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gstreamer/gst/dtmf/gstrtpdtmfsrc.c
Youness Alaoui 26a8292d83 [MOVED FROM GST-P-FARSIGHT] Ported the event queue work from dtmfsrc to rtpdtmfsrc 
Added a queue based system for the rtpdtmfsrc. Now it waits for start/stop messages on the queue, and makes sure that the minimum duty cycle (120ms) is respected between each
tone, including inter-digit silence.

20070822175533-4f0f6-f27414c406f1f7b00c9a9084a988cf3a7930fe5c.gz
2009-02-21 17:48:00 +01:00

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28 KiB
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/* GStreamer RTP DTMF source
*
* gstrtpdtmfsrc.c:
*
* Copyright (C) <2007> Nokia Corporation.
* Contact: Zeeshan Ali <zeeshan.ali@nokia.com>
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000,2005 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.
*/
/**
* SECTION:element-rtpdtmfsrc
* @short_description: Generates RTP DTMF packets
*
* <refsect2>
*
* <para>
* The RTPDTMFSrc element generates RTP DTMF (RFC 2833) event packets on request
* from application. The application communicates the beginning and end of a
* DTMF event using custom upstream gstreamer events. To report a DTMF event, an
* application must send an event of type GST_EVENT_CUSTOM_UPSTREAM, having a
* structure of name "dtmf-event" with fields set according to the following
* table:
* </para>
*
* <para>
* <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>The application uses this field to specify which of the two methods
* specified in RFC 2833 to use. The value should be 0 for tones and 1 for
* named events. This element is only capable of generating named events.
* </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. Can be omitted if start is set to FALSE.
* </entry>
* </row>
* <row>
* <entry>start</entry>
* <entry>G_TYPE_BOOLEAN</entry>
* <entry>True or False</entry>
* <entry>Whether the event is starting or ending.</entry>
* </row>
* <row>
* <entry>method</entry>
* <entry>G_TYPE_INT</entry>
* <entry>1</entry>
* <entry>The method used for sending event, this element will react if this field
* is absent or 1.
* </entry>
* </row>
* </tbody>
* </tgroup>
* </informaltable>
* </para>
*
* <para>For example, the following code informs the pipeline (and in turn, the
* RTPDTMFSrc element inside the pipeline) about the start of an RTP DTMF named
* event '1' of volume -25 dBm0:
* </para>
*
* <para>
* <programlisting>
* structure = gst_structure_new ("dtmf-event",
* "type", G_TYPE_INT, 1,
* "number", G_TYPE_INT, 1,
* "volume", G_TYPE_INT, 25,
* "start", G_TYPE_BOOLEAN, TRUE, NULL);
*
* event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure);
* gst_element_send_event (pipeline, event);
* </programlisting>
* </para>
*
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <glib.h>
#include "gstrtpdtmfsrc.h"
#define GST_RTP_DTMF_TYPE_EVENT 1
#define DEFAULT_PACKET_INTERVAL 50 /* ms */
#define MIN_PACKET_INTERVAL 10 /* ms */
#define MAX_PACKET_INTERVAL 50 /* ms */
#define DEFAULT_SSRC -1
#define DEFAULT_PT 96
#define DEFAULT_TIMESTAMP_OFFSET -1
#define DEFAULT_SEQNUM_OFFSET -1
#define DEFAULT_CLOCK_RATE 8000
#define MIN_EVENT 0
#define MAX_EVENT 16
#define MIN_EVENT_STRING "0"
#define MAX_EVENT_STRING "16"
#define MIN_VOLUME 0
#define MAX_VOLUME 36
#define MIN_EVENT_DURATION 50
#define MIN_INTER_DIGIT_INTERVAL 50
#define MIN_PULSE_DURATION 70
#define MIN_DUTY_CYCLE (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION)
