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. Tones are specified by their frequencies and events are specied
66 * by their number. This element can only take events as input. Do not confuse
67 * with "method" which specified the output.
71 * <entry>number</entry>
72 * <entry>G_TYPE_INT</entry>
74 * <entry>The event number.</entry>
77 * <entry>volume</entry>
78 * <entry>G_TYPE_INT</entry>
80 * <entry>This field describes the power level of the tone, expressed in dBm0
81 * after dropping the sign. Power levels range from 0 to -63 dBm0. The range of
82 * valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE.
86 * <entry>start</entry>
87 * <entry>G_TYPE_BOOLEAN</entry>
88 * <entry>True or False</entry>
89 * <entry>Whether the event is starting or ending.</entry>
92 * <entry>method</entry>
93 * <entry>G_TYPE_INT</entry>
95 * <entry>The method used for sending event, this element will react if this
96 * field is absent or 1.
104 * <para>For example, the following code informs the pipeline (and in turn, the
105 * RTPDTMFSrc element inside the pipeline) about the start of an RTP DTMF named
106 * event '1' of volume -25 dBm0:
111 * structure = gst_structure_new ("dtmf-event",
112 * "type", G_TYPE_INT, 1,
113 * "number", G_TYPE_INT, 1,
114 * "volume", G_TYPE_INT, 25,
115 * "start", G_TYPE_BOOLEAN, TRUE, NULL);
117 * event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure);
118 * gst_element_send_event (pipeline, event);
134 #include "gstrtpdtmfsrc.h"
136 #define GST_RTP_DTMF_TYPE_EVENT 1
137 #define DEFAULT_PACKET_INTERVAL 50 /* ms */
138 #define MIN_PACKET_INTERVAL 10 /* ms */
139 #define MAX_PACKET_INTERVAL 50 /* ms */
140 #define DEFAULT_SSRC -1
141 #define DEFAULT_PT 96
142 #define DEFAULT_TIMESTAMP_OFFSET -1
143 #define DEFAULT_SEQNUM_OFFSET -1
144 #define DEFAULT_CLOCK_RATE 8000
147 #define MIN_EVENT_STRING "0"
148 #define MAX_EVENT_STRING "16"
150 #define MAX_VOLUME 36
152 #define MIN_INTER_DIGIT_INTERVAL 50 /* ms */
153 #define MIN_PULSE_DURATION 70 /* ms */
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 "encoding-name = (string) \"TELEPHONE-EVENT\"")
200 /* "events = (string) \"0-15\" */
205 GST_BOILERPLATE (GstRTPDTMFSrc, gst_rtp_dtmf_src, GstBaseSrc,
209 static void gst_rtp_dtmf_src_base_init (gpointer g_class);
210 static void gst_rtp_dtmf_src_class_init (GstRTPDTMFSrcClass * klass);
211 static void gst_rtp_dtmf_src_finalize (GObject * object);
214 static void gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id,
215 const GValue * value, GParamSpec * pspec);
216 static void gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id,
217 GValue * value, GParamSpec * pspec);
218 static gboolean gst_rtp_dtmf_src_handle_event (GstBaseSrc *basesrc,
220 static GstStateChangeReturn gst_rtp_dtmf_src_change_state (GstElement * element,
221 GstStateChange transition);
222 static void gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc *dtmfsrc,
223 gint event_number, gint event_volume);
224 static void gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc *dtmfsrc);
226 static gboolean gst_rtp_dtmf_src_unlock (GstBaseSrc *src);
227 static gboolean gst_rtp_dtmf_src_unlock_stop (GstBaseSrc *src);
228 static GstFlowReturn gst_rtp_dtmf_src_create (GstBaseSrc * basesrc,
229 guint64 offset, guint length, GstBuffer ** buffer);
