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 specified
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);
158 GST_ELEMENT_REGISTER_DEFINE (rtpdtmfdepay, "rtpdtmfdepay", GST_RANK_MARGINAL,
159 GST_TYPE_RTP_DTMF_DEPAY);
161 static void gst_rtp_dtmf_depay_set_property (GObject * object, guint prop_id,
162 const GValue * value, GParamSpec * pspec);
163 static void gst_rtp_dtmf_depay_get_property (GObject * object, guint prop_id,
164 GValue * value, GParamSpec * pspec);
165 static GstBuffer *gst_rtp_dtmf_depay_process (GstRTPBaseDepayload * depayload,
167 gboolean gst_rtp_dtmf_depay_setcaps (GstRTPBaseDepayload * filter,
171 gst_rtp_dtmf_depay_class_init (GstRtpDTMFDepayClass * klass)
173 GObjectClass *gobject_class;
174 GstElementClass *gstelement_class;
175 GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
177 gobject_class = G_OBJECT_CLASS (klass);
178 gstelement_class = GST_ELEMENT_CLASS (klass);
179 gstrtpbasedepayload_class = GST_RTP_BASE_DEPAYLOAD_CLASS (klass);
181 gst_element_class_add_static_pad_template (gstelement_class,
182 &gst_rtp_dtmf_depay_src_template);
183 gst_element_class_add_static_pad_template (gstelement_class,
184 &gst_rtp_dtmf_depay_sink_template);
186 GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_depay_debug,
187 "rtpdtmfdepay", 0, "rtpdtmfdepay element");
188 gst_element_class_set_static_metadata (gstelement_class,
189 "RTP DTMF packet depayloader", "Codec/Depayloader/Network",
190 "Generates DTMF Sound from telephone-event RTP packets",
191 "Youness Alaoui <youness.alaoui@collabora.co.uk>");
193 gobject_class->set_property =
194 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_set_property);
195 gobject_class->get_property =
196 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_get_property);
198 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_UNIT_TIME,
199 g_param_spec_uint ("unit-time", "Duration unittime",
200 "The smallest unit (ms) the duration must be a multiple of (0 disables it)",
201 MIN_UNIT_TIME, MAX_UNIT_TIME, DEFAULT_UNIT_TIME,
202 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
204 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MAX_DURATION,
205 g_param_spec_uint ("max-duration", "Maximum duration",
206 "The maxumimum duration (ms) of the outgoing soundpacket. "
207 "(0 = no limit)", 0, G_MAXUINT, DEFAULT_MAX_DURATION,
208 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
210 gstrtpbasedepayload_class->process =
211 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_process);
212 gstrtpbasedepayload_class->set_caps =
213 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_setcaps);
218 gst_rtp_dtmf_depay_init (GstRtpDTMFDepay * rtpdtmfdepay)
220 rtpdtmfdepay->unit_time = DEFAULT_UNIT_TIME;
224 gst_rtp_dtmf_depay_set_property (GObject * object, guint prop_id,
225 const GValue * value, GParamSpec * pspec)
227 GstRtpDTMFDepay *rtpdtmfdepay;
229 rtpdtmfdepay = GST_RTP_DTMF_DEPAY (object);
233 rtpdtmfdepay->unit_time = g_value_get_uint (value);
235 case PROP_MAX_DURATION:
236 rtpdtmfdepay->max_duration = g_value_get_uint (value);
239 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
245 gst_rtp_dtmf_depay_get_property (GObject * object, guint prop_id,
246 GValue * value, GParamSpec * pspec)
248 GstRtpDTMFDepay *rtpdtmfdepay;
250 rtpdtmfdepay = GST_RTP_DTMF_DEPAY (object);
254 g_value_set_uint (value, rtpdtmfdepay->unit_time);
