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998 lines
28 KiB
C
998 lines
28 KiB
C
/* GStreamer DTMF source
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*
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* gstdtmfsrc.c:
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*
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* Copyright (C) <2007> Collabora.
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* Contact: Youness Alaoui <youness.alaoui@collabora.co.uk>
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* Copyright (C) <2007> Nokia Corporation.
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* Contact: Zeeshan Ali <zeeshan.ali@nokia.com>
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2000,2005 Wim Taymans <wim@fluendo.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-dtmfsrc
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* @see_also: rtpdtmsrc, rtpdtmfmuxx
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*
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* The DTMFSrc element generates DTMF (ITU-T Q.23 Specification) tone packets on request
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* from application. The application communicates the beginning and end of a
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* DTMF event using custom upstream gstreamer events. To report a DTMF event, an
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* application must send an event of type GST_EVENT_CUSTOM_UPSTREAM, having a
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* structure of name "dtmf-event" with fields set according to the following
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* table:
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*
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* <informaltable>
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* <tgroup cols='4'>
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* <colspec colname='Name' />
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* <colspec colname='Type' />
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* <colspec colname='Possible values' />
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* <colspec colname='Purpose' />
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* <thead>
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* <row>
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* <entry>Name</entry>
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* <entry>GType</entry>
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* <entry>Possible values</entry>
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* <entry>Purpose</entry>
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* </row>
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* </thead>
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* <tbody>
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* <row>
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* <entry>type</entry>
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* <entry>G_TYPE_INT</entry>
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* <entry>0-1</entry>
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* <entry>The application uses this field to specify which of the two methods
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* specified in RFC 2833 to use. The value should be 0 for tones and 1 for
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* named events. Tones are specified by their frequencies and events are specied
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* by their number. This element can only take events as input. Do not confuse
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* with "method" which specified the output.
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* </entry>
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* </row>
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* <row>
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* <entry>number</entry>
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* <entry>G_TYPE_INT</entry>
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* <entry>0-15</entry>
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* <entry>The event number.</entry>
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* </row>
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* <row>
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* <entry>volume</entry>
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* <entry>G_TYPE_INT</entry>
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* <entry>0-36</entry>
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* <entry>This field describes the power level of the tone, expressed in dBm0
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* after dropping the sign. Power levels range from 0 to -63 dBm0. The range of
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* valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE.
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* </entry>
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* </row>
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* <row>
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* <entry>start</entry>
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* <entry>G_TYPE_BOOLEAN</entry>
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* <entry>True or False</entry>
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* <entry>Whether the event is starting or ending.</entry>