230 static gboolean gst_rtp_dtmf_src_negotiate (GstBaseSrc * basesrc);
234 gst_rtp_dtmf_src_base_init (gpointer g_class)
236 GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
238 GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_src_debug,
239 "rtpdtmfsrc", 0, "rtpdtmfsrc element");
241 gst_element_class_add_pad_template (element_class,
242 gst_static_pad_template_get (&gst_rtp_dtmf_src_template));
244 gst_element_class_set_details (element_class, &gst_rtp_dtmf_src_details);
248 gst_rtp_dtmf_src_class_init (GstRTPDTMFSrcClass * klass)
250 GObjectClass *gobject_class;
251 GstBaseSrcClass *gstbasesrc_class;
252 GstElementClass *gstelement_class;
254 gobject_class = G_OBJECT_CLASS (klass);
255 gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
256 gstelement_class = GST_ELEMENT_CLASS (klass);
258 parent_class = g_type_class_peek_parent (klass);
260 gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_finalize);
261 gobject_class->set_property =
262 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_set_property);
263 gobject_class->get_property =
264 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_get_property);
266 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_TIMESTAMP,
267 g_param_spec_uint ("timestamp", "Timestamp",
268 "The RTP timestamp of the last processed packet",
269 0, G_MAXUINT, 0, G_PARAM_READABLE));
270 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM,
271 g_param_spec_uint ("seqnum", "Sequence number",
272 "The RTP sequence number of the last processed packet",
273 0, G_MAXUINT, 0, G_PARAM_READABLE));
274 g_object_class_install_property (G_OBJECT_CLASS (klass),
275 PROP_TIMESTAMP_OFFSET, g_param_spec_int ("timestamp-offset",
277 "Offset to add to all outgoing timestamps (-1 = random)", -1,
278 G_MAXINT, DEFAULT_TIMESTAMP_OFFSET, G_PARAM_READWRITE));
279 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM_OFFSET,
280 g_param_spec_int ("seqnum-offset", "Sequence number Offset",
281 "Offset to add to all outgoing seqnum (-1 = random)", -1, G_MAXINT,
282 DEFAULT_SEQNUM_OFFSET, G_PARAM_READWRITE));
283 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_CLOCK_RATE,
284 g_param_spec_uint ("clock-rate", "clockrate",
285 "The clock-rate at which to generate the dtmf packets",
286 0, G_MAXUINT, DEFAULT_CLOCK_RATE, G_PARAM_READWRITE));
287 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SSRC,
288 g_param_spec_uint ("ssrc", "SSRC",
289 "The SSRC of the packets (-1 == random)",
290 0, G_MAXUINT, DEFAULT_SSRC, G_PARAM_READWRITE));
291 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PT,
292 g_param_spec_uint ("pt", "payload type",
293 "The payload type of the packets",
294 0, 0x80, DEFAULT_PT, G_PARAM_READWRITE));
295 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_INTERVAL,
296 g_param_spec_uint ("interval", "Interval between rtp packets",
297 "Interval in ms between two rtp packets", MIN_PACKET_INTERVAL,
298 MAX_PACKET_INTERVAL, DEFAULT_PACKET_INTERVAL, G_PARAM_READWRITE));
299 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_REDUNDANCY,
300 g_param_spec_uint ("packet-redundancy", "Packet Redundancy",
301 "Number of packets to send to indicate start and stop dtmf events",
302 MIN_PACKET_REDUNDANCY, MAX_PACKET_REDUNDANCY,
303 DEFAULT_PACKET_REDUNDANCY, G_PARAM_READWRITE));
305 gstelement_class->change_state =
