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
150 #define MIN_INTER_DIGIT_INTERVAL 50 /* ms */
151 #define MIN_PULSE_DURATION 70 /* ms */
153 #define DEFAULT_PACKET_REDUNDANCY 1
154 #define MIN_PACKET_REDUNDANCY 1
155 #define MAX_PACKET_REDUNDANCY 5
157 /* elementfactory information */
158 static const GstElementDetails gst_rtp_dtmf_src_details =
159 GST_ELEMENT_DETAILS ("RTP DTMF packet generator",
161 "Generates RTP DTMF packets",
162 "Zeeshan Ali <zeeshan.ali@nokia.com>");
164 GST_DEBUG_CATEGORY_STATIC (gst_rtp_dtmf_src_debug);
165 #define GST_CAT_DEFAULT gst_rtp_dtmf_src_debug
167 /* signals and args */
178 PROP_TIMESTAMP_OFFSET,
188 static GstStaticPadTemplate gst_rtp_dtmf_src_template =
189 GST_STATIC_PAD_TEMPLATE ("src",
192 GST_STATIC_CAPS ("application/x-rtp, "
193 "media = (string) \"audio\", "
194 "payload = (int) [ 96, 127 ], "
195 "clock-rate = (int) [ 0, MAX ], "
196 "ssrc = (int) [ 0, MAX ], "
197 "encoding-name = (string) \"telephone-event\"")
198 /* "events = (string) \"1-16\" */
203 GST_BOILERPLATE (GstRTPDTMFSrc, gst_rtp_dtmf_src, GstBaseSrc,
207 static void gst_rtp_dtmf_src_base_init (gpointer g_class);
208 static void gst_rtp_dtmf_src_class_init (GstRTPDTMFSrcClass * klass);
209 static void gst_rtp_dtmf_src_finalize (GObject * object);
212 static void gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id,
213 const GValue * value, GParamSpec * pspec);
214 static void gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id,
215 GValue * value, GParamSpec * pspec);
216 static gboolean gst_rtp_dtmf_src_handle_event (GstBaseSrc *basesrc,
218 static GstStateChangeReturn gst_rtp_dtmf_src_change_state (GstElement * element,
219 GstStateChange transition);
220 static void gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc *dtmfsrc,
221 gint event_number, gint event_volume);
222 static void gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc *dtmfsrc);
224 static gboolean gst_rtp_dtmf_src_unlock (GstBaseSrc *src);
225 static gboolean gst_rtp_dtmf_src_unlock_stop (GstBaseSrc *src);
226 static GstFlowReturn gst_rtp_dtmf_src_create (GstBaseSrc * basesrc,
227 guint64 offset, guint length, GstBuffer ** buffer);
228 static gboolean gst_rtp_dtmf_src_negotiate (GstBaseSrc * basesrc);
232 gst_rtp_dtmf_src_base_init (gpointer g_class)
234 GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
236 GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_src_debug,
237 "rtpdtmfsrc", 0, "rtpdtmfsrc element");
239 gst_element_class_add_pad_template (element_class,
240 gst_static_pad_template_get (&gst_rtp_dtmf_src_template));
242 gst_element_class_set_details (element_class, &gst_rtp_dtmf_src_details);
246 gst_rtp_dtmf_src_class_init (GstRTPDTMFSrcClass * klass)
248 GObjectClass *gobject_class;
249 GstBaseSrcClass *gstbasesrc_class;
250 GstElementClass *gstelement_class;
252 gobject_class = G_OBJECT_CLASS (klass);
253 gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
254 gstelement_class = GST_ELEMENT_CLASS (klass);
256 parent_class = g_type_class_peek_parent (klass);
258 gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_finalize);
259 gobject_class->set_property =
