1 /* GStreamer DTMF source
5 * Copyright (C) <2007> Collabora.
6 * Contact: Youness Alaoui <youness.alaoui@collabora.co.uk>
7 * Copyright (C) <2007> Nokia Corporation.
8 * Contact: Zeeshan Ali <zeeshan.ali@nokia.com>
9 * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
10 * 2000,2005 Wim Taymans <wim@fluendo.com>
12 * This library is free software; you can redistribute it and/or
13 * modify it under the terms of the GNU Library General Public
14 * License as published by the Free Software Foundation; either
15 * version 2 of the License, or (at your option) any later version.
17 * This library is distributed in the hope that it will be useful,
18 * but WITHOUT ANY WARRANTY; without even the implied warranty of
19 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
20 * Library General Public License for more details.
22 * You should have received a copy of the GNU Library General Public
23 * License along with this library; if not, write to the
24 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
25 * Boston, MA 02111-1307, USA.
29 * SECTION:element-dtmfsrc
30 * @short_description: Generates DTMF packets
35 * The DTMFSrc element generates DTMF (ITU-T Q.23 Specification) tone packets on request
36 * from application. The application communicates the beginning and end of a
37 * DTMF event using custom upstream gstreamer events. To report a DTMF event, an
38 * application must send an event of type GST_EVENT_CUSTOM_UPSTREAM, having a
39 * structure of name "dtmf-event" with fields set according to the following
46 * <colspec colname='Name' />
47 * <colspec colname='Type' />
48 * <colspec colname='Possible values' />
49 * <colspec colname='Purpose' />
54 * <entry>GType</entry>
55 * <entry>Possible values</entry>
56 * <entry>Purpose</entry>
63 * <entry>G_TYPE_INT</entry>
65 * <entry>The application uses this field to specify which of the two methods
66 * specified in RFC 2833 to use. The value should be 0 for tones and 1 for
67 * named events. This element is only capable of generating tones.
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 field
104 * <para>For example, the following code informs the pipeline (and in turn, the
105 * DTMFSrc element inside the pipeline) about the start of a DTMF named
106 * event '1' of volume -25 dBm0:
111 * structure = gst_structure_new ("dtmf-event",
112 * "type", G_TYPE_INT, 0,
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);
136 # define M_PI 3.14159265358979323846 /* pi */
140 #include "gstdtmfsrc.h"
142 #define GST_TONE_DTMF_TYPE_EVENT 0
143 #define DEFAULT_PACKET_INTERVAL 50 /* ms */
144 #define MIN_PACKET_INTERVAL 10 /* ms */
145 #define MAX_PACKET_INTERVAL 50 /* ms */
146 #define SAMPLE_RATE 8000
147 #define SAMPLE_SIZE 16
152 #define MAX_VOLUME 36
153 #define MIN_INTER_DIGIT_INTERVAL 50
154 #define MIN_PULSE_DURATION 70
155 #define MIN_DUTY_CYCLE (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION)
158 typedef struct st_dtmf_key {
162 float high_frequency;
165 static const DTMF_KEY DTMF_KEYS[] = {
166 {"DTMF_KEY_EVENT_0", 0, 941, 1336},
167 {"DTMF_KEY_EVENT_1", 1, 697, 1209},
168 {"DTMF_KEY_EVENT_2", 2, 697, 1336},
169 {"DTMF_KEY_EVENT_3", 3, 697, 1477},
