gstreamer/gst/dtmf/gstdtmfsrc.c

/* GStreamer DTMF source
 *
 * gstdtmfsrc.c:
 *
 * Copyright (C) <2007> Collabora.
 *   Contact: Youness Alaoui <youness.alaoui@collabora.co.uk>
 * 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-dtmfsrc
 * @short_description: Generates DTMF packets
 *
 * <refsect2>
 *
 * <para>
 * The DTMFSrc element generates DTMF (ITU-T Q.23 Specification) tone 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 tones.
 * </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 2.
 * </entry>
 * </row>
 * </tbody>
 * </tgroup>
 * </informaltable>
 * </para>
 *
 * <para>For example, the following code informs the pipeline (and in turn, the
 * DTMFSrc element inside the pipeline) about the start of a DTMF named
 * event '1' of volume -25 dBm0:
 * </para>
 *
 * <para>
 * <programlisting>
 * structure = gst_structure_new ("dtmf-event",
 *                    "type", G_TYPE_INT, 0,
 *                    "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 <math.h>

#ifndef M_PI
# define M_PI           3.14159265358979323846  /* pi */
#endif


#include "gstdtmfsrc.h"

#define GST_TONE_DTMF_TYPE_EVENT 0
#define DEFAULT_PACKET_INTERVAL  50 /* ms */
#define MIN_PACKET_INTERVAL      10 /* ms */
#define MAX_PACKET_INTERVAL      50 /* ms */
#define SAMPLE_RATE              8000
#define SAMPLE_SIZE              16
#define CHANNELS                 1
#define MIN_EVENT                0
#define MAX_EVENT                16
#define MIN_VOLUME               0
#define MAX_VOLUME               36



typedef struct st_dtmf_key {
        char *event_name;
        int event_encoding;
        float low_frequency;
        float high_frequency;
} DTMF_KEY;

static const DTMF_KEY DTMF_KEYS[] = {
        {"DTMF_KEY_EVENT_0",  0, 941, 1336},
        {"DTMF_KEY_EVENT_1",  1, 697, 1209},
        {"DTMF_KEY_EVENT_2",  2, 697, 1336},
        {"DTMF_KEY_EVENT_3",  3, 697, 1477},
        {"DTMF_KEY_EVENT_4",  4, 770, 1209},
        {"DTMF_KEY_EVENT_5",  5, 770, 1336},
        {"DTMF_KEY_EVENT_6",  6, 770, 1477},
        {"DTMF_KEY_EVENT_7",  7, 852, 1209},
        {"DTMF_KEY_EVENT_8",  8, 852, 1336},
        {"DTMF_KEY_EVENT_9",  9, 852, 1477},
        {"DTMF_KEY_EVENT_S", 10, 941, 1209},
        {"DTMF_KEY_EVENT_P", 11, 941, 1477},
        {"DTMF_KEY_EVENT_A", 12, 697, 1633},
        {"DTMF_KEY_EVENT_B", 13, 770, 1633},
        {"DTMF_KEY_EVENT_C", 14, 852, 1633},
        {"DTMF_KEY_EVENT_D", 15, 941, 1633},
};

#define MAX_DTMF_EVENTS 16

enum {
DTMF_KEY_EVENT_1 = 1,
DTMF_KEY_EVENT_2 = 2,
DTMF_KEY_EVENT_3 = 3,
DTMF_KEY_EVENT_4 = 4,
DTMF_KEY_EVENT_5 = 5,
DTMF_KEY_EVENT_6 = 6,
DTMF_KEY_EVENT_7 = 7,
DTMF_KEY_EVENT_8 = 8,
DTMF_KEY_EVENT_9 = 9,
DTMF_KEY_EVENT_0 = 0,
DTMF_KEY_EVENT_STAR = 10,
DTMF_KEY_EVENT_POUND = 11,
DTMF_KEY_EVENT_A = 12,
DTMF_KEY_EVENT_B = 13,
DTMF_KEY_EVENT_C = 14,
DTMF_KEY_EVENT_D = 15,
};

/* elementfactory information */
static const GstElementDetails gst_dtmf_src_details =
GST_ELEMENT_DETAILS ("DTMF tone generator",
    "Source/Audio",
    "Generates DTMF tones",
    "Youness Alaoui <youness.alaoui@collabora.co.uk>");

GST_DEBUG_CATEGORY_STATIC (gst_dtmf_src_debug);
#define GST_CAT_DEFAULT gst_dtmf_src_debug

enum
{
  PROP_0,
  PROP_INTERVAL,
};

static GstStaticPadTemplate gst_dtmf_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
    GST_PAD_SRC,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("audio/x-raw-int, "
        "width = (int) 16, "
        "depth = (int) 16, "
	"endianness = (int) 1234, "
	"signed = (bool) true, "
        "rate = (int) 8000, "
	"channels = (int) 1")
    );

static GstElementClass *parent_class = NULL;

static void gst_dtmf_src_base_init (gpointer g_class);
static void gst_dtmf_src_class_init (GstDTMFSrcClass * klass);
static void gst_dtmf_src_init (GstDTMFSrc * dtmfsrc, gpointer g_class);
static void gst_dtmf_src_finalize (GObject * object);

