gstreamer/gst/dtmf/gstrtpdtmfdepay.c

/* GstRtpDtmfDepay
 *
 * Copyright (C) <2008> Collabora.
 * Copyright (C) <2008> Nokia.
 *   Contact: Youness Alaoui <youness.alaoui@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.
 */

#ifdef HAVE_CONFIG_H
#  include "config.h"
#endif

#include <string.h>
#include <math.h>

#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpdtmfdepay.h"

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


#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
#define MIN_INTER_DIGIT_INTERVAL 100
#define MIN_PULSE_DURATION       250
#define MIN_DUTY_CYCLE           (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION)


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_rtp_dtmfdepay_details =
GST_ELEMENT_DETAILS ("RTP DTMF packet depayloader",
    "Codec/Depayloader/Network",
    "Generates DTMF Sound from telephone-event RTP packets",
    "Youness Alaoui <youness.alaoui@collabora.co.uk>");

GST_DEBUG_CATEGORY_STATIC (gst_rtp_dtmf_depay_debug);
#define GST_CAT_DEFAULT gst_rtp_dtmf_depay_debug

enum
{
  /* FILL ME */
  LAST_SIGNAL
};

enum
{
  ARG_0
};

static GstStaticPadTemplate gst_rtp_dtmf_depay_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 = (boolean) true, "
        "rate = (int) [0, MAX], "
        "channels = (int) 1")
    );

static GstStaticPadTemplate gst_rtp_dtmf_depay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
    GST_PAD_SINK,
    GST_PAD_ALWAYS,
    GST_STATIC_CAPS ("application/x-rtp, "
        "media = (string) \"audio\", "
        "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
        "clock-rate = (int) [ 0, MAX ], "
        "encoding-name = (string) \"TELEPHONE-EVENT\"")
    );

GST_BOILERPLATE (GstRtpDTMFDepay, gst_rtp_dtmf_depay, GstBaseRTPDepayload,
    GST_TYPE_BASE_RTP_DEPAYLOAD);


static GstBuffer *gst_rtp_dtmf_depay_process (GstBaseRTPDepayload * depayload,
    GstBuffer * buf);
gboolean gst_rtp_dtmf_depay_setcaps (GstBaseRTPDepayload * filter,
    GstCaps * caps);

/*static void
gst_rtp_dtmf_depay_set_gst_timestamp (GstBaseRTPDepayload * filter,
    guint32 rtptime, GstBuffer * buf);
*/

static void
gst_rtp_dtmf_depay_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_dtmf_depay_src_template));
  gst_element_class_add_pad_template (element_class,
      gst_static_pad_template_get (&gst_rtp_dtmf_depay_sink_template));


  GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_depay_debug,
          "rtpdtmfdepay", 0, "rtpdtmfdepay element");
  gst_element_class_set_details (element_class, &gst_rtp_dtmfdepay_details);
}

static void
gst_rtp_dtmf_depay_class_init (GstRtpDTMFDepayClass * klass)
{
  GObjectClass *gobject_class;
  GstElementClass *gstelement_class;
  GstBaseRTPDepayloadClass *gstbasertpdepayload_class;

  gobject_class = (GObjectClass *) klass;
  gstelement_class = (GstElementClass *) klass;
  gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;

  parent_class = g_type_class_peek_parent (klass);

  gstbasertpdepayload_class->process = gst_rtp_dtmf_depay_process;
  gstbasertpdepayload_class->set_caps = gst_rtp_dtmf_depay_setcaps;
  //  gstbasertpdepayload_class->set_gst_timestamp = gst_rtp_dtmf_depay_set_gst_timestamp;

}

static void
gst_rtp_dtmf_depay_init (GstRtpDTMFDepay * rtpdtmfdepay,
    GstRtpDTMFDepayClass * klass)
{

