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., 51 Franklin St, Fifth Floor,
23 * Boston, MA 02110-1301, 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 "encoding-name = (string) \"TELEPHONE-EVENT\"")
174 /* "events = (string) \"0-15\" */
178 G_DEFINE_TYPE (GstRTPDTMFSrc, gst_rtp_dtmf_src, GST_TYPE_BASE_SRC);
180 static void gst_rtp_dtmf_src_finalize (GObject * object);
182 static void gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id,
183 const GValue * value, GParamSpec * pspec);
184 static void gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id,
185 GValue * value, GParamSpec * pspec);
186 static gboolean gst_rtp_dtmf_src_handle_event (GstBaseSrc * basesrc,
188 static GstStateChangeReturn gst_rtp_dtmf_src_change_state (GstElement * element,
189 GstStateChange transition);
190 static void gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc * dtmfsrc,
191 gint event_number, gint event_volume);
192 static void gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc * dtmfsrc);
194 static gboolean gst_rtp_dtmf_src_unlock (GstBaseSrc * src);
195 static gboolean gst_rtp_dtmf_src_unlock_stop (GstBaseSrc * src);
196 static GstFlowReturn gst_rtp_dtmf_src_create (GstBaseSrc * basesrc,
197 guint64 offset, guint length, GstBuffer ** buffer);
198 static gboolean gst_rtp_dtmf_src_negotiate (GstBaseSrc * basesrc);
199 static gboolean gst_rtp_dtmf_src_query (GstBaseSrc * basesrc, GstQuery * query);
203 gst_rtp_dtmf_src_class_init (GstRTPDTMFSrcClass * klass)
205 GObjectClass *gobject_class;
206 GstBaseSrcClass *gstbasesrc_class;
207 GstElementClass *gstelement_class;
209 gobject_class = G_OBJECT_CLASS (klass);
210 gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
211 gstelement_class = GST_ELEMENT_CLASS (klass);
213 GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_src_debug,
214 "rtpdtmfsrc", 0, "rtpdtmfsrc element");
216 gst_element_class_add_static_pad_template (gstelement_class,
217 &gst_rtp_dtmf_src_template);
219 gst_element_class_set_static_metadata (gstelement_class,
220 "RTP DTMF packet generator", "Source/Network",
221 "Generates RTP DTMF packets", "Zeeshan Ali <zeeshan.ali@nokia.com>");
223 gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_finalize);
224 gobject_class->set_property =
225 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_set_property);
226 gobject_class->get_property =
227 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_get_property);
229 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_TIMESTAMP,
230 g_param_spec_uint ("timestamp", "Timestamp",
231 "The RTP timestamp of the last processed packet",
232 0, G_MAXUINT, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
233 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM,
234 g_param_spec_uint ("seqnum", "Sequence number",
235 "The RTP sequence number of the last processed packet",
236 0, G_MAXUINT, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
237 g_object_class_install_property (G_OBJECT_CLASS (klass),
238 PROP_TIMESTAMP_OFFSET, g_param_spec_int ("timestamp-offset",
240 "Offset to add to all outgoing timestamps (-1 = random)", -1,
241 G_MAXINT, DEFAULT_TIMESTAMP_OFFSET,
242 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
243 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM_OFFSET,
244 g_param_spec_int ("seqnum-offset", "Sequence number Offset",
245 "Offset to add to all outgoing seqnum (-1 = random)", -1, G_MAXINT,