GType
gst_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 (GstDTMFSrcClass),
      (GBaseInitFunc) gst_dtmf_src_base_init,
      NULL,
      (GClassInitFunc) gst_dtmf_src_class_init,
      NULL,
      NULL,
      sizeof (GstDTMFSrc),
      0,
      (GInstanceInitFunc) gst_dtmf_src_init,
    };

    base_src_type = g_type_register_static (GST_TYPE_ELEMENT,
        "GstDTMFSrc", &base_src_info, 0);
  }
  return base_src_type;
}

static void gst_dtmf_src_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec);
static void gst_dtmf_src_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec);
static gboolean gst_dtmf_src_handle_event (GstPad * pad, GstEvent * event);
static GstStateChangeReturn gst_dtmf_src_change_state (GstElement * element,
    GstStateChange transition);
static void gst_dtmf_src_generate_tone(GstDTMFSrc *dtmfsrc, DTMF_KEY key, float duration,
    GstBuffer * buffer);
static void gst_dtmf_src_push_next_tone_packet (GstDTMFSrc *dtmfsrc);
static void gst_dtmf_src_start (GstDTMFSrc *dtmfsrc, gint event_number,
    gint event_volume);
static void gst_dtmf_src_stop (GstDTMFSrc *dtmfsrc);

static void
gst_dtmf_src_base_init (gpointer g_class)
{
  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);

  GST_DEBUG_CATEGORY_INIT (gst_dtmf_src_debug,
          "dtmfsrc", 0, "dtmfsrc element");

  gst_element_class_add_pad_template (element_class,
      gst_static_pad_template_get (&gst_dtmf_src_template));

  gst_element_class_set_details (element_class, &gst_dtmf_src_details);
}

static void
gst_dtmf_src_class_init (GstDTMFSrcClass * 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_dtmf_src_finalize);
  gobject_class->set_property =
      GST_DEBUG_FUNCPTR (gst_dtmf_src_set_property);
  gobject_class->get_property =
      GST_DEBUG_FUNCPTR (gst_dtmf_src_get_property);

  g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_INTERVAL,
      g_param_spec_int ("interval", "Interval between tone packets",
          "Interval in ms between two tone packets", MIN_PACKET_INTERVAL,
          MAX_PACKET_INTERVAL, DEFAULT_PACKET_INTERVAL, G_PARAM_READWRITE));

  gstelement_class->change_state =
      GST_DEBUG_FUNCPTR (gst_dtmf_src_change_state);
}

static void
gst_dtmf_src_init (GstDTMFSrc * dtmfsrc, gpointer g_class)
{
  dtmfsrc->srcpad =
      gst_pad_new_from_static_template (&gst_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_dtmf_src_handle_event);

  dtmfsrc->interval = DEFAULT_PACKET_INTERVAL;

  dtmfsrc->sample = 0;

  GST_DEBUG_OBJECT (dtmfsrc, "init done");
}

static void
gst_dtmf_src_finalize (GObject * object)
{
  GstDTMFSrc *dtmfsrc;

  dtmfsrc = GST_DTMF_SRC (object);

  G_OBJECT_CLASS (parent_class)->finalize (object);
}

static gboolean
gst_dtmf_src_handle_dtmf_event (GstDTMFSrc *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) ||
      (start == TRUE && event_type != GST_TONE_DTMF_TYPE_EVENT))
    goto failure;

  if (gst_structure_get_int (event_structure, "method", &method)) {
    if (method != 2) {
      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_dtmf_src_start (dtmfsrc, event_number, event_volume);
  }

  else {
    GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
    gst_dtmf_src_stop (dtmfsrc);
  }

  return TRUE;
failure:
  return FALSE;
}

static gboolean
gst_dtmf_src_handle_custom_upstream (GstDTMFSrc *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_dtmf_src_handle_dtmf_event (dtmfsrc, structure);

ret:
  return result;
}

static gboolean
gst_dtmf_src_handle_event (GstPad * pad, GstEvent * event)
{
  GstDTMFSrc *dtmfsrc;
  gboolean result = FALSE;

  dtmfsrc = GST_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_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_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_dtmf_src_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec)
{
  GstDTMFSrc *dtmfsrc;

  dtmfsrc = GST_DTMF_SRC (object);

  switch (prop_id) {
    case PROP_INTERVAL:
      dtmfsrc->interval = g_value_get_int (value);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
gst_dtmf_src_get_property (GObject * object, guint prop_id, GValue * value,
    GParamSpec * pspec)
{
  GstDTMFSrc *dtmfsrc;

  dtmfsrc = GST_DTMF_SRC (object);

  switch (prop_id) {
    case PROP_INTERVAL:
      g_value_set_uint (value, dtmfsrc->interval);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
gst_dtmf_src_set_stream_lock (GstDTMFSrc *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_dtmf_prepare_timestamps (GstDTMFSrc *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;
  }
}

static void
gst_dtmf_src_start (GstDTMFSrc *dtmfsrc,
        gint event_number, gint event_volume)
{
  GstCaps * caps = gst_pad_get_pad_template_caps (dtmfsrc->srcpad);

  dtmfsrc->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
  dtmfsrc->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);

  gst_dtmf_prepare_timestamps (dtmfsrc);

