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 * @short_description: Generates RTP DTMF packets
33 * The RTPDTMFSrc element generates RTP DTMF (RFC 2833) event packets on request
34 * from application. The application communicates the beginning and end of a
35 * DTMF event using custom upstream gstreamer events. To report a DTMF event, an
36 * application must send an event of type GST_EVENT_CUSTOM_UPSTREAM, having a
37 * structure of name "dtmf-event" with fields set according to the following
44 * <colspec colname='Name' />
45 * <colspec colname='Type' />
46 * <colspec colname='Possible values' />
47 * <colspec colname='Purpose' />
52 * <entry>GType</entry>
53 * <entry>Possible values</entry>
54 * <entry>Purpose</entry>
61 * <entry>G_TYPE_INT</entry>
63 * <entry>The application uses this field to specify which of the two methods
64 * specified in RFC 2833 to use. The value should be 0 for tones and 1 for
65 * named events. This element is only capable of generating named events.
69 * <entry>number</entry>
70 * <entry>G_TYPE_INT</entry>
72 * <entry>The event number.</entry>
75 * <entry>volume</entry>
76 * <entry>G_TYPE_INT</entry>
78 * <entry>This field describes the power level of the tone, expressed in dBm0
79 * after dropping the sign. Power levels range from 0 to -63 dBm0. The range of
80 * valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE.
84 * <entry>start</entry>
85 * <entry>G_TYPE_BOOLEAN</entry>
86 * <entry>True or False</entry>
87 * <entry>Whether the event is starting or ending.</entry>
90 * <entry>method</entry>
91 * <entry>G_TYPE_INT</entry>
93 * <entry>The method used for sending event, this element will react if this
94 * field is absent or 1.
102 * <para>For example, the following code informs the pipeline (and in turn, the
103 * RTPDTMFSrc element inside the pipeline) about the start of an RTP DTMF named
104 * event '1' of volume -25 dBm0:
109 * structure = gst_structure_new ("dtmf-event",
110 * "type", G_TYPE_INT, 1,
111 * "number", G_TYPE_INT, 1,
112 * "volume", G_TYPE_INT, 25,
113 * "start", G_TYPE_BOOLEAN, TRUE, NULL);
115 * event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure);
116 * gst_element_send_event (pipeline, event);
132 #include "gstrtpdtmfsrc.h"
134 #define GST_RTP_DTMF_TYPE_EVENT 1
135 #define DEFAULT_PACKET_INTERVAL 50 /* ms */
136 #define MIN_PACKET_INTERVAL 10 /* ms */
137 #define MAX_PACKET_INTERVAL 50 /* ms */
138 #define DEFAULT_SSRC -1
139 #define DEFAULT_PT 96
140 #define DEFAULT_TIMESTAMP_OFFSET -1
141 #define DEFAULT_SEQNUM_OFFSET -1
142 #define DEFAULT_CLOCK_RATE 8000
145 #define MIN_EVENT_STRING "0"
146 #define MAX_EVENT_STRING "16"
148 #define MAX_VOLUME 36
149 #define MIN_EVENT_DURATION 50
151 #define MIN_INTER_DIGIT_INTERVAL 50
152 #define MIN_PULSE_DURATION 70
153 #define MIN_DUTY_CYCLE (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION)
155 #define DEFAULT_PACKET_REDUNDANCY 1
156 #define MIN_PACKET_REDUNDANCY 1
157 #define MAX_PACKET_REDUNDANCY 5
159 /* elementfactory information */
160 static const GstElementDetails gst_rtp_dtmf_src_details =
161 GST_ELEMENT_DETAILS ("RTP DTMF packet generator",
163 "Generates RTP DTMF packets",
