--- /dev/null
+/* 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>
+
+#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
+
+
+
+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(GstDTMFSrc *dtmfsrc, 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, gint event_number,
+ gint event_volume);
+static void gst_dtmf_src_stop (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->sample = 0;
+
+ GST_DEBUG_OBJECT (dtmfsrc, "init done");
+}
+
+static void
+gst_dtmf_src_finalize (GObject * object)
+{
+ GstDTMFSrc *dtmfsrc;
+
+ dtmfsrc = GST_DTMF_SRC (object);
+
+ 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_start (dtmfsrc, event_number, event_volume);
+ }
+
+ else {
+ GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
+ gst_dtmf_src_stop (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,
+ gint event_number, gint event_volume)
+{
+ GstCaps * caps = gst_pad_get_pad_template_caps (dtmfsrc->srcpad);
+
+ dtmfsrc->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
+ dtmfsrc->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
+
+ gst_dtmf_prepare_timestamps (dtmfsrc);
+
+ /* Don't forget to get exclusive access to the stream */
+ gst_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
+
+ 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);
+
+ if (!gst_pad_pause_task (dtmfsrc->srcpad)) {
+ GST_ERROR_OBJECT (dtmfsrc, "Failed to pause task on src pad");
+ return;
+ }
+
+}
+
+static void
+gst_dtmf_src_generate_tone(GstDTMFSrc *dtmfsrc, DTMF_KEY key, float duration, GstBuffer * buffer)
+{
+ gint16 *p;
+ gint tone_size;
+ double i = 0;
+ double amplitude, f1, f2;
+
+ /* 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);
+
+ /*
+ * 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 * (dtmfsrc->sample / SAMPLE_RATE));
+ f2 = sin(2 * M_PI * key.high_frequency * (dtmfsrc->sample / SAMPLE_RATE));
+
+ amplitude = (f1 + f2) / 2;
+
+ /* Make the [-1:1] interval into a [-32767:32767] interval */
+ amplitude *= 32767;
+
+ /* Store it in the data buffer */
+ *(p++) = (gint16) amplitude;
+
+ (dtmfsrc->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)
+{
+ GstBuffer *buf = NULL;
+
+
+ GST_DEBUG_OBJECT (dtmfsrc,
+ "Creating buffer for tone");
+
+ /* create buffer to hold the tone */
+ buf = gst_buffer_new ();
+
+ /* Generate the tone */
+ gst_dtmf_src_generate_tone(dtmfsrc, DTMF_KEYS[dtmfsrc->event], dtmfsrc->interval, buf);
+
+
+ /* 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) /2;
+
+ /* 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;
+
+ buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc);
+
+ 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_SIZE (buf));
+ }
+
+ 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, 6, 30); */
+ 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);
+}
projects / platform / upstream / gstreamer.git / commitdiff