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., 51 Franklin St, Fifth Floor,
20 * Boston, MA 02110-1301, USA.
23 * SECTION:element-rtpdtmfdepay
24 * @title: rtpdtmfdepay
25 * @see_also: rtpdtmfsrc, rtpdtmfmux
27 * This element takes RTP DTMF packets and produces sound. It also emits a
28 * message on the #GstBus.
30 * The message is called "dtmf-event" and has the following fields:
32 * * `type` (G_TYPE_INT, 0-1): Which of the two methods
33 * specified in RFC 2833 to use. The value should be 0 for tones and 1 for
34 * named events. Tones are specified by their frequencies and events are specied
35 * by their number. This element currently only recognizes events.
36 * Do not confuse with "method" which specified the output.
38 * * `number` (G_TYPE_INT, 0-16): The event number.
40 * * `volume` (G_TYPE_INT, 0-36): This field describes the power level of the tone, expressed in dBm0
41 * after dropping the sign. Power levels range from 0 to -63 dBm0. The range of
42 * valid DTMF is from 0 to -36 dBm0.
44 * * `method` (G_TYPE_INT, 1): This field will always been 1 (ie RTP event) from this element.
51 #include "gstrtpdtmfdepay.h"
56 #include <gst/audio/audio.h>
57 #include <gst/rtp/gstrtpbuffer.h>
59 #define DEFAULT_PACKET_INTERVAL 50 /* ms */
60 #define MIN_PACKET_INTERVAL 10 /* ms */
61 #define MAX_PACKET_INTERVAL 50 /* ms */
62 #define SAMPLE_RATE 8000
63 #define SAMPLE_SIZE 16
65 #define MIN_DUTY_CYCLE (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION)
67 #define MIN_UNIT_TIME 0
68 #define MAX_UNIT_TIME 1000
69 #define DEFAULT_UNIT_TIME 0
71 #define DEFAULT_MAX_DURATION 0
73 typedef struct st_dtmf_key
79 static const DTMF_KEY DTMF_KEYS[] = {
98 #define MAX_DTMF_EVENTS 16
102 DTMF_KEY_EVENT_1 = 1,
103 DTMF_KEY_EVENT_2 = 2,
104 DTMF_KEY_EVENT_3 = 3,
105 DTMF_KEY_EVENT_4 = 4,
106 DTMF_KEY_EVENT_5 = 5,
107 DTMF_KEY_EVENT_6 = 6,
108 DTMF_KEY_EVENT_7 = 7,
109 DTMF_KEY_EVENT_8 = 8,
110 DTMF_KEY_EVENT_9 = 9,
111 DTMF_KEY_EVENT_0 = 0,
112 DTMF_KEY_EVENT_STAR = 10,
113 DTMF_KEY_EVENT_POUND = 11,
114 DTMF_KEY_EVENT_A = 12,
115 DTMF_KEY_EVENT_B = 13,
116 DTMF_KEY_EVENT_C = 14,
117 DTMF_KEY_EVENT_D = 15,
120 GST_DEBUG_CATEGORY_STATIC (gst_rtp_dtmf_depay_debug);
121 #define GST_CAT_DEFAULT gst_rtp_dtmf_depay_debug
136 static GstStaticPadTemplate gst_rtp_dtmf_depay_src_template =
137 GST_STATIC_PAD_TEMPLATE ("src",
140 GST_STATIC_CAPS ("audio/x-raw, "
141 "format = (string) \"" GST_AUDIO_NE (S16) "\", "
142 "rate = " GST_AUDIO_RATE_RANGE ", " "channels = (int) 1")
145 static GstStaticPadTemplate gst_rtp_dtmf_depay_sink_template =
146 GST_STATIC_PAD_TEMPLATE ("sink",
149 GST_STATIC_CAPS ("application/x-rtp, "
150 "media = (string) \"audio\", "
151 "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
152 "clock-rate = (int) [ 0, MAX ], "
153 "encoding-name = (string) \"TELEPHONE-EVENT\"")
156 G_DEFINE_TYPE (GstRtpDTMFDepay, gst_rtp_dtmf_depay,
157 GST_TYPE_RTP_BASE_DEPAYLOAD);
159 static void gst_rtp_dtmf_depay_set_property (GObject * object, guint prop_id,
160 const GValue * value, GParamSpec * pspec);
161 static void gst_rtp_dtmf_depay_get_property (GObject * object, guint prop_id,
162 GValue * value, GParamSpec * pspec);
163 static GstBuffer *gst_rtp_dtmf_depay_process (GstRTPBaseDepayload * depayload,
