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935 lines
26 KiB
C
935 lines
26 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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* @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>0-16</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>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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*/
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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 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_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 100
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#define MIN_PULSE_DURATION 250
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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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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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/* 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) " G_STRINGIFY (G_BYTE_ORDER) ", "
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"signed = (bool) true, " "rate = (int) 8000, " "channels = (int) 1")
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);
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GST_BOILERPLATE (GstDTMFSrc, gst_dtmf_src, GstBaseSrc, 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 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 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, "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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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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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_dtmf_src_finalize);
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gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_dtmf_src_set_property);
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gobject_class->get_property = 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_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, 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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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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}
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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, GstDTMFSrcClass * g_class)
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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 (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, GstEvent * event)
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{
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gboolean result = FALSE;
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const GstStructure *structure;
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GstState state;
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GstStateChangeReturn ret;
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ret = gst_element_get_state (GST_ELEMENT (dtmfsrc), &state, NULL, 0);
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if (ret != GST_STATE_CHANGE_SUCCESS || state != 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 (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_DEBUG_OBJECT (dtmfsrc, "Received an event on the src pad");
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if (GST_EVENT_TYPE (event) == GST_EVENT_CUSTOM_UPSTREAM) {
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result = gst_dtmf_src_handle_custom_upstream (dtmfsrc, event);
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}
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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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{
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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:
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dtmfsrc->interval = g_value_get_uint (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_dtmf_src_get_property (GObject * object, guint prop_id, GValue * value,
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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:
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g_value_set_uint (value, dtmfsrc->interval);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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|
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static void
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gst_dtmf_src_set_stream_lock (GstDTMFSrc * dtmfsrc, gboolean lock)
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{
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GstPad *srcpad = GST_BASE_SRC_PAD (dtmfsrc);
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GstEvent *event;
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GstStructure *structure;
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structure = gst_structure_new ("stream-lock",
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"lock", G_TYPE_BOOLEAN, lock, NULL);
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event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_OOB, structure);
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if (!gst_pad_push_event (srcpad, event)) {
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GST_WARNING_OBJECT (dtmfsrc, "stream-lock event not handled");
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}
|
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}
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|
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static void
|
|
gst_dtmf_prepare_timestamps (GstDTMFSrc * dtmfsrc)
|
|
{
|
|