#define DEFAULT_PACKET_REDUNDANCY 1
#define MIN_PACKET_REDUNDANCY 1
#define MAX_PACKET_REDUNDANCY 5
/* elementfactory information */
static const GstElementDetails gst_rtp_dtmf_src_details =
GST_ELEMENT_DETAILS ("RTP DTMF packet generator",
"Source/Network",
"Generates RTP DTMF packets",
"Zeeshan Ali <zeeshan.ali@nokia.com>");
GST_DEBUG_CATEGORY_STATIC (gst_rtp_dtmf_src_debug);
#define GST_CAT_DEFAULT gst_rtp_dtmf_src_debug
/* signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_SSRC,
PROP_TIMESTAMP_OFFSET,
PROP_SEQNUM_OFFSET,
PROP_PT,
PROP_CLOCK_RATE,
PROP_TIMESTAMP,
PROP_SEQNUM,
PROP_INTERVAL,
PROP_REDUNDANCY
};
static GstStaticPadTemplate gst_rtp_dtmf_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) [ 96, 127 ], "
"clock-rate = (int) [ 0, MAX ], "
"ssrc = (int) [ 0, MAX ], "
"events = (int) [ " MIN_EVENT_STRING ", " MAX_EVENT_STRING " ], "
"encoding-name = (string) \"telephone-event\"")
);
static GstElementClass *parent_class = NULL;
static void gst_rtp_dtmf_src_base_init (gpointer g_class);
static void gst_rtp_dtmf_src_class_init (GstRTPDTMFSrcClass * klass);
static void gst_rtp_dtmf_src_init (GstRTPDTMFSrc * dtmfsrc, gpointer g_class);
static void gst_rtp_dtmf_src_finalize (GObject * object);
GType
gst_rtp_dtmf_src_get_type (void)
{
static GType base_src_type = 0;
if (G_UNLIKELY (base_src_type == 0)) {
static const GTypeInfo base_src_info = {
sizeof (GstRTPDTMFSrcClass),
(GBaseInitFunc) gst_rtp_dtmf_src_base_init,
NULL,
(GClassInitFunc) gst_rtp_dtmf_src_class_init,
NULL,
NULL,
sizeof (GstRTPDTMFSrc),
0,
(GInstanceInitFunc) gst_rtp_dtmf_src_init,
};
base_src_type = g_type_register_static (GST_TYPE_ELEMENT,
"GstRTPDTMFSrc", &base_src_info, 0);
}
return base_src_type;
}
static void gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_rtp_dtmf_src_handle_event (GstPad * pad, GstEvent * event);
static GstStateChangeReturn gst_rtp_dtmf_src_change_state (GstElement * element,
GstStateChange transition);
static void gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc);
static void gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc);
static void gst_rtp_dtmf_src_stop (GstRTPDTMFSrc *dtmfsrc);
static void gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc *dtmfsrc,
gint event_number, gint event_volume);
static void gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc *dtmfsrc);
static void gst_rtp_dtmf_src_set_caps (GstRTPDTMFSrc *dtmfsrc);
static void
gst_rtp_dtmf_src_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_src_debug,
"rtpdtmfsrc", 0, "rtpdtmfsrc element");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_dtmf_src_template));
gst_element_class_set_details (element_class, &gst_rtp_dtmf_src_details);
}
static void
gst_rtp_dtmf_src_class_init (GstRTPDTMFSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = G_OBJECT_CLASS (klass);
gstelement_class = GST_ELEMENT_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_finalize);
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_get_property);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_TIMESTAMP,
g_param_spec_uint ("timestamp", "Timestamp",
"The RTP timestamp of the last processed packet",
0, G_MAXUINT, 0, G_PARAM_READABLE));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM,
g_param_spec_uint ("seqnum", "Sequence number",
"The RTP sequence number of the last processed packet",
0, G_MAXUINT, 0, G_PARAM_READABLE));
