3 * Copyright (C) 2008 Collabora Limited
4 * Copyright (C) 2008 Nokia Corporation
5 * Contact: Youness Alaoui <youness.alaoui@collabora.co.uk>
7 * This library is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Library General Public
9 * License as published by the Free Software Foundation; either
10 * version 2 of the License, or (at your option) any later version.
12 * This library is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Library General Public License for more details.
17 * You should have received a copy of the GNU Library General Public
18 * License along with this library; if not, write to the
19 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
20 * Boston, MA 02111-1307, USA.
23 * SECTION:element-rtpdtmfdepay
24 * @see_also: rtpdtmfsrc, rtpdtmfmux
26 * This element takes RTP DTMF packets and produces sound. It also emits a
27 * message on the #GstBus.
29 * The message is called "dtmf-event" and has the following fields
32 * <colspec colname='Name' />
33 * <colspec colname='Type' />
34 * <colspec colname='Possible values' />
35 * <colspec colname='Purpose' />
39 * <entry>GType</entry>
40 * <entry>Possible values</entry>
41 * <entry>Purpose</entry>
47 * <entry>G_TYPE_INT</entry>
49 * <entry>Which of the two methods
50 * specified in RFC 2833 to use. The value should be 0 for tones and 1 for
51 * named events. Tones are specified by their frequencies and events are specied
52 * by their number. This element currently only recognizes events.
53 * Do not confuse with "method" which specified the output.
57 * <entry>number</entry>
58 * <entry>G_TYPE_INT</entry>
60 * <entry>The event number.</entry>
63 * <entry>volume</entry>
64 * <entry>G_TYPE_INT</entry>
66 * <entry>This field describes the power level of the tone, expressed in dBm0
67 * after dropping the sign. Power levels range from 0 to -63 dBm0. The range of
68 * valid DTMF is from 0 to -36 dBm0.
72 * <entry>method</entry>
73 * <entry>G_TYPE_INT</entry>
75 * <entry>This field will always been 1 (ie RTP event) from this element.
87 #include "gstrtpdtmfdepay.h"
92 #include <gst/audio/audio.h>
93 #include <gst/rtp/gstrtpbuffer.h>
95 #define DEFAULT_PACKET_INTERVAL 50 /* ms */
96 #define MIN_PACKET_INTERVAL 10 /* ms */
97 #define MAX_PACKET_INTERVAL 50 /* ms */
98 #define SAMPLE_RATE 8000
99 #define SAMPLE_SIZE 16
101 #define MIN_DUTY_CYCLE (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION)
103 #define MIN_UNIT_TIME 0
104 #define MAX_UNIT_TIME 1000
105 #define DEFAULT_UNIT_TIME 0
107 #define DEFAULT_MAX_DURATION 0
109 typedef struct st_dtmf_key
111 const char *event_name;
114 float high_frequency;
117 static const DTMF_KEY DTMF_KEYS[] = {
118 {"DTMF_KEY_EVENT_0", 0, 941, 1336},
119 {"DTMF_KEY_EVENT_1", 1, 697, 1209},
120 {"DTMF_KEY_EVENT_2", 2, 697, 1336},
121 {"DTMF_KEY_EVENT_3", 3, 697, 1477},
122 {"DTMF_KEY_EVENT_4", 4, 770, 1209},
123 {"DTMF_KEY_EVENT_5", 5, 770, 1336},
124 {"DTMF_KEY_EVENT_6", 6, 770, 1477},
125 {"DTMF_KEY_EVENT_7", 7, 852, 1209},
126 {"DTMF_KEY_EVENT_8", 8, 852, 1336},
127 {"DTMF_KEY_EVENT_9", 9, 852, 1477},
128 {"DTMF_KEY_EVENT_S", 10, 941, 1209},
129 {"DTMF_KEY_EVENT_P", 11, 941, 1477},
130 {"DTMF_KEY_EVENT_A", 12, 697, 1633},
131 {"DTMF_KEY_EVENT_B", 13, 770, 1633},
132 {"DTMF_KEY_EVENT_C", 14, 852, 1633},
133 {"DTMF_KEY_EVENT_D", 15, 941, 1633},
136 #define MAX_DTMF_EVENTS 16
140 DTMF_KEY_EVENT_1 = 1,
141 DTMF_KEY_EVENT_2 = 2,
142 DTMF_KEY_EVENT_3 = 3,
143 DTMF_KEY_EVENT_4 = 4,
144 DTMF_KEY_EVENT_5 = 5,
145 DTMF_KEY_EVENT_6 = 6,
146 DTMF_KEY_EVENT_7 = 7,
147 DTMF_KEY_EVENT_8 = 8,
148 DTMF_KEY_EVENT_9 = 9,
149 DTMF_KEY_EVENT_0 = 0,
150 DTMF_KEY_EVENT_STAR = 10,
