/* GstRtpDtmfDepay * * Copyright (C) 2008 Collabora Limited * Copyright (C) 2008 Nokia Corporation * Contact: Youness Alaoui * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION:element-rtpdtmfdepay * @title: rtpdtmfdepay * @see_also: rtpdtmfsrc, rtpdtmfmux * * This element takes RTP DTMF packets and produces sound. It also emits a * message on the #GstBus. * * The message is called "dtmf-event" and has the following fields: * * * `type` (G_TYPE_INT, 0-1): Which of the two methods * specified in RFC 2833 to use. The value should be 0 for tones and 1 for * named events. Tones are specified by their frequencies and events are specified * by their number. This element currently only recognizes events. * Do not confuse with "method" which specified the output. * * * `number` (G_TYPE_INT, 0-16): The event number. * * * `volume` (G_TYPE_INT, 0-36): This field describes the power level of the tone, expressed in dBm0 * after dropping the sign. Power levels range from 0 to -63 dBm0. The range of * valid DTMF is from 0 to -36 dBm0. * * * `method` (G_TYPE_INT, 1): This field will always been 1 (ie RTP event) from this element. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include "gstrtpdtmfdepay.h" #include #include #include #include #include #define DEFAULT_PACKET_INTERVAL 50 /* ms */ #define MIN_PACKET_INTERVAL 10 /* ms */ #define MAX_PACKET_INTERVAL 50 /* ms */ #define SAMPLE_RATE 8000 #define SAMPLE_SIZE 16 #define CHANNELS 1 #define MIN_DUTY_CYCLE (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION) #define MIN_UNIT_TIME 0 #define MAX_UNIT_TIME 1000 #define DEFAULT_UNIT_TIME 0 #define DEFAULT_MAX_DURATION 0 typedef struct st_dtmf_key { float low_frequency; float high_frequency; } DTMF_KEY; static const DTMF_KEY DTMF_KEYS[] = { {941, 1336}, {697, 1209}, {697, 1336}, {697, 1477}, {770, 1209}, {770, 1336}, {770, 1477}, {852, 1209}, {852, 1336}, {852, 1477}, {941, 1209}, {941, 1477}, {697, 1633}, {770, 1633}, {852, 1633}, {941, 1633}, }; #define MAX_DTMF_EVENTS 16 enum { DTMF_KEY_EVENT_1 = 1, DTMF_KEY_EVENT_2 = 2, DTMF_KEY_EVENT_3 = 3, DTMF_KEY_EVENT_4 = 4, DTMF_KEY_EVENT_5 = 5, DTMF_KEY_EVENT_6 = 6, DTMF_KEY_EVENT_7 = 7, DTMF_KEY_EVENT_8 = 8, DTMF_KEY_EVENT_9 = 9, DTMF_KEY_EVENT_0 = 0, DTMF_KEY_EVENT_STAR = 10, DTMF_KEY_EVENT_POUND = 11, DTMF_KEY_EVENT_A = 12, DTMF_KEY_EVENT_B = 13, DTMF_KEY_EVENT_C = 14, DTMF_KEY_EVENT_D = 15, }; GST_DEBUG_CATEGORY_STATIC (gst_rtp_dtmf_depay_debug); #define GST_CAT_DEFAULT gst_rtp_dtmf_depay_debug enum { /* FILL ME */ LAST_SIGNAL }; enum { PROP_0, PROP_UNIT_TIME, PROP_MAX_DURATION }; static GstStaticPadTemplate gst_rtp_dtmf_depay_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw, " "format = (string) \"" GST_AUDIO_NE (S16) "\", " "rate = " GST_AUDIO_RATE_RANGE ", " "channels = (int) 1") ); static GstStaticPadTemplate gst_rtp_dtmf_depay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) [ 0, MAX ], " "encoding-name = (string) \"TELEPHONE-EVENT\"") ); G_DEFINE_TYPE (GstRtpDTMFDepay, gst_rtp_dtmf_depay, GST_TYPE_RTP_BASE_DEPAYLOAD); GST_ELEMENT_REGISTER_DEFINE (rtpdtmfdepay, "rtpdtmfdepay", GST_RANK_MARGINAL, GST_TYPE_RTP_DTMF_DEPAY); static void gst_rtp_dtmf_depay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtp_dtmf_depay_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstBuffer *gst_rtp_dtmf_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf); gboolean gst_rtp_dtmf_depay_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps); static void gst_rtp_dtmf_depay_class_init (GstRtpDTMFDepayClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstRTPBaseDepayloadClass *gstrtpbasedepayload_class; gobject_class = G_OBJECT_CLASS (klass); gstelement_class = GST_ELEMENT_CLASS (klass); gstrtpbasedepayload_class = GST_RTP_BASE_DEPAYLOAD_CLASS (klass); gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_dtmf_depay_src_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_dtmf_depay_sink_template); GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_depay_debug, "rtpdtmfdepay", 0, "rtpdtmfdepay element"); gst_element_class_set_static_metadata (gstelement_class, "RTP DTMF packet depayloader", "Codec/Depayloader/Network/RTP", "Generates DTMF Sound from telephone-event RTP packets", "Youness Alaoui "); gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_set_property); gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_get_property); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_UNIT_TIME, g_param_spec_uint ("unit-time", "Duration unittime", "The smallest unit (ms) the duration must be a multiple of (0 disables it)", MIN_UNIT_TIME, MAX_UNIT_TIME, DEFAULT_UNIT_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MAX_DURATION, g_param_spec_uint ("max-duration", "Maximum duration", "The maxumimum duration (ms) of the outgoing soundpacket. " "(0 = no limit)", 0, G_MAXUINT, DEFAULT_MAX_DURATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gstrtpbasedepayload_class->process = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_process); gstrtpbasedepayload_class->set_caps = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_setcaps); } static void gst_rtp_dtmf_depay_init (GstRtpDTMFDepay * rtpdtmfdepay) { rtpdtmfdepay->unit_time = DEFAULT_UNIT_TIME; } static void gst_rtp_dtmf_depay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstRtpDTMFDepay *rtpdtmfdepay; rtpdtmfdepay = GST_RTP_DTMF_DEPAY (object); switch (prop_id) { case PROP_UNIT_TIME: rtpdtmfdepay->unit_time = g_value_get_uint (value); break; case PROP_MAX_DURATION: rtpdtmfdepay->max_duration = g_value_get_uint (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtp_dtmf_depay_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRtpDTMFDepay *rtpdtmfdepay; rtpdtmfdepay = GST_RTP_DTMF_DEPAY (object); switch (prop_id) { case PROP_UNIT_TIME: g_value_set_uint (value, rtpdtmfdepay->unit_time); break; case PROP_MAX_DURATION: g_value_set_uint (value, rtpdtmfdepay->max_duration); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } gboolean gst_rtp_dtmf_depay_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps) { GstCaps *filtercaps, *srccaps; GstStructure *structure = gst_caps_get_structure (caps, 0); gint clock_rate = 8000; /* default */ gst_structure_get_int (structure, "clock-rate", &clock_rate); filter->clock_rate = clock_rate; filtercaps = gst_pad_get_pad_template_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter)); filtercaps = gst_caps_make_writable (filtercaps); gst_caps_set_simple (filtercaps, "rate", G_TYPE_INT, clock_rate, NULL); srccaps = gst_pad_peer_query_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter), filtercaps); gst_caps_unref (filtercaps); gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter), srccaps); gst_caps_unref (srccaps); return TRUE; } static GstBuffer * gst_dtmf_src_generate_tone (GstRtpDTMFDepay * rtpdtmfdepay, GstRTPDTMFPayload payload) { GstBuffer *buf; GstMapInfo map; gint16 *p; gint tone_size; double i = 0; double amplitude, f1, f2; double volume_factor; DTMF_KEY key = DTMF_KEYS[payload.event]; guint32 clock_rate; GstRTPBaseDepayload *depayload = GST_RTP_BASE_DEPAYLOAD (rtpdtmfdepay); gint volume; static GstAllocationParams params = { 0, 1, 0, 0, }; clock_rate = depayload->clock_rate; /* Create a buffer for the tone */ tone_size = (payload.duration * SAMPLE_SIZE * CHANNELS) / 8; buf = gst_buffer_new_allocate (NULL, tone_size, ¶ms); GST_BUFFER_DURATION (buf) = payload.duration * GST_SECOND / clock_rate; volume = payload.volume; gst_buffer_map (buf, &map, GST_MAP_WRITE); p = (gint16 *) map.data; volume_factor = pow (10, (-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 * (rtpdtmfdepay->sample / clock_rate)); f2 = sin (2 * M_PI * key.high_frequency * (rtpdtmfdepay->sample / clock_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; (rtpdtmfdepay->sample)++; } gst_buffer_unmap (buf, &map); return buf; } static GstBuffer * gst_rtp_dtmf_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf) { GstRtpDTMFDepay *rtpdtmfdepay = NULL; GstBuffer *outbuf = NULL; guint payload_len; guint8 *payload = NULL; guint32 timestamp; GstRTPDTMFPayload dtmf_payload; gboolean marker; GstStructure *structure = NULL; GstMessage *dtmf_message = NULL; GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT; GstBitReader bitreader; rtpdtmfdepay = GST_RTP_DTMF_DEPAY (depayload); gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuffer); payload_len = gst_rtp_buffer_get_payload_len (&rtpbuffer); payload = gst_rtp_buffer_get_payload (&rtpbuffer); if (payload_len != 4) goto bad_packet; gst_bit_reader_init (&bitreader, payload, payload_len); gst_bit_reader_get_bits_uint8 (&bitreader, &dtmf_payload.event, 8); gst_bit_reader_skip (&bitreader, 2); gst_bit_reader_get_bits_uint8 (&bitreader, &dtmf_payload.volume, 6); gst_bit_reader_get_bits_uint16 (&bitreader, &dtmf_payload.duration, 16); if (dtmf_payload.event > MAX_EVENT) goto bad_packet; marker = gst_rtp_buffer_get_marker (&rtpbuffer); timestamp = gst_rtp_buffer_get_timestamp (&rtpbuffer); /* clip to whole units of unit_time */ if (rtpdtmfdepay->unit_time) { guint unit_time_clock = (rtpdtmfdepay->unit_time * depayload->clock_rate) / 1000; if (dtmf_payload.duration % unit_time_clock) { /* Make sure we don't overflow the duration */ if (dtmf_payload.duration < G_MAXUINT16 - unit_time_clock) dtmf_payload.duration += unit_time_clock - (dtmf_payload.duration % unit_time_clock); else dtmf_payload.duration -= dtmf_payload.duration % unit_time_clock; } } /* clip to max duration */ if (rtpdtmfdepay->max_duration) { guint max_duration_clock = (rtpdtmfdepay->max_duration * depayload->clock_rate) / 1000; if (max_duration_clock < G_MAXUINT16 && dtmf_payload.duration > max_duration_clock) dtmf_payload.duration = max_duration_clock; } GST_DEBUG_OBJECT (depayload, "Received new RTP DTMF packet : " "marker=%d - timestamp=%u - event=%d - duration=%d", marker, timestamp, dtmf_payload.event, dtmf_payload.duration); GST_DEBUG_OBJECT (depayload, "Previous information : timestamp=%u - duration=%d", rtpdtmfdepay->previous_ts, rtpdtmfdepay->previous_duration); /* 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; }