g_object_class_install_property (G_OBJECT_CLASS (klass),
PROP_TIMESTAMP_OFFSET, g_param_spec_int ("timestamp-offset",
"Timestamp Offset",
"Offset to add to all outgoing timestamps (-1 = random)", -1,
G_MAXINT, DEFAULT_TIMESTAMP_OFFSET, G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM_OFFSET,
g_param_spec_int ("seqnum-offset", "Sequence number Offset",
"Offset to add to all outgoing seqnum (-1 = random)", -1, G_MAXINT,
DEFAULT_SEQNUM_OFFSET, G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_CLOCK_RATE,
g_param_spec_uint ("clock-rate", "clockrate",
"The clock-rate at which to generate the dtmf packets",
0, G_MAXUINT, DEFAULT_CLOCK_RATE, G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SSRC,
g_param_spec_uint ("ssrc", "SSRC",
"The SSRC of the packets (-1 == random)",
0, G_MAXUINT, DEFAULT_SSRC, G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PT,
g_param_spec_uint ("pt", "payload type",
"The payload type of the packets",
0, 0x80, DEFAULT_PT, G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_INTERVAL,
g_param_spec_int ("interval", "Interval between rtp packets",
"Interval in ms between two rtp packets", MIN_PACKET_INTERVAL,
MAX_PACKET_INTERVAL, DEFAULT_PACKET_INTERVAL, G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_REDUNDANCY,
g_param_spec_int ("packet-redundancy", "Packet Redundancy",
"Number of packets to send to indicate start and stop dtmf events",
MIN_PACKET_REDUNDANCY, MAX_PACKET_REDUNDANCY,
DEFAULT_PACKET_REDUNDANCY, G_PARAM_READWRITE));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_change_state);
}
static void
gst_rtp_dtmf_src_init (GstRTPDTMFSrc * dtmfsrc, gpointer g_class)
{
dtmfsrc->srcpad =
gst_pad_new_from_static_template (&gst_rtp_dtmf_src_template, "src");
GST_DEBUG_OBJECT (dtmfsrc, "adding src pad");
gst_element_add_pad (GST_ELEMENT (dtmfsrc), dtmfsrc->srcpad);
gst_pad_set_event_function (dtmfsrc->srcpad, gst_rtp_dtmf_src_handle_event);
dtmfsrc->ssrc = DEFAULT_SSRC;
dtmfsrc->seqnum_offset = DEFAULT_SEQNUM_OFFSET;
dtmfsrc->ts_offset = DEFAULT_TIMESTAMP_OFFSET;
dtmfsrc->pt = DEFAULT_PT;
dtmfsrc->clock_rate = DEFAULT_CLOCK_RATE;
dtmfsrc->interval = DEFAULT_PACKET_INTERVAL;
dtmfsrc->packet_redundancy = DEFAULT_PACKET_REDUNDANCY;
dtmfsrc->event_queue = g_async_queue_new ();
dtmfsrc->last_event = NULL;
GST_DEBUG_OBJECT (dtmfsrc, "init done");
}
static void
gst_rtp_dtmf_src_finalize (GObject * object)
{
GstRTPDTMFSrc *dtmfsrc;
if (dtmfsrc->event_queue) {
g_async_queue_unref (dtmfsrc->event_queue);
dtmfsrc->event_queue = NULL;
}
dtmfsrc = GST_RTP_DTMF_SRC (object);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_rtp_dtmf_src_handle_dtmf_event (GstRTPDTMFSrc *dtmfsrc,
const GstStructure * event_structure)
{
gint event_type;
gboolean start;
gint method;
if (!gst_structure_get_int (event_structure, "type", &event_type) ||
!gst_structure_get_boolean (event_structure, "start", &start) ||
event_type != GST_RTP_DTMF_TYPE_EVENT)
goto failure;
if (gst_structure_get_int (event_structure, "method", &method)) {
if (method != 1) {
goto failure;
}
}
if (start) {
gint event_number;
gint event_volume;
if (!gst_structure_get_int (event_structure, "number", &event_number) ||
!gst_structure_get_int (event_structure, "volume", &event_volume))
goto failure;
GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d",
event_number, event_volume);
gst_rtp_dtmf_src_add_start_event (dtmfsrc, event_number, event_volume);
}
else {
GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
