1 /* GStreamer RTP DTMF source
5 * Copyright (C) <2007> Nokia Corporation.
6 * Contact: Zeeshan Ali <zeeshan.ali@nokia.com>
7 * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
8 * 2000,2005 Wim Taymans <wim@fluendo.com>
10 * This library is free software; you can redistribute it and/or
11 * modify it under the terms of the GNU Library General Public
12 * License as published by the Free Software Foundation; either
13 * version 2 of the License, or (at your option) any later version.
15 * This library is distributed in the hope that it will be useful,
16 * but WITHOUT ANY WARRANTY; without even the implied warranty of
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18 * Library General Public License for more details.
20 * You should have received a copy of the GNU Library General Public
21 * License along with this library; if not, write to the
22 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
23 * Boston, MA 02111-1307, USA.
27 * SECTION:element-rtpdtmfsrc
28 * @see_also: dtmfsrc, rtpdtmfdepay, rtpdtmfmux
30 * The RTPDTMFSrc element generates RTP DTMF (RFC 2833) event packets on request
31 * from application. The application communicates the beginning and end of a
32 * DTMF event using custom upstream gstreamer events. To report a DTMF event, an
33 * application must send an event of type GST_EVENT_CUSTOM_UPSTREAM, having a
34 * structure of name "dtmf-event" with fields set according to the following
39 * <colspec colname='Name' />
40 * <colspec colname='Type' />
41 * <colspec colname='Possible values' />
42 * <colspec colname='Purpose' />
46 * <entry>GType</entry>
47 * <entry>Possible values</entry>
48 * <entry>Purpose</entry>
54 * <entry>G_TYPE_INT</entry>
56 * <entry>The application uses this field to specify which of the two methods
57 * specified in RFC 2833 to use. The value should be 0 for tones and 1 for
58 * named events. Tones are specified by their frequencies and events are specied
59 * by their number. This element can only take events as input. Do not confuse
60 * with "method" which specified the output.
64 * <entry>number</entry>
65 * <entry>G_TYPE_INT</entry>
67 * <entry>The event number.</entry>
70 * <entry>volume</entry>
71 * <entry>G_TYPE_INT</entry>
73 * <entry>This field describes the power level of the tone, expressed in dBm0
74 * after dropping the sign. Power levels range from 0 to -63 dBm0. The range of
75 * valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE.
79 * <entry>start</entry>
80 * <entry>G_TYPE_BOOLEAN</entry>
81 * <entry>True or False</entry>
82 * <entry>Whether the event is starting or ending.</entry>
85 * <entry>method</entry>
86 * <entry>G_TYPE_INT</entry>
88 * <entry>The method used for sending event, this element will react if this
89 * field is absent or 1.
96 * For example, the following code informs the pipeline (and in turn, the
97 * RTPDTMFSrc element inside the pipeline) about the start of an RTP DTMF named
98 * event '1' of volume -25 dBm0:
101 * structure = gst_structure_new ("dtmf-event",
102 * "type", G_TYPE_INT, 1,
103 * "number", G_TYPE_INT, 1,
104 * "volume", G_TYPE_INT, 25,
105 * "start", G_TYPE_BOOLEAN, TRUE, NULL);
107 * event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure);
108 * gst_element_send_event (pipeline, event);
111 * When a DTMF tone actually starts or stop, a "dtmf-event-processed"
112 * element #GstMessage with the same fields as the "dtmf-event"
113 * #GstEvent that was used to request the event. Also, if any event
114 * has not been processed when the element goes from the PAUSED to the
115 * READY state, then a "dtmf-event-dropped" message is posted on the
116 * #GstBus in the order that they were received.
