--- /dev/null
+/* GStreamer
+ *
+ * Copyright 2007 Nokia Corporation
+ * Copyright 2007 Collabora Ltd,
+ * @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>
+ *
+ * Copyright (C) <2005> Wim Taymans <wim@fluendo.com>
+ *
+ * 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include <string.h>
+#include <math.h>
+
+#include <gst/rtp/gstrtpbuffer.h>
+#include "gstrtpdtmfdepay.h"
+
+#ifndef M_PI
+# define M_PI 3.14159265358979323846 /* pi */
+#endif
+
+
+#define GST_TONE_DTMF_TYPE_EVENT 0
+#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_EVENT 0
+#define MAX_EVENT 16
+#define MIN_VOLUME 0
+#define MAX_VOLUME 36
+#define MIN_INTER_DIGIT_INTERVAL 100
+#define MIN_PULSE_DURATION 250
+#define MIN_DUTY_CYCLE (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION)
+
+
+typedef struct st_dtmf_key {
+ 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,
+};
+
+/* elementfactory information */
+static const GstElementDetails gst_rtp_dtmfdepay_details =
+GST_ELEMENT_DETAILS ("RTP DTMF packet depayloader",
+ "Codec/Depayloader/Network",
+ "Generates DTMF Sound from telephone-event RTP packets",
+ "Youness Alaoui <youness.alaoui@collabora.co.uk>");
+
+GST_DEBUG_CATEGORY_STATIC (gst_rtp_dtmf_depay_debug);
+#define GST_CAT_DEFAULT gst_rtp_dtmf_depay_debug
+
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+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-int, "
+ "width = (int) 16, "
+ "depth = (int) 16, "
+ "endianness = (int) 1234, "
+ "signed = (boolean) true, "
+ "rate = (int) [0, MAX], "
+ "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\"")
+ );
+
+GST_BOILERPLATE (GstRtpDTMFDepay, gst_rtp_dtmf_depay, GstBaseRTPDepayload,
+ GST_TYPE_BASE_RTP_DEPAYLOAD);
+
+
+static GstBuffer *gst_rtp_dtmf_depay_process (GstBaseRTPDepayload * depayload,
+ GstBuffer * buf);
+gboolean gst_rtp_dtmf_depay_setcaps (GstBaseRTPDepayload * filter,
+ GstCaps * caps);
+
+/*static void
+gst_rtp_dtmf_depay_set_gst_timestamp (GstBaseRTPDepayload * filter,
+ guint32 rtptime, GstBuffer * buf);
+*/
+
+static void
+gst_rtp_dtmf_depay_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtp_dtmf_depay_src_template));
+ gst_element_class_add_pad_template (element_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_details (element_class, &gst_rtp_dtmfdepay_details);
+}
+
+static void
+gst_rtp_dtmf_depay_class_init (GstRtpDTMFDepayClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+ gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+
+ parent_class = g_type_class_peek_parent (klass);
+
+ gstbasertpdepayload_class->process = gst_rtp_dtmf_depay_process;
+ gstbasertpdepayload_class->set_caps = gst_rtp_dtmf_depay_setcaps;
+ // gstbasertpdepayload_class->set_gst_timestamp = gst_rtp_dtmf_depay_set_gst_timestamp;
+
+}
+
+static void
+gst_rtp_dtmf_depay_init (GstRtpDTMFDepay * rtpdtmfdepay,
+ GstRtpDTMFDepayClass * klass)
+{
+
+}
+
+
+gboolean
+gst_rtp_dtmf_depay_setcaps (GstBaseRTPDepayload * filter, GstCaps * caps)
+{
+ GstCaps *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;
+
+ srccaps = gst_caps_new_simple ("audio/x-raw-int",
+ "width", G_TYPE_INT, 16,
+ "depth", G_TYPE_INT, 16,
+ "endianness", G_TYPE_INT, 1234,
+ "signed", G_TYPE_BOOLEAN, TRUE,
+ "channels", G_TYPE_INT, 1,
+ "rate", G_TYPE_INT, clock_rate, NULL);
+ gst_pad_set_caps (GST_BASE_RTP_DEPAYLOAD_SRCPAD (filter), srccaps);
+ gst_caps_unref (srccaps);