gst_rtp_dtmf_src_add_stop_event (dtmfsrc);
}
return TRUE;
failure:
return FALSE;
}
static gboolean
gst_rtp_dtmf_src_handle_custom_upstream (GstRTPDTMFSrc *dtmfsrc,
GstEvent * event)
{
gboolean result = FALSE;
gchar *struct_str;
const GstStructure *structure;
if (GST_STATE (dtmfsrc) != GST_STATE_PLAYING) {
GST_DEBUG_OBJECT (dtmfsrc, "Received event while not in PLAYING state");
goto ret;
}
GST_DEBUG_OBJECT (dtmfsrc, "Received event is of our interest");
structure = gst_event_get_structure (event);
struct_str = gst_structure_to_string (structure);
GST_DEBUG_OBJECT (dtmfsrc, "Event has structure %s", struct_str);
g_free (struct_str);
if (structure && gst_structure_has_name (structure, "dtmf-event"))
result = gst_rtp_dtmf_src_handle_dtmf_event (dtmfsrc, structure);
ret:
return result;
}
static gboolean
gst_rtp_dtmf_src_handle_event (GstPad * pad, GstEvent * event)
{
GstRTPDTMFSrc *dtmfsrc;
gboolean result = FALSE;
dtmfsrc = GST_RTP_DTMF_SRC (GST_PAD_PARENT (pad));
GST_DEBUG_OBJECT (dtmfsrc, "Received an event on the src pad");
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CUSTOM_UPSTREAM:
{
result = gst_rtp_dtmf_src_handle_custom_upstream (dtmfsrc, event);
break;
}
/* Ideally this element should not be flushed but let's handle the event
* just in case it is */
case GST_EVENT_FLUSH_START:
gst_rtp_dtmf_src_stop (dtmfsrc);
result = TRUE;
break;
case GST_EVENT_FLUSH_STOP:
gst_segment_init (&dtmfsrc->segment, GST_FORMAT_UNDEFINED);
break;
case GST_EVENT_NEWSEGMENT:
{
gboolean update;
gdouble rate;
GstFormat fmt;
gint64 start, stop, position;
gst_event_parse_new_segment (event, &update, &rate, &fmt, &start,
&stop, &position);
gst_segment_set_newsegment (&dtmfsrc->segment, update, rate, fmt,
start, stop, position);
}
/* fallthrough */
default:
result = gst_pad_event_default (pad, event);
break;
}
gst_event_unref (event);
return result;
}
static void
gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRTPDTMFSrc *dtmfsrc;
dtmfsrc = GST_RTP_DTMF_SRC (object);
switch (prop_id) {
case PROP_TIMESTAMP_OFFSET:
dtmfsrc->ts_offset = g_value_get_int (value);
break;
case PROP_SEQNUM_OFFSET:
dtmfsrc->seqnum_offset = g_value_get_int (value);
break;
case PROP_CLOCK_RATE:
dtmfsrc->clock_rate = g_value_get_uint (value);
gst_rtp_dtmf_src_set_caps (dtmfsrc);
break;
case PROP_SSRC:
dtmfsrc->ssrc = g_value_get_uint (value);
break;
case PROP_PT:
dtmfsrc->pt = g_value_get_uint (value);
gst_rtp_dtmf_src_set_caps (dtmfsrc);
break;
case PROP_INTERVAL:
dtmfsrc->interval = g_value_get_int (value);
break;
case PROP_REDUNDANCY:
dtmfsrc->packet_redundancy = g_value_get_int (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstRTPDTMFSrc *dtmfsrc;
dtmfsrc = GST_RTP_DTMF_SRC (object);
switch (prop_id) {
case PROP_TIMESTAMP_OFFSET:
g_value_set_int (value, dtmfsrc->ts_offset);
break;
case PROP_SEQNUM_OFFSET:
g_value_set_int (value, dtmfsrc->seqnum_offset);
break;
case PROP_CLOCK_RATE:
g_value_set_uint (value, dtmfsrc->clock_rate);
break;
case PROP_SSRC:
g_value_set_uint (value, dtmfsrc->ssrc);
break;
case PROP_PT:
g_value_set_uint (value, dtmfsrc->pt);
break;
case PROP_TIMESTAMP:
g_value_set_uint (value, dtmfsrc->rtp_timestamp);
break;
case PROP_SEQNUM:
g_value_set_uint (value, dtmfsrc->seqnum);
break;
case PROP_INTERVAL:
g_value_set_uint (value, dtmfsrc->interval);
break;
case PROP_REDUNDANCY:
g_value_set_uint (value, dtmfsrc->packet_redundancy);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_dtmf_src_set_stream_lock (GstRTPDTMFSrc *dtmfsrc, gboolean lock)
{
GstEvent *event;
GstStructure *structure;
structure = gst_structure_new ("stream-lock",
"lock", G_TYPE_BOOLEAN, lock, NULL);
event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_OOB, structure);
gst_pad_push_event (dtmfsrc->srcpad, event);
}
static void
gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc *dtmfsrc)
{
GstClock *clock;
clock = GST_ELEMENT_CLOCK (dtmfsrc);
if (clock != NULL)
dtmfsrc->timestamp = gst_clock_get_time (GST_ELEMENT_CLOCK (dtmfsrc))
+ (MIN_INTER_DIGIT_INTERVAL * GST_MSECOND);
else {
GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s",
GST_ELEMENT_NAME (dtmfsrc));
dtmfsrc->timestamp = GST_CLOCK_TIME_NONE;
}
dtmfsrc->rtp_timestamp = dtmfsrc->ts_base +
gst_util_uint64_scale_int (
gst_segment_to_running_time (&dtmfsrc->segment, GST_FORMAT_TIME,
dtmfsrc->timestamp),
dtmfsrc->clock_rate, GST_SECOND);
}
static void
gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc)
{
gst_rtp_dtmf_src_set_caps (dtmfsrc);
if (!gst_pad_start_task (dtmfsrc->srcpad,
(GstTaskFunction) gst_rtp_dtmf_src_push_next_rtp_packet, dtmfsrc)) {
GST_ERROR_OBJECT (dtmfsrc, "Failed to start task on src pad");
}
}
static void
gst_rtp_dtmf_src_stop (GstRTPDTMFSrc *dtmfsrc)
{
/* Don't forget to release the stream lock */
gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
/* Flushing the event queue */
GstRTPDTMFSrcEvent *event = g_async_queue_try_pop (dtmfsrc->event_queue);
while (event != NULL) {
g_free (event);
event = g_async_queue_try_pop (dtmfsrc->event_queue);
}
if (dtmfsrc->last_event) {
g_free (dtmfsrc->last_event);
dtmfsrc->last_event = NULL;
}
if (!gst_pad_pause_task (dtmfsrc->srcpad)) {
GST_ERROR_OBJECT (dtmfsrc, "Failed to pause task on src pad");
return;
}
}
static void
gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc *dtmfsrc, gint event_number,
gint event_volume)
{
GstRTPDTMFSrcEvent * event = g_malloc (sizeof(GstRTPDTMFSrcEvent));
event->event_type = RTP_DTMF_EVENT_TYPE_START;
event->payload = g_new0 (GstRTPDTMFPayload, 1);
event->payload->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
event->payload->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
g_async_queue_push (dtmfsrc->event_queue, event);
}
static void
gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc *dtmfsrc)
{
GstRTPDTMFSrcEvent * event = g_malloc (sizeof(GstRTPDTMFSrcEvent));
event->event_type = RTP_DTMF_EVENT_TYPE_STOP;
event->payload = g_new0 (GstRTPDTMFPayload, 1);
event->payload->event = 0;
event->payload->volume = 0;
g_async_queue_push (dtmfsrc->event_queue, event);
}
static void
gst_rtp_dtmf_src_wait_for_buffer_ts (GstRTPDTMFSrc *dtmfsrc, GstBuffer * buf)
{
GstClock *clock;
clock = GST_ELEMENT_CLOCK (dtmfsrc);
if (clock != NULL) {
GstClockID clock_id;
GstClockReturn clock_ret;
clock_id = gst_clock_new_single_shot_id (clock, GST_BUFFER_TIMESTAMP (buf));
clock_ret = gst_clock_id_wait (clock_id, NULL);
if (clock_ret != GST_CLOCK_OK && clock_ret != GST_CLOCK_EARLY) {
GST_ERROR_OBJECT (dtmfsrc, "Failed to wait on clock %s",
GST_ELEMENT_NAME (clock));
}
gst_clock_id_unref (clock_id);
}
else {
GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s",
GST_ELEMENT_NAME (dtmfsrc));
}
}
static void
gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc, GstRTPDTMFSrcEvent *event, GstBuffer *buf)
{
gst_rtp_buffer_set_ssrc (buf, dtmfsrc->current_ssrc);
gst_rtp_buffer_set_payload_type (buf, dtmfsrc->pt);