  /* Don't forget to get exclusive access to the stream */
  gst_dtmf_src_set_stream_lock (dtmfsrc, TRUE);

  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);


  if (!gst_pad_start_task (dtmfsrc->srcpad,
      (GstTaskFunction) gst_dtmf_src_push_next_tone_packet, dtmfsrc)) {
    GST_ERROR_OBJECT (dtmfsrc, "Failed to start task on src pad");
  }
}

static void
gst_dtmf_src_stop (GstDTMFSrc *dtmfsrc)
{
  /* Don't forget to release the stream lock */
  gst_dtmf_src_set_stream_lock (dtmfsrc, FALSE);

  if (!gst_pad_pause_task (dtmfsrc->srcpad)) {
    GST_ERROR_OBJECT (dtmfsrc, "Failed to pause task on src pad");
    return;
  }

}

static void
gst_dtmf_src_generate_tone(GstDTMFSrc *dtmfsrc, DTMF_KEY key, float duration, GstBuffer * buffer)
{
  gint16 *p;
  gint tone_size;
  double i = 0;
  double amplitude, f1, f2;

  /* Create a buffer for the tone */
  tone_size = ((duration/1000)*SAMPLE_RATE*SAMPLE_SIZE*CHANNELS)/8;
  GST_BUFFER_SIZE (buffer) = tone_size;
  GST_BUFFER_MALLOCDATA (buffer) = g_malloc(tone_size);
  GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);

  p = (gint16 *) GST_BUFFER_MALLOCDATA (buffer);

  /*
   * For each sample point we calculate 'x' as the
   * the amplitude value.
   */
  for (i = 0; i < (tone_size / (SAMPLE_SIZE/8)); i++) {
    /*
     * We add the fundamental frequencies together.
     */
    f1 = sin(2 * M_PI * key.low_frequency * (dtmfsrc->sample / SAMPLE_RATE));
    f2 = sin(2 * M_PI * key.high_frequency * (dtmfsrc->sample / SAMPLE_RATE));

    amplitude = (f1 + f2) / 2;

    /* Make the [-1:1] interval into a [-32767:32767] interval */
    amplitude *= 32767;

    /* Store it in the data buffer */
    *(p++) = (gint16) amplitude;

    (dtmfsrc->sample)++;
  }
}

static void
gst_dtmf_src_wait_for_buffer_ts (GstDTMFSrc *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 GstBuffer *
gst_dtmf_src_create_next_tone_packet (GstDTMFSrc *dtmfsrc)
{
  GstBuffer *buf = NULL;


  GST_DEBUG_OBJECT (dtmfsrc,
      "Creating buffer for tone");

  /* create buffer to hold the tone */
  buf = gst_buffer_new ();

  /* Generate the tone */
  gst_dtmf_src_generate_tone(dtmfsrc, DTMF_KEYS[dtmfsrc->event], dtmfsrc->interval, buf);


  /* 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) /2;

  /* FIXME: Should we sync to clock ourselves or leave it to sink */
  gst_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_dtmf_src_push_next_tone_packet (GstDTMFSrc *dtmfsrc)
{
  GstBuffer *buf = NULL;
  GstFlowReturn ret;

  buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc);

  gst_buffer_ref(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_buffer_unref(buf);
  GST_DEBUG_OBJECT (dtmfsrc, "pushed DTMF tone on src pad");

}

static GstStateChangeReturn
gst_dtmf_src_change_state (GstElement * element, GstStateChange transition)
{
  GstDTMFSrc *dtmfsrc;
  GstStateChangeReturn result;
  gboolean no_preroll = FALSE;

  dtmfsrc = GST_DTMF_SRC (element);

  switch (transition) {
    case GST_STATE_CHANGE_READY_TO_PAUSED:
      gst_segment_init (&dtmfsrc->segment, GST_FORMAT_UNDEFINED);
      /* Indicate that we don't do PRE_ROLL */
      no_preroll = TRUE;
      break;
    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
      /* gst_dtmf_src_start (dtmfsrc, 6, 30); */
      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 */
      /* gst_dtmf_src_stop (dtmfsrc); */
      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;
  }
}

gboolean
gst_dtmf_src_plugin_init (GstPlugin * plugin)
{
  return gst_element_register (plugin, "dtmfsrc",
      GST_RANK_NONE, GST_TYPE_DTMF_SRC);
}