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* </row>
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* <row>
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* <entry>method</entry>
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* <entry>G_TYPE_INT</entry>
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* <entry>2</entry>
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* <entry>The method used for sending event, this element will react if this
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* field is absent or 2.
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* </entry>
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* </row>
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* </tbody>
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* </tgroup>
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* </informaltable>
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*
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* For example, the following code informs the pipeline (and in turn, the
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* DTMFSrc element inside the pipeline) about the start of a DTMF named
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* event '1' of volume -25 dBm0:
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*
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* <programlisting>
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* structure = gst_structure_new ("dtmf-event",
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* "type", G_TYPE_INT, 1,
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* "number", G_TYPE_INT, 1,
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* "volume", G_TYPE_INT, 25,
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* "start", G_TYPE_BOOLEAN, TRUE, NULL);
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*
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* event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure);
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* gst_element_send_event (pipeline, event);
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* </programlisting>
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*
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* When a DTMF tone actually starts or stop, a "dtmf-event-processed"
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* element #GstMessage with the same fields as the "dtmf-event"
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* #GstEvent that was used to request the event. Also, if any event
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* has not been processed when the element goes from the PAUSED to the
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* READY state, then a "dtmf-event-dropped" message is posted on the
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* #GstBus in the order that they were received.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <stdlib.h>
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#include <string.h>
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#include <math.h>
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#include <glib.h>
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#include "gstdtmfcommon.h"
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#include "gstdtmfsrc.h"
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#include <gst/audio/audio.h>
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#define GST_TONE_DTMF_TYPE_EVENT 1
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#define DEFAULT_PACKET_INTERVAL 50 /* ms */
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#define MIN_PACKET_INTERVAL 10 /* ms */
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#define MAX_PACKET_INTERVAL 50 /* ms */
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#define DEFAULT_SAMPLE_RATE 8000
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#define SAMPLE_SIZE 16
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#define CHANNELS 1
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#define MIN_DUTY_CYCLE (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION)
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typedef struct st_dtmf_key
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{
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const char *event_name;
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int event_encoding;
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float low_frequency;
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float high_frequency;
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} DTMF_KEY;
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static const DTMF_KEY DTMF_KEYS[] = {
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{"DTMF_KEY_EVENT_0", 0, 941, 1336},
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{"DTMF_KEY_EVENT_1", 1, 697, 1209},
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{"DTMF_KEY_EVENT_2", 2, 697, 1336},
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{"DTMF_KEY_EVENT_3", 3, 697, 1477},
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{"DTMF_KEY_EVENT_4", 4, 770, 1209},
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{"DTMF_KEY_EVENT_5", 5, 770, 1336},
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{"DTMF_KEY_EVENT_6", 6, 770, 1477},
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{"DTMF_KEY_EVENT_7", 7, 852, 1209},
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{"DTMF_KEY_EVENT_8", 8, 852, 1336},
