[MOVED FROM GST-P-FARSIGHT] Moved rtpdtmf to dtmf directory

20070820184800-4f0f6-fa33ea974510161de8c9951c39087af3613b65a4.gz

3 files changed, 0 insertions, 982 deletions

diff --git a/gst/rtpdtmf/Makefile.am b/gst/rtpdtmf/Makefile.am
deleted file mode 100644
index 603b32c2..00000000
--- a/gst/rtpdtmf/Makefile.am
+++ /dev/null
@@ -1,9 +0,0 @@
-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
deleted file mode 100644
index 8e4f31b5..00000000
--- a/gst/rtpdtmf/gstrtpdtmfsrc.c
+++ /dev/null
@@ -1,872 +0,0 @@
-/* GStreamer RTP DTMF source
- *
- * gstrtpdtmfsrc.c:
- *
- * Copyright (C) <2007> Nokia Corporation.
- *   Contact: Zeeshan Ali <zeeshan.ali@nokia.com>
- * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
- *               2000,2005 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.
- */
-
-/**
- * SECTION:element-rtpdtmfsrc
- * @short_description: Generates RTP DTMF packets
- *
- * <refsect2>
- *
- * <para>
- * The RTPDTMFSrc element generates RTP DTMF (RFC 2833) event packets on request
- * from application. The application communicates the beginning and end of a
- * DTMF event using custom upstream gstreamer events. To report a DTMF event, an
- * application must send an event of type GST_EVENT_CUSTOM_UPSTREAM, having a
- * structure of name "dtmf-event" with fields set according to the following
- * table:
- * </para>
- *
- * <para>
- * <informaltable>
- * <tgroup cols='4'>
- * <colspec colname='Name' />
- * <colspec colname='Type' />
- * <colspec colname='Possible values' />
- * <colspec colname='Purpose' />
- * 
- * <thead>
- * <row>
- * <entry>Name</entry>
- * <entry>GType</entry>
- * <entry>Possible values</entry>
- * <entry>Purpose</entry>
- * </row>
- * </thead>
- * 
- * <tbody>
- * <row>
- * <entry>type</entry>
- * <entry>G_TYPE_INT</entry>
- * <entry>0-1</entry>
- * <entry>The application uses this field to specify which of the two methods
- * specified in RFC 2833 to use. The value should be 0 for tones and 1 for
- * named events. This element is only capable of generating named events.
- * </entry>
- * </row>
- * <row>
- * <entry>number</entry>
- * <entry>G_TYPE_INT</entry>
- * <entry>0-16</entry>
- * <entry>The event number.</entry>
- * </row>
- * <row>
- * <entry>volume</entry>
- * <entry>G_TYPE_INT</entry>
- * <entry>0-36</entry>
- * <entry>This field describes the power level of the tone, expressed in dBm0
- * after dropping the sign. Power levels range from 0 to -63 dBm0. The range of
- * valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE.
- * </entry>
- * </row>
- * <row>
- * <entry>start</entry>
- * <entry>G_TYPE_BOOLEAN</entry>
- * <entry>True or False</entry>
- * <entry>Whether the event is starting or ending.</entry>
- * </row>
- * <row>
- * <entry>method</entry>
- * <entry>G_TYPE_INT</entry>
- * <entry>1</entry>
- * <entry>The method used for sending event, this element will react if this field
- * is absent or 1.