260 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_set_property);
261 gobject_class->get_property =
262 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_get_property);
264 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_TIMESTAMP,
265 g_param_spec_uint ("timestamp", "Timestamp",
266 "The RTP timestamp of the last processed packet",
267 0, G_MAXUINT, 0, G_PARAM_READABLE));
268 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM,
269 g_param_spec_uint ("seqnum", "Sequence number",
270 "The RTP sequence number of the last processed packet",
271 0, G_MAXUINT, 0, G_PARAM_READABLE));
272 g_object_class_install_property (G_OBJECT_CLASS (klass),
273 PROP_TIMESTAMP_OFFSET, g_param_spec_int ("timestamp-offset",
275 "Offset to add to all outgoing timestamps (-1 = random)", -1,
276 G_MAXINT, DEFAULT_TIMESTAMP_OFFSET, G_PARAM_READWRITE));
277 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM_OFFSET,
278 g_param_spec_int ("seqnum-offset", "Sequence number Offset",
279 "Offset to add to all outgoing seqnum (-1 = random)", -1, G_MAXINT,
280 DEFAULT_SEQNUM_OFFSET, G_PARAM_READWRITE));
281 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_CLOCK_RATE,
282 g_param_spec_uint ("clock-rate", "clockrate",
283 "The clock-rate at which to generate the dtmf packets",
284 0, G_MAXUINT, DEFAULT_CLOCK_RATE, G_PARAM_READWRITE));
285 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SSRC,
286 g_param_spec_uint ("ssrc", "SSRC",
287 "The SSRC of the packets (-1 == random)",
288 0, G_MAXUINT, DEFAULT_SSRC, G_PARAM_READWRITE));
289 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PT,
290 g_param_spec_uint ("pt", "payload type",
291 "The payload type of the packets",
292 0, 0x80, DEFAULT_PT, G_PARAM_READWRITE));
293 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_INTERVAL,
294 g_param_spec_uint ("interval", "Interval between rtp packets",
295 "Interval in ms between two rtp packets", MIN_PACKET_INTERVAL,
296 MAX_PACKET_INTERVAL, DEFAULT_PACKET_INTERVAL, G_PARAM_READWRITE));
297 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_REDUNDANCY,
298 g_param_spec_uint ("packet-redundancy", "Packet Redundancy",
299 "Number of packets to send to indicate start and stop dtmf events",
300 MIN_PACKET_REDUNDANCY, MAX_PACKET_REDUNDANCY,
301 DEFAULT_PACKET_REDUNDANCY, G_PARAM_READWRITE));
303 gstelement_class->change_state =
304 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_change_state);
306 gstbasesrc_class->unlock =
307 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_unlock);
308 gstbasesrc_class->unlock_stop =
309 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_unlock_stop);
311 gstbasesrc_class->event =
312 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_handle_event);
313 gstbasesrc_class->create =
314 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_create);
315 gstbasesrc_class->negotiate =
316 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_negotiate);
320 gst_rtp_dtmf_src_init (GstRTPDTMFSrc * object, GstRTPDTMFSrcClass * g_class)
322 gst_base_src_set_format (GST_BASE_SRC (object), GST_FORMAT_TIME);