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{"DTMF_KEY_EVENT_9", 9, 852, 1477},
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{"DTMF_KEY_EVENT_S", 10, 941, 1209},
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{"DTMF_KEY_EVENT_P", 11, 941, 1477},
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{"DTMF_KEY_EVENT_A", 12, 697, 1633},
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{"DTMF_KEY_EVENT_B", 13, 770, 1633},
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{"DTMF_KEY_EVENT_C", 14, 852, 1633},
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{"DTMF_KEY_EVENT_D", 15, 941, 1633},
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};
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#define MAX_DTMF_EVENTS 16
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enum
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{
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DTMF_KEY_EVENT_1 = 1,
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DTMF_KEY_EVENT_2 = 2,
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DTMF_KEY_EVENT_3 = 3,
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DTMF_KEY_EVENT_4 = 4,
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DTMF_KEY_EVENT_5 = 5,
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DTMF_KEY_EVENT_6 = 6,
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DTMF_KEY_EVENT_7 = 7,
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DTMF_KEY_EVENT_8 = 8,
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DTMF_KEY_EVENT_9 = 9,
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DTMF_KEY_EVENT_0 = 0,
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DTMF_KEY_EVENT_STAR = 10,
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DTMF_KEY_EVENT_POUND = 11,
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DTMF_KEY_EVENT_A = 12,
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DTMF_KEY_EVENT_B = 13,
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DTMF_KEY_EVENT_C = 14,
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DTMF_KEY_EVENT_D = 15,
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};
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GST_DEBUG_CATEGORY_STATIC (gst_dtmf_src_debug);
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#define GST_CAT_DEFAULT gst_dtmf_src_debug
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enum
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{
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PROP_0,
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PROP_INTERVAL,
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};
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static GstStaticPadTemplate gst_dtmf_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) \"" GST_AUDIO_NE (S16) "\", "
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"rate = " GST_AUDIO_RATE_RANGE ", " "channels = (int) 1")
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);
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#define parent_class gst_dtmf_src_parent_class
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G_DEFINE_TYPE (GstDTMFSrc, gst_dtmf_src, GST_TYPE_BASE_SRC);
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static void gst_dtmf_src_finalize (GObject * object);
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static void gst_dtmf_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_dtmf_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_dtmf_src_handle_event (GstBaseSrc * src, GstEvent * event);
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static gboolean gst_dtmf_src_send_event (GstElement * src, GstEvent * event);
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static GstStateChangeReturn gst_dtmf_src_change_state (GstElement * element,
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GstStateChange transition);
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static GstFlowReturn gst_dtmf_src_create (GstBaseSrc * basesrc,
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guint64 offset, guint length, GstBuffer ** buffer);
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static void gst_dtmf_src_add_start_event (GstDTMFSrc * dtmfsrc,
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gint event_number, gint event_volume);
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static void gst_dtmf_src_add_stop_event (GstDTMFSrc * dtmfsrc);
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static gboolean gst_dtmf_src_unlock (GstBaseSrc * src);
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static gboolean gst_dtmf_src_unlock_stop (GstBaseSrc * src);
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static gboolean gst_dtmf_src_negotiate (GstBaseSrc * basesrc);
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static gboolean gst_dtmf_src_query (GstBaseSrc * basesrc, GstQuery * query);
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static void
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gst_dtmf_src_class_init (GstDTMFSrcClass * klass)
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{
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GObjectClass *gobject_class;
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GstBaseSrcClass *gstbasesrc_class;
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GstElementClass *gstelement_class;
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gobject_class = G_OBJECT_CLASS (klass);
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gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
