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539 lines
16 KiB
C
539 lines
16 KiB
C
/* GstRtpDtmfDepay
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*
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* Copyright (C) 2008 Collabora Limited
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* Copyright (C) 2008 Nokia Corporation
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* Contact: Youness Alaoui <youness.alaoui@collabora.co.uk>
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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-rtpdtmfdepay
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* @see_also: rtpdtmfsrc, rtpdtmfmux
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*
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* This element takes RTP DTMF packets and produces sound. It also emits a
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* message on the #GstBus.
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*
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* The message is called "dtmf-event" and has the following fields
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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></entry>
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* <entry>G_TYPE_INT</entry>
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* <entry>0-1</entry>
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* <entry>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 currently only recognizes events.
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* Do not confuse 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.
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* </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>This field will always been 1 (ie RTP event) from this element.
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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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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstrtpdtmfdepay.h"
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#include <string.h>
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#include <math.h>
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#include <gst/audio/audio.h>
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#include <gst/rtp/gstrtpbuffer.h>
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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_DUTY_CYCLE (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION)
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#define MIN_UNIT_TIME 0
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#define MAX_UNIT_TIME 1000
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#define DEFAULT_UNIT_TIME 0
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#define DEFAULT_MAX_DURATION 0
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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_rtp_dtmf_depay_debug);
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#define GST_CAT_DEFAULT gst_rtp_dtmf_depay_debug
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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PROP_0,
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PROP_UNIT_TIME,
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PROP_MAX_DURATION
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};
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enum
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{
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ARG_0
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};
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static GstStaticPadTemplate gst_rtp_dtmf_depay_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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static GstStaticPadTemplate gst_rtp_dtmf_depay_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
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"clock-rate = (int) [ 0, MAX ], "
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"encoding-name = (string) \"TELEPHONE-EVENT\"")
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);
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G_DEFINE_TYPE (GstRtpDTMFDepay, gst_rtp_dtmf_depay,
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GST_TYPE_RTP_BASE_DEPAYLOAD);
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static void gst_rtp_dtmf_depay_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtp_dtmf_depay_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static GstBuffer *gst_rtp_dtmf_depay_process (GstRTPBaseDepayload * depayload,
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GstBuffer * buf);
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gboolean gst_rtp_dtmf_depay_setcaps (GstRTPBaseDepayload * filter,
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GstCaps * caps);
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static void
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gst_rtp_dtmf_depay_class_init (GstRtpDTMFDepayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstRTPBaseDepayloadClass *gstrtpbasedepayload_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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gstrtpbasedepayload_class = GST_RTP_BASE_DEPAYLOAD_CLASS (klass);
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_dtmf_depay_src_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_rtp_dtmf_depay_sink_template));
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GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_depay_debug,
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"rtpdtmfdepay", 0, "rtpdtmfdepay element");
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gst_element_class_set_details_simple (gstelement_class,
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"RTP DTMF packet depayloader", "Codec/Depayloader/Network",
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"Generates DTMF Sound from telephone-event RTP packets",
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"Youness Alaoui <youness.alaoui@collabora.co.uk>");
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gobject_class->set_property =
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GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_set_property);
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gobject_class->get_property =
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GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_get_property);
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_UNIT_TIME,
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g_param_spec_uint ("unit-time", "Duration unittime",
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"The smallest unit (ms) the duration must be a multiple of (0 disables it)",
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MIN_UNIT_TIME, MAX_UNIT_TIME, DEFAULT_UNIT_TIME,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MAX_DURATION,
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g_param_spec_uint ("max-duration", "Maximum duration",
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"The maxumimum duration (ms) of the outgoing soundpacket. "
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"(0 = no limit)", 0, G_MAXUINT, DEFAULT_MAX_DURATION,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gstrtpbasedepayload_class->process =
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GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_process);
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gstrtpbasedepayload_class->set_caps =
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GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_setcaps);
