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 "events = (int) [ " MIN_EVENT_STRING ", " MAX_EVENT_STRING " ], "
198 "encoding-name = (string) \"telephone-event\"")
202 GST_BOILERPLATE (GstRTPDTMFSrc, gst_rtp_dtmf_src, GstBaseSrc,
206 static void gst_rtp_dtmf_src_base_init (gpointer g_class);
207 static void gst_rtp_dtmf_src_class_init (GstRTPDTMFSrcClass * klass);
208 static void gst_rtp_dtmf_src_finalize (GObject * object);
211 static void gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id,
212 const GValue * value, GParamSpec * pspec);
213 static void gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id,
214 GValue * value, GParamSpec * pspec);
215 static gboolean gst_rtp_dtmf_src_handle_event (GstBaseSrc *basesrc,
217 static GstStateChangeReturn gst_rtp_dtmf_src_change_state (GstElement * element,
218 GstStateChange transition);
219 static void gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc *dtmfsrc,
220 gint event_number, gint event_volume);
221 static void gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc *dtmfsrc);
223 static gboolean gst_rtp_dtmf_src_unlock (GstBaseSrc *src);
224 static gboolean gst_rtp_dtmf_src_unlock_stop (GstBaseSrc *src);
225 static GstFlowReturn gst_rtp_dtmf_src_create (GstBaseSrc * basesrc,
226 guint64 offset, guint length, GstBuffer ** buffer);
227 static gboolean gst_rtp_dtmf_src_negotiate (GstBaseSrc * basesrc);
231 gst_rtp_dtmf_src_base_init (gpointer g_class)
233 GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
235 GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_src_debug,
236 "rtpdtmfsrc", 0, "rtpdtmfsrc element");
238 gst_element_class_add_pad_template (element_class,
239 gst_static_pad_template_get (&gst_rtp_dtmf_src_template));
241 gst_element_class_set_details (element_class, &gst_rtp_dtmf_src_details);
245 gst_rtp_dtmf_src_class_init (GstRTPDTMFSrcClass * klass)
247 GObjectClass *gobject_class;
248 GstBaseSrcClass *gstbasesrc_class;
249 GstElementClass *gstelement_class;
251 gobject_class = G_OBJECT_CLASS (klass);
252 gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
253 gstelement_class = GST_ELEMENT_CLASS (klass);
255 parent_class = g_type_class_peek_parent (klass);
257 gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_finalize);
258 gobject_class->set_property =
259 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_set_property);
260 gobject_class->get_property =
261 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_get_property);
263 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_TIMESTAMP,
264 g_param_spec_uint ("timestamp", "Timestamp",
265 "The RTP timestamp of the last processed packet",
266 0, G_MAXUINT, 0, G_PARAM_READABLE));
267 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM,
268 g_param_spec_uint ("seqnum", "Sequence number",
269 "The RTP sequence number of the last processed packet",
270 0, G_MAXUINT, 0, G_PARAM_READABLE));
271 g_object_class_install_property (G_OBJECT_CLASS (klass),
272 PROP_TIMESTAMP_OFFSET, g_param_spec_int ("timestamp-offset",
274 "Offset to add to all outgoing timestamps (-1 = random)", -1,
275 G_MAXINT, DEFAULT_TIMESTAMP_OFFSET, G_PARAM_READWRITE));
276 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM_OFFSET,
277 g_param_spec_int ("seqnum-offset", "Sequence number Offset",
278 "Offset to add to all outgoing seqnum (-1 = random)", -1, G_MAXINT,
279 DEFAULT_SEQNUM_OFFSET, G_PARAM_READWRITE));
280 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_CLOCK_RATE,