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gstelement_class = GST_ELEMENT_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (gst_dtmf_src_debug, "dtmfsrc", 0, "dtmfsrc element");
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gst_element_class_add_static_pad_template (gstelement_class,
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&gst_dtmf_src_template);
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gst_element_class_set_static_metadata (gstelement_class,
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"DTMF tone generator", "Source/Audio", "Generates DTMF tones",
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"Youness Alaoui <youness.alaoui@collabora.co.uk>");
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gobject_class->finalize = gst_dtmf_src_finalize;
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gobject_class->set_property = gst_dtmf_src_set_property;
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gobject_class->get_property = gst_dtmf_src_get_property;
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_INTERVAL,
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g_param_spec_uint ("interval", "Interval between tone packets",
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"Interval in ms between two tone packets", MIN_PACKET_INTERVAL,
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MAX_PACKET_INTERVAL, DEFAULT_PACKET_INTERVAL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_dtmf_src_change_state);
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gstelement_class->send_event = GST_DEBUG_FUNCPTR (gst_dtmf_src_send_event);
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gstbasesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_dtmf_src_unlock);
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gstbasesrc_class->unlock_stop = GST_DEBUG_FUNCPTR (gst_dtmf_src_unlock_stop);
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gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_dtmf_src_handle_event);
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gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_dtmf_src_create);
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gstbasesrc_class->negotiate = GST_DEBUG_FUNCPTR (gst_dtmf_src_negotiate);
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gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_dtmf_src_query);
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}
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static void
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event_free (GstDTMFSrcEvent * event)
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{
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if (event)
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g_slice_free (GstDTMFSrcEvent, event);
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}
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static void
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gst_dtmf_src_init (GstDTMFSrc * dtmfsrc)
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{
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/* we operate in time */
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gst_base_src_set_format (GST_BASE_SRC (dtmfsrc), GST_FORMAT_TIME);
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gst_base_src_set_live (GST_BASE_SRC (dtmfsrc), TRUE);
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dtmfsrc->interval = DEFAULT_PACKET_INTERVAL;
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dtmfsrc->event_queue = g_async_queue_new_full ((GDestroyNotify) event_free);
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dtmfsrc->last_event = NULL;
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dtmfsrc->sample_rate = DEFAULT_SAMPLE_RATE;
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GST_DEBUG_OBJECT (dtmfsrc, "init done");
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}
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static void
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gst_dtmf_src_finalize (GObject * object)
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{
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GstDTMFSrc *dtmfsrc;
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dtmfsrc = GST_DTMF_SRC (object);
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if (dtmfsrc->event_queue) {
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g_async_queue_unref (dtmfsrc->event_queue);
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dtmfsrc->event_queue = NULL;
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}
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G_OBJECT_CLASS (gst_dtmf_src_parent_class)->finalize (object);
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}
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static gboolean
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gst_dtmf_src_handle_dtmf_event (GstDTMFSrc * dtmfsrc, GstEvent * event)
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{
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const GstStructure *event_structure;
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GstStateChangeReturn sret;
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GstState state;
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gint event_type;
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gboolean start;
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gint method;
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GstClockTime last_stop;
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gint event_number;
