_caps_new_simple ("application/x-rtp",
+      "media", G_TYPE_STRING, "audio",
+      "payload", G_TYPE_INT, dtmfsrc->pt,
+      "clock-rate", G_TYPE_INT, dtmfsrc->clock_rate,
+      "encoding-name", G_TYPE_STRING, "telephone-event",
+      "ssrc", G_TYPE_UINT, dtmfsrc->current_ssrc, NULL);
+
+  if (!gst_pad_set_caps (dtmfsrc->srcpad, caps)) {
+    GST_ERROR_OBJECT (dtmfsrc, "Failed to set caps % on src pad",
+            GST_PTR_FORMAT, caps);
+  }
+
+  gst_caps_unref (caps);
+}
+
+static GstStateChangeReturn
+gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition)
+{
+  GstRTPDTMFSrc *dtmfsrc;
+  GstStateChangeReturn result;
+  gboolean no_preroll = TRUE;
+
+  dtmfsrc = GST_RTP_DTMF_SRC (element);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+      break;
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+      /* Indicate that we don't do PRE_ROLL */
+      no_preroll = TRUE;
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+      break;
+    default:
+      break;
+  }
+
+  if ((result =
+          GST_ELEMENT_CLASS (parent_class)->change_state (element,
+              transition)) == GST_STATE_CHANGE_FAILURE)
+    goto failure;
+
+  switch (transition) {
+    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+      /* Indicate that we don't do PRE_ROLL */
+      no_preroll = TRUE;
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      break;
+    case GST_STATE_CHANGE_READY_TO_NULL:
+      break;
+    default:
+      break;
+  }
+
+  if (no_preroll && result == GST_STATE_CHANGE_SUCCESS)
+    result = GST_STATE_CHANGE_NO_PREROLL;
+
+  return result;
+
+  /* ERRORS */
+failure:
+  {
+    GST_ERROR_OBJECT (dtmfsrc, "parent failed state change");
+    return result;
+  }
+}
+
+static gboolean
+gst_rtp_dtmf_src_plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rtpdtmfsrc",
+      GST_RANK_NONE, GST_TYPE_RTP_DTMF_SRC);
+}
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  if (!gst_rtp_dtmf_src_plugin_init (plugin))
+    return FALSE;
+
+  return TRUE;
+}
+
+#define PACKAGE "gstreamer"
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    "dtmf",
+    "DTMF plugins",
+    plugin_init, "0.1" , "LGPL", "DTMF", "");
diff --git a/gst/rtpdtmf/gstrtpdtmfsrc.h b/gst/rtpdtmf/gstrtpdtmfsrc.h
new file mode 100644
index 00000000..399dbcae
--- /dev/null
+++ b/gst/rtpdtmf/gstrtpdtmfsrc.h
@@ -0,0 +1,85 @@
+/* GStreamer RTP DTMF source
+ * Copyright (C) 2007 Zeeshan Ali <zeenix@gstreamer.net>
+ *
+ * gstrtpdtmfsrc.h:
+ *
+ * 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_DTMF_SRC_H__
+#define __GST_RTP_DTMF_SRC_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTP_DTMF_SRC		(gst_rtp_dtmf_src_get_type())
+#define GST_RTP_DTMF_SRC(obj)		(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_DTMF_SRC,GstRTPDTMFSrc))
+#define GST_RTP_DTMF_SRC_CLASS(klass)	(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_DTMF_SRC,GstRTPDTMFSrcClass))
+#define GST_RTP_DTMF_SRC_GET_CLASS(obj)     (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTP_DTMF_SRC, GstRTPDTMFSrcClass))
+#define GST_IS_RTP_DTMF_SRC(obj)		(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_DTMF_SRC))
+#define GST_IS_RTP_DTMF_SRC_CLASS(klass)	(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_DTMF_SRC))
+#define GST_RTP_DTMF_SRC_CAST(obj)		((GstRTPDTMFSrc *)(obj))
+
+typedef struct {
+  unsigned  event:8;         /* Current DTMF event */
+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
+  unsigned  volume:6;        /* power level of the tone, in dBm0 */
+  unsigned  r:1;             /* Reserved-bit */
+  unsigned  e:1;             /* End-bit */
+#elif G_BYTE_ORDER == G_BIG_ENDIAN
+  unsigned  e:1;             /* End-bit */
+  unsigned  r:1;             /* Reserved-bit */
+  unsigned  volume:6;        /* power level of the tone, in dBm0 */
+#else
+#error "G_BYTE_ORDER should be big or little endian."
+#endif
+  unsigned  duration:16;     /* Duration of digit, in timestamp units */
+} GstRTPDTMFPayload;
+
+typedef struct _GstRTPDTMFSrc GstRTPDTMFSrc;
+typedef struct _GstRTPDTMFSrcClass GstRTPDTMFSrcClass;
+
+/**
+ * GstRTPDTMFSrc:
+ * @element: the parent element.
+ *
+ * The opaque #GstRTPDTMFSrc data structure.
+ */
+struct _GstRTPDTMFSrc {
+  GstElement        element;
+
+  GstPad	    *srcpad;
+  GstRTPDTMFPayload *payload;
+
+  GstClockTime      next_ts;
+  guint32           clock_rate;
+  guint             pt;
+  guint             ssrc;
+  guint             current_ssrc;
+  gboolean          first_packet;
+};
+
+struct _GstRTPDTMFSrcClass {
+  GstElementClass parent_class;
+};
+
+GType gst_rtp_dtmf_src_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_DTMF_SRC_H__ */