128 #include "gstrtpdtmfsrc.h"
130 #define GST_RTP_DTMF_TYPE_EVENT 1
131 #define DEFAULT_PTIME 40 /* ms */
132 #define DEFAULT_SSRC -1
133 #define DEFAULT_PT 96
134 #define DEFAULT_TIMESTAMP_OFFSET -1
135 #define DEFAULT_SEQNUM_OFFSET -1
136 #define DEFAULT_CLOCK_RATE 8000
138 #define DEFAULT_PACKET_REDUNDANCY 1
139 #define MIN_PACKET_REDUNDANCY 1
140 #define MAX_PACKET_REDUNDANCY 5
142 GST_DEBUG_CATEGORY_STATIC (gst_rtp_dtmf_src_debug);
143 #define GST_CAT_DEFAULT gst_rtp_dtmf_src_debug
145 /* signals and args */
156 PROP_TIMESTAMP_OFFSET,
165 static GstStaticPadTemplate gst_rtp_dtmf_src_template =
166 GST_STATIC_PAD_TEMPLATE ("src",
169 GST_STATIC_CAPS ("application/x-rtp, "
170 "media = (string) \"audio\", "
171 "payload = (int) [ 96, 127 ], "
172 "clock-rate = (int) [ 0, MAX ], "
173 "ssrc = (int) [ 0, MAX ], "
174 "encoding-name = (string) \"TELEPHONE-EVENT\"")
175 /* "events = (string) \"0-15\" */
179 GST_BOILERPLATE (GstRTPDTMFSrc, gst_rtp_dtmf_src, GstBaseSrc,
182 static void gst_rtp_dtmf_src_finalize (GObject * object);
184 static void gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id,
185 const GValue * value, GParamSpec * pspec);
186 static void gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id,
187 GValue * value, GParamSpec * pspec);
188 static gboolean gst_rtp_dtmf_src_handle_event (GstBaseSrc * basesrc,
190 static GstStateChangeReturn gst_rtp_dtmf_src_change_state (GstElement * element,
191 GstStateChange transition);
192 static void gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc * dtmfsrc,
193 gint event_number, gint event_volume);
194 static void gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc * dtmfsrc);
196 static gboolean gst_rtp_dtmf_src_unlock (GstBaseSrc * src);
197 static gboolean gst_rtp_dtmf_src_unlock_stop (GstBaseSrc * src);
198 static GstFlowReturn gst_rtp_dtmf_src_create (GstBaseSrc * basesrc,
199 guint64 offset, guint length, GstBuffer ** buffer);
200 static gboolean gst_rtp_dtmf_src_negotiate (GstBaseSrc * basesrc);
204 gst_rtp_dtmf_src_base_init (gpointer g_class)
206 GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
208 GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_src_debug,
209 "rtpdtmfsrc", 0, "rtpdtmfsrc element");
211 gst_element_class_add_static_pad_template (element_class,
212 &gst_rtp_dtmf_src_template);
214 gst_element_class_set_details_simple (element_class,
215 "RTP DTMF packet generator", "Source/Network",
216 "Generates RTP DTMF packets", "Zeeshan Ali <zeeshan.ali@nokia.com>");
220 gst_rtp_dtmf_src_class_init (GstRTPDTMFSrcClass * klass)
222 GObjectClass *gobject_class;
223 GstBaseSrcClass *gstbasesrc_class;
224 GstElementClass *gstelement_class;
226 gobject_class = G_OBJECT_CLASS (klass);
227 gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
228 gstelement_class = GST_ELEMENT_CLASS (klass);
230 parent_class = g_type_class_peek_parent (klass);
232 gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_finalize);
233 gobject_class->set_property =
234 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_set_property);
235 gobject_class->get_property =
236 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_get_property);
238 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_TIMESTAMP,
239 g_param_spec_uint ("timestamp", "Timestamp",
240 "The RTP timestamp of the last processed packet",
241 0, G_MAXUINT, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
242 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM,
243 g_param_spec_uint ("seqnum", "Sequence number",
244 "The RTP sequence number of the last processed packet",
245 0, G_MAXUINT, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
246 g_object_class_install_property (G_OBJECT_CLASS (klass),
247 PROP_TIMESTAMP_OFFSET, g_param_spec_int ("timestamp-offset",
249 "Offset to add to all outgoing timestamps (-1 = random)", -1,
250 G_MAXINT, DEFAULT_TIMESTAMP_OFFSET,
251 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
252 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM_OFFSET,