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gint event_volume;
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gboolean correct_order;
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sret = gst_element_get_state (GST_ELEMENT (dtmfsrc), &state, NULL, 0);
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if (sret != GST_STATE_CHANGE_SUCCESS || state != GST_STATE_PLAYING) {
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GST_DEBUG_OBJECT (dtmfsrc, "dtmf-event, but not in PLAYING state");
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goto failure;
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}
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event_structure = gst_event_get_structure (event);
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if (!gst_structure_get_int (event_structure, "type", &event_type) ||
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!gst_structure_get_boolean (event_structure, "start", &start) ||
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(start == TRUE && event_type != GST_TONE_DTMF_TYPE_EVENT))
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goto failure;
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if (gst_structure_get_int (event_structure, "method", &method)) {
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if (method != 2) {
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goto failure;
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}
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}
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if (start)
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if (!gst_structure_get_int (event_structure, "number", &event_number) ||
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!gst_structure_get_int (event_structure, "volume", &event_volume))
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goto failure;
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GST_OBJECT_LOCK (dtmfsrc);
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if (gst_structure_get_clock_time (event_structure, "last-stop", &last_stop))
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dtmfsrc->last_stop = last_stop;
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else
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dtmfsrc->last_stop = GST_CLOCK_TIME_NONE;
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correct_order = (start != dtmfsrc->last_event_was_start);
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dtmfsrc->last_event_was_start = start;
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GST_OBJECT_UNLOCK (dtmfsrc);
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if (!correct_order)
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goto failure;
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if (start) {
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GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d",
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event_number, event_volume);
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gst_dtmf_src_add_start_event (dtmfsrc, event_number, event_volume);
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}
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else {
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GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
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gst_dtmf_src_add_stop_event (dtmfsrc);
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}
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return TRUE;
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failure:
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return FALSE;
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}
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static gboolean
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gst_dtmf_src_handle_event (GstBaseSrc * src, GstEvent * event)
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{
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GstDTMFSrc *dtmfsrc;
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gboolean result = FALSE;
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dtmfsrc = GST_DTMF_SRC (src);
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GST_LOG_OBJECT (dtmfsrc, "Received an %s event on the src pad",
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GST_EVENT_TYPE_NAME (event));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_CUSTOM_UPSTREAM:
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if (gst_event_has_name (event, "dtmf-event")) {
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result = gst_dtmf_src_handle_dtmf_event (dtmfsrc, event);
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break;
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}
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/* fall through */
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default:
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result = GST_BASE_SRC_CLASS (parent_class)->event (src, event);
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break;
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}
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return result;
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}
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static gboolean
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gst_dtmf_src_send_event (GstElement * element, GstEvent * event)
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{