281 g_param_spec_uint ("clock-rate", "clockrate",
282 "The clock-rate at which to generate the dtmf packets",
283 0, G_MAXUINT, DEFAULT_CLOCK_RATE, G_PARAM_READWRITE));
284 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SSRC,
285 g_param_spec_uint ("ssrc", "SSRC",
286 "The SSRC of the packets (-1 == random)",
287 0, G_MAXUINT, DEFAULT_SSRC, G_PARAM_READWRITE));
288 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PT,
289 g_param_spec_uint ("pt", "payload type",
290 "The payload type of the packets",
291 0, 0x80, DEFAULT_PT, G_PARAM_READWRITE));
292 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_INTERVAL,
293 g_param_spec_uint ("interval", "Interval between rtp packets",
294 "Interval in ms between two rtp packets", MIN_PACKET_INTERVAL,
295 MAX_PACKET_INTERVAL, DEFAULT_PACKET_INTERVAL, G_PARAM_READWRITE));
296 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_REDUNDANCY,
297 g_param_spec_uint ("packet-redundancy", "Packet Redundancy",
298 "Number of packets to send to indicate start and stop dtmf events",
299 MIN_PACKET_REDUNDANCY, MAX_PACKET_REDUNDANCY,
300 DEFAULT_PACKET_REDUNDANCY, G_PARAM_READWRITE));
302 gstelement_class->change_state =
303 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_change_state);
305 gstbasesrc_class->unlock =
306 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_unlock);
307 gstbasesrc_class->unlock_stop =
308 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_unlock_stop);
310 gstbasesrc_class->event =
311 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_handle_event);
312 gstbasesrc_class->create =
313 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_create);
314 gstbasesrc_class->negotiate =
315 GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_negotiate);
319 gst_rtp_dtmf_src_init (GstRTPDTMFSrc * object, GstRTPDTMFSrcClass * g_class)
321 gst_base_src_set_format (GST_BASE_SRC (object), GST_FORMAT_TIME);
322 gst_base_src_set_live (GST_BASE_SRC (object), TRUE);
324 object->ssrc = DEFAULT_SSRC;
325 object->seqnum_offset = DEFAULT_SEQNUM_OFFSET;
326 object->ts_offset = DEFAULT_TIMESTAMP_OFFSET;
327 object->pt = DEFAULT_PT;
328 object->clock_rate = DEFAULT_CLOCK_RATE;
329 object->interval = DEFAULT_PACKET_INTERVAL;
330 object->packet_redundancy = DEFAULT_PACKET_REDUNDANCY;
332 object->event_queue = g_async_queue_new ();
333 object->payload = NULL;
335 GST_DEBUG_OBJECT (object, "init done");
339 gst_rtp_dtmf_src_finalize (GObject * object)
341 GstRTPDTMFSrc *dtmfsrc;
343 dtmfsrc = GST_RTP_DTMF_SRC (object);
345 if (dtmfsrc->event_queue) {
346 g_async_queue_unref (dtmfsrc->event_queue);
347 dtmfsrc->event_queue = NULL;
351 G_OBJECT_CLASS (parent_class)->finalize (object);
355 gst_rtp_dtmf_src_handle_dtmf_event (GstRTPDTMFSrc *dtmfsrc,
356 const GstStructure * event_structure)
362 if (!gst_structure_get_int (event_structure, "type", &event_type) ||
363 !gst_structure_get_boolean (event_structure, "start", &start) ||
364 event_type != GST_RTP_DTMF_TYPE_EVENT)
367 if (gst_structure_get_int (event_structure, "method", &method)) {
377 if (!gst_structure_get_int (event_structure, "number", &event_number) ||
378 !gst_structure_get_int (event_structure, "volume", &event_volume))
381 GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d",
382 event_number, event_volume);
383 gst_rtp_dtmf_src_add_start_event (dtmfsrc, event_number, event_volume);
387 GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