170 {"DTMF_KEY_EVENT_4", 4, 770, 1209},
171 {"DTMF_KEY_EVENT_5", 5, 770, 1336},
172 {"DTMF_KEY_EVENT_6", 6, 770, 1477},
173 {"DTMF_KEY_EVENT_7", 7, 852, 1209},
174 {"DTMF_KEY_EVENT_8", 8, 852, 1336},
175 {"DTMF_KEY_EVENT_9", 9, 852, 1477},
176 {"DTMF_KEY_EVENT_S", 10, 941, 1209},
177 {"DTMF_KEY_EVENT_P", 11, 941, 1477},
178 {"DTMF_KEY_EVENT_A", 12, 697, 1633},
179 {"DTMF_KEY_EVENT_B", 13, 770, 1633},
180 {"DTMF_KEY_EVENT_C", 14, 852, 1633},
181 {"DTMF_KEY_EVENT_D", 15, 941, 1633},
184 #define MAX_DTMF_EVENTS 16
187 DTMF_KEY_EVENT_1 = 1,
188 DTMF_KEY_EVENT_2 = 2,
189 DTMF_KEY_EVENT_3 = 3,
190 DTMF_KEY_EVENT_4 = 4,
191 DTMF_KEY_EVENT_5 = 5,
192 DTMF_KEY_EVENT_6 = 6,
193 DTMF_KEY_EVENT_7 = 7,
194 DTMF_KEY_EVENT_8 = 8,
195 DTMF_KEY_EVENT_9 = 9,
196 DTMF_KEY_EVENT_0 = 0,
197 DTMF_KEY_EVENT_STAR = 10,
198 DTMF_KEY_EVENT_POUND = 11,
199 DTMF_KEY_EVENT_A = 12,
200 DTMF_KEY_EVENT_B = 13,
201 DTMF_KEY_EVENT_C = 14,
202 DTMF_KEY_EVENT_D = 15,
205 /* elementfactory information */
206 static const GstElementDetails gst_dtmf_src_details =
207 GST_ELEMENT_DETAILS ("DTMF tone generator",
209 "Generates DTMF tones",
210 "Youness Alaoui <youness.alaoui@collabora.co.uk>");
212 GST_DEBUG_CATEGORY_STATIC (gst_dtmf_src_debug);
213 #define GST_CAT_DEFAULT gst_dtmf_src_debug
221 static GstStaticPadTemplate gst_dtmf_src_template =
222 GST_STATIC_PAD_TEMPLATE ("src",
225 GST_STATIC_CAPS ("audio/x-raw-int, "
228 "endianness = (int) 1234, "
229 "signed = (bool) true, "
230 "rate = (int) 8000, "
231 "channels = (int) 1")
234 static GstElementClass *parent_class = NULL;
236 static void gst_dtmf_src_base_init (gpointer g_class);
237 static void gst_dtmf_src_class_init (GstDTMFSrcClass * klass);
238 static void gst_dtmf_src_init (GstDTMFSrc * dtmfsrc, gpointer g_class);
239 static void gst_dtmf_src_finalize (GObject * object);
242 gst_dtmf_src_get_type (void)
244 static GType base_src_type = 0;
246 if (G_UNLIKELY (base_src_type == 0)) {
247 static const GTypeInfo base_src_info = {
248 sizeof (GstDTMFSrcClass),
249 (GBaseInitFunc) gst_dtmf_src_base_init,
251 (GClassInitFunc) gst_dtmf_src_class_init,
256 (GInstanceInitFunc) gst_dtmf_src_init,
259 base_src_type = g_type_register_static (GST_TYPE_ELEMENT,
260 "GstDTMFSrc", &base_src_info, 0);
262 return base_src_type;
265 static void gst_dtmf_src_set_property (GObject * object, guint prop_id,
266 const GValue * value, GParamSpec * pspec);
267 static void gst_dtmf_src_get_property (GObject * object, guint prop_id,
268 GValue * value, GParamSpec * pspec);
269 static gboolean gst_dtmf_src_handle_event (GstPad * pad, GstEvent * event);
270 static GstStateChangeReturn gst_dtmf_src_change_state (GstElement * element,
271 GstStateChange transition);
272 static void gst_dtmf_src_generate_tone(GstDTMFSrcEvent *event, DTMF_KEY key, float duration,
274 static void gst_dtmf_src_push_next_tone_packet (GstDTMFSrc *dtmfsrc);
275 static void gst_dtmf_src_start (GstDTMFSrc *dtmfsrc);
276 static void gst_dtmf_src_stop (GstDTMFSrc *dtmfsrc);
277 static void gst_dtmf_src_add_start_event (GstDTMFSrc *dtmfsrc, gint event_number,
279 static void gst_dtmf_src_add_stop_event (GstDTMFSrc *dtmfsrc);