246 DEFAULT_SEQNUM_OFFSET, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
247 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_CLOCK_RATE,
248 g_param_spec_uint ("clock-rate", "clockrate",
249 "The clock-rate at which to generate the dtmf packets",
250 0, G_MAXUINT, DEFAULT_CLOCK_RATE,
251 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
252 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SSRC,
253 g_param_spec_uint ("ssrc", "SSRC",
254 "The SSRC of the packets (-1 == random)",
255 0, G_MAXUINT, DEFAULT_SSRC,
256 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
257 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PT,
258 g_param_spec_uint ("pt", "payload type",
259 "The payload type of the packets",
260 0, 0x80, DEFAULT_PT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
261 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_REDUNDANCY,
262 g_param_spec_uint ("packet-redundancy", "Packet Redundancy",
263 "Number of packets to send to indicate start and stop dtmf events",
264 MIN_PACKET_REDUNDANCY, MAX_PACKET_REDUNDANCY,
265 DEFAULT_PACKET_REDUNDANCY,
266 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
268 gstelement_class->change_state =
269 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_change_state);
271 gstbasesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_unlock);
272 gstbasesrc_class->unlock_stop =
273 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_unlock_stop);
275 gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_handle_event);
276 gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_create);
277 gstbasesrc_class->negotiate = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_negotiate);
278 gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_query);
282 gst_rtp_dtmf_src_event_free (GstRTPDTMFSrcEvent * event)
286 g_slice_free (GstRTPDTMFPayload, event->payload);
287 g_slice_free (GstRTPDTMFSrcEvent, event);
292 gst_rtp_dtmf_src_init (GstRTPDTMFSrc * object)
294 gst_base_src_set_format (GST_BASE_SRC (object), GST_FORMAT_TIME);
295 gst_base_src_set_live (GST_BASE_SRC (object), TRUE);
297 object->ssrc = DEFAULT_SSRC;
298 object->seqnum_offset = DEFAULT_SEQNUM_OFFSET;
299 object->ts_offset = DEFAULT_TIMESTAMP_OFFSET;
300 object->pt = DEFAULT_PT;
301 object->clock_rate = DEFAULT_CLOCK_RATE;
302 object->ptime = DEFAULT_PTIME;
303 object->packet_redundancy = DEFAULT_PACKET_REDUNDANCY;
305 object->event_queue =
306 g_async_queue_new_full ((GDestroyNotify) gst_rtp_dtmf_src_event_free);
307 object->payload = NULL;
309 GST_DEBUG_OBJECT (object, "init done");
313 gst_rtp_dtmf_src_finalize (GObject * object)
315 GstRTPDTMFSrc *dtmfsrc;
317 dtmfsrc = GST_RTP_DTMF_SRC (object);
319 if (dtmfsrc->event_queue) {
320 g_async_queue_unref (dtmfsrc->event_queue);
321 dtmfsrc->event_queue = NULL;
325 G_OBJECT_CLASS (gst_rtp_dtmf_src_parent_class)->finalize (object);
329 gst_rtp_dtmf_src_handle_dtmf_event (GstRTPDTMFSrc * dtmfsrc,
330 const GstStructure * event_structure)
335 GstClockTime last_stop;
338 gboolean correct_order;
340 if (!gst_structure_get_int (event_structure, "type", &event_type) ||
341 !gst_structure_get_boolean (event_structure, "start", &start) ||
342 event_type != GST_RTP_DTMF_TYPE_EVENT)
345 if (gst_structure_get_int (event_structure, "method", &method)) {
352 if (!gst_structure_get_int (event_structure, "number", &event_number) ||
353 !gst_structure_get_int (event_structure, "volume", &event_volume))