253 g_param_spec_int ("seqnum-offset", "Sequence number Offset",
254 "Offset to add to all outgoing seqnum (-1 = random)", -1, G_MAXINT,
255 DEFAULT_SEQNUM_OFFSET, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
256 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_CLOCK_RATE,
257 g_param_spec_uint ("clock-rate", "clockrate",
258 "The clock-rate at which to generate the dtmf packets",
259 0, G_MAXUINT, DEFAULT_CLOCK_RATE,
260 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
261 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SSRC,
262 g_param_spec_uint ("ssrc", "SSRC",
263 "The SSRC of the packets (-1 == random)",
264 0, G_MAXUINT, DEFAULT_SSRC,
265 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
266 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PT,
267 g_param_spec_uint ("pt", "payload type",
268 "The payload type of the packets",
269 0, 0x80, DEFAULT_PT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
270 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_REDUNDANCY,
271 g_param_spec_uint ("packet-redundancy", "Packet Redundancy",
272 "Number of packets to send to indicate start and stop dtmf events",
273 MIN_PACKET_REDUNDANCY, MAX_PACKET_REDUNDANCY,
274 DEFAULT_PACKET_REDUNDANCY,
275 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
277 gstelement_class->change_state =
278 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_change_state);
280 gstbasesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_unlock);
281 gstbasesrc_class->unlock_stop =
282 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_unlock_stop);
284 gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_handle_event);
285 gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_create);
286 gstbasesrc_class->negotiate = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_negotiate);
290 gst_rtp_dtmf_src_event_free (GstRTPDTMFSrcEvent * event)
294 g_slice_free (GstRTPDTMFPayload, event->payload);
295 g_slice_free (GstRTPDTMFSrcEvent, event);
300 gst_rtp_dtmf_src_init (GstRTPDTMFSrc * object, GstRTPDTMFSrcClass * g_class)
302 gst_base_src_set_format (GST_BASE_SRC (object), GST_FORMAT_TIME);
303 gst_base_src_set_live (GST_BASE_SRC (object), TRUE);
305 object->ssrc = DEFAULT_SSRC;
306 object->seqnum_offset = DEFAULT_SEQNUM_OFFSET;
307 object->ts_offset = DEFAULT_TIMESTAMP_OFFSET;
308 object->pt = DEFAULT_PT;
309 object->clock_rate = DEFAULT_CLOCK_RATE;
310 object->ptime = DEFAULT_PTIME;
311 object->packet_redundancy = DEFAULT_PACKET_REDUNDANCY;
313 object->event_queue =
314 g_async_queue_new_full ((GDestroyNotify) gst_rtp_dtmf_src_event_free);
315 object->payload = NULL;
317 GST_DEBUG_OBJECT (object, "init done");
321 gst_rtp_dtmf_src_finalize (GObject * object)
323 GstRTPDTMFSrc *dtmfsrc;
325 dtmfsrc = GST_RTP_DTMF_SRC (object);
327 if (dtmfsrc->event_queue) {
328 g_async_queue_unref (dtmfsrc->event_queue);
329 dtmfsrc->event_queue = NULL;
333 G_OBJECT_CLASS (parent_class)->finalize (object);
337 gst_rtp_dtmf_src_handle_dtmf_event (GstRTPDTMFSrc * dtmfsrc,
338 const GstStructure * event_structure)
343 GstClockTime last_stop;
346 gboolean correct_order;
348 if (!gst_structure_get_int (event_structure, "type", &event_type) ||
349 !gst_structure_get_boolean (event_structure, "start", &start) ||
350 event_type != GST_RTP_DTMF_TYPE_EVENT)
353 if (gst_structure_get_int (event_structure, "method", &method)) {
360 if (!gst_structure_get_int (event_structure, "number", &event_number) ||
361 !gst_structure_get_int (event_structure, "volume", &event_volume))
364 GST_OBJECT_LOCK (dtmfsrc);
365 if (gst_structure_get_clock_time (event_structure, "last-stop", &last_stop))
366 dtmfsrc->last_stop = last_stop;
368 dtmfsrc->last_stop = GST_CLOCK_TIME_NONE;
369 correct_order = (start != dtmfsrc->last_event_was_start);
370 dtmfsrc->last_event_was_start = start;
371 GST_OBJECT_UNLOCK (dtmfsrc);
377 if (!gst_structure_get_int (event_structure, "number", &event_number) ||
378 !gst_structure_get_int (event_structure, "volume", &event_volume))
381 GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d",