164 "Zeeshan Ali <zeeshan.ali@nokia.com>");
166 GST_DEBUG_CATEGORY_STATIC (gst_rtp_dtmf_src_debug);
167 #define GST_CAT_DEFAULT gst_rtp_dtmf_src_debug
169 /* signals and args */
180 PROP_TIMESTAMP_OFFSET,
190 static GstStaticPadTemplate gst_rtp_dtmf_src_template =
191 GST_STATIC_PAD_TEMPLATE ("src",
194 GST_STATIC_CAPS ("application/x-rtp, "
195 "media = (string) \"audio\", "
196 "payload = (int) [ 96, 127 ], "
197 "clock-rate = (int) [ 0, MAX ], "
198 "ssrc = (int) [ 0, MAX ], "
199 "events = (int) [ " MIN_EVENT_STRING ", " MAX_EVENT_STRING " ], "
200 "encoding-name = (string) \"telephone-event\"")
203 static GstElementClass *parent_class = NULL;
205 static void gst_rtp_dtmf_src_base_init (gpointer g_class);
206 static void gst_rtp_dtmf_src_class_init (GstRTPDTMFSrcClass * klass);
207 static void gst_rtp_dtmf_src_init (GstRTPDTMFSrc * dtmfsrc, gpointer g_class);
208 static void gst_rtp_dtmf_src_finalize (GObject * object);
211 gst_rtp_dtmf_src_get_type (void)
213 static GType base_src_type = 0;
215 if (G_UNLIKELY (base_src_type == 0)) {
216 static const GTypeInfo base_src_info = {
217 sizeof (GstRTPDTMFSrcClass),
218 (GBaseInitFunc) gst_rtp_dtmf_src_base_init,
220 (GClassInitFunc) gst_rtp_dtmf_src_class_init,
223 sizeof (GstRTPDTMFSrc),
225 (GInstanceInitFunc) gst_rtp_dtmf_src_init,
228 base_src_type = g_type_register_static (GST_TYPE_ELEMENT,
229 "GstRTPDTMFSrc", &base_src_info, 0);
231 return base_src_type;
234 static void gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id,
235 const GValue * value, GParamSpec * pspec);
236 static void gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id,
237 GValue * value, GParamSpec * pspec);
238 static gboolean gst_rtp_dtmf_src_handle_event (GstPad * pad, GstEvent * event);
239 static GstStateChangeReturn gst_rtp_dtmf_src_change_state (GstElement * element,
240 GstStateChange transition);
241 static void gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc);
242 static void gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc);
243 static void gst_rtp_dtmf_src_stop (GstRTPDTMFSrc *dtmfsrc);
244 static void gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc *dtmfsrc,
245 gint event_number, gint event_volume);
246 static void gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc *dtmfsrc);
247 static void gst_rtp_dtmf_src_set_caps (GstRTPDTMFSrc *dtmfsrc);
251 gst_rtp_dtmf_src_base_init (gpointer g_class)
253 GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
255 GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_src_debug,
256 "rtpdtmfsrc", 0, "rtpdtmfsrc element");
258 gst_element_class_add_pad_template (element_class,
259 gst_static_pad_template_get (&gst_rtp_dtmf_src_template));
261 gst_element_class_set_details (element_class, &gst_rtp_dtmf_src_details);
265 gst_rtp_dtmf_src_class_init (GstRTPDTMFSrcClass * klass)
267 GObjectClass *gobject_class;
268 GstElementClass *gstelement_class;
270 gobject_class = G_OBJECT_CLASS (klass);
271 gstelement_class = GST_ELEMENT_CLASS (klass);
273 parent_class = g_type_class_peek_parent (klass);
275 gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_finalize);
276 gobject_class->set_property =