388 gst_rtp_dtmf_src_add_stop_event (dtmfsrc);
397 gst_rtp_dtmf_src_handle_custom_upstream (GstRTPDTMFSrc *dtmfsrc,
400 gboolean result = FALSE;
402 const GstStructure *structure;
405 GstStateChangeReturn ret;
407 ret = gst_element_get_state (GST_ELEMENT (dtmfsrc), &state, NULL, 0);
408 if (ret != GST_STATE_CHANGE_SUCCESS || state != GST_STATE_PLAYING) {
409 GST_DEBUG_OBJECT (dtmfsrc, "Received event while not in PLAYING state");
413 GST_DEBUG_OBJECT (dtmfsrc, "Received event is of our interest");
414 structure = gst_event_get_structure (event);
415 struct_str = gst_structure_to_string (structure);
416 GST_DEBUG_OBJECT (dtmfsrc, "Event has structure %s", struct_str);
418 if (structure && gst_structure_has_name (structure, "dtmf-event"))
419 result = gst_rtp_dtmf_src_handle_dtmf_event (dtmfsrc, structure);
426 gst_rtp_dtmf_src_handle_event (GstBaseSrc *basesrc, GstEvent * event)
428 GstRTPDTMFSrc *dtmfsrc;
429 gboolean result = FALSE;
431 dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
433 GST_DEBUG_OBJECT (dtmfsrc, "Received an event on the src pad");
434 if (GST_EVENT_TYPE (event) == GST_EVENT_CUSTOM_UPSTREAM) {
435 result = gst_rtp_dtmf_src_handle_custom_upstream (dtmfsrc, event);
442 gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id,
443 const GValue * value, GParamSpec * pspec)
445 GstRTPDTMFSrc *dtmfsrc;
447 dtmfsrc = GST_RTP_DTMF_SRC (object);
450 case PROP_TIMESTAMP_OFFSET:
451 dtmfsrc->ts_offset = g_value_get_int (value);
453 case PROP_SEQNUM_OFFSET:
454 dtmfsrc->seqnum_offset = g_value_get_int (value);
456 case PROP_CLOCK_RATE:
457 dtmfsrc->clock_rate = g_value_get_uint (value);
458 dtmfsrc->dirty = TRUE;
461 dtmfsrc->ssrc = g_value_get_uint (value);
464 dtmfsrc->pt = g_value_get_uint (value);
465 dtmfsrc->dirty = TRUE;
468 dtmfsrc->interval = g_value_get_uint (value);
470 case PROP_REDUNDANCY:
471 dtmfsrc->packet_redundancy = g_value_get_uint (value);
474 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
480 gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id, GValue * value,
483 GstRTPDTMFSrc *dtmfsrc;
485 dtmfsrc = GST_RTP_DTMF_SRC (object);
488 case PROP_TIMESTAMP_OFFSET:
489 g_value_set_int (value, dtmfsrc->ts_offset);
491 case PROP_SEQNUM_OFFSET:
492 g_value_set_int (value, dtmfsrc->seqnum_offset);
494 case PROP_CLOCK_RATE:
495 g_value_set_uint (value, dtmfsrc->clock_rate);
498 g_value_set_uint (value, dtmfsrc->ssrc);
501 g_value_set_uint (value, dtmfsrc->pt);
504 g_value_set_uint (value, dtmfsrc->rtp_timestamp);
507 g_value_set_uint (value, dtmfsrc->seqnum);
510 g_value_set_uint (value, dtmfsrc->interval);
512 case PROP_REDUNDANCY:
513 g_value_set_uint (value, dtmfsrc->packet_redundancy);
516 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
522 gst_rtp_dtmf_src_set_stream_lock (GstRTPDTMFSrc *dtmfsrc, gboolean lock)
525 GstStructure *structure;
527 structure = gst_structure_new ("stream-lock",
528 "lock", G_TYPE_BOOLEAN, lock, NULL);
530 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_OOB, structure);
531 if (!gst_pad_push_event (GST_BASE_SRC_PAD (dtmfsrc), event)) {
532 GST_WARNING_OBJECT (dtmfsrc, "stream-lock event not handled");
538 gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc *dtmfsrc)
541 GstClockTime base_time;
543 base_time = gst_element_get_base_time (GST_ELEMENT (dtmfsrc));
545 clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
547 dtmfsrc->timestamp = gst_clock_get_time (clock)
