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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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01f5b39eb2
20070822175444-4f0f6-445ea6ce7a9668d04cf999af772a504ec74fb67a.gz
843 lines
23 KiB
C
843 lines
23 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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-dtmfsrc
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* @short_description: Generates DTMF packets
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*
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* <refsect2>
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*
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* <para>
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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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* </para>
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*
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* <para>
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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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*
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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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*
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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. This element is only capable of generating tones.
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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>0-16</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>1</entry>
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* <entry>The method used for sending event, this element will react if this field
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* 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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* </para>
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*
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* <para>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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* </para>
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*
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* <para>
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* <programlisting>
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* structure = gst_structure_new ("dtmf-event",
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* "type", G_TYPE_INT, 0,
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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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* </para>
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*
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* </refsect2>
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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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#ifndef M_PI
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# define M_PI 3.14159265358979323846 /* pi */
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#endif
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#include "gstdtmfsrc.h"
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#define GST_TONE_DTMF_TYPE_EVENT 0
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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 SAMPLE_RATE 8000
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#define SAMPLE_SIZE 16
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#define CHANNELS 1
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#define MIN_EVENT 0
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#define MAX_EVENT 16
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#define MIN_VOLUME 0
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#define MAX_VOLUME 36
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#define MIN_INTER_DIGIT_INTERVAL 50
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#define MIN_PULSE_DURATION 70
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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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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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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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/* elementfactory information */
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static const GstElementDetails gst_dtmf_src_details =
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GST_ELEMENT_DETAILS ("DTMF tone generator",
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"Source/Audio",
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"Generates DTMF tones",
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"Youness Alaoui <youness.alaoui@collabora.co.uk>");
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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-int, "
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"width = (int) 16, "
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"depth = (int) 16, "
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"endianness = (int) 1234, "
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"signed = (bool) true, "
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"rate = (int) 8000, "
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"channels = (int) 1")
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);
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static GstElementClass *parent_class = NULL;
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static void gst_dtmf_src_base_init (gpointer g_class);
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static void gst_dtmf_src_class_init (GstDTMFSrcClass * klass);
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static void gst_dtmf_src_init (GstDTMFSrc * dtmfsrc, gpointer g_class);
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static void gst_dtmf_src_finalize (GObject * object);
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GType
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gst_dtmf_src_get_type (void)
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{
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static GType base_src_type = 0;