277 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_set_property);
278 gobject_class->get_property =
279 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_get_property);
281 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_TIMESTAMP,
282 g_param_spec_uint ("timestamp", "Timestamp",
283 "The RTP timestamp of the last processed packet",
284 0, G_MAXUINT, 0, G_PARAM_READABLE));
285 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM,
286 g_param_spec_uint ("seqnum", "Sequence number",
287 "The RTP sequence number of the last processed packet",
288 0, G_MAXUINT, 0, G_PARAM_READABLE));
289 g_object_class_install_property (G_OBJECT_CLASS (klass),
290 PROP_TIMESTAMP_OFFSET, g_param_spec_int ("timestamp-offset",
292 "Offset to add to all outgoing timestamps (-1 = random)", -1,
293 G_MAXINT, DEFAULT_TIMESTAMP_OFFSET, G_PARAM_READWRITE));
294 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM_OFFSET,
295 g_param_spec_int ("seqnum-offset", "Sequence number Offset",
296 "Offset to add to all outgoing seqnum (-1 = random)", -1, G_MAXINT,
297 DEFAULT_SEQNUM_OFFSET, G_PARAM_READWRITE));
298 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_CLOCK_RATE,
299 g_param_spec_uint ("clock-rate", "clockrate",
300 "The clock-rate at which to generate the dtmf packets",
301 0, G_MAXUINT, DEFAULT_CLOCK_RATE, G_PARAM_READWRITE));
302 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SSRC,
303 g_param_spec_uint ("ssrc", "SSRC",
304 "The SSRC of the packets (-1 == random)",
305 0, G_MAXUINT, DEFAULT_SSRC, G_PARAM_READWRITE));
306 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PT,
307 g_param_spec_uint ("pt", "payload type",
308 "The payload type of the packets",
309 0, 0x80, DEFAULT_PT, G_PARAM_READWRITE));
310 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_INTERVAL,
311 g_param_spec_int ("interval", "Interval between rtp packets",
312 "Interval in ms between two rtp packets", MIN_PACKET_INTERVAL,
313 MAX_PACKET_INTERVAL, DEFAULT_PACKET_INTERVAL, G_PARAM_READWRITE));
314 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_REDUNDANCY,
315 g_param_spec_int ("packet-redundancy", "Packet Redundancy",
316 "Number of packets to send to indicate start and stop dtmf events",
317 MIN_PACKET_REDUNDANCY, MAX_PACKET_REDUNDANCY,
318 DEFAULT_PACKET_REDUNDANCY, G_PARAM_READWRITE));
320 gstelement_class->change_state =
321 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_change_state);
325 gst_rtp_dtmf_src_init (GstRTPDTMFSrc * dtmfsrc, gpointer g_class)
328 gst_pad_new_from_static_template (&gst_rtp_dtmf_src_template, "src");
329 GST_DEBUG_OBJECT (dtmfsrc, "adding src pad");
330 gst_element_add_pad (GST_ELEMENT (dtmfsrc), dtmfsrc->srcpad);
332 gst_pad_set_event_function (dtmfsrc->srcpad, gst_rtp_dtmf_src_handle_event);
334 dtmfsrc->ssrc = DEFAULT_SSRC;
335 dtmfsrc->seqnum_offset = DEFAULT_SEQNUM_OFFSET;
336 dtmfsrc->ts_offset = DEFAULT_TIMESTAMP_OFFSET;
337 dtmfsrc->pt = DEFAULT_PT;
338 dtmfsrc->clock_rate = DEFAULT_CLOCK_RATE;
339 dtmfsrc->interval = DEFAULT_PACKET_INTERVAL;
340 dtmfsrc->packet_redundancy = DEFAULT_PACKET_REDUNDANCY;
343 dtmfsrc->event_queue = g_async_queue_new ();
344 dtmfsrc->last_event = NULL;
345 dtmfsrc->clock_id = NULL;