548 + (MIN_INTER_DIGIT_INTERVAL * GST_MSECOND) - base_time;
549 dtmfsrc->start_timestamp = dtmfsrc->timestamp;
550 gst_object_unref (clock);
552 gchar *dtmf_name = gst_element_get_name (dtmfsrc);
553 GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s", dtmf_name);
554 dtmfsrc->timestamp = GST_CLOCK_TIME_NONE;
558 dtmfsrc->rtp_timestamp = dtmfsrc->ts_base +
559 gst_util_uint64_scale_int (
560 gst_segment_to_running_time (&GST_BASE_SRC (dtmfsrc)->segment,
561 GST_FORMAT_TIME, dtmfsrc->timestamp),
562 dtmfsrc->clock_rate, GST_SECOND);
567 gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc *dtmfsrc, gint event_number,
571 GstRTPDTMFSrcEvent * event = g_malloc (sizeof(GstRTPDTMFSrcEvent));
572 event->event_type = RTP_DTMF_EVENT_TYPE_START;
574 event->payload = g_new0 (GstRTPDTMFPayload, 1);
575 event->payload->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
576 event->payload->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
577 event->payload->duration = dtmfsrc->interval * dtmfsrc->clock_rate / 1000;
579 g_async_queue_push (dtmfsrc->event_queue, event);
583 gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc *dtmfsrc)
586 GstRTPDTMFSrcEvent * event = g_malloc (sizeof(GstRTPDTMFSrcEvent));
587 event->event_type = RTP_DTMF_EVENT_TYPE_STOP;
589 g_async_queue_push (dtmfsrc->event_queue, event);
594 gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
596 gst_rtp_buffer_set_ssrc (buf, dtmfsrc->current_ssrc);
597 gst_rtp_buffer_set_payload_type (buf, dtmfsrc->pt);
598 /* Only the very first packet gets a marker */
599 if (dtmfsrc->first_packet) {
600 gst_rtp_buffer_set_marker (buf, TRUE);
601 } else if (dtmfsrc->last_packet) {
602 dtmfsrc->payload->e = 1;
606 gst_rtp_buffer_set_seq (buf, dtmfsrc->seqnum);
608 /* timestamp of RTP header */
609 gst_rtp_buffer_set_timestamp (buf, dtmfsrc->rtp_timestamp);
613 gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
615 GstRTPDTMFPayload *payload;
617 gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc, buf);
619 /* timestamp and duration of GstBuffer */
620 /* Redundant buffer have no duration ... */
621 if (dtmfsrc->redundancy_count > 1)
622 GST_BUFFER_DURATION (buf) = 0;
624 GST_BUFFER_DURATION (buf) = dtmfsrc->interval * GST_MSECOND;
625 GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp;
627 dtmfsrc->timestamp += GST_BUFFER_DURATION (buf);
629 payload = (GstRTPDTMFPayload *) gst_rtp_buffer_get_payload (buf);
631 /* copy payload and convert to network-byte order */
632 g_memmove (payload, dtmfsrc->payload, sizeof (GstRTPDTMFPayload));
633 /* Force the packet duration to a certain minumum
634 * if its the end of the event
637 payload->duration < MIN_PULSE_DURATION * dtmfsrc->clock_rate / 1000 )
638 payload->duration = MIN_PULSE_DURATION * dtmfsrc->clock_rate / 1000;
640 payload->duration = g_htons (payload->duration);
643 /* duration of DTMF payloadfor the NEXT packet */
644 /* not updated for redundant packets */
645 if (dtmfsrc->redundancy_count == 0)
646 dtmfsrc->payload->duration +=
647 dtmfsrc->interval * dtmfsrc->clock_rate / 1000;
652 gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
654 GstBuffer *buf = NULL;
656 /* create buffer to hold the payload */
657 buf = gst_rtp_buffer_new_allocate (sizeof (GstRTPDTMFPayload), 0, 0);
659 gst_rtp_dtmf_prepare_buffer_data (dtmfsrc, buf);
661 /* Set caps on the buffer before pushing it */
662 gst_buffer_set_caps (buf, GST_PAD_CAPS (GST_BASE_SRC_PAD (dtmfsrc)));
