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gstreamer/gst/dtmf/gstdtmfsrc.c
Thibault Saunier 0a6a62aa76 docs: Port all docstring to gtk-doc markdown
2019-05-13 10:24:40 -04:00

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/* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-dtmfsrc
* @title: dtmfsrc
* @see_also: rtpdtmsrc, rtpdtmfmuxx
*
* 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:
*
* <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. Tones are specified by their frequencies and events are specied
* by their number. This element can only take events as input. Do not confuse
* with "method" which specified the output.
* </entry>
* </row>
* <row>
* <entry>number</entry>
* <entry>G_TYPE_INT</entry>
* <entry>0-15</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>2</entry>
* <entry>The method used for sending event, this element will react if this
* field is absent or 2.
* </entry>
* </row>
* </tbody>
* </tgroup>
* </informaltable>
*
* 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:
*
* |[
* 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);
* ]|
*
* When a DTMF tone actually starts or stop, a "dtmf-event-processed"
* element #GstMessage with the same fields as the "dtmf-event"
* #GstEvent that was used to request the event. Also, if any event
* has not been processed when the element goes from the PAUSED to the
* READY state, then a "dtmf-event-dropped" message is posted on the
* #GstBus in the order that they were received.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include <glib.h>
#include "gstdtmfcommon.h"
#include "gstdtmfsrc.h"
#include <gst/audio/audio.h>
#define GST_TONE_DTMF_TYPE_EVENT 1
#define DEFAULT_PACKET_INTERVAL 50 /* ms */
#define MIN_PACKET_INTERVAL 10 /* ms */
#define MAX_PACKET_INTERVAL 50 /* ms */
#define DEFAULT_SAMPLE_RATE 8000
#define SAMPLE_SIZE 16
#define CHANNELS 1
#define MIN_DUTY_CYCLE (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION)
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_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, "
"format = (string) \"" GST_AUDIO_NE (S16) "\", "
"rate = " GST_AUDIO_RATE_RANGE ", " "channels = (int) 1, "
"layout = (string)interleaved")
);
#define parent_class gst_dtmf_src_parent_class
G_DEFINE_TYPE (GstDTMFSrc, gst_dtmf_src, GST_TYPE_BASE_SRC);
static void gst_dtmf_src_finalize (GObject * object);
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 (GstBaseSrc * src, GstEvent * event);
static gboolean gst_dtmf_src_send_event (GstElement * src, GstEvent * event);
static GstStateChangeReturn gst_dtmf_src_change_state (GstElement * element,
GstStateChange transition);
static GstFlowReturn gst_dtmf_src_create (GstBaseSrc * basesrc,
guint64 offset, guint length, GstBuffer ** buffer);
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 gboolean gst_dtmf_src_unlock (GstBaseSrc * src);
static gboolean gst_dtmf_src_unlock_stop (GstBaseSrc * src);
static gboolean gst_dtmf_src_negotiate (GstBaseSrc * basesrc);
static gboolean gst_dtmf_src_query (GstBaseSrc * basesrc, GstQuery * query);
static void
gst_dtmf_src_class_init (GstDTMFSrcClass * klass)
{
GObjectClass *gobject_class;
GstBaseSrcClass *gstbasesrc_class;
GstElementClass *gstelement_class;
gobject_class = G_OBJECT_CLASS (klass);
gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
gstelement_class = GST_ELEMENT_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (gst_dtmf_src_debug, "dtmfsrc", 0, "dtmfsrc element");
gst_element_class_add_static_pad_template (gstelement_class,
&gst_dtmf_src_template);
gst_element_class_set_static_metadata (gstelement_class,
"DTMF tone generator", "Source/Audio", "Generates DTMF tones",
"Youness Alaoui <youness.alaoui@collabora.co.uk>");
gobject_class->finalize = gst_dtmf_src_finalize;