347 GST_DEBUG_OBJECT (dtmfsrc, "init done");
351 gst_rtp_dtmf_src_finalize (GObject * object)
353 GstRTPDTMFSrc *dtmfsrc;
355 dtmfsrc = GST_RTP_DTMF_SRC (object);
357 if (dtmfsrc->event_queue) {
358 g_async_queue_unref (dtmfsrc->event_queue);
359 dtmfsrc->event_queue = NULL;
363 G_OBJECT_CLASS (parent_class)->finalize (object);
367 gst_rtp_dtmf_src_handle_dtmf_event (GstRTPDTMFSrc *dtmfsrc,
368 const GstStructure * event_structure)
374 if (!gst_structure_get_int (event_structure, "type", &event_type) ||
375 !gst_structure_get_boolean (event_structure, "start", &start) ||
376 event_type != GST_RTP_DTMF_TYPE_EVENT)
379 if (gst_structure_get_int (event_structure, "method", &method)) {
389 if (!gst_structure_get_int (event_structure, "number", &event_number) ||
390 !gst_structure_get_int (event_structure, "volume", &event_volume))
393 GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d",
394 event_number, event_volume);
395 gst_rtp_dtmf_src_add_start_event (dtmfsrc, event_number, event_volume);
399 GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
400 gst_rtp_dtmf_src_add_stop_event (dtmfsrc);
409 gst_rtp_dtmf_src_handle_custom_upstream (GstRTPDTMFSrc *dtmfsrc,
412 gboolean result = FALSE;
414 const GstStructure *structure;
417 GstStateChangeReturn ret;
419 ret = gst_element_get_state (GST_ELEMENT (dtmfsrc), &state, NULL, 0);
420 if (ret != GST_STATE_CHANGE_SUCCESS || state != GST_STATE_PLAYING) {
421 GST_DEBUG_OBJECT (dtmfsrc, "Received event while not in PLAYING state");
425 GST_DEBUG_OBJECT (dtmfsrc, "Received event is of our interest");
426 structure = gst_event_get_structure (event);
427 struct_str = gst_structure_to_string (structure);
428 GST_DEBUG_OBJECT (dtmfsrc, "Event has structure %s", struct_str);
430 if (structure && gst_structure_has_name (structure, "dtmf-event"))
431 result = gst_rtp_dtmf_src_handle_dtmf_event (dtmfsrc, structure);
438 gst_rtp_dtmf_src_handle_event (GstPad * pad, GstEvent * event)
440 GstRTPDTMFSrc *dtmfsrc;
441 gboolean result = FALSE;
442 GstElement *parent = gst_pad_get_parent_element (pad);
443 dtmfsrc = GST_RTP_DTMF_SRC (parent);
446 GST_DEBUG_OBJECT (dtmfsrc, "Received an event on the src pad");
447 switch (GST_EVENT_TYPE (event)) {
448 case GST_EVENT_CUSTOM_UPSTREAM:
450 result = gst_rtp_dtmf_src_handle_custom_upstream (dtmfsrc, event);
453 /* Ideally this element should not be flushed but let's handle the event
454 * just in case it is */
455 case GST_EVENT_FLUSH_START:
456 gst_rtp_dtmf_src_stop (dtmfsrc);
459 case GST_EVENT_FLUSH_STOP:
460 gst_segment_init (&dtmfsrc->segment, GST_FORMAT_UNDEFINED);
463 result = gst_pad_event_default (pad, event);
467 gst_object_unref (parent);
468 gst_event_unref (event);
473 gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id,
474 const GValue * value, GParamSpec * pspec)
476 GstRTPDTMFSrc *dtmfsrc;
478 dtmfsrc = GST_RTP_DTMF_SRC (object);
481 case PROP_TIMESTAMP_OFFSET:
482 dtmfsrc->ts_offset = g_value_get_int (value);
484 case PROP_SEQNUM_OFFSET:
485 dtmfsrc->seqnum_offset = g_value_get_int (value);
487 case PROP_CLOCK_RATE:
488 dtmfsrc->clock_rate = g_value_get_uint (value);
489 gst_rtp_dtmf_src_set_caps (dtmfsrc);