668 gst_rtp_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
669 guint length, GstBuffer ** buffer)
671 GstRTPDTMFSrcEvent *event;
672 GstRTPDTMFSrc * dtmfsrc;
675 GstClockReturn clockret;
677 dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
681 if (dtmfsrc->payload == NULL) {
682 GST_DEBUG_OBJECT (dtmfsrc, "popping");
683 event = g_async_queue_pop (dtmfsrc->event_queue);
685 GST_DEBUG_OBJECT (dtmfsrc, "popped %d", event->event_type);
687 switch (event->event_type) {
688 case RTP_DTMF_EVENT_TYPE_STOP:
689 GST_WARNING_OBJECT (dtmfsrc,
690 "Received a DTMF stop event when already stopped");
693 case RTP_DTMF_EVENT_TYPE_START:
694 dtmfsrc->first_packet = TRUE;
695 dtmfsrc->last_packet = FALSE;
696 /* Set the redundanc on the first packet */
697 dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
698 gst_rtp_dtmf_prepare_timestamps (dtmfsrc);
700 /* Don't forget to get exclusive access to the stream */
701 gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
703 dtmfsrc->payload = event->payload;
706 case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
708 * We're pushing it back because it has to stay in there until
709 * the task is really paused (and the queue will then be flushed
711 GST_OBJECT_LOCK (dtmfsrc);
712 if (dtmfsrc->paused) {
713 g_async_queue_push (dtmfsrc->event_queue, event);
716 GST_OBJECT_UNLOCK (dtmfsrc);
721 } else if (!dtmfsrc->first_packet && !dtmfsrc->last_packet &&
722 (dtmfsrc->timestamp - dtmfsrc->start_timestamp)/GST_MSECOND >=
723 MIN_PULSE_DURATION) {
724 GST_DEBUG_OBJECT (dtmfsrc, "try popping");
725 event = g_async_queue_try_pop (dtmfsrc->event_queue);
729 GST_DEBUG_OBJECT (dtmfsrc, "try popped %d", event->event_type);
731 switch (event->event_type) {
732 case RTP_DTMF_EVENT_TYPE_START:
733 GST_WARNING_OBJECT (dtmfsrc,
734 "Received two consecutive DTMF start events");
737 case RTP_DTMF_EVENT_TYPE_STOP:
738 dtmfsrc->first_packet = FALSE;
739 dtmfsrc->last_packet = TRUE;
740 /* Set the redundanc on the last packet */
741 dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
744 case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
746 * We're pushing it back because it has to stay in there until
747 * the task is really paused (and the queue will then be flushed)
749 GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
750 GST_OBJECT_LOCK (dtmfsrc);
751 if (dtmfsrc->paused) {
752 g_async_queue_push (dtmfsrc->event_queue, event);
755 GST_OBJECT_UNLOCK (dtmfsrc);
761 } while (dtmfsrc->payload == NULL);
764 GST_DEBUG_OBJECT (dtmfsrc, "Processed events, now lets wait on the clock");
766 clock = gst_element_get_clock (GST_ELEMENT (basesrc));
768 clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
769 gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
770 gst_object_unref (clock);
772 GST_OBJECT_LOCK (dtmfsrc);
773 if (!dtmfsrc->paused) {
774 dtmfsrc->clockid = clockid;
775 GST_OBJECT_UNLOCK (dtmfsrc);
777 clockret = gst_clock_id_wait (clockid, NULL);
779 GST_OBJECT_LOCK (dtmfsrc);
781 clockret = GST_CLOCK_UNSCHEDULED;
783 clockret = GST_CLOCK_UNSCHEDULED;
785 gst_clock_id_unref (clockid);
786 dtmfsrc->clockid = NULL;
787 GST_OBJECT_UNLOCK (dtmfsrc);
789 if (clockret == GST_CLOCK_UNSCHEDULED) {
796 if (!gst_rtp_dtmf_src_negotiate (basesrc))
797 return GST_FLOW_NOT_NEGOTIATED;
799 /* create buffer to hold the payload */
800 *buffer = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc);
802 if (dtmfsrc->redundancy_count)
803 dtmfsrc->redundancy_count--;