323 gst_base_src_set_live (GST_BASE_SRC (object), TRUE);
325 object->ssrc = DEFAULT_SSRC;
326 object->seqnum_offset = DEFAULT_SEQNUM_OFFSET;
327 object->ts_offset = DEFAULT_TIMESTAMP_OFFSET;
328 object->pt = DEFAULT_PT;
329 object->clock_rate = DEFAULT_CLOCK_RATE;
330 object->interval = DEFAULT_PACKET_INTERVAL;
331 object->packet_redundancy = DEFAULT_PACKET_REDUNDANCY;
333 object->event_queue = g_async_queue_new ();
334 object->payload = NULL;
336 GST_DEBUG_OBJECT (object, "init done");
340 gst_rtp_dtmf_src_finalize (GObject * object)
342 GstRTPDTMFSrc *dtmfsrc;
344 dtmfsrc = GST_RTP_DTMF_SRC (object);
346 if (dtmfsrc->event_queue) {
347 g_async_queue_unref (dtmfsrc->event_queue);
348 dtmfsrc->event_queue = NULL;
352 G_OBJECT_CLASS (parent_class)->finalize (object);
356 gst_rtp_dtmf_src_handle_dtmf_event (GstRTPDTMFSrc *dtmfsrc,
357 const GstStructure * event_structure)
363 if (!gst_structure_get_int (event_structure, "type", &event_type) ||
364 !gst_structure_get_boolean (event_structure, "start", &start) ||
365 event_type != GST_RTP_DTMF_TYPE_EVENT)
368 if (gst_structure_get_int (event_structure, "method", &method)) {
378 if (!gst_structure_get_int (event_structure, "number", &event_number) ||
379 !gst_structure_get_int (event_structure, "volume", &event_volume))
382 GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d",
383 event_number, event_volume);
384 gst_rtp_dtmf_src_add_start_event (dtmfsrc, event_number, event_volume);
388 GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
389 gst_rtp_dtmf_src_add_stop_event (dtmfsrc);
398 gst_rtp_dtmf_src_handle_custom_upstream (GstRTPDTMFSrc *dtmfsrc,
401 gboolean result = FALSE;
403 const GstStructure *structure;
406 GstStateChangeReturn ret;
408 ret = gst_element_get_state (GST_ELEMENT (dtmfsrc), &state, NULL, 0);
409 if (ret != GST_STATE_CHANGE_SUCCESS || state != GST_STATE_PLAYING) {
410 GST_DEBUG_OBJECT (dtmfsrc, "Received event while not in PLAYING state");
414 GST_DEBUG_OBJECT (dtmfsrc, "Received event is of our interest");
415 structure = gst_event_get_structure (event);
416 struct_str = gst_structure_to_string (structure);
417 GST_DEBUG_OBJECT (dtmfsrc, "Event has structure %s", struct_str);
419 if (structure && gst_structure_has_name (structure, "dtmf-event"))
420 result = gst_rtp_dtmf_src_handle_dtmf_event (dtmfsrc, structure);
427 gst_rtp_dtmf_src_handle_event (GstBaseSrc *basesrc, GstEvent * event)
429 GstRTPDTMFSrc *dtmfsrc;
430 gboolean result = FALSE;
432 dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
434 GST_DEBUG_OBJECT (dtmfsrc, "Received an event on the src pad");
435 if (GST_EVENT_TYPE (event) == GST_EVENT_CUSTOM_UPSTREAM) {
436 result = gst_rtp_dtmf_src_handle_custom_upstream (dtmfsrc, event);
443 gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id,
444 const GValue * value, GParamSpec * pspec)
446 GstRTPDTMFSrc *dtmfsrc;
448 dtmfsrc = GST_RTP_DTMF_SRC (object);
451 case PROP_TIMESTAMP_OFFSET:
452 dtmfsrc->ts_offset = g_value_get_int (value);
454 case PROP_SEQNUM_OFFSET:
455 dtmfsrc->seqnum_offset = g_value_get_int (value);
457 case PROP_CLOCK_RATE:
458 dtmfsrc->clock_rate = g_value_get_uint (value);