306 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_change_state);
308 gstbasesrc_class->unlock =
309 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_unlock);
310 gstbasesrc_class->unlock_stop =
311 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_unlock_stop);
313 gstbasesrc_class->event =
314 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_handle_event);
315 gstbasesrc_class->create =
316 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_create);
317 gstbasesrc_class->negotiate =
318 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_negotiate);
322 gst_rtp_dtmf_src_init (GstRTPDTMFSrc * object, GstRTPDTMFSrcClass * g_class)
324 gst_base_src_set_format (GST_BASE_SRC (object), GST_FORMAT_TIME);
325 gst_base_src_set_live (GST_BASE_SRC (object), TRUE);
327 object->ssrc = DEFAULT_SSRC;
328 object->seqnum_offset = DEFAULT_SEQNUM_OFFSET;
329 object->ts_offset = DEFAULT_TIMESTAMP_OFFSET;
330 object->pt = DEFAULT_PT;
331 object->clock_rate = DEFAULT_CLOCK_RATE;
332 object->interval = DEFAULT_PACKET_INTERVAL;
333 object->packet_redundancy = DEFAULT_PACKET_REDUNDANCY;
335 object->event_queue = g_async_queue_new ();
336 object->payload = NULL;
338 GST_DEBUG_OBJECT (object, "init done");
342 gst_rtp_dtmf_src_finalize (GObject * object)
344 GstRTPDTMFSrc *dtmfsrc;
346 dtmfsrc = GST_RTP_DTMF_SRC (object);
348 if (dtmfsrc->event_queue) {
349 g_async_queue_unref (dtmfsrc->event_queue);
350 dtmfsrc->event_queue = NULL;
354 G_OBJECT_CLASS (parent_class)->finalize (object);
358 gst_rtp_dtmf_src_handle_dtmf_event (GstRTPDTMFSrc *dtmfsrc,
359 const GstStructure * event_structure)
365 if (!gst_structure_get_int (event_structure, "type", &event_type) ||
366 !gst_structure_get_boolean (event_structure, "start", &start) ||
367 event_type != GST_RTP_DTMF_TYPE_EVENT)
370 if (gst_structure_get_int (event_structure, "method", &method)) {
380 if (!gst_structure_get_int (event_structure, "number", &event_number) ||
381 !gst_structure_get_int (event_structure, "volume", &event_volume))
384 GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d",
385 event_number, event_volume);
386 gst_rtp_dtmf_src_add_start_event (dtmfsrc, event_number, event_volume);
390 GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
391 gst_rtp_dtmf_src_add_stop_event (dtmfsrc);
400 gst_rtp_dtmf_src_handle_custom_upstream (GstRTPDTMFSrc *dtmfsrc,
403 gboolean result = FALSE;
405 const GstStructure *structure;
408 GstStateChangeReturn ret;
410 ret = gst_element_get_state (GST_ELEMENT (dtmfsrc), &state, NULL, 0);
411 if (ret != GST_STATE_CHANGE_SUCCESS || state != GST_STATE_PLAYING) {
412 GST_DEBUG_OBJECT (dtmfsrc, "Received event while not in PLAYING state");
416 GST_DEBUG_OBJECT (dtmfsrc, "Received event is of our interest");
417 structure = gst_event_get_structure (event);
418 struct_str = gst_structure_to_string (structure);
419 GST_DEBUG_OBJECT (dtmfsrc, "Event has structure %s", struct_str);
421 if (structure && gst_structure_has_name (structure, "dtmf-event"))
422 result = gst_rtp_dtmf_src_handle_dtmf_event (dtmfsrc, structure);
429 gst_rtp_dtmf_src_handle_event (GstBaseSrc *basesrc, GstEvent * event)
431 GstRTPDTMFSrc *dtmfsrc;
432 gboolean result = FALSE;
434 dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
436 GST_DEBUG_OBJECT (dtmfsrc, "Received an event on the src pad");