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if (G_UNLIKELY (base_src_type == 0)) {
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static const GTypeInfo base_src_info = {
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sizeof (GstDTMFSrcClass),
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(GBaseInitFunc) gst_dtmf_src_base_init,
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NULL,
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(GClassInitFunc) gst_dtmf_src_class_init,
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NULL,
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NULL,
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sizeof (GstDTMFSrc),
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0,
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(GInstanceInitFunc) gst_dtmf_src_init,
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};
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base_src_type = g_type_register_static (GST_TYPE_ELEMENT,
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"GstDTMFSrc", &base_src_info, 0);
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}
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return base_src_type;
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}
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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 (GstPad * pad, 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 void gst_dtmf_src_generate_tone(GstDTMFSrcEvent *event, DTMF_KEY key, float duration,
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GstBuffer * buffer);
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static void gst_dtmf_src_push_next_tone_packet (GstDTMFSrc *dtmfsrc);
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static void gst_dtmf_src_start (GstDTMFSrc *dtmfsrc);
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static void gst_dtmf_src_stop (GstDTMFSrc *dtmfsrc);
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static void gst_dtmf_src_add_start_event (GstDTMFSrc *dtmfsrc, gint event_number,
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gint event_volume);
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static void gst_dtmf_src_add_stop_event (GstDTMFSrc *dtmfsrc);
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static void
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gst_dtmf_src_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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GST_DEBUG_CATEGORY_INIT (gst_dtmf_src_debug,
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"dtmfsrc", 0, "dtmfsrc element");
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_dtmf_src_template));
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gst_element_class_set_details (element_class, &gst_dtmf_src_details);
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}
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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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GstElementClass *gstelement_class;
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gobject_class = G_OBJECT_CLASS (klass);
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gstelement_class = GST_ELEMENT_CLASS (klass);
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_dtmf_src_finalize);
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gobject_class->set_property =
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GST_DEBUG_FUNCPTR (gst_dtmf_src_set_property);
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gobject_class->get_property =
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GST_DEBUG_FUNCPTR (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_int ("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, G_PARAM_READWRITE));
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_dtmf_src_change_state);
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}
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static void
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gst_dtmf_src_init (GstDTMFSrc * dtmfsrc, gpointer g_class)
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{
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dtmfsrc->srcpad =
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gst_pad_new_from_static_template (&gst_dtmf_src_template, "src");
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GST_DEBUG_OBJECT (dtmfsrc, "adding src pad");
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gst_element_add_pad (GST_ELEMENT (dtmfsrc), dtmfsrc->srcpad);
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gst_pad_set_event_function (dtmfsrc->srcpad, gst_dtmf_src_handle_event);
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dtmfsrc->interval = DEFAULT_PACKET_INTERVAL;
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dtmfsrc->event_queue = g_async_queue_new ();
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dtmfsrc->last_event = NULL;
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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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gst_dtmf_src_stop (dtmfsrc);
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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 (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,
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const GstStructure * event_structure)
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{
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gint event_type;
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gboolean start;
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gint method;
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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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gint event_number;
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gint event_volume;
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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_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_custom_upstream (GstDTMFSrc *dtmfsrc,
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GstEvent * event)
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{
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gboolean result = FALSE;