165 gboolean gst_rtp_dtmf_depay_setcaps (GstRTPBaseDepayload * filter,
169 gst_rtp_dtmf_depay_class_init (GstRtpDTMFDepayClass * klass)
171 GObjectClass *gobject_class;
172 GstElementClass *gstelement_class;
173 GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
175 gobject_class = G_OBJECT_CLASS (klass);
176 gstelement_class = GST_ELEMENT_CLASS (klass);
177 gstrtpbasedepayload_class = GST_RTP_BASE_DEPAYLOAD_CLASS (klass);
179 gst_element_class_add_static_pad_template (gstelement_class,
180 &gst_rtp_dtmf_depay_src_template);
181 gst_element_class_add_static_pad_template (gstelement_class,
182 &gst_rtp_dtmf_depay_sink_template);
184 GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_depay_debug,
185 "rtpdtmfdepay", 0, "rtpdtmfdepay element");
186 gst_element_class_set_static_metadata (gstelement_class,
187 "RTP DTMF packet depayloader", "Codec/Depayloader/Network",
188 "Generates DTMF Sound from telephone-event RTP packets",
189 "Youness Alaoui <youness.alaoui@collabora.co.uk>");
191 gobject_class->set_property =
192 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_set_property);
193 gobject_class->get_property =
194 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_get_property);
196 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_UNIT_TIME,
197 g_param_spec_uint ("unit-time", "Duration unittime",
198 "The smallest unit (ms) the duration must be a multiple of (0 disables it)",
199 MIN_UNIT_TIME, MAX_UNIT_TIME, DEFAULT_UNIT_TIME,
200 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
202 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MAX_DURATION,
203 g_param_spec_uint ("max-duration", "Maximum duration",
204 "The maxumimum duration (ms) of the outgoing soundpacket. "
205 "(0 = no limit)", 0, G_MAXUINT, DEFAULT_MAX_DURATION,
206 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
208 gstrtpbasedepayload_class->process =
209 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_process);
210 gstrtpbasedepayload_class->set_caps =
211 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_setcaps);
216 gst_rtp_dtmf_depay_init (GstRtpDTMFDepay * rtpdtmfdepay)
218 rtpdtmfdepay->unit_time = DEFAULT_UNIT_TIME;
222 gst_rtp_dtmf_depay_set_property (GObject * object, guint prop_id,
223 const GValue * value, GParamSpec * pspec)
225 GstRtpDTMFDepay *rtpdtmfdepay;
227 rtpdtmfdepay = GST_RTP_DTMF_DEPAY (object);
231 rtpdtmfdepay->unit_time = g_value_get_uint (value);
233 case PROP_MAX_DURATION:
234 rtpdtmfdepay->max_duration = g_value_get_uint (value);
237 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
243 gst_rtp_dtmf_depay_get_property (GObject * object, guint prop_id,
244 GValue * value, GParamSpec * pspec)
246 GstRtpDTMFDepay *rtpdtmfdepay;
248 rtpdtmfdepay = GST_RTP_DTMF_DEPAY (object);
252 g_value_set_uint (value, rtpdtmfdepay->unit_time);
254 case PROP_MAX_DURATION:
255 g_value_set_uint (value, rtpdtmfdepay->max_duration);
258 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
264 gst_rtp_dtmf_depay_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps)
266 GstCaps *filtercaps, *srccaps;
267 GstStructure *structure = gst_caps_get_structure (caps, 0);
268 gint clock_rate = 8000; /* default */
270 gst_structure_get_int (structure, "clock-rate", &clock_rate);