356 GST_OBJECT_LOCK (dtmfsrc);
357 if (gst_structure_get_clock_time (event_structure, "last-stop", &last_stop))
358 dtmfsrc->last_stop = last_stop;
360 dtmfsrc->last_stop = GST_CLOCK_TIME_NONE;
361 correct_order = (start != dtmfsrc->last_event_was_start);
362 dtmfsrc->last_event_was_start = start;
363 GST_OBJECT_UNLOCK (dtmfsrc);
369 if (!gst_structure_get_int (event_structure, "number", &event_number) ||
370 !gst_structure_get_int (event_structure, "volume", &event_volume))
373 GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d",
374 event_number, event_volume);
375 gst_rtp_dtmf_src_add_start_event (dtmfsrc, event_number, event_volume);
379 GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
380 gst_rtp_dtmf_src_add_stop_event (dtmfsrc);
389 gst_rtp_dtmf_src_handle_custom_upstream (GstRTPDTMFSrc * dtmfsrc,
392 gboolean result = FALSE;
394 const GstStructure *structure;
397 GstStateChangeReturn ret;
399 ret = gst_element_get_state (GST_ELEMENT (dtmfsrc), &state, NULL, 0);
400 if (ret != GST_STATE_CHANGE_SUCCESS || state != GST_STATE_PLAYING) {
401 GST_DEBUG_OBJECT (dtmfsrc, "Received event while not in PLAYING state");
405 GST_DEBUG_OBJECT (dtmfsrc, "Received event is of our interest");
406 structure = gst_event_get_structure (event);
407 struct_str = gst_structure_to_string (structure);
408 GST_DEBUG_OBJECT (dtmfsrc, "Event has structure %s", struct_str);
410 if (structure && gst_structure_has_name (structure, "dtmf-event"))
411 result = gst_rtp_dtmf_src_handle_dtmf_event (dtmfsrc, structure);
418 gst_rtp_dtmf_src_handle_event (GstBaseSrc * basesrc, GstEvent * event)
420 GstRTPDTMFSrc *dtmfsrc;
421 gboolean result = FALSE;
423 dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
425 GST_DEBUG_OBJECT (dtmfsrc, "Received an event on the src pad");
426 if (GST_EVENT_TYPE (event) == GST_EVENT_CUSTOM_UPSTREAM) {
427 result = gst_rtp_dtmf_src_handle_custom_upstream (dtmfsrc, event);
434 gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id,
435 const GValue * value, GParamSpec * pspec)
437 GstRTPDTMFSrc *dtmfsrc;
439 dtmfsrc = GST_RTP_DTMF_SRC (object);
442 case PROP_TIMESTAMP_OFFSET:
443 dtmfsrc->ts_offset = g_value_get_int (value);
445 case PROP_SEQNUM_OFFSET:
446 dtmfsrc->seqnum_offset = g_value_get_int (value);
448 case PROP_CLOCK_RATE:
449 dtmfsrc->clock_rate = g_value_get_uint (value);
450 dtmfsrc->dirty = TRUE;
453 dtmfsrc->ssrc = g_value_get_uint (value);
456 dtmfsrc->pt = g_value_get_uint (value);
457 dtmfsrc->dirty = TRUE;
459 case PROP_REDUNDANCY:
460 dtmfsrc->packet_redundancy = g_value_get_uint (value);
463 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
469 gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id, GValue * value,
472 GstRTPDTMFSrc *dtmfsrc;
474 dtmfsrc = GST_RTP_DTMF_SRC (object);
477 case PROP_TIMESTAMP_OFFSET:
478 g_value_set_int (value, dtmfsrc->ts_offset);
480 case PROP_SEQNUM_OFFSET:
481 g_value_set_int (value, dtmfsrc->seqnum_offset);
483 case PROP_CLOCK_RATE:
484 g_value_set_uint (value, dtmfsrc->clock_rate);
487 g_value_set_uint (value, dtmfsrc->ssrc);
490 g_value_set_uint (value, dtmfsrc->pt);
493 g_value_set_uint (value, dtmfsrc->rtp_timestamp);
496 g_value_set_uint (value, dtmfsrc->seqnum);
498 case PROP_REDUNDANCY:
499 g_value_set_uint (value, dtmfsrc->packet_redundancy);
502 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
508 gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc * dtmfsrc)
