/* GstRtpDtmfDepay
*
* Copyright (C) 2008 Collabora Limited
* Copyright (C) 2008 Nokia Corporation
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-rtpdtmfdepay
* @see_also: rtpdtmfsrc, rtpdtmfmux
*
* This element takes RTP DTMF packets and produces sound. It also emits a
* message on the #GstBus.
*
* The message is called "dtmf-event" and has the following fields
* <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>Which of the two methods
* specified in RFC 2833 to use. The value should be 0 for tones and 1 for
* named events. Tones are specified by their frequencies and events are specied
* by their number. This element currently only recognizes events.
* Do not confuse with "method" which specified the output.
* </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.
* </entry>
* </row>
* <row>
* <entry>method</entry>
* <entry>G_TYPE_INT</entry>
* <entry>1</entry>
* <entry>This field will always been 1 (ie RTP event) from this element.
* </entry>
* </row>
* </tbody>
* </tgroup>
* </informaltable>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstrtpdtmfdepay.h"
#include <string.h>
#include <math.h>
#include <gst/audio/audio.h>
#include <gst/rtp/gstrtpbuffer.h>
#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_DUTY_CYCLE (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION)
#define MIN_UNIT_TIME 0
#define MAX_UNIT_TIME 1000
#define DEFAULT_UNIT_TIME 0
#define DEFAULT_MAX_DURATION 0
typedef struct st_dtmf_key
{
float low_frequency;
float high_frequency;
} DTMF_KEY;
static const DTMF_KEY DTMF_KEYS[] = {
{941, 1336},
{697, 1209},
{697, 1336},
{697, 1477},
{770, 1209},
{770, 1336},
{770, 1477},
{852, 1209},
{852, 1336},
{852, 1477},
{941, 1209},
{941, 1477},
{697, 1633},
{770, 1633},
{852, 1633},
{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,
};
GST_DEBUG_CATEGORY_STATIC (gst_rtp_dtmf_depay_debug);
#define GST_CAT_DEFAULT gst_rtp_dtmf_depay_debug
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_UNIT_TIME,
PROP_MAX_DURATION
};
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, "
"format = (string) \"" GST_AUDIO_NE (S16) "\", "
"rate = " GST_AUDIO_RATE_RANGE ", " "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\"")
);
G_DEFINE_TYPE (GstRtpDTMFDepay, gst_rtp_dtmf_depay,
GST_TYPE_RTP_BASE_DEPAYLOAD);
static void gst_rtp_dtmf_depay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_dtmf_depay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstBuffer *gst_rtp_dtmf_depay_process (GstRTPBaseDepayload * depayload,
GstBuffer * buf);
gboolean gst_rtp_dtmf_depay_setcaps (GstRTPBaseDepayload * filter,
GstCaps * caps);
static void
gst_rtp_dtmf_depay_class_init (GstRtpDTMFDepayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
gobject_class = G_OBJECT_CLASS (klass);
gstelement_class = GST_ELEMENT_CLASS (klass);
gstrtpbasedepayload_class = GST_RTP_BASE_DEPAYLOAD_CLASS (klass);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_dtmf_depay_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&gst_rtp_dtmf_depay_sink_template);
GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_depay_debug,
"rtpdtmfdepay", 0, "rtpdtmfdepay element");
gst_element_class_set_static_metadata (gstelement_class,
"RTP DTMF packet depayloader", "Codec/Depayloader/Network",
"Generates DTMF Sound from telephone-event RTP packets",
"Youness Alaoui <youness.alaoui@collabora.co.uk>");
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_get_property);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_UNIT_TIME,
g_param_spec_uint ("unit-time", "Duration unittime",
"The smallest unit (ms) the duration must be a multiple of (0 disables it)",
MIN_UNIT_TIME, MAX_UNIT_TIME, DEFAULT_UNIT_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MAX_DURATION,
g_param_spec_uint ("max-duration", "Maximum duration",
"The maxumimum duration (ms) of the outgoing soundpacket. "
"(0 = no limit)", 0, G_MAXUINT, DEFAULT_MAX_DURATION,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstrtpbasedepayload_class->process =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_process);
gstrtpbasedepayload_class->set_caps =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_setcaps);
}
static void
gst_rtp_dtmf_depay_init (GstRtpDTMFDepay * rtpdtmfdepay)
{
rtpdtmfdepay->unit_time = DEFAULT_UNIT_TIME;
}
static void
gst_rtp_dtmf_depay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpDTMFDepay *rtpdtmfdepay;