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const GstStructure *structure;
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if (GST_STATE (dtmfsrc) != GST_STATE_PLAYING) {
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GST_DEBUG_OBJECT (dtmfsrc, "Received event while not in PLAYING state");
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goto ret;
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}
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GST_DEBUG_OBJECT (dtmfsrc, "Received event is of our interest");
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structure = gst_event_get_structure (event);
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if (structure && gst_structure_has_name (structure, "dtmf-event"))
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result = gst_dtmf_src_handle_dtmf_event (dtmfsrc, structure);
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ret:
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return result;
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}
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static gboolean
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gst_dtmf_src_handle_event (GstPad * pad, 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 (GST_PAD_PARENT (pad));
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GST_DEBUG_OBJECT (dtmfsrc, "Received an event on the src pad");
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_CUSTOM_UPSTREAM:
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{
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result = gst_dtmf_src_handle_custom_upstream (dtmfsrc, event);
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break;
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}
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/* Ideally this element should not be flushed but let's handle the event
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* just in case it is */
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case GST_EVENT_FLUSH_START:
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gst_dtmf_src_stop (dtmfsrc);
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result = TRUE;
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break;
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case GST_EVENT_FLUSH_STOP:
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gst_segment_init (&dtmfsrc->segment, GST_FORMAT_UNDEFINED);
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break;
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case GST_EVENT_NEWSEGMENT:
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{
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gboolean update;
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gdouble rate;
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GstFormat fmt;
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gint64 start, stop, position;
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gst_event_parse_new_segment (event, &update, &rate, &fmt, &start,
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&stop, &position);
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gst_segment_set_newsegment (&dtmfsrc->segment, update, rate, fmt,
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start, stop, position);
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}
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/* fallthrough */
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default:
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result = gst_pad_event_default (pad, event);
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break;
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}
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gst_event_unref (event);
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return result;
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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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{
|
|
GstDTMFSrc *dtmfsrc;
|
|
|
|
dtmfsrc = GST_DTMF_SRC (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_INTERVAL:
|
|
dtmfsrc->interval = g_value_get_int (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_src_set_stream_lock (GstDTMFSrc *dtmfsrc, gboolean lock)
|
|
{
|
|
GstEvent *event;
|
|
GstStructure *structure;
|
|
|
|
structure = gst_structure_new ("stream-lock",
|
|
"lock", G_TYPE_BOOLEAN, lock, NULL);
|
|
|
|
event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_OOB, structure);
|
|
gst_pad_push_event (dtmfsrc->srcpad, event);
|
|
}
|
|
|
|
static void
|
|
gst_dtmf_prepare_timestamps (GstDTMFSrc *dtmfsrc)
|
|
{
|
|
GstClock *clock;
|
|
|
|
clock = GST_ELEMENT_CLOCK (dtmfsrc);
|
|
if (clock != NULL)
|
|
dtmfsrc->timestamp = gst_clock_get_time (GST_ELEMENT_CLOCK (dtmfsrc));
|
|
|
|
else {
|
|
GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s",
|
|
GST_ELEMENT_NAME (dtmfsrc));
|
|
dtmfsrc->timestamp = GST_CLOCK_TIME_NONE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_dtmf_src_start (GstDTMFSrc *dtmfsrc)
|
|
{
|
|
GstCaps * caps = gst_pad_get_pad_template_caps (dtmfsrc->srcpad);
|
|
|
|
if (!gst_pad_set_caps (dtmfsrc->srcpad, caps))
|
|
GST_ERROR_OBJECT (dtmfsrc,
|
|
"Failed to set caps %" GST_PTR_FORMAT " on src pad", caps);
|
|
else
|
|
GST_DEBUG_OBJECT (dtmfsrc,
|
|
"caps %" GST_PTR_FORMAT " set on src pad", caps);
|
|
|
|
|
|
if (!gst_pad_start_task (dtmfsrc->srcpad,
|
|
(GstTaskFunction) gst_dtmf_src_push_next_tone_packet, dtmfsrc)) {
|
|
GST_ERROR_OBJECT (dtmfsrc, "Failed to start task on src pad");
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_dtmf_src_stop (GstDTMFSrc *dtmfsrc)
|
|
{
|
|
/* Don't forget to release the stream lock */
|
|
gst_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
|
|
|
|
|
|
/* Flushing the event queue */
|
|
GstDTMFSrcEvent *event = g_async_queue_try_pop (dtmfsrc->event_queue);
|
|
|
|
while (event != NULL) {
|
|
g_free (event);
|
|
event = g_async_queue_try_pop (dtmfsrc->event_queue);
|
|
}
|
|
|
|
if (dtmfsrc->last_event) {
|
|
g_free (dtmfsrc->last_event);
|
|
dtmfsrc->last_event = NULL;
|
|
}
|
|
|
|
if (!gst_pad_pause_task (dtmfsrc->srcpad)) {
|
|
GST_ERROR_OBJECT (dtmfsrc, "Failed to pause task on src pad");
|
|
return;
|
|
}
|
|
|
|
}
|
|
|
|
static void
|
|