282 gst_dtmf_src_base_init (gpointer g_class)
284 GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
286 GST_DEBUG_CATEGORY_INIT (gst_dtmf_src_debug,
287 "dtmfsrc", 0, "dtmfsrc element");
289 gst_element_class_add_pad_template (element_class,
290 gst_static_pad_template_get (&gst_dtmf_src_template));
292 gst_element_class_set_details (element_class, &gst_dtmf_src_details);
296 gst_dtmf_src_class_init (GstDTMFSrcClass * klass)
298 GObjectClass *gobject_class;
299 GstElementClass *gstelement_class;
301 gobject_class = G_OBJECT_CLASS (klass);
302 gstelement_class = GST_ELEMENT_CLASS (klass);
304 parent_class = g_type_class_peek_parent (klass);
306 gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_dtmf_src_finalize);
307 gobject_class->set_property =
308 GST_DEBUG_FUNCPTR (gst_dtmf_src_set_property);
309 gobject_class->get_property =
310 GST_DEBUG_FUNCPTR (gst_dtmf_src_get_property);
312 g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_INTERVAL,
313 g_param_spec_int ("interval", "Interval between tone packets",
314 "Interval in ms between two tone packets", MIN_PACKET_INTERVAL,
315 MAX_PACKET_INTERVAL, DEFAULT_PACKET_INTERVAL, G_PARAM_READWRITE));
317 gstelement_class->change_state =
318 GST_DEBUG_FUNCPTR (gst_dtmf_src_change_state);
322 gst_dtmf_src_init (GstDTMFSrc * dtmfsrc, gpointer g_class)
325 gst_pad_new_from_static_template (&gst_dtmf_src_template, "src");
326 GST_DEBUG_OBJECT (dtmfsrc, "adding src pad");
327 gst_element_add_pad (GST_ELEMENT (dtmfsrc), dtmfsrc->srcpad);
329 gst_pad_set_event_function (dtmfsrc->srcpad, gst_dtmf_src_handle_event);
331 dtmfsrc->interval = DEFAULT_PACKET_INTERVAL;
333 dtmfsrc->event_queue = g_async_queue_new ();
334 dtmfsrc->last_event = NULL;
336 GST_DEBUG_OBJECT (dtmfsrc, "init done");
340 gst_dtmf_src_finalize (GObject * object)
344 dtmfsrc = GST_DTMF_SRC (object);
347 gst_dtmf_src_stop (dtmfsrc);
349 if (dtmfsrc->event_queue) {
350 g_async_queue_unref (dtmfsrc->event_queue);
351 dtmfsrc->event_queue = NULL;
354 G_OBJECT_CLASS (parent_class)->finalize (object);
358 gst_dtmf_src_handle_dtmf_event (GstDTMFSrc *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 (start == TRUE && event_type != GST_TONE_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_dtmf_src_add_start_event (dtmfsrc, event_number, event_volume);
390 GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
391 gst_dtmf_src_add_stop_event (dtmfsrc);
400 gst_dtmf_src_handle_custom_upstream (GstDTMFSrc *dtmfsrc,
403 gboolean result = FALSE;
404 const GstStructure *structure;
406 if (GST_STATE (dtmfsrc) != GST_STATE_PLAYING) {
407 GST_DEBUG_OBJECT (dtmfsrc, "Received event while not in PLAYING state");
411 GST_DEBUG_OBJECT (dtmfsrc, "Received event is of our interest");
412 structure = gst_event_get_structure (event);
413 if (structure && gst_structure_has_name (structure, "dtmf-event"))
414 result = gst_dtmf_src_handle_dtmf_event (dtmfsrc, structure);
421 gst_dtmf_src_handle_event (GstPad * pad, GstEvent * event)
424 gboolean result = FALSE;
426 dtmfsrc = GST_DTMF_SRC (GST_PAD_PARENT (pad));
428 GST_DEBUG_OBJECT (dtmfsrc, "Received an event on the src pad");
429 switch (GST_EVENT_TYPE (event)) {