510 GstClockTime last_stop;
512 GST_OBJECT_LOCK (dtmfsrc);
513 last_stop = dtmfsrc->last_stop;
514 GST_OBJECT_UNLOCK (dtmfsrc);
516 if (GST_CLOCK_TIME_IS_VALID (last_stop)) {
517 dtmfsrc->start_timestamp = last_stop;
519 GstClock *clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
524 dtmfsrc->start_timestamp = gst_clock_get_time (clock)
525 - gst_element_get_base_time (GST_ELEMENT (dtmfsrc));
526 gst_object_unref (clock);
529 /* If the last stop was in the past, then lets add the buffers together */
530 if (dtmfsrc->start_timestamp < dtmfsrc->timestamp)
531 dtmfsrc->start_timestamp = dtmfsrc->timestamp;
533 dtmfsrc->timestamp = dtmfsrc->start_timestamp;
535 dtmfsrc->rtp_timestamp = dtmfsrc->ts_base +
536 gst_util_uint64_scale_int (gst_segment_to_running_time (&GST_BASE_SRC
537 (dtmfsrc)->segment, GST_FORMAT_TIME, dtmfsrc->timestamp),
538 dtmfsrc->clock_rate, GST_SECOND);
545 gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc * dtmfsrc, gint event_number,
549 GstRTPDTMFSrcEvent *event = g_slice_new0 (GstRTPDTMFSrcEvent);
550 event->event_type = RTP_DTMF_EVENT_TYPE_START;
552 event->payload = g_slice_new0 (GstRTPDTMFPayload);
553 event->payload->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
554 event->payload->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
556 g_async_queue_push (dtmfsrc->event_queue, event);
560 gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc * dtmfsrc)
563 GstRTPDTMFSrcEvent *event = g_slice_new0 (GstRTPDTMFSrcEvent);
564 event->event_type = RTP_DTMF_EVENT_TYPE_STOP;
566 g_async_queue_push (dtmfsrc->event_queue, event);
571 gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc * dtmfsrc,
572 GstRTPBuffer * rtpbuf)
574 gst_rtp_buffer_set_ssrc (rtpbuf, dtmfsrc->current_ssrc);
575 gst_rtp_buffer_set_payload_type (rtpbuf, dtmfsrc->pt);
576 /* Only the very first packet gets a marker */
577 if (dtmfsrc->first_packet) {
578 gst_rtp_buffer_set_marker (rtpbuf, TRUE);
579 } else if (dtmfsrc->last_packet) {
580 dtmfsrc->payload->e = 1;
584 gst_rtp_buffer_set_seq (rtpbuf, dtmfsrc->seqnum);
586 /* timestamp of RTP header */
587 gst_rtp_buffer_set_timestamp (rtpbuf, dtmfsrc->rtp_timestamp);
591 gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc * dtmfsrc)
594 GstRTPDTMFPayload *payload;
595 GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
597 buf = gst_rtp_buffer_new_allocate (sizeof (GstRTPDTMFPayload), 0, 0);
599 gst_rtp_buffer_map (buf, GST_MAP_READWRITE, &rtpbuffer);
601 gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc, &rtpbuffer);
603 /* timestamp and duration of GstBuffer */
604 /* Redundant buffer have no duration ... */
605 if (dtmfsrc->redundancy_count > 1)
606 GST_BUFFER_DURATION (buf) = 0;
608 GST_BUFFER_DURATION (buf) = dtmfsrc->ptime * GST_MSECOND;
609 GST_BUFFER_PTS (buf) = dtmfsrc->timestamp;
611 payload = (GstRTPDTMFPayload *) gst_rtp_buffer_get_payload (&rtpbuffer);
613 /* copy payload and convert to network-byte order */
614 memmove (payload, dtmfsrc->payload, sizeof (GstRTPDTMFPayload));
616 payload->duration = g_htons (payload->duration);
618 if (dtmfsrc->redundancy_count <= 1 && dtmfsrc->last_packet) {
619 GstClockTime inter_digit_interval = MIN_INTER_DIGIT_INTERVAL;
621 if (inter_digit_interval % dtmfsrc->ptime != 0)
622 inter_digit_interval += dtmfsrc->ptime -
623 (MIN_INTER_DIGIT_INTERVAL % dtmfsrc->ptime);
625 GST_BUFFER_DURATION (buf) += inter_digit_interval * GST_MSECOND;