GstClock *clock;
|
|
GstClockTime base_time;
|
|
|
|
base_time = gst_element_get_base_time (GST_ELEMENT (dtmfsrc));
|
|
|
|
clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
|
|
if (clock != NULL) {
|
|
#ifdef MAEMO_BROKEN
|
|
dtmfsrc->timestamp = gst_clock_get_time (clock);
|
|
#else
|
|
dtmfsrc->timestamp = gst_clock_get_time (clock) - base_time;
|
|
#endif
|
|
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);
|
|
}
|
|
}
|
|
|
|
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 void
|
|
gst_dtmf_src_generate_silence (GstBuffer * buffer, 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;
|
|
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, gint sample_rate)
|
|
{
|
|
gint16 *p;
|
|
gint tone_size;
|
|
double i = 0;
|
|
double amplitude, f1, f2;
|
|
double volume_factor;
|
|
|
|
/* 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);
|
|
|
|
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)++;
|
|
}
|
|
}
|
|
|
|
|
|
|
|
static GstBuffer *
|
|
gst_dtmf_src_create_next_tone_packet (GstDTMFSrc * dtmfsrc,
|
|
GstDTMFSrcEvent * event)
|
|
{
|
|
GstBuffer *buf = NULL;
|
|
gboolean send_silence = FALSE;
|
|
GstPad *srcpad = GST_BASE_SRC_PAD (dtmfsrc);
|
|
|
|
GST_DEBUG_OBJECT (dtmfsrc, "Creating buffer for tone %s",
|
|
DTMF_KEYS[event->event_number].event_name);
|
|
|
|
/* create buffer to hold the tone */
|
|
buf = gst_buffer_new ();
|
|
|
|
if (event->packet_count * dtmfsrc->interval < MIN_INTER_DIGIT_INTERVAL) {
|
|
send_silence = TRUE;
|
|
}
|
|
|
|
if (send_silence) {
|
|
GST_DEBUG_OBJECT (dtmfsrc, "Generating silence");
|
|
gst_dtmf_src_generate_silence (buf, dtmfsrc->interval,
|
|
dtmfsrc->sample_rate);
|
|
} else {
|
|
GST_DEBUG_OBJECT (dtmfsrc, "Generating tone");
|
|
gst_dtmf_src_generate_tone (event, DTMF_KEYS[event->event_number],
|
|
dtmfsrc->interval, buf, 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;
|
|
dtmfsrc->timestamp += GST_BUFFER_DURATION (buf);
|
|
|
|
/* Set caps on the buffer before pushing it */
|
|
gst_buffer_set_caps (buf, GST_PAD_CAPS (srcpad));
|
|
|
|
return buf;
|
|
}
|
|
|
|
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");
|
|
break;
|
|
case 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;
|
|
event = NULL;
|
|
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");
|
|
break;
|
|
case DTMF_EVENT_TYPE_STOP:
|
|
gst_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
|
|
|
|
g_slice_free (GstDTMFSrcEvent, dtmfsrc->last_event);
|
|
dtmfsrc->last_event = NULL;
|
|
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_DEBUG_OBJECT (dtmfsrc, "end event check, now wait for the proper time");
|
|
|
|
clock = gst_element_get_clock (GST_ELEMENT (basesrc));
|
|
|
|
#ifdef MAEMO_BROKEN
|
|
clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp);
|
|
#else
|
|
clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
|
|
gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
|
|
#endif
|
|
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_DEBUG_OBJECT (dtmfsrc, "Created buffer of size %d",
|
|
GST_BUFFER_SIZE (buf));
|
|
*buffer = buf;
|
|
|
|
GST_DEBUG_OBJECT (dtmfsrc, "returning a buffer");
|
|
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 */
|
|
gst_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
|
|
g_slice_free (GstDTMFSrcEvent, dtmfsrc->last_event);
|
|
dtmfsrc->last_event = NULL;
|
|
}
|
|
|
|
return GST_FLOW_WRONG_STATE;
|
|
|
|
}
|
|
|
|
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)
|
|
{
|
|
GstCaps *srccaps, *peercaps;
|
|
GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (basesrc);
|
|
gboolean ret = FALSE;
|
|
|
|
srccaps = gst_caps_new_simple ("audio/x-raw-int",
|
|
"width", G_TYPE_INT, 16,
|
|
"depth", G_TYPE_INT, 16,
|
|
"endianness", G_TYPE_INT, G_BYTE_ORDER,
|
|
"signed", G_TYPE_BOOLEAN, TRUE, "channels", G_TYPE_INT, 1, NULL);
|
|
|
|
peercaps = gst_pad_peer_get_caps (GST_BASE_SRC_PAD (basesrc));
|
|
|
|
if (peercaps == NULL) {
|
|
/* no peer caps, just add the other properties */
|
|
gst_caps_set_simple (srccaps,
|
|
"rate", G_TYPE_INT, dtmfsrc->sample_rate, NULL);
|
|
} else {
|
|
GstStructure *s;
|
|
gint sample_rate;
|
|
GstCaps *temp = NULL;
|
|
|
|
/* peer provides caps we can use to fixate, intersect. This always returns a
|
|
* writable caps. */
|
|
temp = gst_caps_intersect (srccaps, peercaps);
|
|
gst_caps_unref (srccaps);
|
|
gst_caps_unref (peercaps);
|
|
|
|
if (!temp) {
|
|
GST_DEBUG_OBJECT (dtmfsrc, "Could not get intersection with peer caps");
|
|
return FALSE;
|
|
}
|
|
|
|
if (gst_caps_is_empty (temp)) {
|
|
GST_DEBUG_OBJECT (dtmfsrc, "Intersection with peer caps is empty");
|
|
gst_caps_unref (temp);
|
|
return FALSE;
|
|
}
|
|
|
|
/* now fixate, start by taking the first caps */
|
|
gst_caps_truncate (temp);
|
|
srccaps = temp;
|
|
|
|
/* get first structure */
|
|
s = gst_caps_get_structure (srccaps, 0);
|
|
|
|
if (gst_structure_get_int (s, "rate", &sample_rate)) {
|
|
dtmfsrc->sample_rate = sample_rate;
|
|
GST_LOG_OBJECT (dtmfsrc, "using rate from caps %d", dtmfsrc->sample_rate);
|
|
} else {
|
|
GST_LOG_OBJECT (dtmfsrc, "using existing rate %d", dtmfsrc->sample_rate);
|
|
}
|
|
gst_structure_set (s, "rate", G_TYPE_INT, dtmfsrc->sample_rate, NULL);
|
|
}
|
|
|
|
ret = gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), srccaps);
|
|
|
|
gst_caps_unref (srccaps);
|
|
|
|
return ret;
|
|
}
|
|
|
|
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) {
|
|
g_slice_free (GstDTMFSrcEvent, event);
|
|
event = g_async_queue_try_pop (dtmfsrc->event_queue);
|
|
}
|
|
no_preroll = TRUE;
|
|
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:
|
|
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) {
|
|
g_slice_free (GstDTMFSrcEvent, event);
|
|
event = g_async_queue_try_pop (dtmfsrc->event_queue);
|
|
}
|
|
|
|
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);
|
|
}
|