gstreamer/gst/dtmf/gstdtmfsrc.c
Youness Alaoui d5110d76e9 [MOVED FROM GST-P-FARSIGHT] Changing minimum values to work better on some gateways
20070827172322-4f0f6-5bf2bffa59a8244538dced795fa7d7649452ca91.gz
2009-02-21 17:48:00 +01:00

849 lines
23 KiB
C

/* GStreamer DTMF source
*
* gstdtmfsrc.c:
*
* 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) 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-dtmfsrc
* @short_description: Generates DTMF packets
*
* <refsect2>
*
* <para>
* The DTMFSrc element generates DTMF (ITU-T Q.23 Specification) tone 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 tones.
* </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 2.
* </entry>
* </row>
* </tbody>
* </tgroup>
* </informaltable>
* </para>
*
* <para>For example, the following code informs the pipeline (and in turn, the
* DTMFSrc element inside the pipeline) about the start of a DTMF named
* event '1' of volume -25 dBm0:
* </para>
*
* <para>
* <programlisting>
* structure = gst_structure_new ("dtmf-event",
* "type", G_TYPE_INT, 0,
* "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 <math.h>
#include <glib.h>
#ifndef M_PI
# define M_PI 3.14159265358979323846 /* pi */
#endif
#include "gstdtmfsrc.h"
#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_dtmf_src_details =
GST_ELEMENT_DETAILS ("DTMF tone generator",
"Source/Audio",
"Generates DTMF tones",
"Youness Alaoui <youness.alaoui@collabora.co.uk>");
GST_DEBUG_CATEGORY_STATIC (gst_dtmf_src_debug);
#define GST_CAT_DEFAULT gst_dtmf_src_debug
enum
{
PROP_0,
PROP_INTERVAL,
};
static GstStaticPadTemplate gst_dtmf_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 = (bool) true, "
"rate = (int) 8000, "
"channels = (int) 1")
);
static GstElementClass *parent_class = NULL;
static void gst_dtmf_src_base_init (gpointer g_class);
static void gst_dtmf_src_class_init (GstDTMFSrcClass * klass);
static void gst_dtmf_src_init (GstDTMFSrc * dtmfsrc, gpointer g_class);
static void gst_dtmf_src_finalize (GObject * object);
GType
gst_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 (GstDTMFSrcClass),
(GBaseInitFunc) gst_dtmf_src_base_init,
NULL,
(GClassInitFunc) gst_dtmf_src_class_init,
NULL,
NULL,
sizeof (GstDTMFSrc),
0,
(GInstanceInitFunc) gst_dtmf_src_init,
};
base_src_type = g_type_register_static (GST_TYPE_ELEMENT,
"GstDTMFSrc", &base_src_info, 0);
}
return base_src_type;
}
static void gst_dtmf_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_dtmf_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_dtmf_src_handle_event (GstPad * pad, GstEvent * event);
static GstStateChangeReturn gst_dtmf_src_change_state (GstElement * element,
GstStateChange transition);
static void gst_dtmf_src_generate_tone(GstDTMFSrcEvent *event, DTMF_KEY key, float duration,
GstBuffer * buffer);
static void gst_dtmf_src_push_next_tone_packet (GstDTMFSrc *dtmfsrc);
static void gst_dtmf_src_start (GstDTMFSrc *dtmfsrc);
static void gst_dtmf_src_stop (GstDTMFSrc *dtmfsrc);
static void gst_dtmf_src_add_start_event (GstDTMFSrc *dtmfsrc, gint event_number,
gint event_volume);
static void gst_dtmf_src_add_stop_event (GstDTMFSrc *dtmfsrc);
static void
gst_dtmf_src_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GST_DEBUG_CATEGORY_INIT (gst_dtmf_src_debug,
"dtmfsrc", 0, "dtmfsrc element");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_dtmf_src_template));
gst_element_class_set_details (element_class, &gst_dtmf_src_details);
}
static void
gst_dtmf_src_class_init (GstDTMFSrcClass * 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_dtmf_src_finalize);
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_dtmf_src_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_dtmf_src_get_property);
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_INTERVAL,
g_param_spec_int ("interval", "Interval between tone packets",
"Interval in ms between two tone packets", MIN_PACKET_INTERVAL,
MAX_PACKET_INTERVAL, DEFAULT_PACKET_INTERVAL, G_PARAM_READWRITE));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_dtmf_src_change_state);
}
static void
gst_dtmf_src_init (GstDTMFSrc * dtmfsrc, gpointer g_class)
{
dtmfsrc->srcpad =