628 GST_LOG_OBJECT (dtmfsrc, "Creating new buffer with event %u duration "
629 " gst: %" GST_TIME_FORMAT " at %" GST_TIME_FORMAT "(rtp ts:%u dur:%u)",
630 dtmfsrc->payload->event, GST_TIME_ARGS (GST_BUFFER_DURATION (buf)),
631 GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), dtmfsrc->rtp_timestamp,
632 dtmfsrc->payload->duration);
634 /* duration of DTMF payloadfor the NEXT packet */
635 /* not updated for redundant packets */
636 if (dtmfsrc->redundancy_count <= 1)
637 dtmfsrc->payload->duration += dtmfsrc->ptime * dtmfsrc->clock_rate / 1000;
639 if (GST_CLOCK_TIME_IS_VALID (dtmfsrc->timestamp))
640 dtmfsrc->timestamp += GST_BUFFER_DURATION (buf);
642 gst_rtp_buffer_unmap (&rtpbuffer);
648 gst_dtmf_src_prepare_message (GstRTPDTMFSrc * dtmfsrc,
649 const gchar * message_name, GstRTPDTMFSrcEvent * event)
653 switch (event->event_type) {
654 case RTP_DTMF_EVENT_TYPE_START:
655 s = gst_structure_new (message_name,
656 "type", G_TYPE_INT, 1,
657 "method", G_TYPE_INT, 1,
658 "start", G_TYPE_BOOLEAN, TRUE,
659 "number", G_TYPE_INT, event->payload->event,
660 "volume", G_TYPE_INT, event->payload->volume, NULL);
662 case RTP_DTMF_EVENT_TYPE_STOP:
663 s = gst_structure_new (message_name,
664 "type", G_TYPE_INT, 1, "method", G_TYPE_INT, 1,
665 "start", G_TYPE_BOOLEAN, FALSE, NULL);
667 case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
673 return gst_message_new_element (GST_OBJECT (dtmfsrc), s);
677 gst_dtmf_src_post_message (GstRTPDTMFSrc * dtmfsrc, const gchar * message_name,
678 GstRTPDTMFSrcEvent * event)
680 GstMessage *m = gst_dtmf_src_prepare_message (dtmfsrc, message_name, event);
684 gst_element_post_message (GST_ELEMENT (dtmfsrc), m);
689 gst_rtp_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
690 guint length, GstBuffer ** buffer)
692 GstRTPDTMFSrcEvent *event;
693 GstRTPDTMFSrc *dtmfsrc;
696 GstClockReturn clockret;
698 GQueue messages = G_QUEUE_INIT;
700 dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
704 if (dtmfsrc->payload == NULL) {
705 GST_DEBUG_OBJECT (dtmfsrc, "popping");
706 event = g_async_queue_pop (dtmfsrc->event_queue);
708 GST_DEBUG_OBJECT (dtmfsrc, "popped %d", event->event_type);
710 switch (event->event_type) {
711 case RTP_DTMF_EVENT_TYPE_STOP:
712 GST_WARNING_OBJECT (dtmfsrc,
713 "Received a DTMF stop event when already stopped");
714 gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
717 case RTP_DTMF_EVENT_TYPE_START:
718 dtmfsrc->first_packet = TRUE;
719 dtmfsrc->last_packet = FALSE;
720 /* Set the redundancy on the first packet */
721 dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
722 if (!gst_rtp_dtmf_prepare_timestamps (dtmfsrc))
725 g_queue_push_tail (&messages,
726 gst_dtmf_src_prepare_message (dtmfsrc, "dtmf-event-processed",
728 dtmfsrc->payload = event->payload;
729 dtmfsrc->payload->duration =
730 dtmfsrc->ptime * dtmfsrc->clock_rate / 1000;
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 g_queue_push_tail (&messages,
772 gst_dtmf_src_prepare_message (dtmfsrc, "dtmf-event-processed",
776 case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
778 * We're pushing it back because it has to stay in there until
779 * the task is really paused (and the queue will then be flushed)
781 GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
782 GST_OBJECT_LOCK (dtmfsrc);
783 if (dtmfsrc->paused) {
784 g_async_queue_push (dtmfsrc->event_queue, event);
787 GST_OBJECT_UNLOCK (dtmfsrc);