492 dtmfsrc->ssrc = g_value_get_uint (value);
495 dtmfsrc->pt = g_value_get_uint (value);
496 gst_rtp_dtmf_src_set_caps (dtmfsrc);
499 dtmfsrc->interval = g_value_get_int (value);
501 case PROP_REDUNDANCY:
502 dtmfsrc->packet_redundancy = g_value_get_int (value);
505 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
511 gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id, GValue * value,
514 GstRTPDTMFSrc *dtmfsrc;
516 dtmfsrc = GST_RTP_DTMF_SRC (object);
519 case PROP_TIMESTAMP_OFFSET:
520 g_value_set_int (value, dtmfsrc->ts_offset);
522 case PROP_SEQNUM_OFFSET:
523 g_value_set_int (value, dtmfsrc->seqnum_offset);
525 case PROP_CLOCK_RATE:
526 g_value_set_uint (value, dtmfsrc->clock_rate);
529 g_value_set_uint (value, dtmfsrc->ssrc);
532 g_value_set_uint (value, dtmfsrc->pt);
535 g_value_set_uint (value, dtmfsrc->rtp_timestamp);
538 g_value_set_uint (value, dtmfsrc->seqnum);
541 g_value_set_uint (value, dtmfsrc->interval);
543 case PROP_REDUNDANCY:
544 g_value_set_uint (value, dtmfsrc->packet_redundancy);
547 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
553 gst_rtp_dtmf_src_set_stream_lock (GstRTPDTMFSrc *dtmfsrc, gboolean lock)
556 GstStructure *structure;
558 structure = gst_structure_new ("stream-lock",
559 "lock", G_TYPE_BOOLEAN, lock, NULL);
561 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_OOB, structure);
562 if (!gst_pad_push_event (dtmfsrc->srcpad, event)) {
563 GST_WARNING_OBJECT (dtmfsrc, "stream-lock event not handled");
569 gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc *dtmfsrc)
573 clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
575 dtmfsrc->timestamp = gst_clock_get_time (clock)
576 + (MIN_INTER_DIGIT_INTERVAL * GST_MSECOND);
577 gst_object_unref (clock);
579 gchar *dtmf_name = gst_element_get_name (dtmfsrc);
580 GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s", dtmf_name);
581 dtmfsrc->timestamp = GST_CLOCK_TIME_NONE;
585 dtmfsrc->rtp_timestamp = dtmfsrc->ts_base +
586 gst_util_uint64_scale_int (
587 gst_segment_to_running_time (&dtmfsrc->segment, GST_FORMAT_TIME,
589 dtmfsrc->clock_rate, GST_SECOND);
593 gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc)
595 gst_rtp_dtmf_src_set_caps (dtmfsrc);
597 if (!gst_pad_start_task (dtmfsrc->srcpad,
598 (GstTaskFunction) gst_rtp_dtmf_src_push_next_rtp_packet, dtmfsrc)) {
599 GST_ERROR_OBJECT (dtmfsrc, "Failed to start task on src pad");
604 gst_rtp_dtmf_src_stop (GstRTPDTMFSrc *dtmfsrc)
607 GstRTPDTMFSrcEvent *event = NULL;
609 if (dtmfsrc->clock_id != NULL) {
610 gst_clock_id_unschedule(dtmfsrc->clock_id);
611 gst_clock_id_unref (dtmfsrc->clock_id);
612 dtmfsrc->clock_id = NULL;
615 g_async_queue_lock (dtmfsrc->event_queue);
616 event = g_malloc (sizeof(GstRTPDTMFSrcEvent));
617 event->event_type = RTP_DTMF_EVENT_TYPE_PAUSE_TASK;
618 g_async_queue_push_unlocked (dtmfsrc->event_queue, event);
619 g_async_queue_unlock (dtmfsrc->event_queue);
623 if (!gst_pad_pause_task (dtmfsrc->srcpad)) {
624 GST_ERROR_OBJECT (dtmfsrc, "Failed to pause task on src pad");
629 if (dtmfsrc->last_event) {
630 /* Don't forget to release the stream lock */
631 gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
632 g_free (dtmfsrc->last_event);
