Commit a4c5aa38 authored by Tim-Philipp Müller's avatar Tim-Philipp Müller 🐠

Merge branch 'dtmf-moved-from-bad'

https://bugzilla.gnome.org/show_bug.cgi?id=687416
parents c853d8da fba30e33
plugin_LTLIBRARIES = libgstdtmf.la
libgstdtmf_la_SOURCES = gstdtmfsrc.c \
gstrtpdtmfsrc.c \
gstrtpdtmfdepay.c \
gstdtmf.c
noinst_HEADERS = gstdtmfsrc.h \
gstrtpdtmfsrc.h \
gstrtpdtmfdepay.h \
gstdtmfcommon.h
libgstdtmf_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS)
libgstdtmf_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstrtp-@GST_API_VERSION@ \
$(GST_BASE_LIBS) $(GST_LIBS) $(LIBM)
libgstdtmf_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstdtmf_la_LIBTOOLFLAGS = --tag=disable-static
Android.mk: Makefile.am $(BUILT_SOURCES)
androgenizer \
-:PROJECT libgstdtmf -:SHARED libgstdtmf \
-:TAGS eng debug \
-:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
-:SOURCES $(libgstdtmf_la_SOURCES) \
-:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstdtmf_la_CFLAGS) \
-:LDFLAGS $(libgstdtmf_la_LDFLAGS) \
$(libgstdtmf_la_LIBADD) \
-ldl \
-:PASSTHROUGH LOCAL_ARM_MODE:=arm \
LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
> $@
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstdtmfsrc.h"
#include "gstrtpdtmfsrc.h"
#include "gstrtpdtmfdepay.h"
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_dtmf_src_plugin_init (plugin))
return FALSE;
if (!gst_rtp_dtmf_src_plugin_init (plugin))
return FALSE;
if (!gst_rtp_dtmf_depay_plugin_init (plugin))
return FALSE;
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
dtmf, "DTMF plugins",
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
#ifndef __GST_RTP_DTMF_COMMON_H__
#define __GST_RTP_DTMF_COMMON_H__
#define MIN_INTER_DIGIT_INTERVAL 100 /* ms */
#define MIN_PULSE_DURATION 250 /* ms */
#define MIN_VOLUME 0
#define MAX_VOLUME 36
#define MIN_EVENT 0
#define MAX_EVENT 15
#define MIN_EVENT_STRING "0"
#define MAX_EVENT_STRING "15"
#ifndef M_PI
#define M_PI 3.14159265358979323846 /* pi */
#endif
typedef struct
{
unsigned event:8; /* Current DTMF event */
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
unsigned volume:6; /* power level of the tone, in dBm0 */
unsigned r:1; /* Reserved-bit */
unsigned e:1; /* End-bit */
#elif G_BYTE_ORDER == G_BIG_ENDIAN
unsigned e:1; /* End-bit */
unsigned r:1; /* Reserved-bit */
unsigned volume:6; /* power level of the tone, in dBm0 */
#else
#error "G_BYTE_ORDER should be big or little endian."
#endif
unsigned duration:16; /* Duration of digit, in timestamp units */
} GstRTPDTMFPayload;
#endif /* __GST_RTP_DTMF_COMMON_H__ */
This diff is collapsed.
/* GStreamer DTMF source
*
* gstdtmfsrc.h:
*
* Copyright (C) <2007> Collabora.
* Contact: Youness Alaoui <youness.alaoui@collabora.co.uk>
* Copyright (C) <2007> Nokia Corporation.
* Contact: Zeeshan Ali <zeeshan.ali@nokia.com>
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifndef __GST_DTMF_SRC_H__
#define __GST_DTMF_SRC_H__
#include <gst/gst.h>
#include <gst/gstbuffer.h>
#include <gst/base/gstbasesrc.h>
G_BEGIN_DECLS
#define GST_TYPE_DTMF_SRC (gst_dtmf_src_get_type())
#define GST_DTMF_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_DTMF_SRC,GstDTMFSrc))
#define GST_DTMF_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_DTMF_SRC,GstDTMFSrcClass))
#define GST_DTMF_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_DTMF_SRC, GstDTMFSrcClass))
#define GST_IS_DTMF_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_DTMF_SRC))
#define GST_IS_DTMF_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_DTMF_SRC))
#define GST_DTMF_SRC_CAST(obj) ((GstDTMFSrc *)(obj))
typedef struct _GstDTMFSrc GstDTMFSrc;
typedef struct _GstDTMFSrcClass GstDTMFSrcClass;
enum _GstDTMFEventType
{
DTMF_EVENT_TYPE_START,
DTMF_EVENT_TYPE_STOP,
DTMF_EVENT_TYPE_PAUSE_TASK
};
typedef enum _GstDTMFEventType GstDTMFEventType;
struct _GstDTMFSrcEvent
{
GstDTMFEventType event_type;
double sample;
guint16 event_number;
guint16 volume;
guint32 packet_count;
};
typedef struct _GstDTMFSrcEvent GstDTMFSrcEvent;
/**
* GstDTMFSrc:
* @element: the parent element.
*
* The opaque #GstDTMFSrc data structure.
*/
struct _GstDTMFSrc
{
/*< private >*/
GstBaseSrc parent;
GAsyncQueue *event_queue;
GstDTMFSrcEvent *last_event;
gboolean last_event_was_start;
guint16 interval;
GstClockTime timestamp;
gboolean paused;
GstClockID clockid;
GstClockTime last_stop;
gint sample_rate;
};
struct _GstDTMFSrcClass
{
GstBaseSrcClass parent_class;
};
GType gst_dtmf_src_get_type (void);
gboolean gst_dtmf_src_plugin_init (GstPlugin * plugin);
G_END_DECLS
#endif /* __GST_DTMF_SRC_H__ */
/* 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, &params);
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;
<