gst_pad_new_from_static_template (&gst_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_dtmf_src_handle_event);
dtmfsrc->interval = DEFAULT_PACKET_INTERVAL;
dtmfsrc->event_queue = g_async_queue_new ();
dtmfsrc->last_event = NULL;
GST_DEBUG_OBJECT (dtmfsrc, "init done");
}
static void
gst_dtmf_src_finalize (GObject * object)
{
GstDTMFSrc *dtmfsrc;
dtmfsrc = GST_DTMF_SRC (object);
gst_dtmf_src_stop (dtmfsrc);
if (dtmfsrc->event_queue) {
g_async_queue_unref (dtmfsrc->event_queue);
dtmfsrc->event_queue = NULL;
}
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_dtmf_src_handle_dtmf_event (GstDTMFSrc *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) ||
(start == TRUE && event_type != GST_TONE_DTMF_TYPE_EVENT))
goto failure;
if (gst_structure_get_int (event_structure, "method", &method)) {
if (method != 2) {
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_dtmf_src_add_start_event (dtmfsrc, event_number, event_volume);
}
else {
GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
gst_dtmf_src_add_stop_event (dtmfsrc);
}
return TRUE;
failure:
return FALSE;
}
static gboolean
gst_dtmf_src_handle_custom_upstream (GstDTMFSrc *dtmfsrc,
GstEvent * event)
{
gboolean result = FALSE;
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);
if (structure && gst_structure_has_name (structure, "dtmf-event"))
result = gst_dtmf_src_handle_dtmf_event (dtmfsrc, structure);
ret:
return result;
}
static gboolean
gst_dtmf_src_handle_event (GstPad * pad, GstEvent * event)
{
GstDTMFSrc *dtmfsrc;
gboolean result = FALSE;
dtmfsrc = GST_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_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_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_dtmf_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstDTMFSrc *dtmfsrc;
dtmfsrc = GST_DTMF_SRC (object);
switch (prop_id) {
case PROP_INTERVAL:
dtmfsrc->interval = g_value_get_int (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_dtmf_src_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstDTMFSrc *dtmfsrc;
dtmfsrc = GST_DTMF_SRC (object);
switch (prop_id) {
case PROP_INTERVAL:
g_value_set_uint (value, dtmfsrc->interval);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_dtmf_src_set_stream_lock (GstDTMFSrc *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_dtmf_prepare_timestamps (GstDTMFSrc *dtmfsrc)
{
GstClock *clock;
clock = GST_ELEMENT_CLOCK (dtmfsrc);
if (clock != NULL)
dtmfsrc->timestamp = gst_clock_get_time (GST_ELEMENT_CLOCK (dtmfsrc));
else {
GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s",
GST_ELEMENT_NAME (dtmfsrc));
dtmfsrc->timestamp = GST_CLOCK_TIME_NONE;
}
}
static void
gst_dtmf_src_start (GstDTMFSrc *dtmfsrc)
{
GstCaps * caps = gst_pad_get_pad_template_caps (dtmfsrc->srcpad);
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);
if (!gst_pad_start_task (dtmfsrc->srcpad,
(GstTaskFunction) gst_dtmf_src_push_next_tone_packet, dtmfsrc)) {
GST_ERROR_OBJECT (dtmfsrc, "Failed to start task on src pad");
}
}
static void
gst_dtmf_src_stop (GstDTMFSrc *dtmfsrc)
{
/* Don't forget to release the stream lock */
gst_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
/* Flushing the event queue */
GstDTMFSrcEvent *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_dtmf_src_add_start_event (GstDTMFSrc *dtmfsrc, gint event_number,
gint event_volume)
{
GstDTMFSrcEvent * event = g_malloc (sizeof(GstDTMFSrcEvent));
event->event_type = DTMF_EVENT_TYPE_START;
event->sample = 0;
event->event_number = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
event->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
g_async_queue_push (dtmfsrc->event_queue, event);
}
static void
gst_dtmf_src_add_stop_event (GstDTMFSrc *dtmfsrc)
{
GstDTMFSrcEvent * event = g_malloc (sizeof(GstDTMFSrcEvent));
event->event_type = DTMF_EVENT_TYPE_STOP;
event->sample = 0;
event->event_number = 0;
event->volume = 0;
g_async_queue_push (dtmfsrc->event_queue, event);
}
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);
}
static void
gst_dtmf_src_generate_tone(GstDTMFSrcEvent *event, DTMF_KEY key, float duration, GstBuffer * buffer)
{