382 event_number, event_volume);
383 gst_rtp_dtmf_src_add_start_event (dtmfsrc, event_number, event_volume);
387 GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
388 gst_rtp_dtmf_src_add_stop_event (dtmfsrc);
397 gst_rtp_dtmf_src_handle_custom_upstream (GstRTPDTMFSrc * dtmfsrc,
400 gboolean result = FALSE;
402 const GstStructure *structure;
405 GstStateChangeReturn ret;
407 ret = gst_element_get_state (GST_ELEMENT (dtmfsrc), &state, NULL, 0);
408 if (ret != GST_STATE_CHANGE_SUCCESS || state != GST_STATE_PLAYING) {
409 GST_DEBUG_OBJECT (dtmfsrc, "Received event while not in PLAYING state");
413 GST_DEBUG_OBJECT (dtmfsrc, "Received event is of our interest");
414 structure = gst_event_get_structure (event);
415 struct_str = gst_structure_to_string (structure);
416 GST_DEBUG_OBJECT (dtmfsrc, "Event has structure %s", struct_str);
418 if (structure && gst_structure_has_name (structure, "dtmf-event"))
419 result = gst_rtp_dtmf_src_handle_dtmf_event (dtmfsrc, structure);
426 gst_rtp_dtmf_src_handle_event (GstBaseSrc * basesrc, GstEvent * event)
428 GstRTPDTMFSrc *dtmfsrc;
429 gboolean result = FALSE;
431 dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
433 GST_DEBUG_OBJECT (dtmfsrc, "Received an event on the src pad");
434 if (GST_EVENT_TYPE (event) == GST_EVENT_CUSTOM_UPSTREAM) {
435 result = gst_rtp_dtmf_src_handle_custom_upstream (dtmfsrc, event);
442 gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id,
443 const GValue * value, GParamSpec * pspec)
445 GstRTPDTMFSrc *dtmfsrc;
447 dtmfsrc = GST_RTP_DTMF_SRC (object);
450 case PROP_TIMESTAMP_OFFSET:
451 dtmfsrc->ts_offset = g_value_get_int (value);
453 case PROP_SEQNUM_OFFSET:
454 dtmfsrc->seqnum_offset = g_value_get_int (value);
456 case PROP_CLOCK_RATE:
457 dtmfsrc->clock_rate = g_value_get_uint (value);
458 dtmfsrc->dirty = TRUE;
461 dtmfsrc->ssrc = g_value_get_uint (value);
464 dtmfsrc->pt = g_value_get_uint (value);
465 dtmfsrc->dirty = TRUE;
467 case PROP_REDUNDANCY:
468 dtmfsrc->packet_redundancy = g_value_get_uint (value);
471 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
477 gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id, GValue * value,
480 GstRTPDTMFSrc *dtmfsrc;
482 dtmfsrc = GST_RTP_DTMF_SRC (object);
485 case PROP_TIMESTAMP_OFFSET:
486 g_value_set_int (value, dtmfsrc->ts_offset);
488 case PROP_SEQNUM_OFFSET:
489 g_value_set_int (value, dtmfsrc->seqnum_offset);
491 case PROP_CLOCK_RATE:
492 g_value_set_uint (value, dtmfsrc->clock_rate);
495 g_value_set_uint (value, dtmfsrc->ssrc);
498 g_value_set_uint (value, dtmfsrc->pt);
501 g_value_set_uint (value, dtmfsrc->rtp_timestamp);
504 g_value_set_uint (value, dtmfsrc->seqnum);
506 case PROP_REDUNDANCY:
507 g_value_set_uint (value, dtmfsrc->packet_redundancy);
510 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
516 gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc * dtmfsrc)
518 GstClockTime last_stop;
520 GST_OBJECT_LOCK (dtmfsrc);
521 last_stop = dtmfsrc->last_stop;
522 GST_OBJECT_UNLOCK (dtmfsrc);
524 if (GST_CLOCK_TIME_IS_VALID (last_stop)) {
525 dtmfsrc->start_timestamp = last_stop;
527 GstClock *clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
532 dtmfsrc->start_timestamp = gst_clock_get_time (clock)
533 - gst_element_get_base_time (GST_ELEMENT (dtmfsrc));
534 gst_object_unref (clock);
537 /* If the last stop was in the past, then lets add the buffers together */
538 if (dtmfsrc->start_timestamp < dtmfsrc->timestamp)
539 dtmfsrc->start_timestamp = dtmfsrc->timestamp;
541 dtmfsrc->timestamp = dtmfsrc->start_timestamp;
543 dtmfsrc->rtp_timestamp = dtmfsrc->ts_base +
544 gst_util_uint64_scale_int (gst_segment_to_running_time (&GST_BASE_SRC
545 (dtmfsrc)->segment, GST_FORMAT_TIME, dtmfsrc->timestamp),
546 dtmfsrc->clock_rate, GST_SECOND);
553 gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc * dtmfsrc, gint event_number,
557 GstRTPDTMFSrcEvent *event = g_slice_new0 (GstRTPDTMFSrcEvent);
558 event->event_type = RTP_DTMF_EVENT_TYPE_START;
560 event->payload = g_slice_new0 (GstRTPDTMFPayload);
