-/* GStreamer
+/* GstRtpDtmfDepay
*
- * Copyright 2007 Nokia Corporation
- * Copyright 2007 Collabora Ltd,
- * @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>
- *
- * Copyright (C) <2005> Wim Taymans <wim@fluendo.com>
+ * 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
*
* 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.
+ * 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"
+#include "config.h"
#endif
+#include "gstrtpdtmfdepay.h"
+
#include <string.h>
#include <math.h>
+#include <gst/audio/audio.h>
+#include <gst/base/gstbitreader.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 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)
+#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 {
- char *event_name;
- int event_encoding;
- float low_frequency;
- float high_frequency;
+typedef struct st_dtmf_key
+{
+ 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},
+ {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,
+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
{
- ARG_0
+ 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-int, "
- "width = (int) 16, "
- "depth = (int) 16, "
- "endianness = (int) 1234, "
- "signed = (boolean) true, "
- "rate = (int) [0, MAX], "
- "channels = (int) 1")
+ 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 =
"encoding-name = (string) \"TELEPHONE-EVENT\"")
);
-GST_BOILERPLATE (GstRtpDTMFDepay, gst_rtp_dtmf_depay, GstBaseRTPDepayload,
- GST_TYPE_BASE_RTP_DEPAYLOAD);
-
+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 GstBuffer *gst_rtp_dtmf_depay_process (GstBaseRTPDepayload * 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 (GstBaseRTPDepayload * filter,
+gboolean gst_rtp_dtmf_depay_setcaps (GstRTPBaseDepayload * 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;
+ GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
- gobject_class = (GObjectClass *) klass;
- gstelement_class = (GstElementClass *) klass;
- gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+ gobject_class = G_OBJECT_CLASS (klass);
+ gstelement_class = GST_ELEMENT_CLASS (klass);
+ gstrtpbasedepayload_class = GST_RTP_BASE_DEPAYLOAD_CLASS (klass);
- parent_class = g_type_class_peek_parent (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);
- 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;
+ 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,
- GstRtpDTMFDepayClass * klass)
+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 (GstBaseRTPDepayload * filter, GstCaps * caps)
+gst_rtp_dtmf_depay_setcaps (GstRTPBaseDepayload * filter, GstCaps * caps)
{
- GstCaps *srccaps;
+ GstCaps *filtercaps, *srccaps;
GstStructure *structure = gst_caps_get_structure (caps, 0);
- gint clock_rate = 8000; /* default */
+ 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;
+ filtercaps =
+ gst_pad_get_pad_template_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter));
- event =
- gst_event_new_new_segment_full (FALSE, 1.0,
- 1.0, GST_FORMAT_TIME, 0, stop, position);
+ filtercaps = gst_caps_make_writable (filtercaps);
+ gst_caps_set_simple (filtercaps, "rate", G_TYPE_INT, clock_rate, NULL);
- gst_pad_push_event (filter->srcpad, event);
+ srccaps = gst_pad_peer_query_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter),
+ filtercaps);
+ gst_caps_unref (filtercaps);
- 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);
+ gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (filter), srccaps);
+ gst_caps_unref (srccaps);
+ return TRUE;
}
-#endif
-static void
-gst_dtmf_src_generate_tone(GstRtpDTMFDepay *rtpdtmfdepay,
- GstRTPDTMFPayload payload, GstBuffer * buffer)
+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 */;
- GstBaseRTPDepayload * depayload = GST_BASE_RTP_DEPAYLOAD (rtpdtmfdepay);
+ 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;
- 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;