- * </entry>
- * </row>
- * </tbody>
- * </tgroup>
- * </informaltable>
- * </para>
- * 
- * <para>For example, the following code informs the pipeline (and in turn, the
- * RTPDTMFSrc element inside the pipeline) about the start of an RTP DTMF named
- * event '1' of volume -25 dBm0:
- * </para>
- * 
- * <para>
- * <programlisting>
- * structure = gst_structure_new ("dtmf-event",
- *                    "type", G_TYPE_INT, 1,
- *                    "number", G_TYPE_INT, 1,
- *                    "volume", G_TYPE_INT, 25,
- *                    "start", G_TYPE_BOOLEAN, TRUE, NULL);
- * 
- * event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure);
- * gst_element_send_event (pipeline, event);
- * </programlisting>
- * </para>
- *
- * </refsect2>
- */
-
-#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  50 /* ms */
-#define MIN_PACKET_INTERVAL      10 /* ms */
-#define MAX_PACKET_INTERVAL      50 /* ms */
-#define DEFAULT_SSRC             -1
-#define DEFAULT_PT               96
-#define DEFAULT_TIMESTAMP_OFFSET -1
-#define DEFAULT_SEQNUM_OFFSET    -1
-#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
-#define MIN_EVENT_DURATION       50
-
-#define DEFAULT_PACKET_REDUNDANCY 1
-#define MIN_PACKET_REDUNDANCY 1
-#define MAX_PACKET_REDUNDANCY 5
-
-/* 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_TIMESTAMP_OFFSET,
-  PROP_SEQNUM_OFFSET,
-  PROP_PT,
-  PROP_CLOCK_RATE,
-  PROP_TIMESTAMP,
-  PROP_SEQNUM,
-  PROP_INTERVAL,
-  PROP_REDUNDANCY
-};
-
-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 ], " 
-        "events = (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_TIMESTAMP,
-      g_param_spec_uint ("timestamp", "Timestamp",
-          "The RTP timestamp of the last processed packet",
-          0, G_MAXUINT, 0, G_PARAM_READABLE));
-  g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM,
-      g_param_spec_uint ("seqnum", "Sequence number",
-          "The RTP sequence number of the last processed packet",
-          0, G_MAXUINT, 0, G_PARAM_READABLE));
-  g_object_class_install_property (G_OBJECT_CLASS (klass),
-      PROP_TIMESTAMP_OFFSET, g_param_spec_int ("timestamp-offset",
-          "Timestamp Offset",
-          "Offset to add to all outgoing timestamps (-1 = random)", -1,
-          G_MAXINT, DEFAULT_TIMESTAMP_OFFSET, G_PARAM_READWRITE));
-  g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM_OFFSET,
-      g_param_spec_int ("seqnum-offset", "Sequence number Offset",
-          "Offset to add to all outgoing seqnum (-1 = random)", -1, G_MAXINT,
-          DEFAULT_SEQNUM_OFFSET, G_PARAM_READWRITE));
-  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));
-  g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_INTERVAL,
-      g_param_spec_int ("interval", "Interval between rtp packets",
-          "Interval in ms between two rtp packets", MIN_PACKET_INTERVAL,
-          MAX_PACKET_INTERVAL, DEFAULT_PACKET_INTERVAL, G_PARAM_READWRITE));
-  g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_REDUNDANCY,
-      g_param_spec_int ("packet-redundancy", "Packet Redundancy",
-          "Number of packets to send to indicate start and stop dtmf events",
-          MIN_PACKET_REDUNDANCY, MAX_PACKET_REDUNDANCY,
-          DEFAULT_PACKET_REDUNDANCY, 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->seqnum_offset = DEFAULT_SEQNUM_OFFSET;
-  dtmfsrc->ts_offset = DEFAULT_TIMESTAMP_OFFSET;
-  dtmfsrc->pt = DEFAULT_PT;
-  dtmfsrc->clock_rate = DEFAULT_CLOCK_RATE;
-  dtmfsrc->payload = NULL;
-  dtmfsrc->interval = DEFAULT_PACKET_INTERVAL;
-  dtmfsrc->packet_redundancy = DEFAULT_PACKET_REDUNDANCY;
-  
-  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_dtmf_event (GstRTPDTMFSrc *dtmfsrc,
-        const GstStructure * event_structure)
-{
-  gint event_type;
-  gboolean start;
-  gint method;
-
-  if (!gst_structure_get_int (event_structure, "type", &event_type) ||
-          !gst_structure_get_boolean (event_structure, "start", &start) ||