gobject_class->set_property = gst_dtmf_src_set_property;
gobject_class->get_property = gst_dtmf_src_get_property;
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_INTERVAL,
g_param_spec_uint ("interval", "Interval between tone packets",
"Interval in ms between two tone packets", MIN_PACKET_INTERVAL,
MAX_PACKET_INTERVAL, DEFAULT_PACKET_INTERVAL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_dtmf_src_change_state);
gstelement_class->send_event = GST_DEBUG_FUNCPTR (gst_dtmf_src_send_event);
gstbasesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_dtmf_src_unlock);
gstbasesrc_class->unlock_stop = GST_DEBUG_FUNCPTR (gst_dtmf_src_unlock_stop);
gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_dtmf_src_handle_event);
gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_dtmf_src_create);
gstbasesrc_class->negotiate = GST_DEBUG_FUNCPTR (gst_dtmf_src_negotiate);
gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_dtmf_src_query);
}
static void
event_free (GstDTMFSrcEvent * event)
{
if (event)
g_slice_free (GstDTMFSrcEvent, event);
}
static void
gst_dtmf_src_init (GstDTMFSrc * dtmfsrc)
{
/* we operate in time */
gst_base_src_set_format (GST_BASE_SRC (dtmfsrc), GST_FORMAT_TIME);
gst_base_src_set_live (GST_BASE_SRC (dtmfsrc), TRUE);
dtmfsrc->interval = DEFAULT_PACKET_INTERVAL;
dtmfsrc->event_queue = g_async_queue_new_full ((GDestroyNotify) event_free);
dtmfsrc->last_event = NULL;
dtmfsrc->sample_rate = DEFAULT_SAMPLE_RATE;
GST_DEBUG_OBJECT (dtmfsrc, "init done");
}
static void
gst_dtmf_src_finalize (GObject * object)
{
GstDTMFSrc *dtmfsrc;
dtmfsrc = GST_DTMF_SRC (object);
if (dtmfsrc->event_queue) {
g_async_queue_unref (dtmfsrc->event_queue);
dtmfsrc->event_queue = NULL;
}
G_OBJECT_CLASS (gst_dtmf_src_parent_class)->finalize (object);
}
static gboolean
gst_dtmf_src_handle_dtmf_event (GstDTMFSrc * dtmfsrc, GstEvent * event)
{
const GstStructure *event_structure;
GstStateChangeReturn sret;
GstState state;
gint event_type;
gboolean start;
gint method;
GstClockTime last_stop;
gint event_number;
gint event_volume;
gboolean correct_order;
sret = gst_element_get_state (GST_ELEMENT (dtmfsrc), &state, NULL, 0);
if (sret != GST_STATE_CHANGE_SUCCESS || state != GST_STATE_PLAYING) {
GST_DEBUG_OBJECT (dtmfsrc, "dtmf-event, but not in PLAYING state");
goto failure;
}
event_structure = gst_event_get_structure (event);
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)
if (!gst_structure_get_int (event_structure, "number", &event_number) ||
!gst_structure_get_int (event_structure, "volume", &event_volume))
goto failure;
GST_OBJECT_LOCK (dtmfsrc);
if (gst_structure_get_clock_time (event_structure, "last-stop", &last_stop))
dtmfsrc->last_stop = last_stop;
else
dtmfsrc->last_stop = GST_CLOCK_TIME_NONE;
correct_order = (start != dtmfsrc->last_event_was_start);
dtmfsrc->last_event_was_start = start;
GST_OBJECT_UNLOCK (dtmfsrc);
if (!correct_order)
goto failure;
if (start) {
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_event (GstBaseSrc * src, GstEvent * event)
{
GstDTMFSrc *dtmfsrc;
gboolean result = FALSE;
dtmfsrc = GST_DTMF_SRC (src);
GST_LOG_OBJECT (dtmfsrc, "Received an %s event on the src pad",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CUSTOM_UPSTREAM:
if (gst_event_has_name (event, "dtmf-event")) {
result = gst_dtmf_src_handle_dtmf_event (dtmfsrc, event);
break;
}
/* fall through */
default:
result = GST_BASE_SRC_CLASS (parent_class)->event (src, event);
break;
}
return result;
}
static gboolean
gst_dtmf_src_send_event (GstElement * element, GstEvent * event)
{
GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (element);
gboolean ret;
GST_LOG_OBJECT (dtmfsrc, "Received an %s event via send_event",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CUSTOM_BOTH:
case GST_EVENT_CUSTOM_BOTH_OOB:
case GST_EVENT_CUSTOM_UPSTREAM:
case GST_EVENT_CUSTOM_DOWNSTREAM:
case GST_EVENT_CUSTOM_DOWNSTREAM_OOB:
if (gst_event_has_name (event, "dtmf-event")) {
ret = gst_dtmf_src_handle_dtmf_event (dtmfsrc, event);
break;
}
/* fall through */
default:
ret = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
break;
}
return ret;
}
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_uint (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_prepare_timestamps (GstDTMFSrc * dtmfsrc)
{
GstClockTime last_stop;
GstClockTime timestamp;
GST_OBJECT_LOCK (dtmfsrc);
last_stop = dtmfsrc->last_stop;
GST_OBJECT_UNLOCK (dtmfsrc);
if (GST_CLOCK_TIME_IS_VALID (last_stop)) {
timestamp = last_stop;
} else {
GstClock *clock;
/* If there is no valid start time, lets use now as the start time */
clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
if (clock != NULL) {
timestamp = gst_clock_get_time (clock)
- gst_element_get_base_time (GST_ELEMENT (dtmfsrc));
gst_object_unref (clock);
} else {
gchar *dtmf_name = gst_element_get_name (dtmfsrc);
GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s", dtmf_name);
dtmfsrc->timestamp = GST_CLOCK_TIME_NONE;
g_free (dtmf_name);
return;
}
}
/* Make sure the timestamp always goes forward */
if (timestamp > dtmfsrc->timestamp)
dtmfsrc->timestamp = timestamp;
}
static void
gst_dtmf_src_add_start_event (GstDTMFSrc * dtmfsrc, gint event_number,
gint event_volume)
{
GstDTMFSrcEvent *event = g_slice_new0 (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_slice_new0 (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 GstBuffer *
gst_dtmf_src_generate_silence (float duration, gint sample_rate)
{
gint buf_size;
/* Create a buffer with data set to 0 */
buf_size = ((duration / 1000) * sample_rate * SAMPLE_SIZE * CHANNELS) / 8;
return gst_buffer_new_wrapped (g_malloc0 (buf_size), buf_size);
}
static GstBuffer *
gst_dtmf_src_generate_tone (GstDTMFSrcEvent * event, DTMF_KEY key,
float duration, gint sample_rate)
{
GstBuffer *buffer;
GstMapInfo map;
gint16 *p;
gint tone_size;
double i = 0;
double amplitude, f1, f2;
double volume_factor;
static GstAllocationParams params = { 0, 1, 0, 0, };
/* Create a buffer for the tone */
tone_size = ((duration / 1000) * sample_rate * SAMPLE_SIZE * CHANNELS) / 8;
buffer = gst_buffer_new_allocate (NULL, tone_size, &params);
gst_buffer_map (buffer, &map, GST_MAP_READWRITE);
p = (gint16 *) map.data;
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)++;
}
gst_buffer_unmap (buffer, &map);
return buffer;
}
static GstBuffer *
gst_dtmf_src_create_next_tone_packet (GstDTMFSrc * dtmfsrc,
GstDTMFSrcEvent * event)
{
GstBuffer *buf = NULL;
gboolean send_silence = FALSE;
GST_LOG_OBJECT (dtmfsrc, "Creating buffer for tone %s",
DTMF_KEYS[event->event_number].event_name);
if (event->packet_count * dtmfsrc->interval < MIN_INTER_DIGIT_INTERVAL) {
send_silence = TRUE;
}
if (send_silence) {
GST_LOG_OBJECT (dtmfsrc, "Generating silence");
buf = gst_dtmf_src_generate_silence (dtmfsrc->interval,
dtmfsrc->sample_rate);
} else {
GST_LOG_OBJECT (dtmfsrc, "Generating tone");
buf = gst_dtmf_src_generate_tone (event, DTMF_KEYS[event->event_number],
dtmfsrc->interval, dtmfsrc->sample_rate);
}
event->packet_count++;
/* timestamp and duration of GstBuffer */
GST_BUFFER_DURATION (buf) = dtmfsrc->interval * GST_MSECOND;
GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp;
GST_LOG_OBJECT (dtmfsrc, "Creating new buffer with event %u duration "
" gst: %" GST_TIME_FORMAT " at %" GST_TIME_FORMAT,
event->event_number, GST_TIME_ARGS (GST_BUFFER_DURATION (buf)),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
dtmfsrc->timestamp += GST_BUFFER_DURATION (buf);
return buf;
}
static void
gst_dtmf_src_post_message (GstDTMFSrc * dtmfsrc, const gchar * message_name,
GstDTMFSrcEvent * event)
{
GstStructure *s = NULL;
switch (event->event_type) {