151 DTMF_KEY_EVENT_POUND = 11,
152 DTMF_KEY_EVENT_A = 12,
153 DTMF_KEY_EVENT_B = 13,
154 DTMF_KEY_EVENT_C = 14,
155 DTMF_KEY_EVENT_D = 15,
158 GST_DEBUG_CATEGORY_STATIC (gst_rtp_dtmf_depay_debug);
159 #define GST_CAT_DEFAULT gst_rtp_dtmf_depay_debug
181 static GstStaticPadTemplate gst_rtp_dtmf_depay_src_template =
182 GST_STATIC_PAD_TEMPLATE ("src",
185 GST_STATIC_CAPS ("audio/x-raw, "
186 "format = (string) \"" GST_AUDIO_NE (S16) "\", "
187 "rate = " GST_AUDIO_RATE_RANGE ", " "channels = (int) 1")
190 static GstStaticPadTemplate gst_rtp_dtmf_depay_sink_template =
191 GST_STATIC_PAD_TEMPLATE ("sink",
194 GST_STATIC_CAPS ("application/x-rtp, "
195 "media = (string) \"audio\", "
196 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
197 "clock-rate = (int) [ 0, MAX ], "
198 "encoding-name = (string) \"TELEPHONE-EVENT\"")
201 G_DEFINE_TYPE (GstRtpDTMFDepay, gst_rtp_dtmf_depay,
202 GST_TYPE_RTP_BASE_DEPAYLOAD);
204 static void gst_rtp_dtmf_depay_set_property (GObject * object, guint prop_id,
205 const GValue * value, GParamSpec * pspec);
206 static void gst_rtp_dtmf_depay_get_property (GObject * object, guint prop_id,
207 GValue * value, GParamSpec * pspec);
208 static GstBuffer *gst_rtp_dtmf_depay_process (GstRTPBaseDepayload * depayload,
210 gboolean gst_rtp_dtmf_depay_setcaps (GstRTPBaseDepayload * filter,
214 gst_rtp_dtmf_depay_class_init (GstRtpDTMFDepayClass * klass)
216 GObjectClass *gobject_class;
217 GstElementClass *gstelement_class;
218 GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
220 gobject_class = G_OBJECT_CLASS (klass);
221 gstelement_class = GST_ELEMENT_CLASS (klass);
222 gstrtpbasedepayload_class = GST_RTP_BASE_DEPAYLOAD_CLASS (klass);
224 gst_element_class_add_pad_template (gstelement_class,
225 gst_static_pad_template_get (&gst_rtp_dtmf_depay_src_template));
226 gst_element_class_add_pad_template (gstelement_class,
227 gst_static_pad_template_get (&gst_rtp_dtmf_depay_sink_template));
229 GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_depay_debug,
230 "rtpdtmfdepay", 0, "rtpdtmfdepay element");
231 gst_element_class_set_details_simple (gstelement_class,
232 "RTP DTMF packet depayloader", "Codec/Depayloader/Network",
233 "Generates DTMF Sound from telephone-event RTP packets",
234 "Youness Alaoui <youness.alaoui@collabora.co.uk>");
236 gobject_class->set_property =
237 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_set_property);
238 gobject_class->get_property =
239 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_get_property);
241 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_UNIT_TIME,
242 g_param_spec_uint ("unit-time", "Duration unittime",
243 "The smallest unit (ms) the duration must be a multiple of (0 disables it)",
244 MIN_UNIT_TIME, MAX_UNIT_TIME, DEFAULT_UNIT_TIME,
245 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
247 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MAX_DURATION,
248 g_param_spec_uint ("max-duration", "Maximum duration",
249 "The maxumimum duration (ms) of the outgoing soundpacket. "
250 "(0 = no limit)", 0, G_MAXUINT, DEFAULT_MAX_DURATION,
251 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
253 gstrtpbasedepayload_class->process =
254 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_process);
255 gstrtpbasedepayload_class->set_caps =
256 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_setcaps);
261 gst_rtp_dtmf_depay_init (GstRtpDTMFDepay * rtpdtmfdepay)
263 rtpdtmfdepay->unit_time = DEFAULT_UNIT_TIME;
267 gst_rtp_dtmf_depay_set_property (GObject * object, guint prop_id,
268 const GValue * value, GParamSpec * pspec)
270 GstRtpDTMFDepay *rtpdtmfdepay;
272 rtpdtmfdepay = GST_RTP_DTMF_DEPAY (object);