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GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (element);
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gboolean ret;
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GST_LOG_OBJECT (dtmfsrc, "Received an %s event via send_event",
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GST_EVENT_TYPE_NAME (event));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_CUSTOM_BOTH:
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case GST_EVENT_CUSTOM_BOTH_OOB:
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case GST_EVENT_CUSTOM_UPSTREAM:
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case GST_EVENT_CUSTOM_DOWNSTREAM:
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case GST_EVENT_CUSTOM_DOWNSTREAM_OOB:
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if (gst_event_has_name (event, "dtmf-event")) {
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ret = gst_dtmf_src_handle_dtmf_event (dtmfsrc, event);
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break;
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}
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/* fall through */
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default:
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ret = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
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break;
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}
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return ret;
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}
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static void
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gst_dtmf_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstDTMFSrc *dtmfsrc;
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dtmfsrc = GST_DTMF_SRC (object);
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switch (prop_id) {
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case PROP_INTERVAL:
|
|
dtmfsrc->interval = g_value_get_uint (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_dtmf_src_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstDTMFSrc *dtmfsrc;
|
|
|
|
dtmfsrc = GST_DTMF_SRC (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_INTERVAL:
|
|
g_value_set_uint (value, dtmfsrc->interval);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_dtmf_prepare_timestamps (GstDTMFSrc * dtmfsrc)
|
|
{
|
|
GstClockTime last_stop;
|
|
GstClockTime timestamp;
|
|
|
|
GST_OBJECT_LOCK (dtmfsrc);
|
|
last_stop = dtmfsrc->last_stop;
|
|
GST_OBJECT_UNLOCK (dtmfsrc);
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (last_stop)) {
|
|
timestamp = last_stop;
|
|
} else {
|
|
GstClock *clock;
|
|
|
|
/* If there is no valid start time, lets use now as the start time */
|
|
|
|
clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
|
|
if (clock != NULL) {
|
|
timestamp = gst_clock_get_time (clock)
|
|
- gst_element_get_base_time (GST_ELEMENT (dtmfsrc));
|
|
gst_object_unref (clock);
|
|
} else {
|
|
gchar *dtmf_name = gst_element_get_name (dtmfsrc);
|
|
GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s", dtmf_name);
|
|
dtmfsrc->timestamp = GST_CLOCK_TIME_NONE;
|
|
g_free (dtmf_name);
|
|
return;
|
|
}
|
|
}
|
|
|
|
/* Make sure the timestamp always goes forward */
|
|
if (timestamp > dtmfsrc->timestamp)
|
|
dtmfsrc->timestamp = timestamp;
|
|
}
|
|
|
|
static void
|
|
gst_dtmf_src_add_start_event (GstDTMFSrc * dtmfsrc, gint event_number,
|
|
gint event_volume)
|
|
{
|
|
|
|
GstDTMFSrcEvent *event = g_slice_new0 (GstDTMFSrcEvent);
|
|
event->event_type = DTMF_EVENT_TYPE_START;
|
|
event->sample = 0;
|
|
event->event_number = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
|
|
event->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
|
|
|
|
g_async_queue_push (dtmfsrc->event_queue, event);
|
|
}
|
|
|
|
static void
|
|
gst_dtmf_src_add_stop_event (GstDTMFSrc * dtmfsrc)
|
|
{
|
|
|
|
GstDTMFSrcEvent *event = g_slice_new0 (GstDTMFSrcEvent);
|
|
event->event_type = DTMF_EVENT_TYPE_STOP;
|
|
event->sample = 0;
|
|
event->event_number = 0;
|
|
event->volume = 0;
|
|
|
|
g_async_queue_push (dtmfsrc->event_queue, event);
|
|
}
|
|
|
|
static GstBuffer *
|
|
gst_dtmf_src_generate_silence (float duration, gint sample_rate)
|
|
{
|
|
gint buf_size;
|
|
|
|
/* Create a buffer with data set to 0 */
|
|
buf_size = ((duration / 1000) * sample_rate * SAMPLE_SIZE * CHANNELS) / 8;
|
|
|
|
return gst_buffer_new_wrapped (g_malloc0 (buf_size), buf_size);
|
|
}
|
|
|
|
static GstBuffer *
|
|
gst_dtmf_src_generate_tone (GstDTMFSrcEvent * event, DTMF_KEY key,
|
|
float duration, gint sample_rate)
|
|
{
|
|
GstBuffer *buffer;
|
|
GstMapInfo map;
|
|
gint16 *p;
|
|
gint tone_size;
|
|
double i = 0;
|
|
double amplitude, f1, f2;
|
|
double volume_factor;
|
|
static GstAllocationParams params = { 0, 1, 0, 0, };
|
|
|
|
/* Create a buffer for the tone */
|
|
tone_size = ((duration / 1000) * sample_rate * SAMPLE_SIZE * CHANNELS) / 8;
|
|
|
|
buffer = gst_buffer_new_allocate (NULL, tone_size, ¶ms);
|
|
|
|
gst_buffer_map (buffer, &map, GST_MAP_READWRITE);
|
|
p = (gint16 *) map.data;
|
|
|
|
volume_factor = pow (10, (-event->volume) / 20);
|
|
|
|
/*
|
|
* For each sample point we calculate 'x' as the
|
|
* the amplitude value.
|
|
*/
|
|
for (i = 0; i < (tone_size / (SAMPLE_SIZE / 8)); i++) {
|
|
/*
|
|
* We add the fundamental frequencies together.
|
|
*/
|
|
f1 = sin (2 * M_PI * key.low_frequency * (event->sample / sample_rate));
|
|
f2 = sin (2 * M_PI * key.high_frequency * (event->sample / sample_rate));