case DTMF_EVENT_TYPE_START:
s = gst_structure_new (message_name,
"type", G_TYPE_INT, 1,
"method", G_TYPE_INT, 2,
"start", G_TYPE_BOOLEAN, TRUE,
"number", G_TYPE_INT, event->event_number,
"volume", G_TYPE_INT, event->volume, NULL);
break;
case DTMF_EVENT_TYPE_STOP:
s = gst_structure_new (message_name,
"type", G_TYPE_INT, 1, "method", G_TYPE_INT, 2,
"start", G_TYPE_BOOLEAN, FALSE, NULL);
break;
case DTMF_EVENT_TYPE_PAUSE_TASK:
return;
}
if (s)
gst_element_post_message (GST_ELEMENT (dtmfsrc),
gst_message_new_element (GST_OBJECT (dtmfsrc), s));
}
static GstFlowReturn
gst_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
guint length, GstBuffer ** buffer)
{
GstBuffer *buf = NULL;
GstDTMFSrcEvent *event;
GstDTMFSrc *dtmfsrc;
GstClock *clock;
GstClockID *clockid;
GstClockReturn clockret;
dtmfsrc = GST_DTMF_SRC (basesrc);
do {
if (dtmfsrc->last_event == NULL) {
GST_DEBUG_OBJECT (dtmfsrc, "popping");
event = g_async_queue_pop (dtmfsrc->event_queue);
GST_DEBUG_OBJECT (dtmfsrc, "popped %d", event->event_type);
switch (event->event_type) {
case DTMF_EVENT_TYPE_STOP:
GST_WARNING_OBJECT (dtmfsrc,
"Received a DTMF stop event when already stopped");
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
break;
case DTMF_EVENT_TYPE_START:
gst_dtmf_prepare_timestamps (dtmfsrc);
event->packet_count = 0;
dtmfsrc->last_event = event;
event = NULL;
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-processed",
dtmfsrc->last_event);
break;
case DTMF_EVENT_TYPE_PAUSE_TASK:
/*
* We're pushing it back because it has to stay in there until
* the task is really paused (and the queue will then be flushed)
*/
GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused) {
g_async_queue_push (dtmfsrc->event_queue, event);
goto paused_locked;
}
GST_OBJECT_UNLOCK (dtmfsrc);
break;
}
if (event)
g_slice_free (GstDTMFSrcEvent, event);
} else if (dtmfsrc->last_event->packet_count * dtmfsrc->interval >=
MIN_DUTY_CYCLE) {
event = g_async_queue_try_pop (dtmfsrc->event_queue);
if (event != NULL) {
switch (event->event_type) {
case DTMF_EVENT_TYPE_START:
GST_WARNING_OBJECT (dtmfsrc,
"Received two consecutive DTMF start events");
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
break;
case DTMF_EVENT_TYPE_STOP:
g_slice_free (GstDTMFSrcEvent, dtmfsrc->last_event);
dtmfsrc->last_event = NULL;
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-processed", event);
break;
case DTMF_EVENT_TYPE_PAUSE_TASK:
/*
* We're pushing it back because it has to stay in there until
* the task is really paused (and the queue will then be flushed)
*/
GST_DEBUG_OBJECT (dtmfsrc, "pushing pause_task...");
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused) {
g_async_queue_push (dtmfsrc->event_queue, event);
goto paused_locked;
}
GST_OBJECT_UNLOCK (dtmfsrc);
break;
}
g_slice_free (GstDTMFSrcEvent, event);
}
}
} while (dtmfsrc->last_event == NULL);
GST_LOG_OBJECT (dtmfsrc, "end event check, now wait for the proper time");
clock = gst_element_get_clock (GST_ELEMENT (basesrc));
clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
gst_object_unref (clock);
GST_OBJECT_LOCK (dtmfsrc);
if (!dtmfsrc->paused) {
dtmfsrc->clockid = clockid;
GST_OBJECT_UNLOCK (dtmfsrc);
clockret = gst_clock_id_wait (clockid, NULL);
GST_OBJECT_LOCK (dtmfsrc);
if (dtmfsrc->paused)
clockret = GST_CLOCK_UNSCHEDULED;
} else {
clockret = GST_CLOCK_UNSCHEDULED;
}
gst_clock_id_unref (clockid);
dtmfsrc->clockid = NULL;
GST_OBJECT_UNLOCK (dtmfsrc);
if (clockret == GST_CLOCK_UNSCHEDULED) {
goto paused;
}
buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc, dtmfsrc->last_event);
GST_LOG_OBJECT (dtmfsrc, "Created buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (buf));
*buffer = buf;
return GST_FLOW_OK;
paused_locked:
GST_OBJECT_UNLOCK (dtmfsrc);
paused:
if (dtmfsrc->last_event) {