430 case GST_EVENT_CUSTOM_UPSTREAM:
432 result = gst_dtmf_src_handle_custom_upstream (dtmfsrc, event);
435 /* Ideally this element should not be flushed but let's handle the event
436 * just in case it is */
437 case GST_EVENT_FLUSH_START:
438 gst_dtmf_src_stop (dtmfsrc);
441 case GST_EVENT_FLUSH_STOP:
442 gst_segment_init (&dtmfsrc->segment, GST_FORMAT_UNDEFINED);
444 case GST_EVENT_NEWSEGMENT:
449 gint64 start, stop, position;
451 gst_event_parse_new_segment (event, &update, &rate, &fmt, &start,
453 gst_segment_set_newsegment (&dtmfsrc->segment, update, rate, fmt,
454 start, stop, position);
458 result = gst_pad_event_default (pad, event);
462 gst_event_unref (event);
467 gst_dtmf_src_set_property (GObject * object, guint prop_id,
468 const GValue * value, GParamSpec * pspec)
472 dtmfsrc = GST_DTMF_SRC (object);
476 dtmfsrc->interval = g_value_get_int (value);
479 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
485 gst_dtmf_src_get_property (GObject * object, guint prop_id, GValue * value,
490 dtmfsrc = GST_DTMF_SRC (object);
494 g_value_set_uint (value, dtmfsrc->interval);
497 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
503 gst_dtmf_src_set_stream_lock (GstDTMFSrc *dtmfsrc, gboolean lock)
506 GstStructure *structure;
508 structure = gst_structure_new ("stream-lock",
509 "lock", G_TYPE_BOOLEAN, lock, NULL);
511 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_OOB, structure);
512 gst_pad_push_event (dtmfsrc->srcpad, event);
516 gst_dtmf_prepare_timestamps (GstDTMFSrc *dtmfsrc)
520 clock = GST_ELEMENT_CLOCK (dtmfsrc);
522 dtmfsrc->timestamp = gst_clock_get_time (GST_ELEMENT_CLOCK (dtmfsrc));
525 GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s",
526 GST_ELEMENT_NAME (dtmfsrc));
527 dtmfsrc->timestamp = GST_CLOCK_TIME_NONE;
532 gst_dtmf_src_start (GstDTMFSrc *dtmfsrc)
534 GstCaps * caps = gst_pad_get_pad_template_caps (dtmfsrc->srcpad);
536 if (!gst_pad_set_caps (dtmfsrc->srcpad, caps))
537 GST_ERROR_OBJECT (dtmfsrc,
538 "Failed to set caps %" GST_PTR_FORMAT " on src pad", caps);
540 GST_DEBUG_OBJECT (dtmfsrc,
541 "caps %" GST_PTR_FORMAT " set on src pad", caps);
544 if (!gst_pad_start_task (dtmfsrc->srcpad,
545 (GstTaskFunction) gst_dtmf_src_push_next_tone_packet, dtmfsrc)) {
546 GST_ERROR_OBJECT (dtmfsrc, "Failed to start task on src pad");
551 gst_dtmf_src_stop (GstDTMFSrc *dtmfsrc)
553 /* Don't forget to release the stream lock */
554 gst_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
557 /* Flushing the event queue */
558 GstDTMFSrcEvent *event = g_async_queue_try_pop (dtmfsrc->event_queue);
560 while (event != NULL) {
562 event = g_async_queue_try_pop (dtmfsrc->event_queue);
565 if (dtmfsrc->last_event) {
566 g_free (dtmfsrc->last_event);
567 dtmfsrc->last_event = NULL;
570 if (!gst_pad_pause_task (dtmfsrc->srcpad)) {
571 GST_ERROR_OBJECT (dtmfsrc, "Failed to pause task on src pad");
578 gst_dtmf_src_add_start_event (GstDTMFSrc *dtmfsrc, gint event_number,
582 GstDTMFSrcEvent * event = g_malloc (sizeof(GstDTMFSrcEvent));
583 event->event_type = DTMF_EVENT_TYPE_START;
585 event->event_number = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