if (dtmfsrc->first_packet) {
gst_rtp_buffer_set_marker (buf, TRUE);
dtmfsrc->first_packet = FALSE;
} else if (dtmfsrc->last_packet) {
event->payload->e = 1;
dtmfsrc->last_packet = FALSE;
}
dtmfsrc->seqnum++;
gst_rtp_buffer_set_seq (buf, dtmfsrc->seqnum);
/* timestamp of RTP header */
gst_rtp_buffer_set_timestamp (buf, dtmfsrc->rtp_timestamp);
}
static void
gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, GstRTPDTMFSrcEvent *event,GstBuffer *buf)
{
GstRTPDTMFPayload *payload;
gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc,event, buf);
/* duration of DTMF payload */
event->payload->duration +=
dtmfsrc->interval * dtmfsrc->clock_rate / 1000;
/* timestamp and duration of GstBuffer */
GST_BUFFER_DURATION (buf) = dtmfsrc->interval * GST_MSECOND;
GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp;
dtmfsrc->timestamp += GST_BUFFER_DURATION (buf);
payload = (GstRTPDTMFPayload *) gst_rtp_buffer_get_payload (buf);
/* copy payload and convert to network-byte order */
g_memmove (payload, event->payload, sizeof (GstRTPDTMFPayload));
/* Force the packet duration to a certain minumum
* if its the end of the event
*/
if (payload->e &&
payload->duration < MIN_EVENT_DURATION * dtmfsrc->clock_rate / 1000)
payload->duration = MIN_EVENT_DURATION * dtmfsrc->clock_rate / 1000;
payload->duration = g_htons (payload->duration);
}
static GstBuffer *
gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc, GstRTPDTMFSrcEvent *event)
{
GstBuffer *buf = NULL;
/* create buffer to hold the payload */
buf = gst_rtp_buffer_new_allocate (sizeof (GstRTPDTMFPayload), 0, 0);
gst_rtp_dtmf_prepare_buffer_data (dtmfsrc, event, buf);
/* FIXME: Should we sync to clock ourselves or leave it to sink */
gst_rtp_dtmf_src_wait_for_buffer_ts (dtmfsrc, buf);
event->sent_packets++;
/* Set caps on the buffer before pushing it */
gst_buffer_set_caps (buf, GST_PAD_CAPS (dtmfsrc->srcpad));
return buf;
}
static void
gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
{
GstBuffer *buf = NULL;
GstFlowReturn ret;
gint redundancy_count = 1;
GstRTPDTMFSrcEvent *event;
g_async_queue_ref (dtmfsrc->event_queue);
if (dtmfsrc->last_event == NULL) {
event = g_async_queue_pop (dtmfsrc->event_queue);
if (event->event_type == RTP_DTMF_EVENT_TYPE_STOP) {
GST_WARNING_OBJECT (dtmfsrc, "Received a DTMF stop event when already stopped");
} else if (event->event_type == RTP_DTMF_EVENT_TYPE_START) {
dtmfsrc->first_packet = TRUE;
dtmfsrc->last_packet = FALSE;
gst_rtp_dtmf_prepare_timestamps (dtmfsrc);
/* Don't forget to get exclusive access to the stream */
gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
event->sent_packets = 0;
dtmfsrc->last_event = event;
}
} else if (dtmfsrc->last_event->sent_packets * dtmfsrc->interval >= MIN_PULSE_DURATION){
event = g_async_queue_try_pop (dtmfsrc->event_queue);
if (event != NULL) {
if (event->event_type == RTP_DTMF_EVENT_TYPE_START) {
GST_WARNING_OBJECT (dtmfsrc, "Received two consecutive DTMF start events");
} else if (event->event_type == RTP_DTMF_EVENT_TYPE_STOP) {
dtmfsrc->first_packet = FALSE;
dtmfsrc->last_packet = TRUE;
}
}
}
g_async_queue_unref (dtmfsrc->event_queue);
if (dtmfsrc->last_event) {
if (dtmfsrc->first_packet == TRUE || dtmfsrc->last_packet == TRUE) {
redundancy_count = dtmfsrc->packet_redundancy;
if(dtmfsrc->first_packet == TRUE) {
GST_DEBUG_OBJECT (dtmfsrc,
"redundancy count set to %d due to dtmf start",
redundancy_count);