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}
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static void
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gst_rtp_dtmf_depay_init (GstRtpDTMFDepay * rtpdtmfdepay)
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{
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rtpdtmfdepay->unit_time = DEFAULT_UNIT_TIME;
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}
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static void
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gst_rtp_dtmf_depay_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstRtpDTMFDepay *rtpdtmfdepay;
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rtpdtmfdepay = GST_RTP_DTMF_DEPAY (object);
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switch (prop_id) {
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case PROP_UNIT_TIME:
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rtpdtmfdepay->unit_time = g_value_get_uint (value);
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break;
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case PROP_MAX_DURATION:
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rtpdtmfdepay->max_duration = 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_rtp_dtmf_depay_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstRtpDTMFDepay *rtpdtmfdepay;
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rtpdtmfdepay = GST_RTP_DTMF_DEPAY (object);
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switch (prop_id) {
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case PROP_UNIT_TIME:
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g_value_set_uint (value, rtpdtmfdepay->unit_time);
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break;
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case PROP_MAX_DURATION:
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g_value_set_uint (value, rtpdtmfdepay->max_duration);
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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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gboolean
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gst_rtp_dtmf_depay_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps)
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{
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GstCaps *srccaps;
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GstStructure *structure = gst_caps_get_structure (caps, 0);
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gint clock_rate = 8000; /* default */
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gst_structure_get_int (structure, "clock-rate", &clock_rate);
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filter->clock_rate = clock_rate;
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srccaps = gst_caps_new_simple ("audio/x-raw-int",
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"width", G_TYPE_INT, 16,
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"depth", G_TYPE_INT, 16,
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"endianness", G_TYPE_INT, G_BYTE_ORDER,
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"signed", G_TYPE_BOOLEAN, TRUE,
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"channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, clock_rate, NULL);
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gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter), srccaps);
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gst_caps_unref (srccaps);
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return TRUE;
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}
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static GstBuffer *
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gst_dtmf_src_generate_tone (GstRtpDTMFDepay * rtpdtmfdepay,
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GstRTPDTMFPayload payload)
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{
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GstBuffer *buf;
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GstMapInfo map;
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gint16 *p;
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gint tone_size;
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double i = 0;
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double amplitude, f1, f2;
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double volume_factor;
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DTMF_KEY key = DTMF_KEYS[payload.event];
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guint32 clock_rate = 8000 /* default */ ;
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GstRTPBaseDepayload *depayload = GST_RTP_BASE_DEPAYLOAD (rtpdtmfdepay);
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gint volume;
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static GstAllocationParams params = { 0, 0, 0, 1, };
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clock_rate = depayload->clock_rate;
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/* Create a buffer for the tone */
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tone_size = (payload.duration * SAMPLE_SIZE * CHANNELS) / 8;
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buf = gst_buffer_new_allocate (NULL, tone_size, ¶ms);
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GST_BUFFER_DURATION (buf) = payload.duration * GST_SECOND / clock_rate;
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volume = payload.volume;
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gst_buffer_map (buf, &map, GST_MAP_WRITE);
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p = (gint16 *) map.data;
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volume_factor = pow (10, (-volume) / 20);
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/*
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* For each sample point we calculate 'x' as the
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* the amplitude value.
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*/
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for (i = 0; i < (tone_size / (SAMPLE_SIZE / 8)); i++) {
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/*
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* We add the fundamental frequencies together.
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*/
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f1 = sin (2 * M_PI * key.low_frequency * (rtpdtmfdepay->sample /
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clock_rate));
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f2 = sin (2 * M_PI * key.high_frequency * (rtpdtmfdepay->sample /
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clock_rate));
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amplitude = (f1 + f2) / 2;
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/* Adjust the volume */
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amplitude *= volume_factor;
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/* Make the [-1:1] interval into a [-32767:32767] interval */
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amplitude *= 32767;
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/* Store it in the data buffer */
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*(p++) = (gint16) amplitude;
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(rtpdtmfdepay->sample)++;
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}
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gst_buffer_unmap (buf, &map);
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return buf;