586 event->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
588 g_async_queue_push (dtmfsrc->event_queue, event);
592 gst_dtmf_src_add_stop_event (GstDTMFSrc *dtmfsrc)
595 GstDTMFSrcEvent * event = g_malloc (sizeof(GstDTMFSrcEvent));
596 event->event_type = DTMF_EVENT_TYPE_STOP;
598 event->event_number = 0;
601 g_async_queue_push (dtmfsrc->event_queue, event);
605 gst_dtmf_src_generate_silence(GstBuffer * buffer, float duration)
609 /* Create a buffer with data set to 0 */
610 buf_size = ((duration/1000)*SAMPLE_RATE*SAMPLE_SIZE*CHANNELS)/8;
611 GST_BUFFER_SIZE (buffer) = buf_size;
612 GST_BUFFER_MALLOCDATA (buffer) = g_malloc0(buf_size);
613 GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
618 gst_dtmf_src_generate_tone(GstDTMFSrcEvent *event, DTMF_KEY key, float duration, GstBuffer * buffer)
623 double amplitude, f1, f2;
624 double volume_factor;
626 /* Create a buffer for the tone */
627 tone_size = ((duration/1000)*SAMPLE_RATE*SAMPLE_SIZE*CHANNELS)/8;
628 GST_BUFFER_SIZE (buffer) = tone_size;
629 GST_BUFFER_MALLOCDATA (buffer) = g_malloc(tone_size);
630 GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
632 p = (gint16 *) GST_BUFFER_MALLOCDATA (buffer);
634 volume_factor = pow (10, (-event->volume) / 20);
637 * For each sample point we calculate 'x' as the
638 * the amplitude value.
640 for (i = 0; i < (tone_size / (SAMPLE_SIZE/8)); i++) {
642 * We add the fundamental frequencies together.
644 f1 = sin(2 * M_PI * key.low_frequency * (event->sample / SAMPLE_RATE));
645 f2 = sin(2 * M_PI * key.high_frequency * (event->sample / SAMPLE_RATE));
647 amplitude = (f1 + f2) / 2;
649 /* Adjust the volume */
650 amplitude *= volume_factor;
652 /* Make the [-1:1] interval into a [-32767:32767] interval */
655 /* Store it in the data buffer */
656 *(p++) = (gint16) amplitude;
663 gst_dtmf_src_wait_for_buffer_ts (GstDTMFSrc *dtmfsrc, GstBuffer * buf)
667 clock = GST_ELEMENT_CLOCK (dtmfsrc);
670 GstClockReturn clock_ret;
672 clock_id = gst_clock_new_single_shot_id (clock, GST_BUFFER_TIMESTAMP (buf));
673 clock_ret = gst_clock_id_wait (clock_id, NULL);
674 if (clock_ret != GST_CLOCK_OK && clock_ret != GST_CLOCK_EARLY) {
675 GST_ERROR_OBJECT (dtmfsrc, "Failed to wait on clock %s",
676 GST_ELEMENT_NAME (clock));
678 gst_clock_id_unref (clock_id);
682 GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s",
683 GST_ELEMENT_NAME (dtmfsrc));
689 gst_dtmf_src_create_next_tone_packet (GstDTMFSrc *dtmfsrc, GstDTMFSrcEvent *event)
691 GstBuffer *buf = NULL;
695 GST_DEBUG_OBJECT (dtmfsrc,
696 "Creating buffer for tone");
698 /* create buffer to hold the tone */
699 buf = gst_buffer_new ();
701 /* The first packet must be inter digit silence, then the second and third must be the
702 * minimal pulse duration divided into two packets to make it small
704 switch(event->packet_count) {
706 duration = MIN_INTER_DIGIT_INTERVAL;
707 gst_dtmf_src_generate_silence (buf, duration);
711 /* Generate the tone */
712 duration = MIN_PULSE_DURATION / 2;
713 gst_dtmf_src_generate_tone(event, DTMF_KEYS[event->event_number], duration, buf);
716 duration = dtmfsrc->interval;
717 gst_dtmf_src_generate_tone(event, DTMF_KEYS[event->event_number], duration, buf);