256 case PROP_MAX_DURATION:
257 g_value_set_uint (value, rtpdtmfdepay->max_duration);
260 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
266 gst_rtp_dtmf_depay_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps)
268 GstCaps *filtercaps, *srccaps;
269 GstStructure *structure = gst_caps_get_structure (caps, 0);
270 gint clock_rate = 8000; /* default */
272 gst_structure_get_int (structure, "clock-rate", &clock_rate);
273 filter->clock_rate = clock_rate;
276 gst_pad_get_pad_template_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter));
278 filtercaps = gst_caps_make_writable (filtercaps);
279 gst_caps_set_simple (filtercaps, "rate", G_TYPE_INT, clock_rate, NULL);
281 srccaps = gst_pad_peer_query_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter),
283 gst_caps_unref (filtercaps);
285 gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter), srccaps);
286 gst_caps_unref (srccaps);
292 gst_dtmf_src_generate_tone (GstRtpDTMFDepay * rtpdtmfdepay,
293 GstRTPDTMFPayload payload)
300 double amplitude, f1, f2;
301 double volume_factor;
302 DTMF_KEY key = DTMF_KEYS[payload.event];
304 GstRTPBaseDepayload *depayload = GST_RTP_BASE_DEPAYLOAD (rtpdtmfdepay);
306 static GstAllocationParams params = { 0, 1, 0, 0, };
308 clock_rate = depayload->clock_rate;
310 /* Create a buffer for the tone */
311 tone_size = (payload.duration * SAMPLE_SIZE * CHANNELS) / 8;
312 buf = gst_buffer_new_allocate (NULL, tone_size, ¶ms);
313 GST_BUFFER_DURATION (buf) = payload.duration * GST_SECOND / clock_rate;
314 volume = payload.volume;
316 gst_buffer_map (buf, &map, GST_MAP_WRITE);
317 p = (gint16 *) map.data;
319 volume_factor = pow (10, (-volume) / 20);
322 * For each sample point we calculate 'x' as the
323 * the amplitude value.
325 for (i = 0; i < (tone_size / (SAMPLE_SIZE / 8)); i++) {
327 * We add the fundamental frequencies together.
329 f1 = sin (2 * M_PI * key.low_frequency * (rtpdtmfdepay->sample /
331 f2 = sin (2 * M_PI * key.high_frequency * (rtpdtmfdepay->sample /
334 amplitude = (f1 + f2) / 2;
336 /* Adjust the volume */
337 amplitude *= volume_factor;
339 /* Make the [-1:1] interval into a [-32767:32767] interval */
342 /* Store it in the data buffer */
343 *(p++) = (gint16) amplitude;
345 (rtpdtmfdepay->sample)++;
348 gst_buffer_unmap (buf, &map);
355 gst_rtp_dtmf_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
358 GstRtpDTMFDepay *rtpdtmfdepay = NULL;
359 GstBuffer *outbuf = NULL;
361 guint8 *payload = NULL;
363 GstRTPDTMFPayload dtmf_payload;
365 GstStructure *structure = NULL;
366 GstMessage *dtmf_message = NULL;
367 GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
369 rtpdtmfdepay = GST_RTP_DTMF_DEPAY (depayload);
371 gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuffer);
373 payload_len = gst_rtp_buffer_get_payload_len (&rtpbuffer);
374 payload = gst_rtp_buffer_get_payload (&rtpbuffer);
376 if (payload_len != sizeof (GstRTPDTMFPayload))
379 memcpy (&dtmf_payload, payload, sizeof (GstRTPDTMFPayload));
381 if (dtmf_payload.event > MAX_EVENT)
384 marker = gst_rtp_buffer_get_marker (&rtpbuffer);
386 timestamp = gst_rtp_buffer_get_timestamp (&rtpbuffer);
388 dtmf_payload.duration = g_ntohs (dtmf_payload.duration);
390 /* clip to whole units of unit_time */
391 if (rtpdtmfdepay->unit_time) {
392 guint unit_time_clock =
393 (rtpdtmfdepay->unit_time * depayload->clock_rate) / 1000;