} else if(dtmfsrc->last_packet == TRUE) {
GST_DEBUG_OBJECT (dtmfsrc,
"redundancy count set to %d due to dtmf stop",
redundancy_count);
}
}
/* create buffer to hold the payload */
buf = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc, dtmfsrc->last_event);
while ( redundancy_count-- ) {
gst_buffer_ref(buf);
GST_DEBUG_OBJECT (dtmfsrc,
"pushing buffer on src pad of size %d with redundancy count %d",
GST_BUFFER_SIZE (buf), redundancy_count);
ret = gst_pad_push (dtmfsrc->srcpad, buf);
if (ret != GST_FLOW_OK)
GST_ERROR_OBJECT (dtmfsrc,
"Failed to push buffer on src pad");
/* Make sure only the first packet sent has the marker set */
gst_rtp_buffer_set_marker (buf, FALSE);
}
gst_buffer_unref(buf);
GST_DEBUG_OBJECT (dtmfsrc,
"pushed DTMF event '%d' on src pad", event->payload->event);
if (dtmfsrc->last_event->payload->e) {
/* Don't forget to release the stream lock */
gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
g_free (dtmfsrc->last_event->payload);
event->payload = NULL;
g_free (dtmfsrc->last_event);
dtmfsrc->last_event = NULL;
}
}
}
static void
gst_rtp_dtmf_src_set_caps (GstRTPDTMFSrc *dtmfsrc)
{
GstCaps *caps;
caps = gst_caps_new_simple ("application/x-rtp",
"media", G_TYPE_STRING, "audio",
"payload", G_TYPE_INT, dtmfsrc->pt,
"clock-rate", G_TYPE_INT, dtmfsrc->clock_rate,
"encoding-name", G_TYPE_STRING, "telephone-event",
"ssrc", G_TYPE_UINT, dtmfsrc->current_ssrc,
"clock-base", G_TYPE_UINT, dtmfsrc->ts_base,
"seqnum-base", G_TYPE_UINT, dtmfsrc->seqnum_base, NULL);
if (!gst_pad_set_caps (dtmfsrc->srcpad, caps))
GST_ERROR_OBJECT (dtmfsrc,
"Failed to set caps %" GST_PTR_FORMAT " on src pad", caps);
else
GST_DEBUG_OBJECT (dtmfsrc,
"caps %" GST_PTR_FORMAT " set on src pad", caps);
gst_caps_unref (caps);
}
static void
gst_rtp_dtmf_src_ready_to_paused (GstRTPDTMFSrc *dtmfsrc)
{
gst_segment_init (&dtmfsrc->segment, GST_FORMAT_UNDEFINED);
if (dtmfsrc->ssrc == -1)
dtmfsrc->current_ssrc = g_random_int ();
else
dtmfsrc->current_ssrc = dtmfsrc->ssrc;
if (dtmfsrc->seqnum_offset == -1)
dtmfsrc->seqnum_base = g_random_int_range (0, G_MAXUINT16);
else
dtmfsrc->seqnum_base = dtmfsrc->seqnum_offset;
dtmfsrc->seqnum = dtmfsrc->seqnum_base;
if (dtmfsrc->ts_offset == -1)
dtmfsrc->ts_base = g_random_int ();
else
dtmfsrc->ts_base = dtmfsrc->ts_offset;
}
static GstStateChangeReturn
gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition)
{
GstRTPDTMFSrc *dtmfsrc;
GstStateChangeReturn result;
gboolean no_preroll = FALSE;
dtmfsrc = GST_RTP_DTMF_SRC (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_rtp_dtmf_src_ready_to_paused (dtmfsrc);
/* Indicate that we don't do PRE_ROLL */
no_preroll = TRUE;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
gst_rtp_dtmf_src_start (dtmfsrc);
break;
default:
break;
}
if ((result =
GST_ELEMENT_CLASS (parent_class)->change_state (element,
transition)) == GST_STATE_CHANGE_FAILURE)
goto failure;
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* Indicate that we don't do PRE_ROLL */
no_preroll = TRUE;
gst_rtp_dtmf_src_stop (dtmfsrc);
break;
default:
break;
}
if (no_preroll && result == GST_STATE_CHANGE_SUCCESS)
result = GST_STATE_CHANGE_NO_PREROLL;
return result;
/* ERRORS */
failure:
{
GST_ERROR_OBJECT (dtmfsrc, "parent failed state change");
return result;
}
}
gboolean
gst_rtp_dtmf_src_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpdtmfsrc",
GST_RANK_NONE, GST_TYPE_RTP_DTMF_SRC);
}
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