|
|
|
|
amplitude = (f1 + f2) / 2;
|
|
|
|
/* Adjust the volume */
|
|
amplitude *= volume_factor;
|
|
|
|
/* Make the [-1:1] interval into a [-32767:32767] interval */
|
|
amplitude *= 32767;
|
|
|
|
/* Store it in the data buffer */
|
|
*(p++) = (gint16) amplitude;
|
|
|
|
(event->sample)++;
|
|
}
|
|
|
|
gst_buffer_unmap (buffer, &map);
|
|
|
|
return buffer;
|
|
}
|
|
|
|
|
|
|
|
static GstBuffer *
|
|
gst_dtmf_src_create_next_tone_packet (GstDTMFSrc * dtmfsrc,
|
|
GstDTMFSrcEvent * event)
|
|
{
|
|
GstBuffer *buf = NULL;
|
|
gboolean send_silence = FALSE;
|
|
|
|
GST_LOG_OBJECT (dtmfsrc, "Creating buffer for tone %s",
|
|
DTMF_KEYS[event->event_number].event_name);
|
|
|
|
if (event->packet_count * dtmfsrc->interval < MIN_INTER_DIGIT_INTERVAL) {
|
|
send_silence = TRUE;
|
|
}
|
|
|
|
if (send_silence) {
|
|
GST_LOG_OBJECT (dtmfsrc, "Generating silence");
|
|
buf = gst_dtmf_src_generate_silence (dtmfsrc->interval,
|
|
dtmfsrc->sample_rate);
|
|
} else {
|
|
GST_LOG_OBJECT (dtmfsrc, "Generating tone");
|
|
buf = gst_dtmf_src_generate_tone (event, DTMF_KEYS[event->event_number],
|
|
dtmfsrc->interval, dtmfsrc->sample_rate);
|
|
}
|
|
event->packet_count++;
|
|
|
|
|
|
/* timestamp and duration of GstBuffer */
|
|
GST_BUFFER_DURATION (buf) = dtmfsrc->interval * GST_MSECOND;
|
|
GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp;
|
|
|
|
GST_LOG_OBJECT (dtmfsrc, "Creating new buffer with event %u duration "
|
|
" gst: %" GST_TIME_FORMAT " at %" GST_TIME_FORMAT,
|
|
event->event_number, GST_TIME_ARGS (GST_BUFFER_DURATION (buf)),
|
|
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
|
|
|
|
dtmfsrc->timestamp += GST_BUFFER_DURATION (buf);
|
|
|
|
return buf;
|
|
}
|
|
|
|
static void
|
|
gst_dtmf_src_post_message (GstDTMFSrc * dtmfsrc, const gchar * message_name,
|
|
GstDTMFSrcEvent * event)
|
|
{
|
|
GstStructure *s = NULL;
|
|
|
|
switch (event->event_type) {
|
|
case DTMF_EVENT_TYPE_START:
|
|
s = gst_structure_new (message_name,
|
|
"type", G_TYPE_INT, 1,
|
|
"method", G_TYPE_INT, 2,
|
|
"start", G_TYPE_BOOLEAN, TRUE,
|
|
"number", G_TYPE_INT, event->event_number,
|
|
"volume", G_TYPE_INT, event->volume, NULL);
|
|
break;
|
|
case DTMF_EVENT_TYPE_STOP:
|
|
s = gst_structure_new (message_name,
|
|
"type", G_TYPE_INT, 1, "method", G_TYPE_INT, 2,
|
|
"start", G_TYPE_BOOLEAN, FALSE, NULL);
|
|
break;
|
|
case DTMF_EVENT_TYPE_PAUSE_TASK:
|
|
return;
|
|
}
|
|
|
|
if (s)
|
|
gst_element_post_message (GST_ELEMENT (dtmfsrc),
|
|
gst_message_new_element (GST_OBJECT (dtmfsrc), s));
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
|
|
guint length, GstBuffer ** buffer)
|
|
{
|
|
GstBuffer *buf = NULL;
|
|
GstDTMFSrcEvent *event;
|
|
GstDTMFSrc *dtmfsrc;
|
|
GstClock *clock;
|
|
GstClockID *clockid;
|
|
GstClockReturn clockret;
|
|
|
|
dtmfsrc = GST_DTMF_SRC (basesrc);
|
|
|
|
do {
|
|
|
|
if (dtmfsrc->last_event == NULL) {
|
|
GST_DEBUG_OBJECT (dtmfsrc, "popping");
|
|
event = g_async_queue_pop (dtmfsrc->event_queue);
|
|
|
|
GST_DEBUG_OBJECT (dtmfsrc, "popped %d", event->event_type);
|
|
|
|
switch (event->event_type) {
|
|
case DTMF_EVENT_TYPE_STOP:
|
|
GST_WARNING_OBJECT (dtmfsrc,
|
|
"Received a DTMF stop event when already stopped");
|
|
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
|
|
break;
|
|
case DTMF_EVENT_TYPE_START:
|
|
gst_dtmf_prepare_timestamps (dtmfsrc);
|
|
|
|
event->packet_count = 0;
|
|
dtmfsrc->last_event = event;
|
|
event = NULL;
|
|
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-processed",
|
|
dtmfsrc->last_event);
|
|
break;
|
|
case DTMF_EVENT_TYPE_PAUSE_TASK:
|
|
/*
|
|
* We're pushing it back because it has to stay in there until
|
|
* the task is really paused (and the queue will then be flushed)
|
|
*/
|
|
GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
|
|
GST_OBJECT_LOCK (dtmfsrc);
|
|
if (dtmfsrc->paused) {
|
|
g_async_queue_push (dtmfsrc->event_queue, event);
|
|
goto paused_locked;
|
|
}
|
|
GST_OBJECT_UNLOCK (dtmfsrc);
|
|
break;
|
|
}
|
|
if (event)
|
|
g_slice_free (GstDTMFSrcEvent, event);
|
|
} else if (dtmfsrc->last_event->packet_count * dtmfsrc->interval >=
|
|
MIN_DUTY_CYCLE) {
|
|
event = g_async_queue_try_pop (dtmfsrc->event_queue);
|
|
|
|
if (event != NULL) {
|
|
|
|
switch (event->event_type) {
|
|
case DTMF_EVENT_TYPE_START:
|
|
GST_WARNING_OBJECT (dtmfsrc,
|
|
"Received two consecutive DTMF start events");
|
|
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
|
|
break;
|
|
case DTMF_EVENT_TYPE_STOP:
|
|
g_slice_free (GstDTMFSrcEvent, dtmfsrc->last_event);
|
|
dtmfsrc->last_event = NULL;
|
|
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-processed", event);
|
|
break;
|
|
case DTMF_EVENT_TYPE_PAUSE_TASK:
|
|
/*
|
|
* We're pushing it back because it has to stay in there until
|
|
* the task is really paused (and the queue will then be flushed)
|
|
*/
|
|
GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
|
|
|
|
GST_OBJECT_LOCK (dtmfsrc);
|
|
if (dtmfsrc->paused) {
|
|
g_async_queue_push (dtmfsrc->event_queue, event);
|
|
goto paused_locked;
|
|
}
|
|
GST_OBJECT_UNLOCK (dtmfsrc);
|
|
|
|
break;
|
|
}
|
|
g_slice_free (GstDTMFSrcEvent, event);
|
|
}
|
|
}
|
|
} while (dtmfsrc->last_event == NULL);
|
|
|
|
GST_LOG_OBJECT (dtmfsrc, "end event check, now wait for the proper time");
|
|
|
|
clock = gst_element_get_clock (GST_ELEMENT (basesrc));
|
|
|
|
clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
|
|
gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
|
|
gst_object_unref (clock);
|
|
|
|
GST_OBJECT_LOCK (dtmfsrc);
|
|
if (!dtmfsrc->paused) {
|
|
dtmfsrc->clockid = clockid;