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}
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static GstBuffer *
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gst_rtp_dtmf_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
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{
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GstRtpDTMFDepay *rtpdtmfdepay = NULL;
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GstBuffer *outbuf = NULL;
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gint payload_len;
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guint8 *payload = NULL;
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guint32 timestamp;
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GstRTPDTMFPayload dtmf_payload;
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gboolean marker;
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GstStructure *structure = NULL;
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GstMessage *dtmf_message = NULL;
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GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
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rtpdtmfdepay = GST_RTP_DTMF_DEPAY (depayload);
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gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuffer);
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payload_len = gst_rtp_buffer_get_payload_len (&rtpbuffer);
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payload = gst_rtp_buffer_get_payload (&rtpbuffer);
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if (payload_len != sizeof (GstRTPDTMFPayload))
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goto bad_packet;
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memcpy (&dtmf_payload, payload, sizeof (GstRTPDTMFPayload));
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if (dtmf_payload.event > MAX_EVENT)
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goto bad_packet;
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marker = gst_rtp_buffer_get_marker (&rtpbuffer);
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timestamp = gst_rtp_buffer_get_timestamp (&rtpbuffer);
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dtmf_payload.duration = g_ntohs (dtmf_payload.duration);
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/* clip to whole units of unit_time */
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if (rtpdtmfdepay->unit_time) {
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guint unit_time_clock =
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(rtpdtmfdepay->unit_time * depayload->clock_rate) / 1000;
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if (dtmf_payload.duration % unit_time_clock) {
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/* Make sure we don't overflow the duration */
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if (dtmf_payload.duration < G_MAXUINT16 - unit_time_clock)
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dtmf_payload.duration += unit_time_clock -
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(dtmf_payload.duration % unit_time_clock);
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else
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dtmf_payload.duration -= dtmf_payload.duration % unit_time_clock;
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}
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}
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/* clip to max duration */
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if (rtpdtmfdepay->max_duration) {
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guint max_duration_clock =
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(rtpdtmfdepay->max_duration * depayload->clock_rate) / 1000;
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if (max_duration_clock < G_MAXUINT16 &&
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dtmf_payload.duration > max_duration_clock)
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dtmf_payload.duration = max_duration_clock;
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}
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GST_DEBUG_OBJECT (depayload, "Received new RTP DTMF packet : "
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"marker=%d - timestamp=%u - event=%d - duration=%d",
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marker, timestamp, dtmf_payload.event, dtmf_payload.duration);
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GST_DEBUG_OBJECT (depayload,
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"Previous information : timestamp=%u - duration=%d",
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rtpdtmfdepay->previous_ts, rtpdtmfdepay->previous_duration);
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/* First packet */
|
|
if (marker || rtpdtmfdepay->previous_ts != timestamp) {
|
|
rtpdtmfdepay->sample = 0;
|
|
rtpdtmfdepay->previous_ts = timestamp;
|
|
rtpdtmfdepay->previous_duration = dtmf_payload.duration;
|
|
rtpdtmfdepay->first_gst_ts = GST_BUFFER_PTS (buf);
|
|
|
|
structure = gst_structure_new ("dtmf-event",
|
|
"number", G_TYPE_INT, dtmf_payload.event,
|
|
"volume", G_TYPE_INT, dtmf_payload.volume,
|
|
"type", G_TYPE_INT, 1, "method", G_TYPE_INT, 1, NULL);
|
|
if (structure) {
|
|
dtmf_message =
|
|
gst_message_new_element (GST_OBJECT (depayload), structure);
|
|
if (dtmf_message) {
|
|
if (!gst_element_post_message (GST_ELEMENT (depayload), dtmf_message)) {
|
|
GST_ERROR_OBJECT (depayload,
|
|
"Unable to send dtmf-event message to bus");
|
|
}
|
|
} else {
|
|
GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event message");
|
|
}
|
|
} else {
|
|
GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event structure");
|
|
}
|
|
} else {
|
|
guint16 duration = dtmf_payload.duration;
|
|
dtmf_payload.duration -= rtpdtmfdepay->previous_duration;
|
|
/* If late buffer, ignore */
|
|
if (duration > rtpdtmfdepay->previous_duration)
|
|
rtpdtmfdepay->previous_duration = duration;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (depayload, "new previous duration : %d - new duration : %d"
|
|
" - diff : %d - clock rate : %d - timestamp : %" G_GUINT64_FORMAT,
|
|
rtpdtmfdepay->previous_duration, dtmf_payload.duration,
|
|
(rtpdtmfdepay->previous_duration - dtmf_payload.duration),
|
|
depayload->clock_rate, GST_BUFFER_TIMESTAMP (buf));
|
|
|
|
/* If late or duplicate packet (like the redundant end packet). Ignore */
|
|
if (dtmf_payload.duration > 0) {
|
|
outbuf = gst_dtmf_src_generate_tone (rtpdtmfdepay, dtmf_payload);
|
|
|
|
|
|
GST_BUFFER_PTS (outbuf) = rtpdtmfdepay->first_gst_ts +
|
|
(rtpdtmfdepay->previous_duration - dtmf_payload.duration) *
|
|
GST_SECOND / depayload->clock_rate;
|
|
GST_BUFFER_OFFSET (outbuf) =
|
|
(rtpdtmfdepay->previous_duration - dtmf_payload.duration) *
|
|
GST_SECOND / depayload->clock_rate;
|
|
GST_BUFFER_OFFSET_END (outbuf) = rtpdtmfdepay->previous_duration *
|
|
GST_SECOND / depayload->clock_rate;
|
|
|
|
GST_DEBUG_OBJECT (depayload,
|
|
"timestamp : %" G_GUINT64_FORMAT " - time %" GST_TIME_FORMAT,
|
|
GST_BUFFER_TIMESTAMP (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
|
|
|
|
}
|
|
|
|
gst_rtp_buffer_unmap (&rtpbuffer);
|
|
|
|
return outbuf;
|
|
|
|
bad_packet:
|
|
GST_ELEMENT_WARNING (rtpdtmfdepay, STREAM, DECODE,
|
|
("Packet did not validate"), (NULL));
|
|
|
|
if (rtpbuffer.buffer != NULL)
|
|
gst_rtp_buffer_unmap (&rtpbuffer);
|
|
|
|
return NULL;
|
|
}
|
|
|
|
gboolean
|
|
gst_rtp_dtmf_depay_plugin_init (GstPlugin * plugin)
|
|
{
|
|
return gst_element_register (plugin, "rtpdtmfdepay",
|
|
GST_RANK_MARGINAL, GST_TYPE_RTP_DTMF_DEPAY);
|
|
}
|