561 event->payload->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
562 event->payload->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
563 event->payload->duration = dtmfsrc->ptime * dtmfsrc->clock_rate / 1000;
565 g_async_queue_push (dtmfsrc->event_queue, event);
569 gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc * dtmfsrc)
572 GstRTPDTMFSrcEvent *event = g_slice_new0 (GstRTPDTMFSrcEvent);
573 event->event_type = RTP_DTMF_EVENT_TYPE_STOP;
575 g_async_queue_push (dtmfsrc->event_queue, event);
580 gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc * dtmfsrc, GstBuffer * buf)
582 gst_rtp_buffer_set_ssrc (buf, dtmfsrc->current_ssrc);
583 gst_rtp_buffer_set_payload_type (buf, dtmfsrc->pt);
584 /* Only the very first packet gets a marker */
585 if (dtmfsrc->first_packet) {
586 gst_rtp_buffer_set_marker (buf, TRUE);
587 } else if (dtmfsrc->last_packet) {
588 dtmfsrc->payload->e = 1;
592 gst_rtp_buffer_set_seq (buf, dtmfsrc->seqnum);
594 /* timestamp of RTP header */
595 gst_rtp_buffer_set_timestamp (buf, dtmfsrc->rtp_timestamp);
599 gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc * dtmfsrc, GstBuffer * buf)
601 GstRTPDTMFPayload *payload;
603 gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc, buf);
605 /* timestamp and duration of GstBuffer */
606 /* Redundant buffer have no duration ... */
607 if (dtmfsrc->redundancy_count > 1)
608 GST_BUFFER_DURATION (buf) = 0;
610 GST_BUFFER_DURATION (buf) = dtmfsrc->ptime * GST_MSECOND;
611 GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp;
614 payload = (GstRTPDTMFPayload *) gst_rtp_buffer_get_payload (buf);
616 /* copy payload and convert to network-byte order */
617 g_memmove (payload, dtmfsrc->payload, sizeof (GstRTPDTMFPayload));
619 payload->duration = g_htons (payload->duration);
621 if (dtmfsrc->redundancy_count <= 1 && dtmfsrc->last_packet) {
622 GstClockTime inter_digit_interval = MIN_INTER_DIGIT_INTERVAL;
624 if (inter_digit_interval % dtmfsrc->ptime != 0)
625 inter_digit_interval += dtmfsrc->ptime -
626 (MIN_INTER_DIGIT_INTERVAL % dtmfsrc->ptime);
628 GST_BUFFER_DURATION (buf) += inter_digit_interval * GST_MSECOND;
631 GST_LOG_OBJECT (dtmfsrc, "Creating new buffer with event %u duration "
632 " gst: %" GST_TIME_FORMAT " at %" GST_TIME_FORMAT "(rtp ts:%u dur:%u)",
633 dtmfsrc->payload->event, GST_TIME_ARGS (GST_BUFFER_DURATION (buf)),
634 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), dtmfsrc->rtp_timestamp,
635 dtmfsrc->payload->duration);
637 /* duration of DTMF payloadfor the NEXT packet */
638 /* not updated for redundant packets */
639 if (dtmfsrc->redundancy_count <= 1)
640 dtmfsrc->payload->duration += dtmfsrc->ptime * dtmfsrc->clock_rate / 1000;
642 if (GST_CLOCK_TIME_IS_VALID (dtmfsrc->timestamp))
643 dtmfsrc->timestamp += GST_BUFFER_DURATION (buf);
648 gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc * dtmfsrc)
650 GstBuffer *buf = NULL;
652 /* create buffer to hold the payload */
653 buf = gst_rtp_buffer_new_allocate (sizeof (GstRTPDTMFPayload), 0, 0);
655 gst_rtp_dtmf_prepare_buffer_data (dtmfsrc, buf);
657 /* Set caps on the buffer before pushing it */
658 gst_buffer_set_caps (buf, GST_PAD_CAPS (GST_BASE_SRC_PAD (dtmfsrc)));
665 gst_dtmf_src_post_message (GstRTPDTMFSrc * dtmfsrc, const gchar * message_name,
666 GstRTPDTMFSrcEvent * event)
668 GstStructure *s = NULL;
670 switch (event->event_type) {
671 case RTP_DTMF_EVENT_TYPE_START:
672 s = gst_structure_new (message_name,
673 "type", G_TYPE_INT, 1,
674 "method", G_TYPE_INT, 1,
675 "start", G_TYPE_BOOLEAN, TRUE,
676 "number", G_TYPE_INT, event->payload->event,
677 "volume", G_TYPE_INT, event->payload->volume, NULL);
679 case RTP_DTMF_EVENT_TYPE_STOP:
680 s = gst_structure_new (message_name,
681 "type", G_TYPE_INT, 1, "method", G_TYPE_INT, 1,
682 "start", G_TYPE_BOOLEAN, FALSE, NULL);
684 case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
689 gst_element_post_message (GST_ELEMENT (dtmfsrc),
690 gst_message_new_element (GST_OBJECT (dtmfsrc), s));
695 gst_rtp_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