437 if (GST_EVENT_TYPE (event) == GST_EVENT_CUSTOM_UPSTREAM) {
438 result = gst_rtp_dtmf_src_handle_custom_upstream (dtmfsrc, event);
445 gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id,
446 const GValue * value, GParamSpec * pspec)
448 GstRTPDTMFSrc *dtmfsrc;
450 dtmfsrc = GST_RTP_DTMF_SRC (object);
453 case PROP_TIMESTAMP_OFFSET:
454 dtmfsrc->ts_offset = g_value_get_int (value);
456 case PROP_SEQNUM_OFFSET:
457 dtmfsrc->seqnum_offset = g_value_get_int (value);
459 case PROP_CLOCK_RATE:
460 dtmfsrc->clock_rate = g_value_get_uint (value);
461 dtmfsrc->dirty = TRUE;
464 dtmfsrc->ssrc = g_value_get_uint (value);
467 dtmfsrc->pt = g_value_get_uint (value);
468 dtmfsrc->dirty = TRUE;
471 dtmfsrc->interval = g_value_get_uint (value);
473 case PROP_REDUNDANCY:
474 dtmfsrc->packet_redundancy = g_value_get_uint (value);
477 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
483 gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id, GValue * value,
486 GstRTPDTMFSrc *dtmfsrc;
488 dtmfsrc = GST_RTP_DTMF_SRC (object);
491 case PROP_TIMESTAMP_OFFSET:
492 g_value_set_int (value, dtmfsrc->ts_offset);
494 case PROP_SEQNUM_OFFSET:
495 g_value_set_int (value, dtmfsrc->seqnum_offset);
497 case PROP_CLOCK_RATE:
498 g_value_set_uint (value, dtmfsrc->clock_rate);
501 g_value_set_uint (value, dtmfsrc->ssrc);
504 g_value_set_uint (value, dtmfsrc->pt);
507 g_value_set_uint (value, dtmfsrc->rtp_timestamp);
510 g_value_set_uint (value, dtmfsrc->seqnum);
513 g_value_set_uint (value, dtmfsrc->interval);
515 case PROP_REDUNDANCY:
516 g_value_set_uint (value, dtmfsrc->packet_redundancy);
519 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
525 gst_rtp_dtmf_src_set_stream_lock (GstRTPDTMFSrc *dtmfsrc, gboolean lock)
528 GstStructure *structure;
530 structure = gst_structure_new ("stream-lock",
531 "lock", G_TYPE_BOOLEAN, lock, NULL);
533 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_OOB, structure);
534 if (!gst_pad_push_event (GST_BASE_SRC_PAD (dtmfsrc), event)) {
535 GST_WARNING_OBJECT (dtmfsrc, "stream-lock event not handled");
541 gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc *dtmfsrc)
544 GstClockTime base_time;
549 base_time = gst_element_get_base_time (GST_ELEMENT (dtmfsrc));
552 clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
554 dtmfsrc->timestamp = gst_clock_get_time (clock)
555 + (MIN_INTER_DIGIT_INTERVAL * GST_MSECOND) - base_time;
556 dtmfsrc->start_timestamp = dtmfsrc->timestamp;
557 gst_object_unref (clock);
559 gchar *dtmf_name = gst_element_get_name (dtmfsrc);
560 GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s", dtmf_name);
561 dtmfsrc->timestamp = GST_CLOCK_TIME_NONE;
565 dtmfsrc->rtp_timestamp = dtmfsrc->ts_base +
566 gst_util_uint64_scale_int (
567 gst_segment_to_running_time (&GST_BASE_SRC (dtmfsrc)->segment,
568 GST_FORMAT_TIME, dtmfsrc->timestamp),
569 dtmfsrc->clock_rate, GST_SECOND);
574 gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc *dtmfsrc, gint event_number,
578 GstRTPDTMFSrcEvent * event = g_malloc (sizeof(GstRTPDTMFSrcEvent));
579 event->event_type = RTP_DTMF_EVENT_TYPE_START;
581 event->payload = g_new0 (GstRTPDTMFPayload, 1);
582 event->payload->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
583 event->payload->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