gst_dtmf_src_add_start_event (GstDTMFSrc *dtmfsrc, gint event_number,
|
|
gint event_volume)
|
|
{
|
|
|
|
GstDTMFSrcEvent * event = g_malloc (sizeof(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_malloc (sizeof(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 void
|
|
gst_dtmf_src_generate_silence(GstBuffer * buffer, float duration)
|
|
{
|
|
gint buf_size;
|
|
|
|
/* Create a buffer with data set to 0 */
|
|
buf_size = ((duration/1000)*SAMPLE_RATE*SAMPLE_SIZE*CHANNELS)/8;
|
|
GST_BUFFER_SIZE (buffer) = buf_size;
|
|
GST_BUFFER_MALLOCDATA (buffer) = g_malloc0(buf_size);
|
|
GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
|
|
|
|
}
|
|
|
|
static void
|
|
gst_dtmf_src_generate_tone(GstDTMFSrcEvent *event, DTMF_KEY key, float duration, GstBuffer * buffer)
|
|
{
|
|
gint16 *p;
|
|
gint tone_size;
|
|
double i = 0;
|
|
double amplitude, f1, f2;
|
|
|
|
/* Create a buffer for the tone */
|
|
tone_size = ((duration/1000)*SAMPLE_RATE*SAMPLE_SIZE*CHANNELS)/8;
|
|
GST_BUFFER_SIZE (buffer) = tone_size;
|
|
GST_BUFFER_MALLOCDATA (buffer) = g_malloc(tone_size);
|
|
GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
|
|
|
|
p = (gint16 *) GST_BUFFER_MALLOCDATA (buffer);
|
|
|
|
/*
|
|
* 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;
|
|
|
|
/* Make the [-1:1] interval into a [-32767:32767] interval */
|
|
amplitude *= 32767;
|
|
|
|
/* Store it in the data buffer */
|
|
*(p++) = (gint16) amplitude;
|
|
|
|
(event->sample)++;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_dtmf_src_wait_for_buffer_ts (GstDTMFSrc *dtmfsrc, GstBuffer * buf)
|
|
{
|
|
GstClock *clock;
|
|
|
|
clock = GST_ELEMENT_CLOCK (dtmfsrc);
|
|
if (clock != NULL) {
|
|
GstClockID clock_id;
|
|
GstClockReturn clock_ret;
|
|
|
|
clock_id = gst_clock_new_single_shot_id (clock, GST_BUFFER_TIMESTAMP (buf));
|
|
clock_ret = gst_clock_id_wait (clock_id, NULL);
|
|
if (clock_ret != GST_CLOCK_OK && clock_ret != GST_CLOCK_EARLY) {
|
|
GST_ERROR_OBJECT (dtmfsrc, "Failed to wait on clock %s",
|
|
GST_ELEMENT_NAME (clock));
|
|
}
|
|
gst_clock_id_unref (clock_id);
|
|
}
|
|
|
|
else {
|
|
GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s",
|
|
GST_ELEMENT_NAME (dtmfsrc));
|
|
}
|
|
}
|
|
|
|
|
|
static GstBuffer *
|
|
gst_dtmf_src_create_next_tone_packet (GstDTMFSrc *dtmfsrc, GstDTMFSrcEvent *event)
|
|
{
|
|
GstBuffer *buf = NULL;
|
|
guint32 duration;
|
|
|
|
|
|
GST_DEBUG_OBJECT (dtmfsrc,
|
|
"Creating buffer for tone");
|
|
|
|
/* create buffer to hold the tone */
|
|
buf = gst_buffer_new ();
|
|
|
|
/* The first packet must be inter digit silence, then the second and third must be the
|
|
* minimal pulse duration divided into two packets to make it small
|
|
*/
|
|
switch(event->packet_count) {
|
|
case 0:
|
|
duration = MIN_INTER_DIGIT_INTERVAL;
|
|
gst_dtmf_src_generate_silence (buf, duration);
|
|
break;
|
|
case 1:
|
|
case 2:
|
|
/* Generate the tone */
|
|
duration = MIN_PULSE_DURATION / 2;
|
|
gst_dtmf_src_generate_tone(event, DTMF_KEYS[event->event_number], duration, buf);
|
|
break;
|
|
default:
|
|
duration = dtmfsrc->interval;
|
|
gst_dtmf_src_generate_tone(event, DTMF_KEYS[event->event_number], duration, buf);
|
|
break;
|
|
}
|
|
event->packet_count++;
|
|
|
|
|
|
/* timestamp and duration of GstBuffer */
|
|
GST_BUFFER_DURATION (buf) = duration * GST_MSECOND;
|
|
GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp;
|
|
dtmfsrc->timestamp += GST_BUFFER_DURATION (buf);
|
|
|
|
/* FIXME: Should we sync to clock ourselves or leave it to sink */
|
|
gst_dtmf_src_wait_for_buffer_ts (dtmfsrc, buf);
|
|
|
|
/* Set caps on the buffer before pushing it */
|
|
gst_buffer_set_caps (buf, GST_PAD_CAPS (dtmfsrc->srcpad));
|
|
|
|
return buf;
|
|
}
|
|
|
|
static void
|
|
gst_dtmf_src_push_next_tone_packet (GstDTMFSrc *dtmfsrc)
|
|
{
|
|
GstBuffer *buf = NULL;
|
|
GstFlowReturn ret;
|
|
GstDTMFSrcEvent *event;
|
|
|
|
g_async_queue_ref (dtmfsrc->event_queue);
|
|
|
|
if (dtmfsrc->last_event == NULL) {
|
|
event = g_async_queue_pop (dtmfsrc->event_queue);
|
|
|
|
if (event->event_type == DTMF_EVENT_TYPE_STOP) {
|
|
GST_WARNING_OBJECT (dtmfsrc, "Received a DTMF stop event when already stopped");
|
|
} else if (event->event_type == DTMF_EVENT_TYPE_START) {
|
|
gst_dtmf_prepare_timestamps (dtmfsrc);
|
|
|
|
/* Don't forget to get exclusive access to the stream */
|
|
gst_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
|
|
|
|
event->packet_count = 0;
|
|
dtmfsrc->last_event = event;
|
|
}
|
|
} else if (dtmfsrc->last_event->packet_count >= 3) {
|
|
event = g_async_queue_try_pop (dtmfsrc->event_queue);
|
|
|
|
if (event != NULL) {
|
|
if (event->event_type == DTMF_EVENT_TYPE_START) {
|
|
GST_WARNING_OBJECT (dtmfsrc, "Received two consecutive DTMF start events");
|
|
} else if (event->event_type == DTMF_EVENT_TYPE_STOP) {
|
|
gst_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
|
|
g_free (dtmfsrc->last_event);
|
|
dtmfsrc->last_event = NULL;
|
|
}
|
|
}
|
|
}
|
|
g_async_queue_unref (dtmfsrc->event_queue);
|
|
|
|
if (dtmfsrc->last_event) {
|
|
buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc, dtmfsrc->last_event);
|
|
|
|
gst_buffer_ref(buf);
|
|
|
|
GST_DEBUG_OBJECT (dtmfsrc,
|
|
"pushing buffer on src pad of size %d", GST_BUFFER_SIZE (buf));
|
|
ret = gst_pad_push (dtmfsrc->srcpad, buf);
|
|
if (ret != GST_FLOW_OK) {
|
|
GST_ERROR_OBJECT (dtmfsrc, "Failed to push buffer on src pad");
|
|
}
|
|
|
|
gst_buffer_unref(buf);
|
|
GST_DEBUG_OBJECT (dtmfsrc, "pushed DTMF tone on src pad");
|
|
}
|
|
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_dtmf_src_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstDTMFSrc *dtmfsrc;
|
|
GstStateChangeReturn result;
|
|
gboolean no_preroll = FALSE;
|
|
|
|
dtmfsrc = GST_DTMF_SRC (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
gst_segment_init (&dtmfsrc->segment, GST_FORMAT_UNDEFINED);
|
|
/* Indicate that we don't do PRE_ROLL */
|
|
no_preroll = TRUE;
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
gst_dtmf_src_start (dtmfsrc);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if ((result =
|
|
GST_ELEMENT_CLASS (parent_class)->change_state (element,
|
|
transition)) == GST_STATE_CHANGE_FAILURE)
|
|
goto failure;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
/* Indicate that we don't do PRE_ROLL */
|
|
gst_dtmf_src_stop (dtmfsrc);
|
|
no_preroll = TRUE;
|
|
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);
|
|
}
|
|
|