394 if (dtmf_payload.duration % unit_time_clock) {
395 /* Make sure we don't overflow the duration */
396 if (dtmf_payload.duration < G_MAXUINT16 - unit_time_clock)
397 dtmf_payload.duration += unit_time_clock -
398 (dtmf_payload.duration % unit_time_clock);
400 dtmf_payload.duration -= dtmf_payload.duration % unit_time_clock;
404 /* clip to max duration */
405 if (rtpdtmfdepay->max_duration) {
406 guint max_duration_clock =
407 (rtpdtmfdepay->max_duration * depayload->clock_rate) / 1000;
409 if (max_duration_clock < G_MAXUINT16 &&
410 dtmf_payload.duration > max_duration_clock)
411 dtmf_payload.duration = max_duration_clock;
414 GST_DEBUG_OBJECT (depayload, "Received new RTP DTMF packet : "
415 "marker=%d - timestamp=%u - event=%d - duration=%d",
416 marker, timestamp, dtmf_payload.event, dtmf_payload.duration);
418 GST_DEBUG_OBJECT (depayload,
419 "Previous information : timestamp=%u - duration=%d",
420 rtpdtmfdepay->previous_ts, rtpdtmfdepay->previous_duration);
423 if (marker || rtpdtmfdepay->previous_ts != timestamp) {
424 rtpdtmfdepay->sample = 0;
425 rtpdtmfdepay->previous_ts = timestamp;
426 rtpdtmfdepay->previous_duration = dtmf_payload.duration;
427 rtpdtmfdepay->first_gst_ts = GST_BUFFER_PTS (buf);
429 structure = gst_structure_new ("dtmf-event",
430 "number", G_TYPE_INT, dtmf_payload.event,
431 "volume", G_TYPE_INT, dtmf_payload.volume,
432 "type", G_TYPE_INT, 1, "method", G_TYPE_INT, 1, NULL);
435 gst_message_new_element (GST_OBJECT (depayload), structure);
437 if (!gst_element_post_message (GST_ELEMENT (depayload), dtmf_message)) {
438 GST_ERROR_OBJECT (depayload,
439 "Unable to send dtmf-event message to bus");
442 GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event message");
445 GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event structure");
448 guint16 duration = dtmf_payload.duration;
449 dtmf_payload.duration -= rtpdtmfdepay->previous_duration;
450 /* If late buffer, ignore */
451 if (duration > rtpdtmfdepay->previous_duration)
452 rtpdtmfdepay->previous_duration = duration;
455 GST_DEBUG_OBJECT (depayload, "new previous duration : %d - new duration : %d"
456 " - diff : %d - clock rate : %d - timestamp : %" G_GUINT64_FORMAT,
457 rtpdtmfdepay->previous_duration, dtmf_payload.duration,
458 (rtpdtmfdepay->previous_duration - dtmf_payload.duration),
459 depayload->clock_rate, GST_BUFFER_TIMESTAMP (buf));
461 /* If late or duplicate packet (like the redundant end packet). Ignore */
462 if (dtmf_payload.duration > 0) {
463 outbuf = gst_dtmf_src_generate_tone (rtpdtmfdepay, dtmf_payload);
466 GST_BUFFER_PTS (outbuf) = rtpdtmfdepay->first_gst_ts +
467 (rtpdtmfdepay->previous_duration - dtmf_payload.duration) *
468 GST_SECOND / depayload->clock_rate;
469 GST_BUFFER_OFFSET (outbuf) =
470 (rtpdtmfdepay->previous_duration - dtmf_payload.duration) *
471 GST_SECOND / depayload->clock_rate;
472 GST_BUFFER_OFFSET_END (outbuf) = rtpdtmfdepay->previous_duration *
473 GST_SECOND / depayload->clock_rate;
475 GST_DEBUG_OBJECT (depayload,
476 "timestamp : %" G_GUINT64_FORMAT " - time %" GST_TIME_FORMAT,
477 GST_BUFFER_TIMESTAMP (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
481 gst_rtp_buffer_unmap (&rtpbuffer);
486 GST_ELEMENT_WARNING (rtpdtmfdepay, STREAM, DECODE,
487 ("Packet did not validate"), (NULL));
489 if (rtpbuffer.buffer != NULL)
490 gst_rtp_buffer_unmap (&rtpbuffer);