*
* 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
- * <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></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>
+ * 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 DEFAULT_PACKET_INTERVAL 50 /* ms */
#define MIN_PACKET_INTERVAL 10 /* 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
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},
+ {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
{
-
-
/* FILL ME */
LAST_SIGNAL
};
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-int, "
- "width = (int) 16, "
- "depth = (int) 16, "
- "endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", "
- "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 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 (GstBaseRTPDepayload * depayload,
+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_base_init (gpointer klass)
+gst_rtp_dtmf_depay_class_init (GstRtpDTMFDepayClass * klass)
{
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
- 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));
+ 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_details_simple (element_class,
+ 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>");
-}
-
-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);
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_set_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));
+ 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_READWRITE | G_PARAM_STATIC_STRINGS));
- gstbasertpdepayload_class->process =
+ gstrtpbasedepayload_class->process =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_depay_process);
- gstbasertpdepayload_class->set_caps =
+ 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;
}
}
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 */
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, G_BYTE_ORDER,
- "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);
+ 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 void
+static GstBuffer *
gst_dtmf_src_generate_tone (GstRtpDTMFDepay * rtpdtmfdepay,
- GstRTPDTMFPayload payload, GstBuffer * buffer)
+ 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;
+ 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, (-volume) / 20);
(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);
- marker = gst_rtp_buffer_get_marker (buf);
-
- timestamp = gst_rtp_buffer_get_timestamp (buf);
-
- dtmf_payload.duration = g_ntohs (dtmf_payload.duration);
+ timestamp = gst_rtp_buffer_get_timestamp (&rtpbuffer);
/* clip to whole units of unit_time */
if (rtpdtmfdepay->unit_time) {
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,
/* 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) =
}
- 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