-          event_type != GST_RTP_DTMF_TYPE_EVENT)
-    goto failure;
-
-  if (gst_structure_get_int (event_structure, "method", &method)) {
-    if (method != 1) {
-      goto failure;
-    }
-  }
-
-  if (start) {
-    gint event_number;
-    gint event_volume;
-
-    if (!gst_structure_get_int (event_structure, "number", &event_number) ||
-            !gst_structure_get_int (event_structure, "volume", &event_volume))
-      goto failure;
-
-    GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d",
-            event_number, event_volume);
-    gst_rtp_dtmf_src_start (dtmfsrc, event_number, event_volume);
-  }
-
-  else {
-    GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
-    gst_rtp_dtmf_src_stop (dtmfsrc);
-  }
-
-  return TRUE;
-failure:
-  return FALSE;
-}
-
-static gboolean
-gst_rtp_dtmf_src_handle_custom_upstream (GstRTPDTMFSrc *dtmfsrc,
-    GstEvent * event)
-{
-  gboolean result = FALSE;
-  const GstStructure *structure;
-
-  if (GST_STATE (dtmfsrc) != GST_STATE_PLAYING) {
-    GST_DEBUG_OBJECT (dtmfsrc, "Received event while not in PLAYING state");
-    goto ret;
-  }
-
-  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"))
-    result = gst_rtp_dtmf_src_handle_dtmf_event (dtmfsrc, structure);
-
-ret:
-  return result;
-}
-
-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:
-    {
-      result = gst_rtp_dtmf_src_handle_custom_upstream (dtmfsrc, event);
-      break;
-    }
-    /* Ideally this element should not be flushed but let's handle the event
-     * just in case it is */
-    case GST_EVENT_FLUSH_START:
-      gst_rtp_dtmf_src_stop (dtmfsrc);
-      result = TRUE;
-      break;
-    case GST_EVENT_FLUSH_STOP:
-      gst_segment_init (&dtmfsrc->segment, GST_FORMAT_UNDEFINED);
-      break;
-    case GST_EVENT_NEWSEGMENT:
-      {
-        gboolean update;
-        gdouble rate;
-        GstFormat fmt;
-        gint64 start, stop, position;
-
-        gst_event_parse_new_segment (event, &update, &rate, &fmt, &start,
-            &stop, &position);
-        gst_segment_set_newsegment (&dtmfsrc->segment, update, rate, fmt,
-            start, stop, position);
-      }
-      /* fallthrough */
-    default:
-      result = gst_pad_event_default (pad, event);
-      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_TIMESTAMP_OFFSET:
-      dtmfsrc->ts_offset = g_value_get_int (value);
-      break;
-    case PROP_SEQNUM_OFFSET:
-      dtmfsrc->seqnum_offset = g_value_get_int (value);
-      break;
-    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;
-    case PROP_INTERVAL:
-      dtmfsrc->interval = g_value_get_int (value);
-      break;
-    case PROP_REDUNDANCY:
-      dtmfsrc->packet_redundancy = g_value_get_int (value);
-      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_TIMESTAMP_OFFSET:
-      g_value_set_int (value, dtmfsrc->ts_offset);
-      break;
-    case PROP_SEQNUM_OFFSET:
-      g_value_set_int (value, dtmfsrc->seqnum_offset);
-      break;
-    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;
-    case PROP_TIMESTAMP:
-      g_value_set_uint (value, dtmfsrc->rtp_timestamp);
-      break;
-    case PROP_SEQNUM:
-      g_value_set_uint (value, dtmfsrc->seqnum);
-      break;
-    case PROP_INTERVAL:
-      g_value_set_uint (value, dtmfsrc->interval);
-      break;
-    case PROP_REDUNDANCY:
-      g_value_set_uint (value, dtmfsrc->packet_redundancy);
-      break;
-    default:
-      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
-      break;
-  }
-}
-
-static void
-gst_rtp_dtmf_src_set_stream_lock (GstRTPDTMFSrc *dtmfsrc, gboolean lock)
-{
-   GstEvent *event;
-   GstStructure *structure;
-
-   structure = gst_structure_new ("stream-lock",
-                      "lock", G_TYPE_BOOLEAN, lock, NULL);
-
-   event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_OOB, structure);
-   gst_pad_push_event (dtmfsrc->srcpad, event);
-}
-
-static void
-gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc *dtmfsrc)