696 guint length, GstBuffer ** buffer)
698 GstRTPDTMFSrcEvent *event;
699 GstRTPDTMFSrc *dtmfsrc;
702 GstClockReturn clockret;
704 dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
708 if (dtmfsrc->payload == NULL) {
709 GST_DEBUG_OBJECT (dtmfsrc, "popping");
710 event = g_async_queue_pop (dtmfsrc->event_queue);
712 GST_DEBUG_OBJECT (dtmfsrc, "popped %d", event->event_type);
714 switch (event->event_type) {
715 case RTP_DTMF_EVENT_TYPE_STOP:
716 GST_WARNING_OBJECT (dtmfsrc,
717 "Received a DTMF stop event when already stopped");
718 gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
721 case RTP_DTMF_EVENT_TYPE_START:
722 dtmfsrc->first_packet = TRUE;
723 dtmfsrc->last_packet = FALSE;
724 /* Set the redundancy on the first packet */
725 dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
726 if (!gst_rtp_dtmf_prepare_timestamps (dtmfsrc))
729 gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-processed", event);
730 dtmfsrc->payload = event->payload;
731 event->payload = NULL;
734 case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
736 * We're pushing it back because it has to stay in there until
737 * the task is really paused (and the queue will then be flushed
739 GST_OBJECT_LOCK (dtmfsrc);
740 if (dtmfsrc->paused) {
741 g_async_queue_push (dtmfsrc->event_queue, event);
744 GST_OBJECT_UNLOCK (dtmfsrc);
748 gst_rtp_dtmf_src_event_free (event);
749 } else if (!dtmfsrc->first_packet && !dtmfsrc->last_packet &&
750 (dtmfsrc->timestamp - dtmfsrc->start_timestamp) / GST_MSECOND >=
751 MIN_PULSE_DURATION) {
752 GST_DEBUG_OBJECT (dtmfsrc, "try popping");
753 event = g_async_queue_try_pop (dtmfsrc->event_queue);
757 GST_DEBUG_OBJECT (dtmfsrc, "try popped %d", event->event_type);
759 switch (event->event_type) {
760 case RTP_DTMF_EVENT_TYPE_START:
761 GST_WARNING_OBJECT (dtmfsrc,
762 "Received two consecutive DTMF start events");
763 gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
766 case RTP_DTMF_EVENT_TYPE_STOP:
767 dtmfsrc->first_packet = FALSE;
768 dtmfsrc->last_packet = TRUE;
769 /* Set the redundancy on the last packet */
770 dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
771 gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-processed", event);
774 case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
776 * We're pushing it back because it has to stay in there until
777 * the task is really paused (and the queue will then be flushed)
779 GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
780 GST_OBJECT_LOCK (dtmfsrc);
781 if (dtmfsrc->paused) {
782 g_async_queue_push (dtmfsrc->event_queue, event);
785 GST_OBJECT_UNLOCK (dtmfsrc);
788 gst_rtp_dtmf_src_event_free (event);
791 } while (dtmfsrc->payload == NULL);
794 GST_DEBUG_OBJECT (dtmfsrc, "Processed events, now lets wait on the clock");
796 clock = gst_element_get_clock (GST_ELEMENT (basesrc));
799 clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
800 gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
801 gst_object_unref (clock);
803 GST_OBJECT_LOCK (dtmfsrc);
804 if (!dtmfsrc->paused) {
805 dtmfsrc->clockid = clockid;
806 GST_OBJECT_UNLOCK (dtmfsrc);
808 clockret = gst_clock_id_wait (clockid, NULL);
810 GST_OBJECT_LOCK (dtmfsrc);
812 clockret = GST_CLOCK_UNSCHEDULED;
814 clockret = GST_CLOCK_UNSCHEDULED;
816 gst_clock_id_unref (clockid);
817 dtmfsrc->clockid = NULL;
818 GST_OBJECT_UNLOCK (dtmfsrc);
820 if (clockret == GST_CLOCK_UNSCHEDULED) {
827 if (!gst_rtp_dtmf_src_negotiate (basesrc))
828 return GST_FLOW_NOT_NEGOTIATED;
830 /* create buffer to hold the payload */
831 *buffer = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc);
833 if (dtmfsrc->redundancy_count)
834 dtmfsrc->redundancy_count--;
836 /* Only the very first one has a marker */
837 dtmfsrc->first_packet = FALSE;
839 /* This is the end of the event */
840 if (dtmfsrc->last_packet == TRUE && dtmfsrc->redundancy_count == 0) {
842 g_slice_free (GstRTPDTMFPayload, dtmfsrc->payload);
843 dtmfsrc->payload = NULL;
845 dtmfsrc->last_packet = FALSE;