459 dtmfsrc->dirty = TRUE;
462 dtmfsrc->ssrc = g_value_get_uint (value);
465 dtmfsrc->pt = g_value_get_uint (value);
466 dtmfsrc->dirty = TRUE;
469 dtmfsrc->interval = g_value_get_uint (value);
471 case PROP_REDUNDANCY:
472 dtmfsrc->packet_redundancy = g_value_get_uint (value);
475 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
481 gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id, GValue * value,
484 GstRTPDTMFSrc *dtmfsrc;
486 dtmfsrc = GST_RTP_DTMF_SRC (object);
489 case PROP_TIMESTAMP_OFFSET:
490 g_value_set_int (value, dtmfsrc->ts_offset);
492 case PROP_SEQNUM_OFFSET:
493 g_value_set_int (value, dtmfsrc->seqnum_offset);
495 case PROP_CLOCK_RATE:
496 g_value_set_uint (value, dtmfsrc->clock_rate);
499 g_value_set_uint (value, dtmfsrc->ssrc);
502 g_value_set_uint (value, dtmfsrc->pt);
505 g_value_set_uint (value, dtmfsrc->rtp_timestamp);
508 g_value_set_uint (value, dtmfsrc->seqnum);
511 g_value_set_uint (value, dtmfsrc->interval);
513 case PROP_REDUNDANCY:
514 g_value_set_uint (value, dtmfsrc->packet_redundancy);
517 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
523 gst_rtp_dtmf_src_set_stream_lock (GstRTPDTMFSrc *dtmfsrc, gboolean lock)
526 GstStructure *structure;
528 structure = gst_structure_new ("stream-lock",
529 "lock", G_TYPE_BOOLEAN, lock, NULL);
531 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_OOB, structure);
532 if (!gst_pad_push_event (GST_BASE_SRC_PAD (dtmfsrc), event)) {
533 GST_WARNING_OBJECT (dtmfsrc, "stream-lock event not handled");
539 gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc *dtmfsrc)
542 GstClockTime base_time;
547 base_time = gst_element_get_base_time (GST_ELEMENT (dtmfsrc));
550 clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
552 dtmfsrc->timestamp = gst_clock_get_time (clock)
553 + (MIN_INTER_DIGIT_INTERVAL * GST_MSECOND) - base_time;
554 dtmfsrc->start_timestamp = dtmfsrc->timestamp;
555 gst_object_unref (clock);
557 gchar *dtmf_name = gst_element_get_name (dtmfsrc);
558 GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s", dtmf_name);
559 dtmfsrc->timestamp = GST_CLOCK_TIME_NONE;
563 dtmfsrc->rtp_timestamp = dtmfsrc->ts_base +
564 gst_util_uint64_scale_int (
565 gst_segment_to_running_time (&GST_BASE_SRC (dtmfsrc)->segment,
566 GST_FORMAT_TIME, dtmfsrc->timestamp),
567 dtmfsrc->clock_rate, GST_SECOND);
572 gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc *dtmfsrc, gint event_number,
576 GstRTPDTMFSrcEvent * event = g_malloc (sizeof(GstRTPDTMFSrcEvent));
577 event->event_type = RTP_DTMF_EVENT_TYPE_START;
579 event->payload = g_new0 (GstRTPDTMFPayload, 1);
580 event->payload->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
581 event->payload->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
582 event->payload->duration = dtmfsrc->interval * dtmfsrc->clock_rate / 1000;
584 g_async_queue_push (dtmfsrc->event_queue, event);
588 gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc *dtmfsrc)
591 GstRTPDTMFSrcEvent * event = g_malloc (sizeof(GstRTPDTMFSrcEvent));
592 event->event_type = RTP_DTMF_EVENT_TYPE_STOP;
594 g_async_queue_push (dtmfsrc->event_queue, event);
