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390 lines
12 KiB
C
390 lines
12 KiB
C
/* GstRtpDtmfDepay
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*
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* Copyright (C) <2008> Collabora.
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* Copyright (C) <2008> Nokia.
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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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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <string.h>
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#include <math.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpdtmfdepay.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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#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 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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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_rtp_dtmfdepay_details =
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GST_ELEMENT_DETAILS ("RTP DTMF packet depayloader",
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"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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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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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-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 = (boolean) true, "
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"rate = (int) [0, MAX], "
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"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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GST_BOILERPLATE (GstRtpDTMFDepay, gst_rtp_dtmf_depay, GstBaseRTPDepayload,
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GST_TYPE_BASE_RTP_DEPAYLOAD);
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static GstBuffer *gst_rtp_dtmf_depay_process (GstBaseRTPDepayload * depayload,
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GstBuffer * buf);
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gboolean gst_rtp_dtmf_depay_setcaps (GstBaseRTPDepayload * filter,
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GstCaps * caps);
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static void
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gst_rtp_dtmf_depay_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_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 (element_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 (element_class, &gst_rtp_dtmfdepay_details);
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}
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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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GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gstbasertpdepayload_class->process = gst_rtp_dtmf_depay_process;
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gstbasertpdepayload_class->set_caps = 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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GstRtpDTMFDepayClass * klass)
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{
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}
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gboolean
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gst_rtp_dtmf_depay_setcaps (GstBaseRTPDepayload * 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, 1234,
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"signed", G_TYPE_BOOLEAN, TRUE,
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"channels", G_TYPE_INT, 1,
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"rate", G_TYPE_INT, clock_rate, NULL);
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gst_pad_set_caps (GST_BASE_RTP_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 void
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gst_dtmf_src_generate_tone(GstRtpDTMFDepay *rtpdtmfdepay,
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GstRTPDTMFPayload payload, GstBuffer * buffer)
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{
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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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GstBaseRTPDepayload * depayload = GST_BASE_RTP_DEPAYLOAD (rtpdtmfdepay);
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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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GST_BUFFER_SIZE (buffer) = tone_size;
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GST_BUFFER_MALLOCDATA (buffer) = g_malloc(tone_size);
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GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
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GST_BUFFER_DURATION (buffer) = payload.duration * GST_SECOND / clock_rate;
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p = (gint16 *) GST_BUFFER_MALLOCDATA (buffer);
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volume_factor = pow (10, (-payload.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 / clock_rate));
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f2 = sin(2 * M_PI * key.high_frequency * (rtpdtmfdepay->sample / 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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}
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static GstBuffer *
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gst_rtp_dtmf_depay_process (GstBaseRTPDepayload * 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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rtpdtmfdepay = GST_RTP_DTMF_DEPAY (depayload);
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if (!gst_rtp_buffer_validate (buf))
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goto bad_packet;
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payload_len = gst_rtp_buffer_get_payload_len (buf);
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payload = gst_rtp_buffer_get_payload (buf);
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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 (buf);
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timestamp = gst_rtp_buffer_get_timestamp (buf);
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dtmf_payload.duration = g_ntohs (dtmf_payload.duration);
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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, "Previous information : timestamp=%u - duration=%d",
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rtpdtmfdepay->previous_ts, rtpdtmfdepay->previous_duration);
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/* First packet */
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if (marker || rtpdtmfdepay->previous_ts != timestamp) {
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rtpdtmfdepay->sample = 0;
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rtpdtmfdepay->previous_ts = timestamp;
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rtpdtmfdepay->previous_duration = dtmf_payload.duration;
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rtpdtmfdepay->first_gst_ts = GST_BUFFER_TIMESTAMP (buf);
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structure = gst_structure_new ("dtmf-event",
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"number", G_TYPE_INT, dtmf_payload.event,
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"volume", G_TYPE_INT, dtmf_payload.volume,
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"type", G_TYPE_INT, 1,
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"method", G_TYPE_INT, 1,
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NULL);
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if (structure) {
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dtmf_message = gst_message_new_element (GST_OBJECT (depayload), structure);
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if (dtmf_message) {
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if (!gst_element_post_message (GST_ELEMENT (depayload), dtmf_message)) {
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GST_DEBUG_OBJECT (depayload, "Unable to send dtmf-event message to bus");
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}
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} else {
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GST_DEBUG_OBJECT (depayload, "Unable to create dtmf-event message");
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}
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} else {
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GST_DEBUG_OBJECT (depayload, "Unable to create dtmf-event structure");
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}
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} else {
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guint16 duration = dtmf_payload.duration;
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dtmf_payload.duration -= rtpdtmfdepay->previous_duration;
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/* If late buffer, ignore */
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if (duration > rtpdtmfdepay->previous_duration)
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rtpdtmfdepay->previous_duration = duration;
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}
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GST_DEBUG_OBJECT (depayload, "new previous duration : %d - new duration : %d"
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" - diff : %d - clock rate : %d - timestamp : %llu",
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rtpdtmfdepay->previous_duration, dtmf_payload.duration,
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(rtpdtmfdepay->previous_duration - dtmf_payload.duration),
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depayload->clock_rate, GST_BUFFER_TIMESTAMP (buf));
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/* If late or duplicate packet (like the redundant end packet). Ignore */
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if (dtmf_payload.duration > 0) {
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outbuf = gst_buffer_new ();
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gst_dtmf_src_generate_tone(rtpdtmfdepay, dtmf_payload, outbuf);
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GST_BUFFER_TIMESTAMP (outbuf) = rtpdtmfdepay->first_gst_ts +
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(rtpdtmfdepay->previous_duration - dtmf_payload.duration) *
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GST_SECOND / depayload->clock_rate;
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GST_BUFFER_OFFSET (outbuf) =
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(rtpdtmfdepay->previous_duration - dtmf_payload.duration) *
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GST_SECOND / depayload->clock_rate;
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GST_BUFFER_OFFSET_END (outbuf) = rtpdtmfdepay->previous_duration *
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GST_SECOND / depayload->clock_rate;
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GST_DEBUG_OBJECT (depayload, "timestamp : %llu - time %" GST_TIME_FORMAT,
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GST_BUFFER_TIMESTAMP (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
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}
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return outbuf;
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bad_packet:
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GST_ELEMENT_WARNING (rtpdtmfdepay, STREAM, DECODE,
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("Packet did not validate"), (NULL));
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return NULL;
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}
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gboolean
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gst_rtp_dtmf_depay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpdtmfdepay",
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GST_RANK_MARGINAL, GST_TYPE_RTP_DTMF_DEPAY);
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}
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