805 /* Only the very first one has a marker */
806 dtmfsrc->first_packet = FALSE;
808 /* This is the end of the event */
809 if (dtmfsrc->last_packet == TRUE && dtmfsrc->redundancy_count == 0) {
811 /* Don't forget to release the stream lock */
812 gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
814 g_free (dtmfsrc->payload);
815 dtmfsrc->payload = NULL;
817 dtmfsrc->last_packet = FALSE;
824 GST_OBJECT_UNLOCK (dtmfsrc);
828 if (dtmfsrc->payload) {
829 dtmfsrc->first_packet = FALSE;
830 dtmfsrc->last_packet = TRUE;
831 /* Set the redundanc on the last packet */
832 dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
835 return GST_FLOW_WRONG_STATE;
841 gst_rtp_dtmf_src_negotiate (GstBaseSrc * basesrc)
843 GstCaps *srccaps, *peercaps;
844 GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (basesrc);
847 /* fill in the defaults, there properties cannot be negotiated. */
848 srccaps = gst_caps_new_simple ("application/x-rtp",
849 "media", G_TYPE_STRING, "audio",
850 "clock-rate", G_TYPE_INT, dtmfsrc->clock_rate,
851 "encoding-name", G_TYPE_STRING, "telephone-event", NULL);
853 /* the peer caps can override some of the defaults */
854 peercaps = gst_pad_peer_get_caps (GST_BASE_SRC_PAD (basesrc));
855 if (peercaps == NULL) {
856 /* no peer caps, just add the other properties */
857 gst_caps_set_simple (srccaps,
858 "payload", G_TYPE_INT, dtmfsrc->pt,
859 "ssrc", G_TYPE_UINT, dtmfsrc->current_ssrc,
860 "clock-base", G_TYPE_UINT, dtmfsrc->ts_base,
861 "seqnum-base", G_TYPE_UINT, dtmfsrc->seqnum_base, NULL);
863 GST_DEBUG_OBJECT (dtmfsrc, "no peer caps: %" GST_PTR_FORMAT, srccaps);
870 /* peer provides caps we can use to fixate, intersect. This always returns a
872 temp = gst_caps_intersect (srccaps, peercaps);
873 gst_caps_unref (srccaps);
874 gst_caps_unref (peercaps);
877 GST_DEBUG_OBJECT (dtmfsrc, "Could not get intersection with peer caps");
881 if (gst_caps_is_empty (temp)) {
882 GST_DEBUG_OBJECT (dtmfsrc, "Intersection with peer caps is empty");
883 gst_caps_unref (temp);
887 /* now fixate, start by taking the first caps */
888 gst_caps_truncate (temp);
891 /* get first structure */
892 s = gst_caps_get_structure (srccaps, 0);
894 if (gst_structure_get_int (s, "dtmfsrc", &pt)) {
897 GST_LOG_OBJECT (dtmfsrc, "using peer pt %d", pt);
899 if (gst_structure_has_field (s, "payload")) {
900 /* can only fixate if there is a field */
901 gst_structure_fixate_field_nearest_int (s, "payload",
903 gst_structure_get_int (s, "payload", &pt);
904 GST_LOG_OBJECT (dtmfsrc, "using peer pt %d", pt);
906 /* no pt field, use the internal pt */
908 gst_structure_set (s, "payload", G_TYPE_INT, pt, NULL);
909 GST_LOG_OBJECT (dtmfsrc, "using internal pt", pt);
913 if (gst_structure_has_field_typed (s, "ssrc", G_TYPE_UINT)) {
914 value = gst_structure_get_value (s, "ssrc");
915 dtmfsrc->current_ssrc = g_value_get_uint (value);
916 GST_LOG_OBJECT (dtmfsrc, "using peer ssrc %08x", dtmfsrc->current_ssrc);
918 /* FIXME, fixate_nearest_uint would be even better */
919 gst_structure_set (s, "ssrc", G_TYPE_UINT, dtmfsrc->current_ssrc, NULL);
920 GST_LOG_OBJECT (dtmfsrc, "using internal ssrc %08x",
921 dtmfsrc->current_ssrc);
924 if (gst_structure_has_field_typed (s, "clock-base", G_TYPE_UINT)) {
925 value = gst_structure_get_value (s, "clock-base");
926 dtmfsrc->ts_base = g_value_get_uint (value);
927 GST_LOG_OBJECT (dtmfsrc, "using peer clock-base %u", dtmfsrc->ts_base);
929 /* FIXME, fixate_nearest_uint would be even better */