599 gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
601 gst_rtp_buffer_set_ssrc (buf, dtmfsrc->current_ssrc);
602 gst_rtp_buffer_set_payload_type (buf, dtmfsrc->pt);
603 /* Only the very first packet gets a marker */
604 if (dtmfsrc->first_packet) {
605 gst_rtp_buffer_set_marker (buf, TRUE);
606 } else if (dtmfsrc->last_packet) {
607 dtmfsrc->payload->e = 1;
611 gst_rtp_buffer_set_seq (buf, dtmfsrc->seqnum);
613 /* timestamp of RTP header */
614 gst_rtp_buffer_set_timestamp (buf, dtmfsrc->rtp_timestamp);
618 gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
620 GstRTPDTMFPayload *payload;
622 gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc, buf);
624 /* timestamp and duration of GstBuffer */
625 /* Redundant buffer have no duration ... */
626 if (dtmfsrc->redundancy_count > 1)
627 GST_BUFFER_DURATION (buf) = 0;
629 GST_BUFFER_DURATION (buf) = dtmfsrc->interval * GST_MSECOND;
630 GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp;
632 dtmfsrc->timestamp += GST_BUFFER_DURATION (buf);
634 payload = (GstRTPDTMFPayload *) gst_rtp_buffer_get_payload (buf);
636 /* copy payload and convert to network-byte order */
637 g_memmove (payload, dtmfsrc->payload, sizeof (GstRTPDTMFPayload));
638 /* Force the packet duration to a certain minumum
639 * if its the end of the event
642 payload->duration < MIN_PULSE_DURATION * dtmfsrc->clock_rate / 1000 )
643 payload->duration = MIN_PULSE_DURATION * dtmfsrc->clock_rate / 1000;
645 payload->duration = g_htons (payload->duration);
648 /* duration of DTMF payloadfor the NEXT packet */
649 /* not updated for redundant packets */
650 if (dtmfsrc->redundancy_count == 0)
651 dtmfsrc->payload->duration +=
652 dtmfsrc->interval * dtmfsrc->clock_rate / 1000;
657 gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
659 GstBuffer *buf = NULL;
661 /* create buffer to hold the payload */
662 buf = gst_rtp_buffer_new_allocate (sizeof (GstRTPDTMFPayload), 0, 0);
664 gst_rtp_dtmf_prepare_buffer_data (dtmfsrc, buf);
666 /* Set caps on the buffer before pushing it */
667 gst_buffer_set_caps (buf, GST_PAD_CAPS (GST_BASE_SRC_PAD (dtmfsrc)));
673 gst_rtp_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
674 guint length, GstBuffer ** buffer)
676 GstRTPDTMFSrcEvent *event;
677 GstRTPDTMFSrc * dtmfsrc;
680 GstClockReturn clockret;
682 dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
686 if (dtmfsrc->payload == NULL) {
687 GST_DEBUG_OBJECT (dtmfsrc, "popping");
688 event = g_async_queue_pop (dtmfsrc->event_queue);
690 GST_DEBUG_OBJECT (dtmfsrc, "popped %d", event->event_type);
692 switch (event->event_type) {
693 case RTP_DTMF_EVENT_TYPE_STOP:
694 GST_WARNING_OBJECT (dtmfsrc,
695 "Received a DTMF stop event when already stopped");
698 case RTP_DTMF_EVENT_TYPE_START:
699 dtmfsrc->first_packet = TRUE;
700 dtmfsrc->last_packet = FALSE;
701 /* Set the redundancy on the first packet */
702 dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
703 gst_rtp_dtmf_prepare_timestamps (dtmfsrc);
705 /* Don't forget to get exclusive access to the stream */
706 gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
708 dtmfsrc->payload = event->payload;