584 event->payload->duration = dtmfsrc->interval * dtmfsrc->clock_rate / 1000;
586 g_async_queue_push (dtmfsrc->event_queue, event);
590 gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc *dtmfsrc)
593 GstRTPDTMFSrcEvent * event = g_malloc (sizeof(GstRTPDTMFSrcEvent));
594 event->event_type = RTP_DTMF_EVENT_TYPE_STOP;
596 g_async_queue_push (dtmfsrc->event_queue, event);
601 gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
603 gst_rtp_buffer_set_ssrc (buf, dtmfsrc->current_ssrc);
604 gst_rtp_buffer_set_payload_type (buf, dtmfsrc->pt);
605 /* Only the very first packet gets a marker */
606 if (dtmfsrc->first_packet) {
607 gst_rtp_buffer_set_marker (buf, TRUE);
608 } else if (dtmfsrc->last_packet) {
609 dtmfsrc->payload->e = 1;
613 gst_rtp_buffer_set_seq (buf, dtmfsrc->seqnum);
615 /* timestamp of RTP header */
616 gst_rtp_buffer_set_timestamp (buf, dtmfsrc->rtp_timestamp);
620 gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
622 GstRTPDTMFPayload *payload;
624 gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc, buf);
626 /* timestamp and duration of GstBuffer */
627 /* Redundant buffer have no duration ... */
628 if (dtmfsrc->redundancy_count > 1)
629 GST_BUFFER_DURATION (buf) = 0;
631 GST_BUFFER_DURATION (buf) = dtmfsrc->interval * GST_MSECOND;
632 GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp;
634 dtmfsrc->timestamp += GST_BUFFER_DURATION (buf);
636 payload = (GstRTPDTMFPayload *) gst_rtp_buffer_get_payload (buf);
638 /* copy payload and convert to network-byte order */
639 g_memmove (payload, dtmfsrc->payload, sizeof (GstRTPDTMFPayload));
640 /* Force the packet duration to a certain minumum
641 * if its the end of the event
644 payload->duration < MIN_PULSE_DURATION * dtmfsrc->clock_rate / 1000 )
645 payload->duration = MIN_PULSE_DURATION * dtmfsrc->clock_rate / 1000;
647 payload->duration = g_htons (payload->duration);
650 /* duration of DTMF payloadfor the NEXT packet */
651 /* not updated for redundant packets */
652 if (dtmfsrc->redundancy_count == 0)
653 dtmfsrc->payload->duration +=
654 dtmfsrc->interval * dtmfsrc->clock_rate / 1000;
659 gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
661 GstBuffer *buf = NULL;
663 /* create buffer to hold the payload */
664 buf = gst_rtp_buffer_new_allocate (sizeof (GstRTPDTMFPayload), 0, 0);
666 gst_rtp_dtmf_prepare_buffer_data (dtmfsrc, buf);
668 /* Set caps on the buffer before pushing it */
669 gst_buffer_set_caps (buf, GST_PAD_CAPS (GST_BASE_SRC_PAD (dtmfsrc)));
675 gst_rtp_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
676 guint length, GstBuffer ** buffer)
678 GstRTPDTMFSrcEvent *event;
679 GstRTPDTMFSrc * dtmfsrc;
682 GstClockReturn clockret;
684 dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
688 if (dtmfsrc->payload == NULL) {
689 GST_DEBUG_OBJECT (dtmfsrc, "popping");
690 event = g_async_queue_pop (dtmfsrc->event_queue);
692 GST_DEBUG_OBJECT (dtmfsrc, "popped %d", event->event_type);
694 switch (event->event_type) {
695 case RTP_DTMF_EVENT_TYPE_STOP:
696 GST_WARNING_OBJECT (dtmfsrc,
697 "Received a DTMF stop event when already stopped");
700 case RTP_DTMF_EVENT_TYPE_START:
701 dtmfsrc->first_packet = TRUE;
702 dtmfsrc->last_packet = FALSE;
703 /* Set the redundancy on the first packet */
704 dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