-{
-  GstClock *clock;
-
-  clock = GST_ELEMENT_CLOCK (dtmfsrc);
-  if (clock != NULL)
-    dtmfsrc->timestamp = gst_clock_get_time (GST_ELEMENT_CLOCK (dtmfsrc));
-
-  else {
-    GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s",
-        GST_ELEMENT_NAME (dtmfsrc));
-    dtmfsrc->timestamp = GST_CLOCK_TIME_NONE;
-  }
-
-  dtmfsrc->rtp_timestamp = dtmfsrc->ts_base +
-      gst_util_uint64_scale_int (
-          gst_segment_to_running_time (&dtmfsrc->segment, GST_FORMAT_TIME,
-              dtmfsrc->timestamp),
-          dtmfsrc->clock_rate, GST_SECOND);
-}
-
-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->first_packet = TRUE;
-  dtmfsrc->last_packet = FALSE;
-
-  gst_rtp_dtmf_prepare_timestamps (dtmfsrc);
-  gst_rtp_dtmf_src_set_caps (dtmfsrc);
-
-  /* Don't forget to get exclusive access to the stream */
-  gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
-  
-  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);
-
-  /* Push the last packet with e-bit set */
-  /* Next packet sent will be the last */
-  dtmfsrc->last_packet = TRUE;
-
-}
-
-static void
-gst_rtp_dtmf_src_wait_for_buffer_ts (GstRTPDTMFSrc *dtmfsrc, GstBuffer * buf)
-{
-  GstClock *clock;
-  
-  clock = GST_ELEMENT_CLOCK (dtmfsrc);
-  if (clock != NULL) {
-    GstClockID clock_id;
-    GstClockReturn clock_ret;
-
-    clock_id = gst_clock_new_single_shot_id (clock, GST_BUFFER_TIMESTAMP (buf));
-    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));
-  }
-}
-
-static void
-gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
-{
-  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;
-  } else if (dtmfsrc->last_packet) {
-    dtmfsrc->payload->e = 1;
-    dtmfsrc->last_packet = FALSE;
-  }
-
-  dtmfsrc->seqnum++;
-  gst_rtp_buffer_set_seq (buf, dtmfsrc->seqnum);
-  
-  /* timestamp of RTP header */
-  gst_rtp_buffer_set_timestamp (buf, dtmfsrc->rtp_timestamp);
-}
-
-static void
-gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
-{
-  GstRTPDTMFPayload *payload;
-  
-  gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc, buf);
-
-  /* duration of DTMF payload */
-  dtmfsrc->payload->duration +=
-      dtmfsrc->interval * dtmfsrc->clock_rate / 1000;
-
-  /* timestamp and duration of GstBuffer */ 
-  GST_BUFFER_DURATION (buf) = dtmfsrc->interval * GST_MSECOND;
-  GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp;
-  dtmfsrc->timestamp += GST_BUFFER_DURATION (buf);
-  
-  payload = (GstRTPDTMFPayload *) gst_rtp_buffer_get_payload (buf);
-  
-  /* copy payload and convert to network-byte order */
-  g_memmove (payload, dtmfsrc->payload, sizeof (GstRTPDTMFPayload));
-  /* Force the packet duration to a certain minumum
-   * if its the end of the event
-   */
-  if (payload->e &&
-      payload->duration < MIN_EVENT_DURATION * dtmfsrc->clock_rate / 1000)
-    payload->duration = MIN_EVENT_DURATION * dtmfsrc->clock_rate / 1000;
-
-  payload->duration = g_htons (payload->duration);
-}
-
-static GstBuffer *
-gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
-{
-  GstBuffer *buf = NULL;
-  
-  /* create buffer to hold the payload */
-  buf = gst_rtp_buffer_new_allocate (sizeof (GstRTPDTMFPayload), 0, 0);
-
-  gst_rtp_dtmf_prepare_buffer_data (dtmfsrc, buf);
-
-  /* FIXME: Should we sync to clock ourselves or leave it to sink */
-  gst_rtp_dtmf_src_wait_for_buffer_ts (dtmfsrc, buf);
-
-  /* Set caps on the buffer before pushing it */
-  gst_buffer_set_caps (buf, GST_PAD_CAPS (dtmfsrc->srcpad));
-
-  return buf;
-}
-
-static void
-gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
-{
-  GstBuffer *buf = NULL;
-  GstFlowReturn ret;
-  gint redundancy_count = 1;
-
-  if (dtmfsrc->first_packet == TRUE || dtmfsrc->last_packet == TRUE) {
-    redundancy_count = dtmfsrc->packet_redundancy;
-
-    if(dtmfsrc->first_packet == TRUE) {
-      GST_DEBUG_OBJECT (dtmfsrc,
-          "redundancy count set to %d due to dtmf start",
-          redundancy_count);