711 case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
713 * We're pushing it back because it has to stay in there until
714 * the task is really paused (and the queue will then be flushed
716 GST_OBJECT_LOCK (dtmfsrc);
717 if (dtmfsrc->paused) {
718 g_async_queue_push (dtmfsrc->event_queue, event);
721 GST_OBJECT_UNLOCK (dtmfsrc);
726 } else if (!dtmfsrc->first_packet && !dtmfsrc->last_packet &&
727 (dtmfsrc->timestamp - dtmfsrc->start_timestamp)/GST_MSECOND >=
728 MIN_PULSE_DURATION) {
729 GST_DEBUG_OBJECT (dtmfsrc, "try popping");
730 event = g_async_queue_try_pop (dtmfsrc->event_queue);
734 GST_DEBUG_OBJECT (dtmfsrc, "try popped %d", event->event_type);
736 switch (event->event_type) {
737 case RTP_DTMF_EVENT_TYPE_START:
738 GST_WARNING_OBJECT (dtmfsrc,
739 "Received two consecutive DTMF start events");
742 case RTP_DTMF_EVENT_TYPE_STOP:
743 dtmfsrc->first_packet = FALSE;
744 dtmfsrc->last_packet = TRUE;
745 /* Set the redundancy on the last packet */
746 dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
749 case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
751 * We're pushing it back because it has to stay in there until
752 * the task is really paused (and the queue will then be flushed)
754 GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
755 GST_OBJECT_LOCK (dtmfsrc);
756 if (dtmfsrc->paused) {
757 g_async_queue_push (dtmfsrc->event_queue, event);
760 GST_OBJECT_UNLOCK (dtmfsrc);
766 } while (dtmfsrc->payload == NULL);
769 GST_DEBUG_OBJECT (dtmfsrc, "Processed events, now lets wait on the clock");
771 clock = gst_element_get_clock (GST_ELEMENT (basesrc));
774 clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp);
776 clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
777 gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
779 gst_object_unref (clock);
781 GST_OBJECT_LOCK (dtmfsrc);
782 if (!dtmfsrc->paused) {
783 dtmfsrc->clockid = clockid;
784 GST_OBJECT_UNLOCK (dtmfsrc);
786 clockret = gst_clock_id_wait (clockid, NULL);
788 GST_OBJECT_LOCK (dtmfsrc);
790 clockret = GST_CLOCK_UNSCHEDULED;
792 clockret = GST_CLOCK_UNSCHEDULED;
794 gst_clock_id_unref (clockid);
795 dtmfsrc->clockid = NULL;
796 GST_OBJECT_UNLOCK (dtmfsrc);
798 if (clockret == GST_CLOCK_UNSCHEDULED) {
805 if (!gst_rtp_dtmf_src_negotiate (basesrc))
806 return GST_FLOW_NOT_NEGOTIATED;
808 /* create buffer to hold the payload */
809 *buffer = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc);
811 if (dtmfsrc->redundancy_count)
812 dtmfsrc->redundancy_count--;
814 /* Only the very first one has a marker */
815 dtmfsrc->first_packet = FALSE;
817 /* This is the end of the event */
818 if (dtmfsrc->last_packet == TRUE && dtmfsrc->redundancy_count == 0) {
820 /* Don't forget to release the stream lock */
821 gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
823 g_free (dtmfsrc->payload);
824 dtmfsrc->payload = NULL;
826 dtmfsrc->last_packet = FALSE;
833 GST_OBJECT_UNLOCK (dtmfsrc);
837 if (dtmfsrc->payload) {
838 dtmfsrc->first_packet = FALSE;
839 dtmfsrc->last_packet = TRUE;
840 /* Set the redundanc on the last packet */
841 dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