271 filter->clock_rate = clock_rate;
274 gst_pad_get_pad_template_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter));
276 filtercaps = gst_caps_make_writable (filtercaps);
277 gst_caps_set_simple (filtercaps, "rate", G_TYPE_INT, clock_rate, NULL);
279 srccaps = gst_pad_peer_query_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter),
281 gst_caps_unref (filtercaps);
283 gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter), srccaps);
284 gst_caps_unref (srccaps);
290 gst_dtmf_src_generate_tone (GstRtpDTMFDepay * rtpdtmfdepay,
291 GstRTPDTMFPayload payload)
298 double amplitude, f1, f2;
299 double volume_factor;
300 DTMF_KEY key = DTMF_KEYS[payload.event];
302 GstRTPBaseDepayload *depayload = GST_RTP_BASE_DEPAYLOAD (rtpdtmfdepay);
304 static GstAllocationParams params = { 0, 1, 0, 0, };
306 clock_rate = depayload->clock_rate;
308 /* Create a buffer for the tone */
309 tone_size = (payload.duration * SAMPLE_SIZE * CHANNELS) / 8;
310 buf = gst_buffer_new_allocate (NULL, tone_size, ¶ms);
311 GST_BUFFER_DURATION (buf) = payload.duration * GST_SECOND / clock_rate;
312 volume = payload.volume;
314 gst_buffer_map (buf, &map, GST_MAP_WRITE);
315 p = (gint16 *) map.data;
317 volume_factor = pow (10, (-volume) / 20);
320 * For each sample point we calculate 'x' as the
321 * the amplitude value.
323 for (i = 0; i < (tone_size / (SAMPLE_SIZE / 8)); i++) {
325 * We add the fundamental frequencies together.
327 f1 = sin (2 * M_PI * key.low_frequency * (rtpdtmfdepay->sample /
329 f2 = sin (2 * M_PI * key.high_frequency * (rtpdtmfdepay->sample /
332 amplitude = (f1 + f2) / 2;
334 /* Adjust the volume */
335 amplitude *= volume_factor;
337 /* Make the [-1:1] interval into a [-32767:32767] interval */
340 /* Store it in the data buffer */
341 *(p++) = (gint16) amplitude;
343 (rtpdtmfdepay->sample)++;
346 gst_buffer_unmap (buf, &map);
353 gst_rtp_dtmf_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
356 GstRtpDTMFDepay *rtpdtmfdepay = NULL;
357 GstBuffer *outbuf = NULL;
359 guint8 *payload = NULL;
361 GstRTPDTMFPayload dtmf_payload;
363 GstStructure *structure = NULL;
364 GstMessage *dtmf_message = NULL;
365 GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
367 rtpdtmfdepay = GST_RTP_DTMF_DEPAY (depayload);
369 gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuffer);
371 payload_len = gst_rtp_buffer_get_payload_len (&rtpbuffer);
372 payload = gst_rtp_buffer_get_payload (&rtpbuffer);
374 if (payload_len != sizeof (GstRTPDTMFPayload))
377 memcpy (&dtmf_payload, payload, sizeof (GstRTPDTMFPayload));
379 if (dtmf_payload.event > MAX_EVENT)
382 marker = gst_rtp_buffer_get_marker (&rtpbuffer);
384 timestamp = gst_rtp_buffer_get_timestamp (&rtpbuffer);
386 dtmf_payload.duration = g_ntohs (dtmf_payload.duration);
388 /* clip to whole units of unit_time */
389 if (rtpdtmfdepay->unit_time) {
390 guint unit_time_clock =
391 (rtpdtmfdepay->unit_time * depayload->clock_rate) / 1000;
392 if (dtmf_payload.duration % unit_time_clock) {
393 /* Make sure we don't overflow the duration */
394 if (dtmf_payload.duration < G_MAXUINT16 - unit_time_clock)
395 dtmf_payload.duration += unit_time_clock -
396 (dtmf_payload.duration % unit_time_clock);