705 gst_rtp_dtmf_prepare_timestamps (dtmfsrc);
707 /* Don't forget to get exclusive access to the stream */
708 gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
710 dtmfsrc->payload = event->payload;
713 case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
715 * We're pushing it back because it has to stay in there until
716 * the task is really paused (and the queue will then be flushed
718 GST_OBJECT_LOCK (dtmfsrc);
719 if (dtmfsrc->paused) {
720 g_async_queue_push (dtmfsrc->event_queue, event);
723 GST_OBJECT_UNLOCK (dtmfsrc);
728 } else if (!dtmfsrc->first_packet && !dtmfsrc->last_packet &&
729 (dtmfsrc->timestamp - dtmfsrc->start_timestamp)/GST_MSECOND >=
730 MIN_PULSE_DURATION) {
731 GST_DEBUG_OBJECT (dtmfsrc, "try popping");
732 event = g_async_queue_try_pop (dtmfsrc->event_queue);
736 GST_DEBUG_OBJECT (dtmfsrc, "try popped %d", event->event_type);
738 switch (event->event_type) {
739 case RTP_DTMF_EVENT_TYPE_START:
740 GST_WARNING_OBJECT (dtmfsrc,
741 "Received two consecutive DTMF start events");
744 case RTP_DTMF_EVENT_TYPE_STOP:
745 dtmfsrc->first_packet = FALSE;
746 dtmfsrc->last_packet = TRUE;
747 /* Set the redundancy on the last packet */
748 dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
751 case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
753 * We're pushing it back because it has to stay in there until
754 * the task is really paused (and the queue will then be flushed)
756 GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
757 GST_OBJECT_LOCK (dtmfsrc);
758 if (dtmfsrc->paused) {
759 g_async_queue_push (dtmfsrc->event_queue, event);
762 GST_OBJECT_UNLOCK (dtmfsrc);
768 } while (dtmfsrc->payload == NULL);
771 GST_DEBUG_OBJECT (dtmfsrc, "Processed events, now lets wait on the clock");
773 clock = gst_element_get_clock (GST_ELEMENT (basesrc));
776 clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp);
778 clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
779 gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
781 gst_object_unref (clock);
783 GST_OBJECT_LOCK (dtmfsrc);
784 if (!dtmfsrc->paused) {
785 dtmfsrc->clockid = clockid;
786 GST_OBJECT_UNLOCK (dtmfsrc);
788 clockret = gst_clock_id_wait (clockid, NULL);
790 GST_OBJECT_LOCK (dtmfsrc);
792 clockret = GST_CLOCK_UNSCHEDULED;
794 clockret = GST_CLOCK_UNSCHEDULED;
796 gst_clock_id_unref (clockid);
797 dtmfsrc->clockid = NULL;
798 GST_OBJECT_UNLOCK (dtmfsrc);
800 if (clockret == GST_CLOCK_UNSCHEDULED) {
807 if (!gst_rtp_dtmf_src_negotiate (basesrc))
808 return GST_FLOW_NOT_NEGOTIATED;
810 /* create buffer to hold the payload */
811 *buffer = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc);
813 if (dtmfsrc->redundancy_count)
814 dtmfsrc->redundancy_count--;
816 /* Only the very first one has a marker */
817 dtmfsrc->first_packet = FALSE;
819 /* This is the end of the event */
820 if (dtmfsrc->last_packet == TRUE && dtmfsrc->redundancy_count == 0) {
822 /* Don't forget to release the stream lock */
823 gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
825 g_free (dtmfsrc->payload);
826 dtmfsrc->payload = NULL;
828 dtmfsrc->last_packet = FALSE;
835 GST_OBJECT_UNLOCK (dtmfsrc);
839 if (dtmfsrc->payload) {
840 dtmfsrc->first_packet = FALSE;
841 dtmfsrc->last_packet = TRUE;
842 /* Set the redundanc on the last packet */
843 dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