276 rtpdtmfdepay->unit_time = g_value_get_uint (value);
278 case PROP_MAX_DURATION:
279 rtpdtmfdepay->max_duration = g_value_get_uint (value);
282 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
288 gst_rtp_dtmf_depay_get_property (GObject * object, guint prop_id,
289 GValue * value, GParamSpec * pspec)
291 GstRtpDTMFDepay *rtpdtmfdepay;
293 rtpdtmfdepay = GST_RTP_DTMF_DEPAY (object);
297 g_value_set_uint (value, rtpdtmfdepay->unit_time);
299 case PROP_MAX_DURATION:
300 g_value_set_uint (value, rtpdtmfdepay->max_duration);
303 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
309 gst_rtp_dtmf_depay_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps)
312 GstStructure *structure = gst_caps_get_structure (caps, 0);
313 gint clock_rate = 8000; /* default */
315 gst_structure_get_int (structure, "clock-rate", &clock_rate);
316 filter->clock_rate = clock_rate;
318 srccaps = gst_caps_new_simple ("audio/x-raw-int",
319 "width", G_TYPE_INT, 16,
320 "depth", G_TYPE_INT, 16,
321 "endianness", G_TYPE_INT, G_BYTE_ORDER,
322 "signed", G_TYPE_BOOLEAN, TRUE,
323 "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, clock_rate, NULL);
324 gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter), srccaps);
325 gst_caps_unref (srccaps);
331 gst_dtmf_src_generate_tone (GstRtpDTMFDepay * rtpdtmfdepay,
332 GstRTPDTMFPayload payload)
339 double amplitude, f1, f2;
340 double volume_factor;
341 DTMF_KEY key = DTMF_KEYS[payload.event];
342 guint32 clock_rate = 8000 /* default */ ;
343 GstRTPBaseDepayload *depayload = GST_RTP_BASE_DEPAYLOAD (rtpdtmfdepay);
345 static GstAllocationParams params = { 0, 1, 0, 0, };
347 clock_rate = depayload->clock_rate;
349 /* Create a buffer for the tone */
350 tone_size = (payload.duration * SAMPLE_SIZE * CHANNELS) / 8;
351 buf = gst_buffer_new_allocate (NULL, tone_size, ¶ms);
352 GST_BUFFER_DURATION (buf) = payload.duration * GST_SECOND / clock_rate;
353 volume = payload.volume;
355 gst_buffer_map (buf, &map, GST_MAP_WRITE);
356 p = (gint16 *) map.data;
358 volume_factor = pow (10, (-volume) / 20);
361 * For each sample point we calculate 'x' as the
362 * the amplitude value.
364 for (i = 0; i < (tone_size / (SAMPLE_SIZE / 8)); i++) {
366 * We add the fundamental frequencies together.
368 f1 = sin (2 * M_PI * key.low_frequency * (rtpdtmfdepay->sample /
370 f2 = sin (2 * M_PI * key.high_frequency * (rtpdtmfdepay->sample /
373 amplitude = (f1 + f2) / 2;
375 /* Adjust the volume */
376 amplitude *= volume_factor;
378 /* Make the [-1:1] interval into a [-32767:32767] interval */
381 /* Store it in the data buffer */
382 *(p++) = (gint16) amplitude;
384 (rtpdtmfdepay->sample)++;
387 gst_buffer_unmap (buf, &map);
394 gst_rtp_dtmf_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
397 GstRtpDTMFDepay *rtpdtmfdepay = NULL;
398 GstBuffer *outbuf = NULL;
400 guint8 *payload = NULL;
402 GstRTPDTMFPayload dtmf_payload;
404 GstStructure *structure = NULL;
405 GstMessage *dtmf_message = NULL;
406 GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
408 rtpdtmfdepay = GST_RTP_DTMF_DEPAY (depayload);
410 gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuffer);
412 payload_len = gst_rtp_buffer_get_payload_len (&rtpbuffer);
413 payload = gst_rtp_buffer_get_payload (&rtpbuffer);
415 if (payload_len != sizeof (GstRTPDTMFPayload))
418 memcpy (&dtmf_payload, payload, sizeof (GstRTPDTMFPayload));
420 if (dtmf_payload.event > MAX_EVENT)
423 marker = gst_rtp_buffer_get_marker (&rtpbuffer);
425 timestamp = gst_rtp_buffer_get_timestamp (&rtpbuffer);
427 dtmf_payload.duration = g_ntohs (dtmf_payload.duration);
429 /* clip to whole units of unit_time */