852 GST_OBJECT_UNLOCK (dtmfsrc);
856 if (dtmfsrc->payload) {
857 dtmfsrc->first_packet = FALSE;
858 dtmfsrc->last_packet = TRUE;
859 /* Set the redundanc on the last packet */
860 dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
863 return GST_FLOW_WRONG_STATE;
867 GST_ELEMENT_ERROR (dtmfsrc, STREAM, MUX, ("No available clock"),
868 ("No available clock"));
869 gst_pad_pause_task (GST_BASE_SRC_PAD (dtmfsrc));
870 return GST_FLOW_ERROR;
875 gst_rtp_dtmf_src_negotiate (GstBaseSrc * basesrc)
877 GstCaps *srccaps, *peercaps;
878 GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
881 /* fill in the defaults, there properties cannot be negotiated. */
882 srccaps = gst_caps_new_simple ("application/x-rtp",
883 "media", G_TYPE_STRING, "audio",
884 "encoding-name", G_TYPE_STRING, "TELEPHONE-EVENT", NULL);
886 /* the peer caps can override some of the defaults */
887 peercaps = gst_pad_peer_get_caps (GST_BASE_SRC_PAD (basesrc));
888 if (peercaps == NULL) {
889 /* no peer caps, just add the other properties */
890 gst_caps_set_simple (srccaps,
891 "payload", G_TYPE_INT, dtmfsrc->pt,
892 "ssrc", G_TYPE_UINT, dtmfsrc->current_ssrc,
893 "clock-base", G_TYPE_UINT, dtmfsrc->ts_base,
894 "clock-rate", G_TYPE_INT, dtmfsrc->clock_rate,
895 "seqnum-base", G_TYPE_UINT, dtmfsrc->seqnum_base, NULL);
897 GST_DEBUG_OBJECT (dtmfsrc, "no peer caps: %" GST_PTR_FORMAT, srccaps);
905 /* peer provides caps we can use to fixate, intersect. This always returns a
907 temp = gst_caps_intersect (srccaps, peercaps);
908 gst_caps_unref (srccaps);
909 gst_caps_unref (peercaps);
912 GST_DEBUG_OBJECT (dtmfsrc, "Could not get intersection with peer caps");
916 if (gst_caps_is_empty (temp)) {
917 GST_DEBUG_OBJECT (dtmfsrc, "Intersection with peer caps is empty");
918 gst_caps_unref (temp);
922 /* now fixate, start by taking the first caps */
923 gst_caps_truncate (temp);
926 /* get first structure */
927 s = gst_caps_get_structure (srccaps, 0);
929 if (gst_structure_get_int (s, "payload", &pt)) {
932 GST_LOG_OBJECT (dtmfsrc, "using peer pt %d", pt);
934 if (gst_structure_has_field (s, "payload")) {
935 /* can only fixate if there is a field */
936 gst_structure_fixate_field_nearest_int (s, "payload", dtmfsrc->pt);
937 gst_structure_get_int (s, "payload", &pt);
938 GST_LOG_OBJECT (dtmfsrc, "using peer pt %d", pt);
940 /* no pt field, use the internal pt */
942 gst_structure_set (s, "payload", G_TYPE_INT, pt, NULL);
943 GST_LOG_OBJECT (dtmfsrc, "using internal pt %d", pt);
947 if (gst_structure_get_int (s, "clock-rate", &clock_rate)) {
948 dtmfsrc->clock_rate = clock_rate;
949 GST_LOG_OBJECT (dtmfsrc, "using clock-rate from caps %d",
950 dtmfsrc->clock_rate);
952 GST_LOG_OBJECT (dtmfsrc, "using existing clock-rate %d",
953 dtmfsrc->clock_rate);
955 gst_structure_set (s, "clock-rate", G_TYPE_INT, dtmfsrc->clock_rate, NULL);
958 if (gst_structure_has_field_typed (s, "ssrc", G_TYPE_UINT)) {
959 value = gst_structure_get_value (s, "ssrc");
960 dtmfsrc->current_ssrc = g_value_get_uint (value);
961 GST_LOG_OBJECT (dtmfsrc, "using peer ssrc %08x", dtmfsrc->current_ssrc);
963 /* FIXME, fixate_nearest_uint would be even better */
964 gst_structure_set (s, "ssrc", G_TYPE_UINT, dtmfsrc->current_ssrc, NULL);
965 GST_LOG_OBJECT (dtmfsrc, "using internal ssrc %08x",
966 dtmfsrc->current_ssrc);
969 if (gst_structure_has_field_typed (s, "clock-base", G_TYPE_UINT)) {
970 value = gst_structure_get_value (s, "clock-base");
971 dtmfsrc->ts_base = g_value_get_uint (value);
972 GST_LOG_OBJECT (dtmfsrc, "using peer clock-base %u", dtmfsrc->ts_base);
974 /* FIXME, fixate_nearest_uint would be even better */
975 gst_structure_set (s, "clock-base", G_TYPE_UINT, dtmfsrc->ts_base, NULL);
976 GST_LOG_OBJECT (dtmfsrc, "using internal clock-base %u",
979 if (gst_structure_has_field_typed (s, "seqnum-base", G_TYPE_UINT)) {
980 value = gst_structure_get_value (s, "seqnum-base");
981 dtmfsrc->seqnum_base = g_value_get_uint (value);
982 GST_LOG_OBJECT (dtmfsrc, "using peer seqnum-base %u",
983 dtmfsrc->seqnum_base);