633 dtmfsrc->last_event = NULL;
636 /* Flushing the event queue */
637 event = g_async_queue_try_pop (dtmfsrc->event_queue);
639 while (event != NULL) {
641 event = g_async_queue_try_pop (dtmfsrc->event_queue);
650 gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc *dtmfsrc, gint event_number,
654 GstRTPDTMFSrcEvent * event = g_malloc (sizeof(GstRTPDTMFSrcEvent));
655 event->event_type = RTP_DTMF_EVENT_TYPE_START;
657 event->payload = g_new0 (GstRTPDTMFPayload, 1);
658 event->payload->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
659 event->payload->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
661 g_async_queue_push (dtmfsrc->event_queue, event);
665 gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc *dtmfsrc)
668 GstRTPDTMFSrcEvent * event = g_malloc (sizeof(GstRTPDTMFSrcEvent));
669 event->event_type = RTP_DTMF_EVENT_TYPE_STOP;
670 event->payload = g_new0 (GstRTPDTMFPayload, 1);
671 event->payload->event = 0;
672 event->payload->volume = 0;
674 g_async_queue_push (dtmfsrc->event_queue, event);
679 gst_rtp_dtmf_src_wait_for_buffer_ts (GstRTPDTMFSrc *dtmfsrc, GstBuffer * buf)
683 clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
685 GstClockReturn clock_ret;
687 dtmfsrc->clock_id = gst_clock_new_single_shot_id (clock, GST_BUFFER_TIMESTAMP (buf));
688 gst_object_unref (clock);
690 clock_ret = gst_clock_id_wait (dtmfsrc->clock_id, NULL);
691 if (clock_ret == GST_CLOCK_UNSCHEDULED) {
692 GST_DEBUG_OBJECT (dtmfsrc, "Clock wait unscheduled");
693 /* we don't free anything in case of an unscheduled, because it would be unscheduled
694 * by the stop function which will do the free itself. We can't handle it here
695 * in case we stop the task before the unref is done
698 if (clock_ret != GST_CLOCK_OK && clock_ret != GST_CLOCK_EARLY) {
699 gchar *clock_name = NULL;
701 clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
702 clock_name = gst_element_get_name (clock);
703 gst_object_unref (clock);
705 GST_ERROR_OBJECT (dtmfsrc, "Failed to wait on clock %s", clock_name);
708 gst_clock_id_unref (dtmfsrc->clock_id);
713 gchar *dtmf_name = gst_element_get_name (dtmfsrc);
714 GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s", dtmf_name);
720 gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc,
721 GstRTPDTMFSrcEvent *event, GstBuffer *buf)
723 gst_rtp_buffer_set_ssrc (buf, dtmfsrc->current_ssrc);
724 gst_rtp_buffer_set_payload_type (buf, dtmfsrc->pt);
725 if (dtmfsrc->first_packet) {
726 gst_rtp_buffer_set_marker (buf, TRUE);
727 dtmfsrc->first_packet = FALSE;
728 } else if (dtmfsrc->last_packet) {
729 event->payload->e = 1;
730 dtmfsrc->last_packet = FALSE;
734 gst_rtp_buffer_set_seq (buf, dtmfsrc->seqnum);
736 /* timestamp of RTP header */
737 gst_rtp_buffer_set_timestamp (buf, dtmfsrc->rtp_timestamp);
741 gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc,
742 GstRTPDTMFSrcEvent *event,GstBuffer *buf)
744 GstRTPDTMFPayload *payload;
746 gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc,event, buf);
748 /* duration of DTMF payload */
749 event->payload->duration +=
750 dtmfsrc->interval * dtmfsrc->clock_rate / 1000;
752 /* timestamp and duration of GstBuffer */