GST_DEBUG_OBJECT (dtmfsrc, "Stopping current event");
/* Don't forget to release the stream lock */
g_slice_free (GstDTMFSrcEvent, dtmfsrc->last_event);
dtmfsrc->last_event = NULL;
}
return GST_FLOW_FLUSHING;
}
static gboolean
gst_dtmf_src_unlock (GstBaseSrc * src)
{
GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (src);
GstDTMFSrcEvent *event = NULL;
GST_DEBUG_OBJECT (dtmfsrc, "Called unlock");
GST_OBJECT_LOCK (dtmfsrc);
dtmfsrc->paused = TRUE;
if (dtmfsrc->clockid) {
gst_clock_id_unschedule (dtmfsrc->clockid);
}
GST_OBJECT_UNLOCK (dtmfsrc);
GST_DEBUG_OBJECT (dtmfsrc, "Pushing the PAUSE_TASK event on unlock request");
event = g_slice_new0 (GstDTMFSrcEvent);
event->event_type = DTMF_EVENT_TYPE_PAUSE_TASK;
g_async_queue_push (dtmfsrc->event_queue, event);
return TRUE;
}
static gboolean
gst_dtmf_src_unlock_stop (GstBaseSrc * src)
{
GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (src);
GST_DEBUG_OBJECT (dtmfsrc, "Unlock stopped");
GST_OBJECT_LOCK (dtmfsrc);
dtmfsrc->paused = FALSE;
GST_OBJECT_UNLOCK (dtmfsrc);
return TRUE;
}
static gboolean
gst_dtmf_src_negotiate (GstBaseSrc * basesrc)
{
GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (basesrc);
GstCaps *caps;
GstStructure *s;
gboolean ret;
caps = gst_pad_get_allowed_caps (GST_BASE_SRC_PAD (basesrc));
if (!caps)
caps = gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD (basesrc));
if (gst_caps_is_empty (caps)) {
gst_caps_unref (caps);
return FALSE;
}
caps = gst_caps_truncate (caps);
caps = gst_caps_make_writable (caps);
s = gst_caps_get_structure (caps, 0);
gst_structure_fixate_field_nearest_int (s, "rate", DEFAULT_SAMPLE_RATE);
if (!gst_structure_get_int (s, "rate", &dtmfsrc->sample_rate)) {
GST_ERROR_OBJECT (dtmfsrc, "Could not get rate");
gst_caps_unref (caps);
return FALSE;
}
ret = gst_pad_set_caps (GST_BASE_SRC_PAD (basesrc), caps);
gst_caps_unref (caps);
return ret;
}
static gboolean
gst_dtmf_src_query (GstBaseSrc * basesrc, GstQuery * query)
{
GstDTMFSrc *dtmfsrc = GST_DTMF_SRC (basesrc);
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
GstClockTime latency;
latency = dtmfsrc->interval * GST_MSECOND;
gst_query_set_latency (query, gst_base_src_is_live (basesrc), latency,
GST_CLOCK_TIME_NONE);
GST_DEBUG_OBJECT (dtmfsrc, "Reporting latency of %" GST_TIME_FORMAT,
GST_TIME_ARGS (latency));
res = TRUE;
}
break;
default:
res = GST_BASE_SRC_CLASS (parent_class)->query (basesrc, query);
break;
}
return res;
}
static GstStateChangeReturn
gst_dtmf_src_change_state (GstElement * element, GstStateChange transition)
{
GstDTMFSrc *dtmfsrc;
GstStateChangeReturn result;
gboolean no_preroll = FALSE;
GstDTMFSrcEvent *event = NULL;
dtmfsrc = GST_DTMF_SRC (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
/* Flushing the event queue */
event = g_async_queue_try_pop (dtmfsrc->event_queue);
while (event != NULL) {
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
g_slice_free (GstDTMFSrcEvent, event);
event = g_async_queue_try_pop (dtmfsrc->event_queue);
}
dtmfsrc->last_event_was_start = FALSE;
dtmfsrc->timestamp = 0;
no_preroll = TRUE;
break;
default:
break;
}
if ((result =
GST_ELEMENT_CLASS (gst_dtmf_src_parent_class)->change_state (element,
transition)) == GST_STATE_CHANGE_FAILURE)
goto failure;
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
no_preroll = TRUE;
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG_OBJECT (dtmfsrc, "Flushing event queue");
/* Flushing the event queue */
event = g_async_queue_try_pop (dtmfsrc->event_queue);
while (event != NULL) {
gst_dtmf_src_post_message (dtmfsrc, "dtmf-event-dropped", event);
g_slice_free (GstDTMFSrcEvent, event);
event = g_async_queue_try_pop (dtmfsrc->event_queue);
}
dtmfsrc->last_event_was_start = FALSE;
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);
}
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