844 return GST_FLOW_WRONG_STATE;
850 gst_rtp_dtmf_src_negotiate (GstBaseSrc * basesrc)
852 GstCaps *srccaps, *peercaps;
853 GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
856 /* fill in the defaults, there properties cannot be negotiated. */
857 srccaps = gst_caps_new_simple ("application/x-rtp",
858 "media", G_TYPE_STRING, "audio",
859 "clock-rate", G_TYPE_INT, dtmfsrc->clock_rate,
860 "encoding-name", G_TYPE_STRING, "telephone-event", NULL);
862 /* the peer caps can override some of the defaults */
863 peercaps = gst_pad_peer_get_caps (GST_BASE_SRC_PAD (basesrc));
864 if (peercaps == NULL) {
865 /* no peer caps, just add the other properties */
866 gst_caps_set_simple (srccaps,
867 "payload", G_TYPE_INT, dtmfsrc->pt,
868 "ssrc", G_TYPE_UINT, dtmfsrc->current_ssrc,
869 "clock-base", G_TYPE_UINT, dtmfsrc->ts_base,
870 "seqnum-base", G_TYPE_UINT, dtmfsrc->seqnum_base, NULL);
872 GST_DEBUG_OBJECT (dtmfsrc, "no peer caps: %" GST_PTR_FORMAT, srccaps);
879 /* peer provides caps we can use to fixate, intersect. This always returns a
881 temp = gst_caps_intersect (srccaps, peercaps);
882 gst_caps_unref (srccaps);
883 gst_caps_unref (peercaps);
886 GST_DEBUG_OBJECT (dtmfsrc, "Could not get intersection with peer caps");
890 if (gst_caps_is_empty (temp)) {
891 GST_DEBUG_OBJECT (dtmfsrc, "Intersection with peer caps is empty");
892 gst_caps_unref (temp);
896 /* now fixate, start by taking the first caps */
897 gst_caps_truncate (temp);
900 /* get first structure */
901 s = gst_caps_get_structure (srccaps, 0);
903 if (gst_structure_get_int (s, "payload", &pt)) {
906 GST_LOG_OBJECT (dtmfsrc, "using peer pt %d", pt);
908 if (gst_structure_has_field (s, "payload")) {
909 /* can only fixate if there is a field */
910 gst_structure_fixate_field_nearest_int (s, "payload",
912 gst_structure_get_int (s, "payload", &pt);
913 GST_LOG_OBJECT (dtmfsrc, "using peer pt %d", pt);
915 /* no pt field, use the internal pt */
917 gst_structure_set (s, "payload", G_TYPE_INT, pt, NULL);
918 GST_LOG_OBJECT (dtmfsrc, "using internal pt", pt);
922 if (gst_structure_has_field_typed (s, "ssrc", G_TYPE_UINT)) {
923 value = gst_structure_get_value (s, "ssrc");
924 dtmfsrc->current_ssrc = g_value_get_uint (value);
925 GST_LOG_OBJECT (dtmfsrc, "using peer ssrc %08x", dtmfsrc->current_ssrc);
927 /* FIXME, fixate_nearest_uint would be even better */
928 gst_structure_set (s, "ssrc", G_TYPE_UINT, dtmfsrc->current_ssrc, NULL);
929 GST_LOG_OBJECT (dtmfsrc, "using internal ssrc %08x",
930 dtmfsrc->current_ssrc);
933 if (gst_structure_has_field_typed (s, "clock-base", G_TYPE_UINT)) {
934 value = gst_structure_get_value (s, "clock-base");
935 dtmfsrc->ts_base = g_value_get_uint (value);
936 GST_LOG_OBJECT (dtmfsrc, "using peer clock-base %u", dtmfsrc->ts_base);
938 /* FIXME, fixate_nearest_uint would be even better */
939 gst_structure_set (s, "clock-base", G_TYPE_UINT, dtmfsrc->ts_base, NULL);
940 GST_LOG_OBJECT (dtmfsrc, "using internal clock-base %u",
943 if (gst_structure_has_field_typed (s, "seqnum-base", G_TYPE_UINT)) {
944 value = gst_structure_get_value (s, "seqnum-base");
945 dtmfsrc->seqnum_base = g_value_get_uint (value);