+
+ return TRUE;
+}
+
+void
+gst_rtp_dtmf_depay_set_gst_timestamp (GstBaseRTPDepayload * filter,
+ guint32 rtptime, GstBuffer * buf)
+{
+ GstClockTime timestamp, duration;
+
+
+ timestamp = GST_BUFFER_TIMESTAMP (buf);
+ duration = GST_BUFFER_DURATION (buf);
+
+ /* if this is the first buffer send a NEWSEGMENT */
+ if (filter->need_newsegment) {
+ GstEvent *event;
+ GstClockTime stop, position;
+
+ stop = -1;
+
+ position = 0;
+
+ event =
+ gst_event_new_new_segment_full (FALSE, 1.0,
+ 1.0, GST_FORMAT_TIME, 0, stop, position);
+
+ gst_pad_push_event (filter->srcpad, event);
+
+ filter->need_newsegment = FALSE;
+ GST_DEBUG_OBJECT (filter, "Pushed newsegment event on this first buffer");
+ }
+}
+
+#if 0
+static void
+gst_dtmf_src_generate_silence(GstBuffer * buffer, float duration)
+{
+ gint buf_size;
+
+ /* Create a buffer with data set to 0 */
+ buf_size = ((duration/1000)*SAMPLE_RATE*SAMPLE_SIZE*CHANNELS)/8;
+ GST_BUFFER_SIZE (buffer) = buf_size;
+ GST_BUFFER_MALLOCDATA (buffer) = g_malloc0(buf_size);
+ GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
+
+}
+#endif
+
+static void
+gst_dtmf_src_generate_tone(GstRtpDTMFDepay *rtpdtmfdepay,
+ GstRTPDTMFPayload payload, GstBuffer * buffer)
+{
+ 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 */;
+ GstBaseRTPDepayload * depayload = GST_BASE_RTP_DEPAYLOAD (rtpdtmfdepay);
+
+ clock_rate = depayload->clock_rate;
+
+ /* Create a buffer for the tone */
+ tone_size = (payload.duration*SAMPLE_SIZE*CHANNELS)/8;
+ GST_BUFFER_SIZE (buffer) = tone_size;
+ GST_BUFFER_MALLOCDATA (buffer) = g_malloc(tone_size);
+ GST_BUFFER_DATA (buffer) = GST_BUFFER_MALLOCDATA (buffer);
+ GST_BUFFER_DURATION (buffer) = payload.duration * GST_SECOND / clock_rate;
+
+ p = (gint16 *) GST_BUFFER_MALLOCDATA (buffer);
+
+ volume_factor = pow (10, (-payload.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)++;
+ }
+}
+
+
+static GstBuffer *
+gst_rtp_dtmf_depay_process (GstBaseRTPDepayload * 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;
+
+ rtpdtmfdepay = GST_RTP_DTMF_DEPAY (depayload);
+
+ if (!gst_rtp_buffer_validate (buf))
+ goto bad_packet;
+
+ payload_len = gst_rtp_buffer_get_payload_len (buf);
+ payload = gst_rtp_buffer_get_payload (buf);
+
+ 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 (buf);
+
+ timestamp = gst_rtp_buffer_get_timestamp (buf);
+
+ dtmf_payload.duration = g_ntohs (dtmf_payload.duration);
+
+ 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_TIMESTAMP (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_DEBUG_OBJECT (depayload, "Unable to send dtmf-event message to bus");
+ }
+ } else {
+ GST_DEBUG_OBJECT (depayload, "Unable to create dtmf-event message");
+ }
+ } else {
+ GST_DEBUG_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 : %llu",
+ 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_buffer_new ();
+ gst_dtmf_src_generate_tone(rtpdtmfdepay, dtmf_payload, outbuf);
+
+
+ GST_BUFFER_TIMESTAMP (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 : %llu - time %" GST_TIME_FORMAT,
+ GST_BUFFER_TIMESTAMP (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
+
+ }
+
+ return outbuf;
+
+
+bad_packet:
+ GST_ELEMENT_WARNING (rtpdtmfdepay, STREAM, DECODE,
+ ("Packet did not validate"), (NULL));
+ 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);
+}
+
projects / platform / upstream / gstreamer.git / commitdiff