985 /* FIXME, fixate_nearest_uint would be even better */
986 gst_structure_set (s, "seqnum-base", G_TYPE_UINT, dtmfsrc->seqnum_base,
988 GST_LOG_OBJECT (dtmfsrc, "using internal seqnum-base %u",
989 dtmfsrc->seqnum_base);
992 if (gst_structure_has_field_typed (s, "ptime", G_TYPE_UINT)) {
993 value = gst_structure_get_value (s, "ptime");
994 dtmfsrc->ptime = g_value_get_uint (value);
995 GST_LOG_OBJECT (dtmfsrc, "using peer ptime %u", dtmfsrc->ptime);
996 } else if (gst_structure_has_field_typed (s, "maxptime", G_TYPE_UINT)) {
997 value = gst_structure_get_value (s, "maxptime");
998 dtmfsrc->ptime = g_value_get_uint (value);
999 GST_LOG_OBJECT (dtmfsrc, "using peer maxptime as ptime %u",
1002 /* FIXME, fixate_nearest_uint would be even better */
1003 gst_structure_set (s, "ptime", G_TYPE_UINT, dtmfsrc->ptime, NULL);
1004 GST_LOG_OBJECT (dtmfsrc, "using internal ptime %u", dtmfsrc->ptime);
1008 GST_DEBUG_OBJECT (dtmfsrc, "with peer caps: %" GST_PTR_FORMAT, srccaps);
1011 ret = gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), srccaps);
1012 gst_caps_unref (srccaps);
1014 dtmfsrc->dirty = FALSE;
1022 gst_rtp_dtmf_src_ready_to_paused (GstRTPDTMFSrc * dtmfsrc)
1024 if (dtmfsrc->ssrc == -1)
1025 dtmfsrc->current_ssrc = g_random_int ();
1027 dtmfsrc->current_ssrc = dtmfsrc->ssrc;
1029 if (dtmfsrc->seqnum_offset == -1)
1030 dtmfsrc->seqnum_base = g_random_int_range (0, G_MAXUINT16);
1032 dtmfsrc->seqnum_base = dtmfsrc->seqnum_offset;
1033 dtmfsrc->seqnum = dtmfsrc->seqnum_base;
1035 if (dtmfsrc->ts_offset == -1)
1036 dtmfsrc->ts_base = g_random_int ();
1038 dtmfsrc->ts_base = dtmfsrc->ts_offset;
1040 dtmfsrc->timestamp = 0;
1043 static GstStateChangeReturn
1044 gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition)
1046 GstRTPDTMFSrc *dtmfsrc;
1047 GstStateChangeReturn result;
1048 gboolean no_preroll = FALSE;
1049 GstRTPDTMFSrcEvent *event = NULL;
1051 dtmfsrc = GST_RTP_DTMF_SRC (element);
1053 switch (transition) {
1054 case GST_STATE_CHANGE_READY_TO_PAUSED:
1055 gst_rtp_dtmf_src_ready_to_paused (dtmfsrc);
1057 /* Flushing the event queue */
1058 while ((event = g_async_queue_try_pop (dtmfsrc->event_queue)) != NULL) {
1059 gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
1060 gst_rtp_dtmf_src_event_free (event);
1062 dtmfsrc->last_event_was_start = FALSE;
1071 GST_ELEMENT_CLASS (parent_class)->change_state (element,
1072 transition)) == GST_STATE_CHANGE_FAILURE)
1075 switch (transition) {
1076 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1079 case GST_STATE_CHANGE_PAUSED_TO_READY:
1081 /* Flushing the event queue */
1082 while ((event = g_async_queue_try_pop (dtmfsrc->event_queue)) != NULL) {
1083 gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
1084 gst_rtp_dtmf_src_event_free (event);
1086 dtmfsrc->last_event_was_start = FALSE;
1088 /* Indicate that we don't do PRE_ROLL */
1095 if (no_preroll && result == GST_STATE_CHANGE_SUCCESS)
1096 result = GST_STATE_CHANGE_NO_PREROLL;
1103 GST_ERROR_OBJECT (dtmfsrc, "parent failed state change");
1110 gst_rtp_dtmf_src_unlock (GstBaseSrc * src)
1112 GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (src);
1113 GstRTPDTMFSrcEvent *event = NULL;
1115 GST_DEBUG_OBJECT (dtmfsrc, "Called unlock");
1117 GST_OBJECT_LOCK (dtmfsrc);
1118 dtmfsrc->paused = TRUE;
1119 if (dtmfsrc->clockid) {
1120 gst_clock_id_unschedule (dtmfsrc->clockid);
1122 GST_OBJECT_UNLOCK (dtmfsrc);
1124 GST_DEBUG_OBJECT (dtmfsrc, "Pushing the PAUSE_TASK event on unlock request");
1125 event = g_slice_new0 (GstRTPDTMFSrcEvent);
1126 event->event_type = RTP_DTMF_EVENT_TYPE_PAUSE_TASK;
1127 g_async_queue_push (dtmfsrc->event_queue, event);
1134 gst_rtp_dtmf_src_unlock_stop (GstBaseSrc * src)
1136 GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (src);
1138 GST_DEBUG_OBJECT (dtmfsrc, "Unlock stopped");
1140 GST_OBJECT_LOCK (dtmfsrc);
1141 dtmfsrc->paused = FALSE;
1142 GST_OBJECT_UNLOCK (dtmfsrc);
1148 gst_rtp_dtmf_src_plugin_init (GstPlugin * plugin)
1150 return gst_element_register (plugin, "rtpdtmfsrc",
1151 GST_RANK_NONE, GST_TYPE_RTP_DTMF_SRC);