gint16 *p;
gint tone_size;
double i = 0;
double amplitude, f1, f2;
double volume_factor;
/* Create a buffer for the tone */
tone_size = ((duration/1000)*SAMPLE_RATE*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);
p = (gint16 *) GST_BUFFER_MALLOCDATA (buffer);
volume_factor = pow (10, (-event->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 * (event->sample / SAMPLE_RATE));
f2 = sin(2 * M_PI * key.high_frequency * (event->sample / SAMPLE_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;
(event->sample)++;
}
}
static void
gst_dtmf_src_wait_for_buffer_ts (GstDTMFSrc *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 GstBuffer *
gst_dtmf_src_create_next_tone_packet (GstDTMFSrc *dtmfsrc, GstDTMFSrcEvent *event)
{
GstBuffer *buf = NULL;
guint32 duration;
GST_DEBUG_OBJECT (dtmfsrc,
"Creating buffer for tone");
/* create buffer to hold the tone */
buf = gst_buffer_new ();
/* The first packet must be inter digit silence, then the second and third must be the
* minimal pulse duration divided into two packets to make it small
*/
switch(event->packet_count) {
case 0:
duration = MIN_INTER_DIGIT_INTERVAL;
gst_dtmf_src_generate_silence (buf, duration);
break;
case 1:
case 2:
/* Generate the tone */
duration = MIN_PULSE_DURATION / 2;
gst_dtmf_src_generate_tone(event, DTMF_KEYS[event->event_number], duration, buf);
break;
default:
duration = dtmfsrc->interval;
gst_dtmf_src_generate_tone(event, DTMF_KEYS[event->event_number], duration, buf);
break;
}
event->packet_count++;
/* timestamp and duration of GstBuffer */
GST_BUFFER_DURATION (buf) = duration * GST_MSECOND;
GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp;
dtmfsrc->timestamp += GST_BUFFER_DURATION (buf);
/* FIXME: Should we sync to clock ourselves or leave it to sink */
gst_dtmf_src_wait_for_buffer_ts (dtmfsrc, buf);
/* Set caps on the buffer before pushing it */
gst_buffer_set_caps (buf, GST_PAD_CAPS (dtmfsrc->srcpad));
return buf;
}
static void
gst_dtmf_src_push_next_tone_packet (GstDTMFSrc *dtmfsrc)
{
GstBuffer *buf = NULL;
GstFlowReturn ret;
GstDTMFSrcEvent *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 == DTMF_EVENT_TYPE_STOP) {
GST_WARNING_OBJECT (dtmfsrc, "Received a DTMF stop event when already stopped");
} else if (event->event_type == DTMF_EVENT_TYPE_START) {
gst_dtmf_prepare_timestamps (dtmfsrc);
/* Don't forget to get exclusive access to the stream */
gst_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
event->packet_count = 0;
dtmfsrc->last_event = event;
}
} else if (dtmfsrc->last_event->packet_count >= 3) {
event = g_async_queue_try_pop (dtmfsrc->event_queue);
if (event != NULL) {
if (event->event_type == DTMF_EVENT_TYPE_START) {
GST_WARNING_OBJECT (dtmfsrc, "Received two consecutive DTMF start events");
} else if (event->event_type == DTMF_EVENT_TYPE_STOP) {
gst_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
g_free (dtmfsrc->last_event);
dtmfsrc->last_event = NULL;
}
}
}
g_async_queue_unref (dtmfsrc->event_queue);
if (dtmfsrc->last_event) {
buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc, dtmfsrc->last_event);
gst_buffer_ref(buf);
GST_DEBUG_OBJECT (dtmfsrc,
"pushing buffer on src pad of size %d", GST_BUFFER_SIZE (buf));
ret = gst_pad_push (dtmfsrc->srcpad, buf);
if (ret != GST_FLOW_OK) {
GST_ERROR_OBJECT (dtmfsrc, "Failed to push buffer on src pad");
}
gst_buffer_unref(buf);
GST_DEBUG_OBJECT (dtmfsrc, "pushed DTMF tone on src pad");
}
}
static GstStateChangeReturn
gst_dtmf_src_change_state (GstElement * element, GstStateChange transition)
{
GstDTMFSrc *dtmfsrc;
GstStateChangeReturn result;
gboolean no_preroll = FALSE;
dtmfsrc = GST_DTMF_SRC (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_segment_init (&dtmfsrc->segment, GST_FORMAT_UNDEFINED);
/* Indicate that we don't do PRE_ROLL */
no_preroll = TRUE;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
gst_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 */
gst_dtmf_src_stop (dtmfsrc);
no_preroll = TRUE;
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_dtmf_src_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "dtmfsrc",
GST_RANK_NONE, GST_TYPE_DTMF_SRC);
}