-    } else if(dtmfsrc->last_packet == TRUE) {
-      GST_DEBUG_OBJECT (dtmfsrc,
-          "redundancy count set to %d due to dtmf stop",
-          redundancy_count);
-    }
-
-  }
-
-  /* create buffer to hold the payload */
-  buf = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc);
-
-  while ( redundancy_count-- ) {
-    gst_buffer_ref(buf);
-
-    GST_DEBUG_OBJECT (dtmfsrc,
-        "pushing buffer on src pad of size %d with redundancy count %d",
-        GST_BUFFER_SIZE (buf), redundancy_count);
-    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));
-
-    /* Make sure only the first packet sent has the marker set */
-    gst_rtp_buffer_set_marker (buf, FALSE);
-  }
-
-  gst_buffer_unref(buf);
-  GST_DEBUG_OBJECT (dtmfsrc,
-          "pushed DTMF event '%d' on src pad", dtmfsrc->payload->event);
-
-  if (dtmfsrc->payload->e) {
-    /* Don't forget to release the stream lock */
-    gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
-
-    g_free (dtmfsrc->payload);
-    dtmfsrc->payload = NULL;
-
-    if (!gst_pad_pause_task (dtmfsrc->srcpad)) {
-      GST_ERROR_OBJECT (dtmfsrc, "Failed to pause task on src pad");
-      return;
-    }
-
-  }
-
-}
-
-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,
-      "clock-base", G_TYPE_UINT, dtmfsrc->ts_base,
-      "seqnum-base", G_TYPE_UINT, dtmfsrc->seqnum_base, NULL);
-
-  if (!gst_pad_set_caps (dtmfsrc->srcpad, caps))
-    GST_ERROR_OBJECT (dtmfsrc,
-            "Failed to set caps %" GST_PTR_FORMAT " on src pad", caps);
-  else
-    GST_DEBUG_OBJECT (dtmfsrc,
-            "caps %" GST_PTR_FORMAT " set on src pad", caps);
-
-  gst_caps_unref (caps);
-}
-
-static void
-gst_rtp_dtmf_src_ready_to_paused (GstRTPDTMFSrc *dtmfsrc)
-{
-  gst_segment_init (&dtmfsrc->segment, GST_FORMAT_UNDEFINED);
-  
-  if (dtmfsrc->ssrc == -1)
-    dtmfsrc->current_ssrc = g_random_int ();
-  else
-    dtmfsrc->current_ssrc = dtmfsrc->ssrc;
-
-  if (dtmfsrc->seqnum_offset == -1)
-    dtmfsrc->seqnum_base = g_random_int_range (0, G_MAXUINT16);
-  else
-    dtmfsrc->seqnum_base = dtmfsrc->seqnum_offset;
-  dtmfsrc->seqnum = dtmfsrc->seqnum_base;
-  
-  if (dtmfsrc->ts_offset == -1)
-    dtmfsrc->ts_base = g_random_int ();
-  else
-    dtmfsrc->ts_base = dtmfsrc->ts_offset;
-}
-
-static GstStateChangeReturn
-gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition)
-{
-  GstRTPDTMFSrc *dtmfsrc;
-  GstStateChangeReturn result;
-  gboolean no_preroll = FALSE;
-
-  dtmfsrc = GST_RTP_DTMF_SRC (element);
-
-  switch (transition) {
-    case GST_STATE_CHANGE_READY_TO_PAUSED:
-      gst_rtp_dtmf_src_ready_to_paused (dtmfsrc);
-      /* Indicate that we don't do PRE_ROLL */
-      no_preroll = TRUE;
-      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;
-    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;
-}
-
-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
deleted file mode 100644
index 797526ea..00000000
--- a/gst/rtpdtmf/gstrtpdtmfsrc.h
+++ /dev/null
@@ -1,101 +0,0 @@
-/* GStreamer RTP DTMF source
- *
- * gstrtpdtmfsrc.h:
- *
- * Copyright (C) <2007> Nokia Corporation.
- *   Contact: Zeeshan Ali <zeeshan.ali@nokia.com>
- * 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_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;
-
-  guint32           ts_base;
-  guint16           seqnum_base;
-
-  gint16            seqnum_offset;
-  guint16           seqnum;
-  gint32            ts_offset;
-  guint32           rtp_timestamp;
-  guint32           clock_rate;
-  guint             pt;
-  guint             ssrc;
-  guint             current_ssrc;
-  gboolean          first_packet;
-  gboolean          last_packet;
-  
-  GstClockTime      timestamp;
-  GstSegment        segment;
-
-  guint16	    interval;
-  guint16	    packet_redundancy;
-};
-
-struct _GstRTPDTMFSrcClass {
-  GstElementClass parent_class;
-};
-
-GType gst_rtp_dtmf_src_get_type (void);
-
-G_END_DECLS
-
-#endif /* __GST_RTP_DTMF_SRC_H__ */