rtpdtmfdepay = GST_RTP_DTMF_DEPAY (object);
switch (prop_id) {
case PROP_UNIT_TIME:
rtpdtmfdepay->unit_time = g_value_get_uint (value);
break;
case PROP_MAX_DURATION:
rtpdtmfdepay->max_duration = g_value_get_uint (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_dtmf_depay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpDTMFDepay *rtpdtmfdepay;
rtpdtmfdepay = GST_RTP_DTMF_DEPAY (object);
switch (prop_id) {
case PROP_UNIT_TIME:
g_value_set_uint (value, rtpdtmfdepay->unit_time);
break;
case PROP_MAX_DURATION:
g_value_set_uint (value, rtpdtmfdepay->max_duration);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
gboolean
gst_rtp_dtmf_depay_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps)
{
GstCaps *filtercaps, *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;
filtercaps =
gst_pad_get_pad_template_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter));
filtercaps = gst_caps_make_writable (filtercaps);
gst_caps_set_simple (filtercaps, "rate", G_TYPE_INT, clock_rate, NULL);
srccaps = gst_pad_peer_query_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter),
filtercaps);
gst_caps_unref (filtercaps);
gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter), srccaps);
gst_caps_unref (srccaps);
return TRUE;
}
static GstBuffer *
gst_dtmf_src_generate_tone (GstRtpDTMFDepay * rtpdtmfdepay,
GstRTPDTMFPayload payload)
{
GstBuffer *buf;
GstMapInfo map;
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;
GstRTPBaseDepayload *depayload = GST_RTP_BASE_DEPAYLOAD (rtpdtmfdepay);
gint volume;
static GstAllocationParams params = { 0, 1, 0, 0, };
clock_rate = depayload->clock_rate;
/* Create a buffer for the tone */
tone_size = (payload.duration * SAMPLE_SIZE * CHANNELS) / 8;
buf = gst_buffer_new_allocate (NULL, tone_size, ¶ms);
GST_BUFFER_DURATION (buf) = payload.duration * GST_SECOND / clock_rate;
volume = payload.volume;
gst_buffer_map (buf, &map, GST_MAP_WRITE);
p = (gint16 *) map.data;
volume_factor = pow (10, (-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)++;
}
gst_buffer_unmap (buf, &map);
return buf;
}
static GstBuffer *
gst_rtp_dtmf_depay_process (GstRTPBaseDepayload * 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;
GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
rtpdtmfdepay = GST_RTP_DTMF_DEPAY (depayload);
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuffer);
payload_len = gst_rtp_buffer_get_payload_len (&rtpbuffer);
payload = gst_rtp_buffer_get_payload (&rtpbuffer);
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 (&rtpbuffer);
timestamp = gst_rtp_buffer_get_timestamp (&rtpbuffer);
dtmf_payload.duration = g_ntohs (dtmf_payload.duration);
/* clip to whole units of unit_time */
if (rtpdtmfdepay->unit_time) {
guint unit_time_clock =
(rtpdtmfdepay->unit_time * depayload->clock_rate) / 1000;
if (dtmf_payload.duration % unit_time_clock) {
/* Make sure we don't overflow the duration */
if (dtmf_payload.duration < G_MAXUINT16 - unit_time_clock)
dtmf_payload.duration += unit_time_clock -
(dtmf_payload.duration % unit_time_clock);
else
dtmf_payload.duration -= dtmf_payload.duration % unit_time_clock;
}
}
/* clip to max duration */
if (rtpdtmfdepay->max_duration) {
guint max_duration_clock =
(rtpdtmfdepay->max_duration * depayload->clock_rate) / 1000;
if (max_duration_clock < G_MAXUINT16 &&
dtmf_payload.duration > max_duration_clock)
dtmf_payload.duration = max_duration_clock;
}
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_PTS (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_ERROR_OBJECT (depayload,
"Unable to send dtmf-event message to bus");
}
} else {
GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event message");
}
} else {
GST_ERROR_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 : %" G_GUINT64_FORMAT,
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_dtmf_src_generate_tone (rtpdtmfdepay, dtmf_payload);
GST_BUFFER_PTS (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 : %" G_GUINT64_FORMAT " - time %" GST_TIME_FORMAT,
GST_BUFFER_TIMESTAMP (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
}
gst_rtp_buffer_unmap (&rtpbuffer);
return outbuf;
bad_packet:
GST_ELEMENT_WARNING (rtpdtmfdepay, STREAM, DECODE,
("Packet did not validate"), (NULL));
if (rtpbuffer.buffer != NULL)
gst_rtp_buffer_unmap (&rtpbuffer);
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);
}