753 GST_BUFFER_DURATION (buf) = dtmfsrc->interval * GST_MSECOND;
754 GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp;
755 dtmfsrc->timestamp += GST_BUFFER_DURATION (buf);
757 payload = (GstRTPDTMFPayload *) gst_rtp_buffer_get_payload (buf);
759 /* copy payload and convert to network-byte order */
760 g_memmove (payload, event->payload, sizeof (GstRTPDTMFPayload));
761 /* Force the packet duration to a certain minumum
762 * if its the end of the event
765 payload->duration < MIN_EVENT_DURATION * dtmfsrc->clock_rate / 1000)
766 payload->duration = MIN_EVENT_DURATION * dtmfsrc->clock_rate / 1000;
768 payload->duration = g_htons (payload->duration);
772 gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc,
773 GstRTPDTMFSrcEvent *event)
775 GstBuffer *buf = NULL;
777 /* create buffer to hold the payload */
778 buf = gst_rtp_buffer_new_allocate (sizeof (GstRTPDTMFPayload), 0, 0);
780 gst_rtp_dtmf_prepare_buffer_data (dtmfsrc, event, buf);
782 /* FIXME: Should we sync to clock ourselves or leave it to sink */
783 gst_rtp_dtmf_src_wait_for_buffer_ts (dtmfsrc, buf);
785 event->sent_packets++;
787 /* Set caps on the buffer before pushing it */
788 gst_buffer_set_caps (buf, GST_PAD_CAPS (dtmfsrc->srcpad));
794 gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
796 GstBuffer *buf = NULL;
798 gint redundancy_count = 1;
799 GstRTPDTMFSrcEvent *event;
801 g_async_queue_ref (dtmfsrc->event_queue);
803 if (dtmfsrc->last_event == NULL) {
804 event = g_async_queue_pop (dtmfsrc->event_queue);
806 if (event->event_type == RTP_DTMF_EVENT_TYPE_STOP) {
807 GST_WARNING_OBJECT (dtmfsrc,
808 "Received a DTMF stop event when already stopped");
809 } else if (event->event_type == RTP_DTMF_EVENT_TYPE_START) {
811 dtmfsrc->first_packet = TRUE;
812 dtmfsrc->last_packet = FALSE;
813 gst_rtp_dtmf_prepare_timestamps (dtmfsrc);
815 /* Don't forget to get exclusive access to the stream */
816 gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
818 event->sent_packets = 0;
820 dtmfsrc->last_event = event;
821 } else if (event->event_type == RTP_DTMF_EVENT_TYPE_PAUSE_TASK) {
823 g_async_queue_unref (dtmfsrc->event_queue);
826 } else if (dtmfsrc->last_event->sent_packets * dtmfsrc->interval >=
828 event = g_async_queue_try_pop (dtmfsrc->event_queue);
831 if (event->event_type == RTP_DTMF_EVENT_TYPE_START) {
832 GST_WARNING_OBJECT (dtmfsrc,
833 "Received two consecutive DTMF start events");
834 } else if (event->event_type == RTP_DTMF_EVENT_TYPE_STOP) {
835 dtmfsrc->first_packet = FALSE;
836 dtmfsrc->last_packet = TRUE;
840 g_async_queue_unref (dtmfsrc->event_queue);
842 if (dtmfsrc->last_event) {
844 if (dtmfsrc->first_packet == TRUE || dtmfsrc->last_packet == TRUE) {
845 redundancy_count = dtmfsrc->packet_redundancy;
847 if(dtmfsrc->first_packet == TRUE) {
848 GST_DEBUG_OBJECT (dtmfsrc,
849 "redundancy count set to %d due to dtmf start",
851 } else if(dtmfsrc->last_packet == TRUE) {
852 GST_DEBUG_OBJECT (dtmfsrc,
853 "redundancy count set to %d due to dtmf stop",
859 /* create buffer to hold the payload */
860 buf = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc,
861 dtmfsrc->last_event);
863 while ( redundancy_count-- ) {