}


gboolean
gst_rtp_dtmf_depay_setcaps (GstBaseRTPDepayload * filter, GstCaps * caps)
{
  GstCaps *srccaps;
  GstStructure *structure = gst_caps_get_structure (caps, 0);
  gint clock_rate = 8000;      /* default */

  gst_structure_get_int (structure, "clock-rate", &clock_rate);
  filter->clock_rate = clock_rate;

  srccaps = gst_caps_new_simple ("audio/x-raw-int",
      "width", G_TYPE_INT, 16,
      "depth", G_TYPE_INT, 16,
      "endianness", G_TYPE_INT, 1234,
      "signed", G_TYPE_BOOLEAN, TRUE,
      "channels", G_TYPE_INT, 1,
      "rate", G_TYPE_INT, clock_rate, NULL);
  gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (filter), srccaps);
  gst_caps_unref (srccaps);

  return TRUE;
}

void
gst_rtp_dtmf_depay_set_gst_timestamp (GstBaseRTPDepayload * filter,
    guint32 rtptime, GstBuffer * buf)
{
  GstClockTime timestamp, duration;


  timestamp = GST_BUFFER_TIMESTAMP (buf);
  duration = GST_BUFFER_DURATION (buf);

  /* if this is the first buffer send a NEWSEGMENT */
  if (filter->need_newsegment) {
    GstEvent *event;
    GstClockTime stop, position;

    stop = -1;

    position = 0;

    event =
        gst_event_new_new_segment_full (FALSE, 1.0,
        1.0, GST_FORMAT_TIME, 0, stop, position);

    gst_pad_push_event (filter->srcpad, event);

    filter->need_newsegment = FALSE;
    GST_DEBUG_OBJECT (filter, "Pushed newsegment event on this first buffer");
  }
}

#if 0
static void
gst_dtmf_src_generate_silence(GstBuffer * buffer, float duration)
{
  gint buf_size;

  /* Create a buffer with data set to 0 */
  buf_size = ((duration/1000)*SAMPLE_RATE*SAMPLE_SIZE*CHANNELS)/8;
  GST_BUFFER_SIZE (buffer) = buf_size;
  GST_BUFFER_MALLOCDATA (buffer) = g_malloc0(buf_size);
  GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);

}
#endif

static void
gst_dtmf_src_generate_tone(GstRtpDTMFDepay *rtpdtmfdepay,
    GstRTPDTMFPayload payload, GstBuffer * buffer)
{
  gint16 *p;
  gint tone_size;
  double i = 0;
  double amplitude, f1, f2;
  double volume_factor;
  DTMF_KEY key = DTMF_KEYS[payload.event];
  guint32 clock_rate = 8000 /* default */;
  GstBaseRTPDepayload * depayload = GST_BASE_RTP_DEPAYLOAD (rtpdtmfdepay);

  clock_rate = depayload->clock_rate;

  /* Create a buffer for the tone */
  tone_size = (payload.duration*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);
  GST_BUFFER_DURATION (buffer) = payload.duration * GST_SECOND / clock_rate;

  p = (gint16 *) GST_BUFFER_MALLOCDATA (buffer);

  volume_factor = pow (10, (-payload.volume) / 20);

  /*
   * 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 * (rtpdtmfdepay->sample / clock_rate));
    f2 = sin(2 * M_PI * key.high_frequency * (rtpdtmfdepay->sample / clock_rate));

    amplitude = (f1 + f2) / 2;

    /* Adjust the volume */
    amplitude *= volume_factor;

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

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

    (rtpdtmfdepay->sample)++;
  }
}


static GstBuffer *
gst_rtp_dtmf_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
{

  GstRtpDTMFDepay *rtpdtmfdepay = NULL;
  GstBuffer *outbuf = NULL;
  gint payload_len;
  guint8 *payload = NULL;
  guint32 timestamp;
  GstRTPDTMFPayload dtmf_payload;
  gboolean marker;
  GstStructure *structure = NULL;
  GstMessage *dtmf_message = NULL;

  rtpdtmfdepay = GST_RTP_DTMF_DEPAY (depayload);

  if (!gst_rtp_buffer_validate (buf))
    goto bad_packet;

  payload_len = gst_rtp_buffer_get_payload_len (buf);
  payload = gst_rtp_buffer_get_payload (buf);

  if (payload_len != sizeof(GstRTPDTMFPayload) )
    goto bad_packet;

  memcpy (&dtmf_payload, payload, sizeof (GstRTPDTMFPayload));

  if (dtmf_payload.event > MAX_EVENT)
    goto bad_packet;


  marker = gst_rtp_buffer_get_marker (buf);

  timestamp = gst_rtp_buffer_get_timestamp (buf);

  dtmf_payload.duration = g_ntohs (dtmf_payload.duration);