846 return GST_FLOW_WRONG_STATE;
852 gst_rtp_dtmf_src_negotiate (GstBaseSrc * basesrc)
854 GstCaps *srccaps, *peercaps;
855 GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
858 /* fill in the defaults, there properties cannot be negotiated. */
859 srccaps = gst_caps_new_simple ("application/x-rtp",
860 "media", G_TYPE_STRING, "audio",
861 "encoding-name", G_TYPE_STRING, "TELEPHONE-EVENT", NULL);
863 /* the peer caps can override some of the defaults */
864 peercaps = gst_pad_peer_get_caps (GST_BASE_SRC_PAD (basesrc));
865 if (peercaps == NULL) {
866 /* no peer caps, just add the other properties */
867 gst_caps_set_simple (srccaps,
868 "payload", G_TYPE_INT, dtmfsrc->pt,
869 "ssrc", G_TYPE_UINT, dtmfsrc->current_ssrc,
870 "clock-base", G_TYPE_UINT, dtmfsrc->ts_base,
871 "clock-rate", G_TYPE_INT, dtmfsrc->clock_rate,
872 "seqnum-base", G_TYPE_UINT, dtmfsrc->seqnum_base, NULL);
874 GST_DEBUG_OBJECT (dtmfsrc, "no peer caps: %" GST_PTR_FORMAT, srccaps);
882 /* peer provides caps we can use to fixate, intersect. This always returns a
884 temp = gst_caps_intersect (srccaps, peercaps);
885 gst_caps_unref (srccaps);
886 gst_caps_unref (peercaps);
889 GST_DEBUG_OBJECT (dtmfsrc, "Could not get intersection with peer caps");
893 if (gst_caps_is_empty (temp)) {
894 GST_DEBUG_OBJECT (dtmfsrc, "Intersection with peer caps is empty");
895 gst_caps_unref (temp);
899 /* now fixate, start by taking the first caps */
900 gst_caps_truncate (temp);
903 /* get first structure */
904 s = gst_caps_get_structure (srccaps, 0);
906 if (gst_structure_get_int (s, "payload", &pt)) {
909 GST_LOG_OBJECT (dtmfsrc, "using peer pt %d", pt);
911 if (gst_structure_has_field (s, "payload")) {
912 /* can only fixate if there is a field */
913 gst_structure_fixate_field_nearest_int (s, "payload",
915 gst_structure_get_int (s, "payload", &pt);
916 GST_LOG_OBJECT (dtmfsrc, "using peer pt %d", pt);
918 /* no pt field, use the internal pt */
920 gst_structure_set (s, "payload", G_TYPE_INT, pt, NULL);
921 GST_LOG_OBJECT (dtmfsrc, "using internal pt", pt);
925 if (gst_structure_get_int (s, "clock-rate", &clock_rate))
927 dtmfsrc->clock_rate = clock_rate;
928 GST_LOG_OBJECT (dtmfsrc, "using clock-rate from caps %d",
929 dtmfsrc->clock_rate);
931 GST_LOG_OBJECT (dtmfsrc, "using existing clock-rate %d",
932 dtmfsrc->clock_rate);
934 gst_structure_set (s, "clock-rate", G_TYPE_INT, dtmfsrc->clock_rate,
938 if (gst_structure_has_field_typed (s, "ssrc", G_TYPE_UINT)) {
939 value = gst_structure_get_value (s, "ssrc");
940 dtmfsrc->current_ssrc = g_value_get_uint (value);
941 GST_LOG_OBJECT (dtmfsrc, "using peer ssrc %08x", dtmfsrc->current_ssrc);
943 /* FIXME, fixate_nearest_uint would be even better */
944 gst_structure_set (s, "ssrc", G_TYPE_UINT, dtmfsrc->current_ssrc, NULL);
945 GST_LOG_OBJECT (dtmfsrc, "using internal ssrc %08x",
946 dtmfsrc->current_ssrc);
949 if (gst_structure_has_field_typed (s, "clock-base", G_TYPE_UINT)) {
950 value = gst_structure_get_value (s, "clock-base");
951 dtmfsrc->ts_base = g_value_get_uint (value);
952 GST_LOG_OBJECT (dtmfsrc, "using peer clock-base %u", dtmfsrc->ts_base);
954 /* FIXME, fixate_nearest_uint would be even better */
955 gst_structure_set (s, "clock-base", G_TYPE_UINT, dtmfsrc->ts_base, NULL);
956 GST_LOG_OBJECT (dtmfsrc, "using internal clock-base %u",