+ 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;
- p = (gint16 *) GST_BUFFER_MALLOCDATA (buffer);
+ gst_buffer_map (buf, &map, GST_MAP_WRITE);
+ p = (gint16 *) map.data;
- volume_factor = pow (10, (-payload.volume) / 20);
+ 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++) {
+ 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));
+ 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;
(rtpdtmfdepay->sample)++;
}
+
+ gst_buffer_unmap (buf, &map);
+
+ return buf;
}
static GstBuffer *
-gst_rtp_dtmf_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+gst_rtp_dtmf_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
{
GstRtpDTMFDepay *rtpdtmfdepay = NULL;
GstBuffer *outbuf = NULL;
- gint payload_len;
+ 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);
- if (!gst_rtp_buffer_validate (buf))
- goto bad_packet;
+ gst_rtp_buffer_map (buf, GST_MAP_READ, &rtpbuffer);
- payload_len = gst_rtp_buffer_get_payload_len (buf);
- payload = gst_rtp_buffer_get_payload (buf);
+ payload_len = gst_rtp_buffer_get_payload_len (&rtpbuffer);
+ payload = gst_rtp_buffer_get_payload (&rtpbuffer);
- if (payload_len != sizeof(GstRTPDTMFPayload) )
+ if (payload_len != 4)
goto bad_packet;
- memcpy (&dtmf_payload, payload, sizeof (GstRTPDTMFPayload));
+ 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;
+ }
+ }
- marker = gst_rtp_buffer_get_marker (buf);
-
- timestamp = gst_rtp_buffer_get_timestamp (buf);
+ /* clip to max duration */
+ if (rtpdtmfdepay->max_duration) {
+ guint max_duration_clock =
+ (rtpdtmfdepay->max_duration * depayload->clock_rate) / 1000;
- dtmf_payload.duration = g_ntohs (dtmf_payload.duration);
+ 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",
+ GST_DEBUG_OBJECT (depayload,
+ "Previous information : timestamp=%u - duration=%d",
rtpdtmfdepay->previous_ts, rtpdtmfdepay->previous_duration);
/* First packet */
rtpdtmfdepay->sample = 0;
rtpdtmfdepay->previous_ts = timestamp;
rtpdtmfdepay->previous_duration = dtmf_payload.duration;
- rtpdtmfdepay->first_gst_ts = GST_BUFFER_TIMESTAMP (buf);
+ 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);
+ "type", G_TYPE_INT, 1, "method", G_TYPE_INT, 1, NULL);
if (structure) {
- dtmf_message = gst_message_new_element (GST_OBJECT (depayload), 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");
+ GST_ERROR_OBJECT (depayload,
+ "Unable to send dtmf-event message to bus");
}
} else {
- GST_DEBUG_OBJECT (depayload, "Unable to create dtmf-event message");
+ GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event message");
}
} else {
- GST_DEBUG_OBJECT (depayload, "Unable to create dtmf-event structure");
+ GST_ERROR_OBJECT (depayload, "Unable to create dtmf-event structure");
}
} else {
guint16 duration = dtmf_payload.duration;
}
GST_DEBUG_OBJECT (depayload, "new previous duration : %d - new duration : %d"
- " - diff : %d - clock rate : %d - timestamp : %llu",
+ " - 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_buffer_new ();
- gst_dtmf_src_generate_tone(rtpdtmfdepay, dtmf_payload, outbuf);
+ outbuf = gst_dtmf_src_generate_tone (rtpdtmfdepay, dtmf_payload);
- GST_BUFFER_TIMESTAMP (outbuf) = rtpdtmfdepay->first_gst_ts +
+ 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_SECOND / depayload->clock_rate;
GST_BUFFER_OFFSET_END (outbuf) = rtpdtmfdepay->previous_duration *
- GST_SECOND / depayload->clock_rate;
+ GST_SECOND / depayload->clock_rate;
- GST_DEBUG_OBJECT (depayload, "timestamp : %llu - time %" GST_TIME_FORMAT,
+ GST_DEBUG_OBJECT (depayload,
+ "timestamp : %" G_GUINT64_FORMAT " - time %" GST_TIME_FORMAT,
GST_BUFFER_TIMESTAMP (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
}
- return outbuf;
+ gst_rtp_buffer_unmap (&rtpbuffer);
+ 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);
-}
+ if (rtpbuffer.buffer != NULL)
+ gst_rtp_buffer_unmap (&rtpbuffer);
+ return NULL;
+}
projects / platform / upstream / gstreamer.git / blobdiff