  GST_DEBUG_OBJECT (depayload, "Received new RTP DTMF packet : "
      "marker=%d - timestamp=%u - event=%d - duration=%d",
      marker, timestamp, dtmf_payload.event, dtmf_payload.duration);

  GST_DEBUG_OBJECT (depayload, "Previous information : timestamp=%u - duration=%d",
      rtpdtmfdepay->previous_ts, rtpdtmfdepay->previous_duration);

  /* First packet */
  if (marker || rtpdtmfdepay->previous_ts != timestamp) {
    rtpdtmfdepay->sample = 0;
    rtpdtmfdepay->previous_ts = timestamp;
    rtpdtmfdepay->previous_duration = dtmf_payload.duration;
    rtpdtmfdepay->first_gst_ts = GST_BUFFER_TIMESTAMP (buf);

    structure = gst_structure_new ("dtmf-event",
        "number", G_TYPE_INT, dtmf_payload.event,
        "volume", G_TYPE_INT, dtmf_payload.volume,
        "type", G_TYPE_INT, 1,
        "method", G_TYPE_INT, 1,
        NULL);
    if (structure) {
      dtmf_message = gst_message_new_element (GST_OBJECT (depayload), structure);
      if (dtmf_message) {
        if (!gst_element_post_message (GST_ELEMENT (depayload), dtmf_message)) {
          GST_DEBUG_OBJECT (depayload, "Unable to send dtmf-event message to bus");
        }
      } else {
        GST_DEBUG_OBJECT (depayload, "Unable to create dtmf-event message");
      }
    } else {
      GST_DEBUG_OBJECT (depayload, "Unable to create dtmf-event structure");
    }
  } else {
    guint16 duration = dtmf_payload.duration;
    dtmf_payload.duration -= rtpdtmfdepay->previous_duration;
    /* If late buffer, ignore */
    if (duration > rtpdtmfdepay->previous_duration)
      rtpdtmfdepay->previous_duration = duration;
  }

  GST_DEBUG_OBJECT (depayload, "new previous duration : %d - new duration : %d"
      " - diff  : %d - clock rate : %d - timestamp : %llu",
      rtpdtmfdepay->previous_duration, dtmf_payload.duration,
      (rtpdtmfdepay->previous_duration - dtmf_payload.duration),
      depayload->clock_rate, GST_BUFFER_TIMESTAMP (buf));

  /* If late or duplicate packet (like the redundant end packet). Ignore */
  if (dtmf_payload.duration > 0) {
    outbuf = gst_buffer_new ();
    gst_dtmf_src_generate_tone(rtpdtmfdepay, dtmf_payload, outbuf);


    GST_BUFFER_TIMESTAMP (outbuf) = rtpdtmfdepay->first_gst_ts +
        (rtpdtmfdepay->previous_duration - dtmf_payload.duration) *
        GST_SECOND / depayload->clock_rate;
    GST_BUFFER_OFFSET (outbuf) =
        (rtpdtmfdepay->previous_duration - dtmf_payload.duration) *
          GST_SECOND / depayload->clock_rate;
    GST_BUFFER_OFFSET_END (outbuf) = rtpdtmfdepay->previous_duration *
          GST_SECOND / depayload->clock_rate;

    GST_DEBUG_OBJECT (depayload, "timestamp : %llu - time %" GST_TIME_FORMAT,
        GST_BUFFER_TIMESTAMP (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));

  }

  return outbuf;


bad_packet:
  GST_ELEMENT_WARNING (rtpdtmfdepay, STREAM, DECODE,
      ("Packet did not validate"), (NULL));
  return NULL;
}

gboolean
gst_rtp_dtmf_depay_plugin_init (GstPlugin * plugin)
{
  return gst_element_register (plugin, "rtpdtmfdepay",
      GST_RANK_MARGINAL, GST_TYPE_RTP_DTMF_DEPAY);
}