790 gst_rtp_dtmf_src_event_free (event);
793 } while (dtmfsrc->payload == NULL);
796 GST_DEBUG_OBJECT (dtmfsrc, "Processed events, now lets wait on the clock");
798 clock = gst_element_get_clock (GST_ELEMENT (basesrc));
801 clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
802 gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
803 gst_object_unref (clock);
805 GST_OBJECT_LOCK (dtmfsrc);
806 if (!dtmfsrc->paused) {
807 dtmfsrc->clockid = clockid;
808 GST_OBJECT_UNLOCK (dtmfsrc);
810 clockret = gst_clock_id_wait (clockid, NULL);
812 GST_OBJECT_LOCK (dtmfsrc);
814 clockret = GST_CLOCK_UNSCHEDULED;
816 clockret = GST_CLOCK_UNSCHEDULED;
818 gst_clock_id_unref (clockid);
819 dtmfsrc->clockid = NULL;
820 GST_OBJECT_UNLOCK (dtmfsrc);
822 while ((message = g_queue_pop_head (&messages)) != NULL)
823 gst_element_post_message (GST_ELEMENT (dtmfsrc), message);
825 if (clockret == GST_CLOCK_UNSCHEDULED) {
832 if (!gst_rtp_dtmf_src_negotiate (basesrc))
833 return GST_FLOW_NOT_NEGOTIATED;
835 /* create buffer to hold the payload */
836 *buffer = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc);
838 if (dtmfsrc->redundancy_count)
839 dtmfsrc->redundancy_count--;
841 /* Only the very first one has a marker */
842 dtmfsrc->first_packet = FALSE;
844 /* This is the end of the event */
845 if (dtmfsrc->last_packet == TRUE && dtmfsrc->redundancy_count == 0) {
847 g_slice_free (GstRTPDTMFPayload, dtmfsrc->payload);
848 dtmfsrc->payload = NULL;
850 dtmfsrc->last_packet = FALSE;
857 GST_OBJECT_UNLOCK (dtmfsrc);
861 if (dtmfsrc->payload) {
862 dtmfsrc->first_packet = FALSE;
863 dtmfsrc->last_packet = TRUE;
864 /* Set the redundanc on the last packet */
865 dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
868 return GST_FLOW_FLUSHING;
872 GST_ELEMENT_ERROR (dtmfsrc, STREAM, MUX, ("No available clock"),
873 ("No available clock"));
874 gst_pad_pause_task (GST_BASE_SRC_PAD (dtmfsrc));
875 return GST_FLOW_ERROR;
880 gst_rtp_dtmf_src_negotiate (GstBaseSrc * basesrc)
882 GstCaps *srccaps, *peercaps;
883 GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
886 /* fill in the defaults, there properties cannot be negotiated. */
887 srccaps = gst_caps_new_simple ("application/x-rtp",
888 "media", G_TYPE_STRING, "audio",
889 "encoding-name", G_TYPE_STRING, "TELEPHONE-EVENT", NULL);
891 /* the peer caps can override some of the defaults */
892 peercaps = gst_pad_peer_query_caps (GST_BASE_SRC_PAD (basesrc), NULL);
893 if (peercaps == NULL) {
894 /* no peer caps, just add the other properties */
895 gst_caps_set_simple (srccaps,
896 "payload", G_TYPE_INT, dtmfsrc->pt,
897 "ssrc", G_TYPE_UINT, dtmfsrc->current_ssrc,
898 "timestamp-offset", G_TYPE_UINT, dtmfsrc->ts_base,
899 "clock-rate", G_TYPE_INT, dtmfsrc->clock_rate,
900 "seqnum-offset", G_TYPE_UINT, dtmfsrc->seqnum_base, NULL);
902 GST_DEBUG_OBJECT (dtmfsrc, "no peer caps: %" GST_PTR_FORMAT, srccaps);
910 /* peer provides caps we can use to fixate, intersect. This always returns a
912 temp = gst_caps_intersect (srccaps, peercaps);
913 gst_caps_unref (srccaps);
914 gst_caps_unref (peercaps);
917 GST_DEBUG_OBJECT (dtmfsrc, "Could not get intersection with peer caps");
921 if (gst_caps_is_empty (temp)) {
922 GST_DEBUG_OBJECT (dtmfsrc, "Intersection with peer caps is empty");
923 gst_caps_unref (temp);
927 /* now fixate, start by taking the first caps */
928 temp = gst_caps_truncate (temp);
929 temp = gst_caps_make_writable (temp);