930 gst_structure_set (s, "clock-base", G_TYPE_UINT, dtmfsrc->ts_base, NULL);
931 GST_LOG_OBJECT (dtmfsrc, "using internal clock-base %u",
934 if (gst_structure_has_field_typed (s, "seqnum-base", G_TYPE_UINT)) {
935 value = gst_structure_get_value (s, "seqnum-base");
936 dtmfsrc->seqnum_base = g_value_get_uint (value);
937 GST_LOG_OBJECT (dtmfsrc, "using peer seqnum-base %u",
938 dtmfsrc->seqnum_base);
940 /* FIXME, fixate_nearest_uint would be even better */
941 gst_structure_set (s, "seqnum-base", G_TYPE_UINT, dtmfsrc->seqnum_base,
943 GST_LOG_OBJECT (dtmfsrc, "using internal seqnum-base %u",
944 dtmfsrc->seqnum_base);
946 GST_DEBUG_OBJECT (dtmfsrc, "with peer caps: %" GST_PTR_FORMAT, srccaps);
949 ret = gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), srccaps);
950 gst_caps_unref (srccaps);
952 dtmfsrc->dirty = FALSE;
960 gst_rtp_dtmf_src_ready_to_paused (GstRTPDTMFSrc *dtmfsrc)
962 if (dtmfsrc->ssrc == -1)
963 dtmfsrc->current_ssrc = g_random_int ();
965 dtmfsrc->current_ssrc = dtmfsrc->ssrc;
967 if (dtmfsrc->seqnum_offset == -1)
968 dtmfsrc->seqnum_base = g_random_int_range (0, G_MAXUINT16);
970 dtmfsrc->seqnum_base = dtmfsrc->seqnum_offset;
971 dtmfsrc->seqnum = dtmfsrc->seqnum_base;
973 if (dtmfsrc->ts_offset == -1)
974 dtmfsrc->ts_base = g_random_int ();
976 dtmfsrc->ts_base = dtmfsrc->ts_offset;
980 static GstStateChangeReturn
981 gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition)
983 GstRTPDTMFSrc *dtmfsrc;
984 GstStateChangeReturn result;
985 gboolean no_preroll = FALSE;
986 GstRTPDTMFSrcEvent *event= NULL;
988 dtmfsrc = GST_RTP_DTMF_SRC (element);
990 switch (transition) {
991 case GST_STATE_CHANGE_READY_TO_PAUSED:
992 gst_rtp_dtmf_src_ready_to_paused (dtmfsrc);
994 /* Flushing the event queue */
995 while ((event = g_async_queue_try_pop (dtmfsrc->event_queue)) != NULL)
1005 GST_ELEMENT_CLASS (parent_class)->change_state (element,
1006 transition)) == GST_STATE_CHANGE_FAILURE)
1009 switch (transition) {
1010 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1013 case GST_STATE_CHANGE_PAUSED_TO_READY:
1015 /* Flushing the event queue */
1016 while ((event = g_async_queue_try_pop (dtmfsrc->event_queue)) != NULL)
1019 /* Indicate that we don't do PRE_ROLL */
1026 if (no_preroll && result == GST_STATE_CHANGE_SUCCESS)
1027 result = GST_STATE_CHANGE_NO_PREROLL;
1034 GST_ERROR_OBJECT (dtmfsrc, "parent failed state change");
1041 gst_rtp_dtmf_src_unlock (GstBaseSrc *src) {
1042 GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (src);
1043 GstRTPDTMFSrcEvent *event = NULL;
1045 GST_DEBUG_OBJECT (dtmfsrc, "Called unlock");
1047 GST_OBJECT_LOCK (dtmfsrc);
1048 dtmfsrc->paused = TRUE;
1049 if (dtmfsrc->clockid) {
1050 gst_clock_id_unschedule (dtmfsrc->clockid);
1052 GST_OBJECT_UNLOCK (dtmfsrc);
1054 GST_DEBUG_OBJECT (dtmfsrc, "Pushing the PAUSE_TASK event on unlock request");
1055 event = g_malloc (sizeof(GstRTPDTMFSrcEvent));
1056 event->event_type = RTP_DTMF_EVENT_TYPE_PAUSE_TASK;
1057 g_async_queue_push (dtmfsrc->event_queue, event);
1064 gst_rtp_dtmf_src_unlock_stop (GstBaseSrc *src) {
1065 GstRTPDTMFSrc *dtmfsrc = GST_RTP_DTMF_SRC (src);
1067 GST_DEBUG_OBJECT (dtmfsrc, "Unlock stopped");
1069 GST_OBJECT_LOCK (dtmfsrc);
1070 dtmfsrc->paused = FALSE;
1071 GST_OBJECT_UNLOCK (dtmfsrc);
1077 gst_rtp_dtmf_src_plugin_init (GstPlugin * plugin)
1079 return gst_element_register (plugin, "rtpdtmfsrc",
1080 GST_RANK_NONE, GST_TYPE_RTP_DTMF_SRC);