|
|
GST_OBJECT_UNLOCK (dtmfsrc);
|
|
|
|
clockret = gst_clock_id_wait (clockid, NULL);
|
|
|
|
GST_OBJECT_LOCK (dtmfsrc);
|
|
if (dtmfsrc->paused)
|
|
clockret = GST_CLOCK_UNSCHEDULED;
|
|
} else {
|
|
clockret = GST_CLOCK_UNSCHEDULED;
|
|
}
|
|
gst_clock_id_unref (clockid);
|
|
dtmfsrc->clockid = NULL;
|
|
GST_OBJECT_UNLOCK (dtmfsrc);
|
|
|
|
if (clockret == GST_CLOCK_UNSCHEDULED) {
|
|
goto paused;
|
|
}
|
|
|
|
buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc, dtmfsrc->last_event);
|
|
|
|
GST_LOG_OBJECT (dtmfsrc, "Created buffer of size %" G_GSIZE_FORMAT,
|
|
gst_buffer_get_size (buf));
|
|
*buffer = buf;
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
paused_locked:
|
|
GST_OBJECT_UNLOCK (dtmfsrc);
|
|
|
|
paused:
|
|
|
|
if (dtmfsrc->last_event) {
|
|
GST_DEBUG_OBJECT (dtmfsrc, "Stopping current event");
|
|
/* Don't forget to release the stream lock */
|
|
g_slice_free (GstDTMFSrcEvent, dtmfsrc->last_event);
|
|
dtmfsrc->last_event = NULL;
|
|
}
|
|
|
|
return GST_FLOW_FLUSHING;
|
|
|
|
}
|
|
|
|
static gboolean
|
|
gst_dtmf_src_unlock (GstBaseSrc * src)
|
|
{
|
|
GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (src);
|
|
GstDTMFSrcEvent *event = NULL;
|
|
|
|
GST_DEBUG_OBJECT (dtmfsrc, "Called unlock");
|
|
|
|
GST_OBJECT_LOCK (dtmfsrc);
|
|
dtmfsrc->paused = TRUE;
|
|
if (dtmfsrc->clockid) {
|
|
gst_clock_id_unschedule (dtmfsrc->clockid);
|
|
}
|
|
GST_OBJECT_UNLOCK (dtmfsrc);
|
|
|
|
GST_DEBUG_OBJECT (dtmfsrc, "Pushing the PAUSE_TASK event on unlock request");
|
|
event = g_slice_new0 (GstDTMFSrcEvent);
|
|
event->event_type = DTMF_EVENT_TYPE_PAUSE_TASK;
|
|
g_async_queue_push (dtmfsrc->event_queue, event);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
|
|
static gboolean
|
|
gst_dtmf_src_unlock_stop (GstBaseSrc * src)
|
|
{
|
|
GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (src);
|
|
|
|
GST_DEBUG_OBJECT (dtmfsrc, "Unlock stopped");
|
|
|
|
GST_OBJECT_LOCK (dtmfsrc);
|
|
dtmfsrc->paused = FALSE;
|
|
GST_OBJECT_UNLOCK (dtmfsrc);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
|
|
static gboolean
|
|
gst_dtmf_src_negotiate (GstBaseSrc * basesrc)
|
|
{
|
|
GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (basesrc);
|
|
GstCaps *caps;
|
|
GstStructure *s;
|
|
gboolean ret;
|
|
|
|
caps = gst_pad_get_allowed_caps (GST_BASE_SRC_PAD (basesrc));
|
|
|
|
if (!caps)
|
|
caps = gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD (basesrc));
|
|
|
|
if (gst_caps_is_empty (caps)) {
|
|
gst_caps_unref (caps);
|
|
return FALSE;
|
|
}
|
|
|
|
caps = gst_caps_truncate (caps);
|
|
|
|
caps = gst_caps_make_writable (caps);
|
|
s = gst_caps_get_structure (caps, 0);
|
|
|
|
gst_structure_fixate_field_nearest_int (s, "rate", DEFAULT_SAMPLE_RATE);
|
|
|
|
if (!gst_structure_get_int (s, "rate", &dtmfsrc->sample_rate)) {
|
|
GST_ERROR_OBJECT (dtmfsrc, "Could not get rate");
|
|
gst_caps_unref (caps);
|
|
return FALSE;
|
|
}
|
|
|
|
ret = gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), caps);
|
|
|
|
gst_caps_unref (caps);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_dtmf_src_query (GstBaseSrc * basesrc, GstQuery * query)
|
|
{
|
|
GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (basesrc);
|
|
gboolean res = FALSE;
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_LATENCY:
|
|
{
|
|
GstClockTime latency;
|
|
|
|
latency = dtmfsrc->interval * GST_MSECOND;
|
|
gst_query_set_latency (query, gst_base_src_is_live (basesrc), latency,
|
|
GST_CLOCK_TIME_NONE);
|
|
GST_DEBUG_OBJECT (dtmfsrc, "Reporting latency of %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (latency));
|
|
res = TRUE;
|
|
}
|
|
break;
|
|
default:
|
|
res = GST_BASE_SRC_CLASS (parent_class)->query (basesrc, query);
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_dtmf_src_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstDTMFSrc *dtmfsrc;
|
|
GstStateChangeReturn result;
|
|
gboolean no_preroll = FALSE;
|
|
GstDTMFSrcEvent *event = NULL;
|
|
|
|
dtmfsrc = GST_DTMF_SRC (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
/* Flushing the event queue */
|
|
event = g_async_queue_try_pop (dtmfsrc->event_queue);
|
|
|
|
while (event != NULL) {
|
|
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
|
|
g_slice_free (GstDTMFSrcEvent, event);
|
|
event = g_async_queue_try_pop (dtmfsrc->event_queue);
|
|
}
|
|
dtmfsrc->last_event_was_start = FALSE;
|
|
dtmfsrc->timestamp = 0;
|
|
no_preroll = TRUE;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if ((result =
|
|
GST_ELEMENT_CLASS (gst_dtmf_src_parent_class)->change_state (element,
|
|
transition)) == GST_STATE_CHANGE_FAILURE)
|
|
goto failure;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
no_preroll = TRUE;
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
GST_DEBUG_OBJECT (dtmfsrc, "Flushing event queue");
|
|
/* Flushing the event queue */
|
|
event = g_async_queue_try_pop (dtmfsrc->event_queue);
|
|
|
|
while (event != NULL) {
|
|
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
|
|
g_slice_free (GstDTMFSrcEvent, event);
|
|
event = g_async_queue_try_pop (dtmfsrc->event_queue);
|
|
}
|
|
dtmfsrc->last_event_was_start = FALSE;
|
|
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if (no_preroll && result == GST_STATE_CHANGE_SUCCESS)
|
|
result = GST_STATE_CHANGE_NO_PREROLL;
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
failure:
|
|
{
|
|
GST_ERROR_OBJECT (dtmfsrc, "parent failed state change");
|
|
return result;
|
|
}
|
|
}
|
|
|
|
gboolean
|
|
gst_dtmf_src_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "dtmfsrc",
|
|
GST_RANK_NONE, GST_TYPE_DTMF_SRC);
|
|
}
|