959 if (gst_structure_has_field_typed (s, "seqnum-base", G_TYPE_UINT)) {
960 value = gst_structure_get_value (s, "seqnum-base");
961 dtmfsrc->seqnum_base = g_value_get_uint (value);
962 GST_LOG_OBJECT (dtmfsrc, "using peer seqnum-base %u",
963 dtmfsrc->seqnum_base);
965 /* FIXME, fixate_nearest_uint would be even better */
966 gst_structure_set (s, "seqnum-base", G_TYPE_UINT, dtmfsrc->seqnum_base,
968 GST_LOG_OBJECT (dtmfsrc, "using internal seqnum-base %u",
969 dtmfsrc->seqnum_base);
971 GST_DEBUG_OBJECT (dtmfsrc, "with peer caps: %" GST_PTR_FORMAT, srccaps);
974 ret = gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), srccaps);
975 gst_caps_unref (srccaps);
977 dtmfsrc->dirty = FALSE;
985 gst_rtp_dtmf_src_ready_to_paused (GstRTPDTMFSrc *dtmfsrc)
987 if (dtmfsrc->ssrc == -1)
988 dtmfsrc->current_ssrc = g_random_int ();
990 dtmfsrc->current_ssrc = dtmfsrc->ssrc;
992 if (dtmfsrc->seqnum_offset == -1)
993 dtmfsrc->seqnum_base = g_random_int_range (0, G_MAXUINT16);
995 dtmfsrc->seqnum_base = dtmfsrc->seqnum_offset;
996 dtmfsrc->seqnum = dtmfsrc->seqnum_base;
998 if (dtmfsrc->ts_offset == -1)
999 dtmfsrc->ts_base = g_random_int ();
1001 dtmfsrc->ts_base = dtmfsrc->ts_offset;
1005 static GstStateChangeReturn
1006 gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition)
1008 GstRTPDTMFSrc *dtmfsrc;
1009 GstStateChangeReturn result;
1010 gboolean no_preroll = FALSE;
1011 GstRTPDTMFSrcEvent *event= NULL;
1013 dtmfsrc = GST_RTP_DTMF_SRC (element);
1015 switch (transition) {
1016 case GST_STATE_CHANGE_READY_TO_PAUSED:
1017 gst_rtp_dtmf_src_ready_to_paused (dtmfsrc);
1019 /* Flushing the event queue */
1020 while ((event = g_async_queue_try_pop (dtmfsrc->event_queue)) != NULL)
1030 GST_ELEMENT_CLASS (parent_class)->change_state (element,
1031 transition)) == GST_STATE_CHANGE_FAILURE)
1034 switch (transition) {
1035 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1038 case GST_STATE_CHANGE_PAUSED_TO_READY:
1040 /* Flushing the event queue */
1041 while ((event = g_async_queue_try_pop (dtmfsrc->event_queue)) != NULL)
1044 /* Indicate that we don't do PRE_ROLL */
1051 if (no_preroll && result == GST_STATE_CHANGE_SUCCESS)
1052 result = GST_STATE_CHANGE_NO_PREROLL;
1059 GST_ERROR_OBJECT (dtmfsrc, "parent failed state change");
1066 gst_rtp_dtmf_src_unlock (GstBaseSrc *src) {
1067 GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (src);
1068 GstRTPDTMFSrcEvent *event = NULL;
1070 GST_DEBUG_OBJECT (dtmfsrc, "Called unlock");
1072 GST_OBJECT_LOCK (dtmfsrc);
1073 dtmfsrc->paused = TRUE;
1074 if (dtmfsrc->clockid) {
1075 gst_clock_id_unschedule (dtmfsrc->clockid);
1077 GST_OBJECT_UNLOCK (dtmfsrc);
1079 GST_DEBUG_OBJECT (dtmfsrc, "Pushing the PAUSE_TASK event on unlock request");
1080 event = g_malloc (sizeof(GstRTPDTMFSrcEvent));
1081 event->event_type = RTP_DTMF_EVENT_TYPE_PAUSE_TASK;
1082 g_async_queue_push (dtmfsrc->event_queue, event);
1089 gst_rtp_dtmf_src_unlock_stop (GstBaseSrc *src) {
1090 GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (src);
1092 GST_DEBUG_OBJECT (dtmfsrc, "Unlock stopped");
1094 GST_OBJECT_LOCK (dtmfsrc);
1095 dtmfsrc->paused = FALSE;
1096 GST_OBJECT_UNLOCK (dtmfsrc);
1102 gst_rtp_dtmf_src_plugin_init (GstPlugin * plugin)
1104 return gst_element_register (plugin, "rtpdtmfsrc",
1105 GST_RANK_NONE, GST_TYPE_RTP_DTMF_SRC);