| /* GstRtpDtmfDepay |
| * |
| * Copyright (C) 2008 Collabora Limited |
| * Copyright (C) 2008 Nokia Corporation |
| * Contact: Youness Alaoui <youness.alaoui@collabora.co.uk> |
| * |
| * 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 |
| * @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 |
| * <informaltable> |
| * <tgroup cols='4'> |
| * <colspec colname='Name' /> |
| * <colspec colname='Type' /> |
| * <colspec colname='Possible values' /> |
| * <colspec colname='Purpose' /> |
| * <thead> |
| * <row> |
| * <entry>Name</entry> |
| * <entry>GType</entry> |
| * <entry>Possible values</entry> |
| * <entry>Purpose</entry> |
| * </row> |
| * </thead> |
| * <tbody> |
| * <row> |
| * <entry>type</entry> |
| * <entry>G_TYPE_INT</entry> |
| * <entry>0-1</entry> |
| * <entry>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 specied |
| * by their number. This element currently only recognizes events. |
| * Do not confuse with "method" which specified the output. |
| * </entry> |
| * </row> |
| * <row> |
| * <entry>number</entry> |
| * <entry>G_TYPE_INT</entry> |
| * <entry>0-16</entry> |
| * <entry>The event number.</entry> |
| * </row> |
| * <row> |
| * <entry>volume</entry> |
| * <entry>G_TYPE_INT</entry> |
| * <entry>0-36</entry> |
| * <entry>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. |
| * </entry> |
| * </row> |
| * <row> |
| * <entry>method</entry> |
| * <entry>G_TYPE_INT</entry> |
| * <entry>1</entry> |
| * <entry>This field will always been 1 (ie RTP event) from this element. |
| * </entry> |
| * </row> |
| * </tbody> |
| * </tgroup> |
| * </informaltable> |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| #include "config.h" |
| #endif |
| |
| #include "gstrtpdtmfdepay.h" |
| |
| #include <string.h> |
| #include <math.h> |
| |
| #include <gst/audio/audio.h> |
| #include <gst/rtp/gstrtpbuffer.h> |
| |
| #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 |
| { |
| const char *event_name; |
| int event_encoding; |
| float low_frequency; |
| float high_frequency; |
| } DTMF_KEY; |
| |
| static const DTMF_KEY DTMF_KEYS[] = { |
| {"DTMF_KEY_EVENT_0", 0, 941, 1336}, |
| {"DTMF_KEY_EVENT_1", 1, 697, 1209}, |
| {"DTMF_KEY_EVENT_2", 2, 697, 1336}, |
| {"DTMF_KEY_EVENT_3", 3, 697, 1477}, |
| {"DTMF_KEY_EVENT_4", 4, 770, 1209}, |
| {"DTMF_KEY_EVENT_5", 5, 770, 1336}, |
| {"DTMF_KEY_EVENT_6", 6, 770, 1477}, |
| {"DTMF_KEY_EVENT_7", 7, 852, 1209}, |
| {"DTMF_KEY_EVENT_8", 8, 852, 1336}, |
| {"DTMF_KEY_EVENT_9", 9, 852, 1477}, |
| {"DTMF_KEY_EVENT_S", 10, 941, 1209}, |
| {"DTMF_KEY_EVENT_P", 11, 941, 1477}, |
| {"DTMF_KEY_EVENT_A", 12, 697, 1633}, |
| {"DTMF_KEY_EVENT_B", 13, 770, 1633}, |
| {"DTMF_KEY_EVENT_C", 14, 852, 1633}, |
| {"DTMF_KEY_EVENT_D", 15, 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 |
| }; |
| |
| enum |
| { |
| ARG_0 |
| }; |
| |
| 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); |
| |
| 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_pad_template (gstelement_class, |
| gst_static_pad_template_get (&gst_rtp_dtmf_depay_src_template)); |
| gst_element_class_add_pad_template (gstelement_class, |
| gst_static_pad_template_get (&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", |
| "Generates DTMF Sound from telephone-event RTP packets", |
| "Youness Alaoui <youness.alaoui@collabora.co.uk>"); |
| |
| 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 = 8000 /* default */ ; |
| 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; |
| gint 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; |
| |
| 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 != sizeof (GstRTPDTMFPayload)) |
| goto bad_packet; |
| |
| memcpy (&dtmf_payload, payload, sizeof (GstRTPDTMFPayload)); |
| |
| if (dtmf_payload.event > MAX_EVENT) |
| goto bad_packet; |
| |
| marker = gst_rtp_buffer_get_marker (&rtpbuffer); |
| |
| timestamp = gst_rtp_buffer_get_timestamp (&rtpbuffer); |
| |
| dtmf_payload.duration = g_ntohs (dtmf_payload.duration); |
| |
| /* 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; |
| } |
| |
| gboolean |
| gst_rtp_dtmf_depay_plugin_init (GstPlugin * plugin) |
| { |
| return gst_element_register (plugin, "rtpdtmfdepay", |
| GST_RANK_MARGINAL, GST_TYPE_RTP_DTMF_DEPAY); |
| } |