430 if (rtpdtmfdepay->unit_time) {
431 guint unit_time_clock =
432 (rtpdtmfdepay->unit_time * depayload->clock_rate) / 1000;
433 if (dtmf_payload.duration % unit_time_clock) {
434 /* Make sure we don't overflow the duration */
435 if (dtmf_payload.duration < G_MAXUINT16 - unit_time_clock)
436 dtmf_payload.duration += unit_time_clock -
437 (dtmf_payload.duration % unit_time_clock);
439 dtmf_payload.duration -= dtmf_payload.duration % unit_time_clock;
443 /* clip to max duration */
444 if (rtpdtmfdepay->max_duration) {
445 guint max_duration_clock =
446 (rtpdtmfdepay->max_duration * depayload->clock_rate) / 1000;
448 if (max_duration_clock < G_MAXUINT16 &&
449 dtmf_payload.duration > max_duration_clock)
450 dtmf_payload.duration = max_duration_clock;
453 GST_DEBUG_OBJECT (depayload, "Received new RTP DTMF packet : "
454 "marker=%d - timestamp=%u - event=%d - duration=%d",
455 marker, timestamp, dtmf_payload.event, dtmf_payload.duration);
457 GST_DEBUG_OBJECT (depayload,
458 "Previous information : timestamp=%u - duration=%d",
459 rtpdtmfdepay->previous_ts, rtpdtmfdepay->previous_duration);
462 if (marker || rtpdtmfdepay->previous_ts != timestamp) {
463 rtpdtmfdepay->sample = 0;
464 rtpdtmfdepay->previous_ts = timestamp;
465 rtpdtmfdepay->previous_duration = dtmf_payload.duration;
466 rtpdtmfdepay->first_gst_ts = GST_BUFFER_PTS (buf);
468 structure = gst_structure_new ("dtmf-event",
469 "number", G_TYPE_INT, dtmf_payload.event,
470 "volume", G_TYPE_INT, dtmf_payload.volume,
471 "type", G_TYPE_INT, 1, "method", G_TYPE_INT, 1, NULL);
474 gst_message_new_element (GST_OBJECT (depayload), structure);
476 if (!gst_element_post_message (GST_ELEMENT (depayload), dtmf_message)) {
477 GST_ERROR_OBJECT (depayload,
478 "Unable to send dtmf-event message to bus");
481 GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event message");
484 GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event structure");
487 guint16 duration = dtmf_payload.duration;
488 dtmf_payload.duration -= rtpdtmfdepay->previous_duration;
489 /* If late buffer, ignore */
490 if (duration > rtpdtmfdepay->previous_duration)
491 rtpdtmfdepay->previous_duration = duration;
494 GST_DEBUG_OBJECT (depayload, "new previous duration : %d - new duration : %d"
495 " - diff : %d - clock rate : %d - timestamp : %" G_GUINT64_FORMAT,
496 rtpdtmfdepay->previous_duration, dtmf_payload.duration,
497 (rtpdtmfdepay->previous_duration - dtmf_payload.duration),
498 depayload->clock_rate, GST_BUFFER_TIMESTAMP (buf));
500 /* If late or duplicate packet (like the redundant end packet). Ignore */
501 if (dtmf_payload.duration > 0) {
502 outbuf = gst_dtmf_src_generate_tone (rtpdtmfdepay, dtmf_payload);
505 GST_BUFFER_PTS (outbuf) = rtpdtmfdepay->first_gst_ts +
506 (rtpdtmfdepay->previous_duration - dtmf_payload.duration) *
507 GST_SECOND / depayload->clock_rate;
508 GST_BUFFER_OFFSET (outbuf) =
509 (rtpdtmfdepay->previous_duration - dtmf_payload.duration) *
510 GST_SECOND / depayload->clock_rate;
511 GST_BUFFER_OFFSET_END (outbuf) = rtpdtmfdepay->previous_duration *
512 GST_SECOND / depayload->clock_rate;
514 GST_DEBUG_OBJECT (depayload,
515 "timestamp : %" G_GUINT64_FORMAT " - time %" GST_TIME_FORMAT,
516 GST_BUFFER_TIMESTAMP (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
520 gst_rtp_buffer_unmap (&rtpbuffer);
525 GST_ELEMENT_WARNING (rtpdtmfdepay, STREAM, DECODE,
526 ("Packet did not validate"), (NULL));
528 if (rtpbuffer.buffer != NULL)
529 gst_rtp_buffer_unmap (&rtpbuffer);
535 gst_rtp_dtmf_depay_plugin_init (GstPlugin * plugin)
537 return gst_element_register (plugin, "rtpdtmfdepay",
538 GST_RANK_MARGINAL, GST_TYPE_RTP_DTMF_DEPAY);