720 event->packet_count++;
723 /* timestamp and duration of GstBuffer */
724 GST_BUFFER_DURATION (buf) = duration * GST_MSECOND;
725 GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp;
726 dtmfsrc->timestamp += GST_BUFFER_DURATION (buf);
728 /* FIXME: Should we sync to clock ourselves or leave it to sink */
729 gst_dtmf_src_wait_for_buffer_ts (dtmfsrc, buf);
731 /* Set caps on the buffer before pushing it */
732 gst_buffer_set_caps (buf, GST_PAD_CAPS (dtmfsrc->srcpad));
738 gst_dtmf_src_push_next_tone_packet (GstDTMFSrc *dtmfsrc)
740 GstBuffer *buf = NULL;
742 GstDTMFSrcEvent *event;
744 g_async_queue_ref (dtmfsrc->event_queue);
746 if (dtmfsrc->last_event == NULL) {
747 event = g_async_queue_pop (dtmfsrc->event_queue);
749 if (event->event_type == DTMF_EVENT_TYPE_STOP) {
750 GST_WARNING_OBJECT (dtmfsrc, "Received a DTMF stop event when already stopped");
751 } else if (event->event_type == DTMF_EVENT_TYPE_START) {
752 gst_dtmf_prepare_timestamps (dtmfsrc);
754 /* Don't forget to get exclusive access to the stream */
755 gst_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
757 event->packet_count = 0;
758 dtmfsrc->last_event = event;
760 } else if (dtmfsrc->last_event->packet_count >= 3) {
761 event = g_async_queue_try_pop (dtmfsrc->event_queue);
764 if (event->event_type == DTMF_EVENT_TYPE_START) {
765 GST_WARNING_OBJECT (dtmfsrc, "Received two consecutive DTMF start events");
766 } else if (event->event_type == DTMF_EVENT_TYPE_STOP) {
767 gst_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
768 g_free (dtmfsrc->last_event);
769 dtmfsrc->last_event = NULL;
773 g_async_queue_unref (dtmfsrc->event_queue);
775 if (dtmfsrc->last_event) {
776 buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc, dtmfsrc->last_event);
780 GST_DEBUG_OBJECT (dtmfsrc,
781 "pushing buffer on src pad of size %d", GST_BUFFER_SIZE (buf));
782 ret = gst_pad_push (dtmfsrc->srcpad, buf);
783 if (ret != GST_FLOW_OK) {
784 GST_ERROR_OBJECT (dtmfsrc, "Failed to push buffer on src pad");
787 gst_buffer_unref(buf);
788 GST_DEBUG_OBJECT (dtmfsrc, "pushed DTMF tone on src pad");
793 static GstStateChangeReturn
794 gst_dtmf_src_change_state (GstElement * element, GstStateChange transition)
797 GstStateChangeReturn result;
798 gboolean no_preroll = FALSE;
800 dtmfsrc = GST_DTMF_SRC (element);
802 switch (transition) {
803 case GST_STATE_CHANGE_READY_TO_PAUSED:
804 gst_segment_init (&dtmfsrc->segment, GST_FORMAT_UNDEFINED);
805 /* Indicate that we don't do PRE_ROLL */
808 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
809 gst_dtmf_src_start (dtmfsrc);
816 GST_ELEMENT_CLASS (parent_class)->change_state (element,
817 transition)) == GST_STATE_CHANGE_FAILURE)
820 switch (transition) {
821 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
822 /* Indicate that we don't do PRE_ROLL */
823 gst_dtmf_src_stop (dtmfsrc);
830 if (no_preroll && result == GST_STATE_CHANGE_SUCCESS)
831 result = GST_STATE_CHANGE_NO_PREROLL;
838 GST_ERROR_OBJECT (dtmfsrc, "parent failed state change");
844 gst_dtmf_src_plugin_init (GstPlugin * plugin)
846 return gst_element_register (plugin, "dtmfsrc",
847 GST_RANK_NONE, GST_TYPE_DTMF_SRC);