932 /* get first structure */
933 s = gst_caps_get_structure (srccaps, 0);
935 if (gst_structure_get_int (s, "payload", &pt)) {
938 GST_LOG_OBJECT (dtmfsrc, "using peer pt %d", pt);
940 if (gst_structure_has_field (s, "payload")) {
941 /* can only fixate if there is a field */
942 gst_structure_fixate_field_nearest_int (s, "payload", dtmfsrc->pt);
943 gst_structure_get_int (s, "payload", &pt);
944 GST_LOG_OBJECT (dtmfsrc, "using peer pt %d", pt);
946 /* no pt field, use the internal pt */
948 gst_structure_set (s, "payload", G_TYPE_INT, pt, NULL);
949 GST_LOG_OBJECT (dtmfsrc, "using internal pt %d", pt);
953 if (gst_structure_get_int (s, "clock-rate", &clock_rate)) {
954 dtmfsrc->clock_rate = clock_rate;
955 GST_LOG_OBJECT (dtmfsrc, "using clock-rate from caps %d",
956 dtmfsrc->clock_rate);
958 GST_LOG_OBJECT (dtmfsrc, "using existing clock-rate %d",
959 dtmfsrc->clock_rate);
961 gst_structure_set (s, "clock-rate", G_TYPE_INT, dtmfsrc->clock_rate, NULL);
964 if (gst_structure_has_field_typed (s, "ssrc", G_TYPE_UINT)) {
965 value = gst_structure_get_value (s, "ssrc");
966 dtmfsrc->current_ssrc = g_value_get_uint (value);
967 GST_LOG_OBJECT (dtmfsrc, "using peer ssrc %08x", dtmfsrc->current_ssrc);
969 /* FIXME, fixate_nearest_uint would be even better */
970 gst_structure_set (s, "ssrc", G_TYPE_UINT, dtmfsrc->current_ssrc, NULL);
971 GST_LOG_OBJECT (dtmfsrc, "using internal ssrc %08x",
972 dtmfsrc->current_ssrc);
975 if (gst_structure_has_field_typed (s, "timestamp-offset", G_TYPE_UINT)) {
976 value = gst_structure_get_value (s, "timestamp-offset");
977 dtmfsrc->ts_base = g_value_get_uint (value);
978 GST_LOG_OBJECT (dtmfsrc, "using peer timestamp-offset %u",
981 /* FIXME, fixate_nearest_uint would be even better */
982 gst_structure_set (s, "timestamp-offset", G_TYPE_UINT, dtmfsrc->ts_base,
984 GST_LOG_OBJECT (dtmfsrc, "using internal timestamp-offset %u",
987 if (gst_structure_has_field_typed (s, "seqnum-offset", G_TYPE_UINT)) {
988 value = gst_structure_get_value (s, "seqnum-offset");
989 dtmfsrc->seqnum_base = g_value_get_uint (value);
990 GST_LOG_OBJECT (dtmfsrc, "using peer seqnum-offset %u",
991 dtmfsrc->seqnum_base);
993 /* FIXME, fixate_nearest_uint would be even better */
994 gst_structure_set (s, "seqnum-offset", G_TYPE_UINT, dtmfsrc->seqnum_base,
996 GST_LOG_OBJECT (dtmfsrc, "using internal seqnum-offset %u",
997 dtmfsrc->seqnum_base);
1000 if (gst_structure_has_field_typed (s, "ptime", G_TYPE_UINT)) {
1001 value = gst_structure_get_value (s, "ptime");
1002 dtmfsrc->ptime = g_value_get_uint (value);
1003 GST_LOG_OBJECT (dtmfsrc, "using peer ptime %u", dtmfsrc->ptime);
1004 } else if (gst_structure_has_field_typed (s, "maxptime", G_TYPE_UINT)) {
1005 value = gst_structure_get_value (s, "maxptime");
1006 dtmfsrc->ptime = g_value_get_uint (value);
1007 GST_LOG_OBJECT (dtmfsrc, "using peer maxptime as ptime %u",
1010 /* FIXME, fixate_nearest_uint would be even better */
1011 gst_structure_set (s, "ptime", G_TYPE_UINT, dtmfsrc->ptime, NULL);
1012 GST_LOG_OBJECT (dtmfsrc, "using internal ptime %u", dtmfsrc->ptime);
1016 GST_DEBUG_OBJECT (dtmfsrc, "with peer caps: %" GST_PTR_FORMAT, srccaps);
1019 ret = gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), srccaps);
1020 gst_caps_unref (srccaps);
1022 dtmfsrc->dirty = FALSE;
1029 gst_rtp_dtmf_src_query (GstBaseSrc * basesrc, GstQuery * query)
1031 GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