866 GST_DEBUG_OBJECT (dtmfsrc,
867 "pushing buffer on src pad of size %d with redundancy count %d",
868 GST_BUFFER_SIZE (buf), redundancy_count);
869 ret = gst_pad_push (dtmfsrc->srcpad, buf);
870 if (ret != GST_FLOW_OK)
871 GST_ERROR_OBJECT (dtmfsrc,
872 "Failed to push buffer on src pad");
874 /* Make sure only the first packet sent has the marker set */
875 gst_rtp_buffer_set_marker (buf, FALSE);
878 gst_buffer_unref(buf);
879 GST_DEBUG_OBJECT (dtmfsrc,
880 "pushed DTMF event '%d' on src pad", dtmfsrc->last_event->payload->event);
882 if (dtmfsrc->last_event->payload->e) {
883 /* Don't forget to release the stream lock */
884 gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
886 g_free (dtmfsrc->last_event->payload);
887 dtmfsrc->last_event->payload = NULL;
889 g_free (dtmfsrc->last_event);
890 dtmfsrc->last_event = NULL;
897 gst_rtp_dtmf_src_set_caps (GstRTPDTMFSrc *dtmfsrc)
901 caps = gst_caps_new_simple ("application/x-rtp",
902 "media", G_TYPE_STRING, "audio",
903 "payload", G_TYPE_INT, dtmfsrc->pt,
904 "clock-rate", G_TYPE_INT, dtmfsrc->clock_rate,
905 "encoding-name", G_TYPE_STRING, "telephone-event",
906 "ssrc", G_TYPE_UINT, dtmfsrc->current_ssrc,
907 "clock-base", G_TYPE_UINT, dtmfsrc->ts_base,
908 "seqnum-base", G_TYPE_UINT, dtmfsrc->seqnum_base, NULL);
910 if (!gst_pad_set_caps (dtmfsrc->srcpad, caps))
911 GST_ERROR_OBJECT (dtmfsrc,
912 "Failed to set caps %" GST_PTR_FORMAT " on src pad", caps);
914 GST_DEBUG_OBJECT (dtmfsrc,
915 "caps %" GST_PTR_FORMAT " set on src pad", caps);
917 gst_caps_unref (caps);
921 gst_rtp_dtmf_src_ready_to_paused (GstRTPDTMFSrc *dtmfsrc)
923 gst_segment_init (&dtmfsrc->segment, GST_FORMAT_UNDEFINED);
925 if (dtmfsrc->ssrc == -1)
926 dtmfsrc->current_ssrc = g_random_int ();
928 dtmfsrc->current_ssrc = dtmfsrc->ssrc;
930 if (dtmfsrc->seqnum_offset == -1)
931 dtmfsrc->seqnum_base = g_random_int_range (0, G_MAXUINT16);
933 dtmfsrc->seqnum_base = dtmfsrc->seqnum_offset;
934 dtmfsrc->seqnum = dtmfsrc->seqnum_base;
936 if (dtmfsrc->ts_offset == -1)
937 dtmfsrc->ts_base = g_random_int ();
939 dtmfsrc->ts_base = dtmfsrc->ts_offset;
942 static GstStateChangeReturn
943 gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition)
945 GstRTPDTMFSrc *dtmfsrc;
946 GstStateChangeReturn result;
947 gboolean no_preroll = FALSE;
949 dtmfsrc = GST_RTP_DTMF_SRC (element);
951 switch (transition) {
952 case GST_STATE_CHANGE_READY_TO_PAUSED:
953 gst_rtp_dtmf_src_ready_to_paused (dtmfsrc);
954 /* Indicate that we don't do PRE_ROLL */
957 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
958 gst_rtp_dtmf_src_start (dtmfsrc);
965 GST_ELEMENT_CLASS (parent_class)->change_state (element,
966 transition)) == GST_STATE_CHANGE_FAILURE)
969 switch (transition) {
970 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
971 /* Indicate that we don't do PRE_ROLL */
973 gst_rtp_dtmf_src_stop (dtmfsrc);
979 if (no_preroll && result == GST_STATE_CHANGE_SUCCESS)
980 result = GST_STATE_CHANGE_NO_PREROLL;
987 GST_ERROR_OBJECT (dtmfsrc, "parent failed state change");
993 gst_rtp_dtmf_src_plugin_init (GstPlugin * plugin)
995 return gst_element_register (plugin, "rtpdtmfsrc",
996 GST_RANK_NONE, GST_TYPE_RTP_DTMF_SRC);