946 GST_LOG_OBJECT (dtmfsrc, "using peer seqnum-base %u",
947 dtmfsrc->seqnum_base);
949 /* FIXME, fixate_nearest_uint would be even better */
950 gst_structure_set (s, "seqnum-base", G_TYPE_UINT, dtmfsrc->seqnum_base,
952 GST_LOG_OBJECT (dtmfsrc, "using internal seqnum-base %u",
953 dtmfsrc->seqnum_base);
955 GST_DEBUG_OBJECT (dtmfsrc, "with peer caps: %" GST_PTR_FORMAT, srccaps);
958 ret = gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), srccaps);
959 gst_caps_unref (srccaps);
961 dtmfsrc->dirty = FALSE;
969 gst_rtp_dtmf_src_ready_to_paused (GstRTPDTMFSrc *dtmfsrc)
971 if (dtmfsrc->ssrc == -1)
972 dtmfsrc->current_ssrc = g_random_int ();
974 dtmfsrc->current_ssrc = dtmfsrc->ssrc;
976 if (dtmfsrc->seqnum_offset == -1)
977 dtmfsrc->seqnum_base = g_random_int_range (0, G_MAXUINT16);
979 dtmfsrc->seqnum_base = dtmfsrc->seqnum_offset;
980 dtmfsrc->seqnum = dtmfsrc->seqnum_base;
982 if (dtmfsrc->ts_offset == -1)
983 dtmfsrc->ts_base = g_random_int ();
985 dtmfsrc->ts_base = dtmfsrc->ts_offset;
989 static GstStateChangeReturn
990 gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition)
992 GstRTPDTMFSrc *dtmfsrc;
993 GstStateChangeReturn result;
994 gboolean no_preroll = FALSE;
995 GstRTPDTMFSrcEvent *event= NULL;
997 dtmfsrc = GST_RTP_DTMF_SRC (element);
999 switch (transition) {
1000 case GST_STATE_CHANGE_READY_TO_PAUSED:
1001 gst_rtp_dtmf_src_ready_to_paused (dtmfsrc);
1003 /* Flushing the event queue */
1004 while ((event = g_async_queue_try_pop (dtmfsrc->event_queue)) != NULL)
1014 GST_ELEMENT_CLASS (parent_class)->change_state (element,
1015 transition)) == GST_STATE_CHANGE_FAILURE)
1018 switch (transition) {
1019 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1022 case GST_STATE_CHANGE_PAUSED_TO_READY:
1024 /* Flushing the event queue */
1025 while ((event = g_async_queue_try_pop (dtmfsrc->event_queue)) != NULL)
1028 /* Indicate that we don't do PRE_ROLL */
1035 if (no_preroll && result == GST_STATE_CHANGE_SUCCESS)
1036 result = GST_STATE_CHANGE_NO_PREROLL;
1043 GST_ERROR_OBJECT (dtmfsrc, "parent failed state change");
1050 gst_rtp_dtmf_src_unlock (GstBaseSrc *src) {
1051 GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (src);
1052 GstRTPDTMFSrcEvent *event = NULL;
1054 GST_DEBUG_OBJECT (dtmfsrc, "Called unlock");
1056 GST_OBJECT_LOCK (dtmfsrc);
1057 dtmfsrc->paused = TRUE;
1058 if (dtmfsrc->clockid) {
1059 gst_clock_id_unschedule (dtmfsrc->clockid);
1061 GST_OBJECT_UNLOCK (dtmfsrc);
1063 GST_DEBUG_OBJECT (dtmfsrc, "Pushing the PAUSE_TASK event on unlock request");
1064 event = g_malloc (sizeof(GstRTPDTMFSrcEvent));
1065 event->event_type = RTP_DTMF_EVENT_TYPE_PAUSE_TASK;
1066 g_async_queue_push (dtmfsrc->event_queue, event);
1073 gst_rtp_dtmf_src_unlock_stop (GstBaseSrc *src) {
1074 GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (src);
1076 GST_DEBUG_OBJECT (dtmfsrc, "Unlock stopped");
1078 GST_OBJECT_LOCK (dtmfsrc);
1079 dtmfsrc->paused = FALSE;
1080 GST_OBJECT_UNLOCK (dtmfsrc);
1086 gst_rtp_dtmf_src_plugin_init (GstPlugin * plugin)
1088 return gst_element_register (plugin, "rtpdtmfsrc",
1089 GST_RANK_NONE, GST_TYPE_RTP_DTMF_SRC);