1032 gboolean res = FALSE;
1034 switch (GST_QUERY_TYPE (query)) {
1035 case GST_QUERY_LATENCY:
1037 GstClockTime latency;
1039 latency = dtmfsrc->ptime * GST_MSECOND;
1040 gst_query_set_latency (query, gst_base_src_is_live (basesrc), latency,
1041 GST_CLOCK_TIME_NONE);
1042 GST_DEBUG_OBJECT (dtmfsrc, "Reporting latency of %" GST_TIME_FORMAT,
1043 GST_TIME_ARGS (latency));
1048 res = GST_BASE_SRC_CLASS (gst_rtp_dtmf_src_parent_class)->query (basesrc,
1057 gst_rtp_dtmf_src_ready_to_paused (GstRTPDTMFSrc * dtmfsrc)
1059 if (dtmfsrc->ssrc == -1)
1060 dtmfsrc->current_ssrc = g_random_int ();
1062 dtmfsrc->current_ssrc = dtmfsrc->ssrc;
1064 if (dtmfsrc->seqnum_offset == -1)
1065 dtmfsrc->seqnum_base = g_random_int_range (0, G_MAXUINT16);
1067 dtmfsrc->seqnum_base = dtmfsrc->seqnum_offset;
1068 dtmfsrc->seqnum = dtmfsrc->seqnum_base;
1070 if (dtmfsrc->ts_offset == -1)
1071 dtmfsrc->ts_base = g_random_int ();
1073 dtmfsrc->ts_base = dtmfsrc->ts_offset;
1075 dtmfsrc->timestamp = 0;
1078 static GstStateChangeReturn
1079 gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition)
1081 GstRTPDTMFSrc *dtmfsrc;
1082 GstStateChangeReturn result;
1083 gboolean no_preroll = FALSE;
1084 GstRTPDTMFSrcEvent *event = NULL;
1086 dtmfsrc = GST_RTP_DTMF_SRC (element);
1088 switch (transition) {
1089 case GST_STATE_CHANGE_READY_TO_PAUSED:
1090 gst_rtp_dtmf_src_ready_to_paused (dtmfsrc);
1092 /* Flushing the event queue */
1093 while ((event = g_async_queue_try_pop (dtmfsrc->event_queue)) != NULL) {
1094 gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
1095 gst_rtp_dtmf_src_event_free (event);
1097 dtmfsrc->last_event_was_start = FALSE;
1106 GST_ELEMENT_CLASS (gst_rtp_dtmf_src_parent_class)->change_state
1107 (element, transition)) == GST_STATE_CHANGE_FAILURE)
1110 switch (transition) {
1111 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1114 case GST_STATE_CHANGE_PAUSED_TO_READY:
1116 /* Flushing the event queue */
1117 while ((event = g_async_queue_try_pop (dtmfsrc->event_queue)) != NULL) {
1118 gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
1119 gst_rtp_dtmf_src_event_free (event);
1121 dtmfsrc->last_event_was_start = FALSE;
1123 /* Indicate that we don't do PRE_ROLL */
1130 if (no_preroll && result == GST_STATE_CHANGE_SUCCESS)
1131 result = GST_STATE_CHANGE_NO_PREROLL;
1138 GST_ERROR_OBJECT (dtmfsrc, "parent failed state change");
1145 gst_rtp_dtmf_src_unlock (GstBaseSrc * src)
1147 GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (src);
1148 GstRTPDTMFSrcEvent *event = NULL;
1150 GST_DEBUG_OBJECT (dtmfsrc, "Called unlock");
1152 GST_OBJECT_LOCK (dtmfsrc);
1153 dtmfsrc->paused = TRUE;
1154 if (dtmfsrc->clockid) {
1155 gst_clock_id_unschedule (dtmfsrc->clockid);
1157 GST_OBJECT_UNLOCK (dtmfsrc);
1159 GST_DEBUG_OBJECT (dtmfsrc, "Pushing the PAUSE_TASK event on unlock request");
1160 event = g_slice_new0 (GstRTPDTMFSrcEvent);
1161 event->event_type = RTP_DTMF_EVENT_TYPE_PAUSE_TASK;
1162 g_async_queue_push (dtmfsrc->event_queue, event);
1169 gst_rtp_dtmf_src_unlock_stop (GstBaseSrc * src)
1171 GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (src);
1173 GST_DEBUG_OBJECT (dtmfsrc, "Unlock stopped");
1175 GST_OBJECT_LOCK (dtmfsrc);
1176 dtmfsrc->paused = FALSE;
1177 GST_OBJECT_UNLOCK (dtmfsrc);
1183 gst_rtp_dtmf_src_plugin_init (GstPlugin * plugin)
1185 return gst_element_register (plugin, "rtpdtmfsrc",
1186 GST_RANK_NONE, GST_TYPE_RTP_DTMF_SRC);