398 dtmf_payload.duration -= dtmf_payload.duration % unit_time_clock;
402 /* clip to max duration */
403 if (rtpdtmfdepay->max_duration) {
404 guint max_duration_clock =
405 (rtpdtmfdepay->max_duration * depayload->clock_rate) / 1000;
407 if (max_duration_clock < G_MAXUINT16 &&
408 dtmf_payload.duration > max_duration_clock)
409 dtmf_payload.duration = max_duration_clock;
412 GST_DEBUG_OBJECT (depayload, "Received new RTP DTMF packet : "
413 "marker=%d - timestamp=%u - event=%d - duration=%d",
414 marker, timestamp, dtmf_payload.event, dtmf_payload.duration);
416 GST_DEBUG_OBJECT (depayload,
417 "Previous information : timestamp=%u - duration=%d",
418 rtpdtmfdepay->previous_ts, rtpdtmfdepay->previous_duration);
421 if (marker || rtpdtmfdepay->previous_ts != timestamp) {
422 rtpdtmfdepay->sample = 0;
423 rtpdtmfdepay->previous_ts = timestamp;
424 rtpdtmfdepay->previous_duration = dtmf_payload.duration;
425 rtpdtmfdepay->first_gst_ts = GST_BUFFER_PTS (buf);
427 structure = gst_structure_new ("dtmf-event",
428 "number", G_TYPE_INT, dtmf_payload.event,
429 "volume", G_TYPE_INT, dtmf_payload.volume,
430 "type", G_TYPE_INT, 1, "method", G_TYPE_INT, 1, NULL);
433 gst_message_new_element (GST_OBJECT (depayload), structure);
435 if (!gst_element_post_message (GST_ELEMENT (depayload), dtmf_message)) {
436 GST_ERROR_OBJECT (depayload,
437 "Unable to send dtmf-event message to bus");
440 GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event message");
443 GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event structure");
446 guint16 duration = dtmf_payload.duration;
447 dtmf_payload.duration -= rtpdtmfdepay->previous_duration;
448 /* If late buffer, ignore */
449 if (duration > rtpdtmfdepay->previous_duration)
450 rtpdtmfdepay->previous_duration = duration;
453 GST_DEBUG_OBJECT (depayload, "new previous duration : %d - new duration : %d"
454 " - diff : %d - clock rate : %d - timestamp : %" G_GUINT64_FORMAT,
455 rtpdtmfdepay->previous_duration, dtmf_payload.duration,
456 (rtpdtmfdepay->previous_duration - dtmf_payload.duration),
457 depayload->clock_rate, GST_BUFFER_TIMESTAMP (buf));
459 /* If late or duplicate packet (like the redundant end packet). Ignore */
460 if (dtmf_payload.duration > 0) {
461 outbuf = gst_dtmf_src_generate_tone (rtpdtmfdepay, dtmf_payload);
464 GST_BUFFER_PTS (outbuf) = rtpdtmfdepay->first_gst_ts +
465 (rtpdtmfdepay->previous_duration - dtmf_payload.duration) *
466 GST_SECOND / depayload->clock_rate;
467 GST_BUFFER_OFFSET (outbuf) =
468 (rtpdtmfdepay->previous_duration - dtmf_payload.duration) *
469 GST_SECOND / depayload->clock_rate;
470 GST_BUFFER_OFFSET_END (outbuf) = rtpdtmfdepay->previous_duration *
471 GST_SECOND / depayload->clock_rate;
473 GST_DEBUG_OBJECT (depayload,
474 "timestamp : %" G_GUINT64_FORMAT " - time %" GST_TIME_FORMAT,
475 GST_BUFFER_TIMESTAMP (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
479 gst_rtp_buffer_unmap (&rtpbuffer);
484 GST_ELEMENT_WARNING (rtpdtmfdepay, STREAM, DECODE,
485 ("Packet did not validate"), (NULL));
487 if (rtpbuffer.buffer != NULL)
488 gst_rtp_buffer_unmap (&rtpbuffer);
494 gst_rtp_dtmf_depay_plugin_init (GstPlugin * plugin)
496 return gst_element_register (plugin, "rtpdtmfdepay",
497 GST_RANK_MARGINAL, GST_TYPE_RTP_DTMF_DEPAY);