/* GStreamer DTMF source * * gstdtmfsrc.c: * * Copyright (C) <2007> Collabora. * Contact: Youness Alaoui * Copyright (C) <2007> Nokia Corporation. * Contact: Zeeshan Ali * Copyright (C) 1999,2000 Erik Walthinsen * 2000,2005 Wim Taymans * * 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 * * * * * 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: * * * * * * * * * * * * * Name * GType * Possible values * Purpose * * * * * * type * G_TYPE_INT * 0-1 * 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. * * * * number * G_TYPE_INT * 0-16 * The event number. * * * volume * G_TYPE_INT * 0-36 * This field describes the power level of the tone, expressed in dBm0 * after dropping the sign. Power levels range from 0 to -63 dBm0. The range of * valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE. * * * * start * G_TYPE_BOOLEAN * True or False * Whether the event is starting or ending. * * * method * G_TYPE_INT * 1 * The method used for sending event, this element will react if this field * is absent or 2. * * * * * * * * 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, 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); * * * * */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #include #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 "); 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") ); GST_BOILERPLATE (GstDTMFSrc, gst_dtmf_src, GstBaseSrc, 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 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 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 void gst_dtmf_src_get_times (GstBaseSrc * basesrc, GstBuffer * buffer, GstClockTime * start, GstClockTime * end); 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; 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); 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); gstbasesrc_class->event = GST_DEBUG_FUNCPTR (gst_dtmf_src_handle_event); gstbasesrc_class->get_times = GST_DEBUG_FUNCPTR (gst_dtmf_src_get_times); gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_dtmf_src_create); } static void gst_dtmf_src_init (GstDTMFSrc * dtmfsrc, GstDTMFSrcClass *g_class) { /* 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 (); dtmfsrc->last_event = NULL; dtmfsrc->clock_id = NULL; 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 (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; GstState state; GstStateChangeReturn ret; ret = gst_element_get_state (GST_ELEMENT (dtmfsrc), &state, NULL, 0); if (ret != GST_STATE_CHANGE_SUCCESS || state != 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 (GstBaseSrc * src, GstEvent * event) { GstDTMFSrc *dtmfsrc; gboolean result = FALSE; dtmfsrc = GST_DTMF_SRC (src); GST_DEBUG_OBJECT (dtmfsrc, "Received an event on the src pad"); if (GST_EVENT_TYPE (event) == GST_EVENT_CUSTOM_UPSTREAM) { result = gst_dtmf_src_handle_custom_upstream (dtmfsrc, 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) { GstPad *srcpad = GST_BASE_SRC_PAD (dtmfsrc); 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); if (!gst_pad_push_event (srcpad, event)) { GST_WARNING_OBJECT (dtmfsrc, "stream-lock event not handled"); } } static void gst_dtmf_prepare_timestamps (GstDTMFSrc *dtmfsrc) { GstClock *clock; GstClockTime base_time; base_time = GST_ELEMENT_CAST (dtmfsrc)->base_time; clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc)); if (clock != NULL) { dtmfsrc->timestamp = gst_clock_get_time (clock) - base_time; 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); } } 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_get_times (GstBaseSrc * basesrc, GstBuffer * buffer, GstClockTime * start, GstClockTime * end) { /* for live sources, sync on the timestamp of the buffer */ if (gst_base_src_is_live (basesrc)) { GstClockTime timestamp = GST_BUFFER_TIMESTAMP (buffer); if (GST_CLOCK_TIME_IS_VALID (timestamp)) { /* get duration to calculate end time */ GstClockTime duration = GST_BUFFER_DURATION (buffer); *start = timestamp; if (GST_CLOCK_TIME_IS_VALID (duration)) { *end = *start + duration; } } } else { *start = -1; *end = -1; } } static GstBuffer * gst_dtmf_src_create_next_tone_packet (GstDTMFSrc *dtmfsrc, GstDTMFSrcEvent *event) { GstBuffer *buf = NULL; gboolean send_silence = FALSE; GstPad *srcpad = GST_BASE_SRC_PAD (dtmfsrc); GST_DEBUG_OBJECT (dtmfsrc, "Creating buffer for tone %s", DTMF_KEYS[event->event_number].event_name); /* create buffer to hold the tone */ buf = gst_buffer_new (); if (event->packet_count * dtmfsrc->interval < MIN_INTER_DIGIT_INTERVAL) { send_silence = TRUE; } if (send_silence) { GST_DEBUG_OBJECT (dtmfsrc, "Generating silence"); gst_dtmf_src_generate_silence (buf, dtmfsrc->interval); } else { GST_DEBUG_OBJECT (dtmfsrc, "Generating tone"); gst_dtmf_src_generate_tone(event, DTMF_KEYS[event->event_number], dtmfsrc->interval, buf); } event->packet_count++; /* 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); /* Set caps on the buffer before pushing it */ gst_buffer_set_caps (buf, GST_PAD_CAPS (srcpad)); return buf; } static GstFlowReturn gst_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset, guint length, GstBuffer ** buffer) { GstBuffer *buf = NULL; GstFlowReturn ret; GstDTMFSrcEvent *event; GstDTMFSrc * dtmfsrc; dtmfsrc = GST_DTMF_SRC (basesrc); g_async_queue_ref (dtmfsrc->event_queue); start: 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); 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 (event->event_type == 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..."); g_async_queue_push (dtmfsrc->event_queue, event); g_async_queue_unref (dtmfsrc->event_queue); } } else if (dtmfsrc->last_event->packet_count * dtmfsrc->interval >= MIN_DUTY_CYCLE) { 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; goto start; } else if (event->event_type == 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..."); g_async_queue_push (dtmfsrc->event_queue, event); g_async_queue_unref (dtmfsrc->event_queue); } } } g_async_queue_unref (dtmfsrc->event_queue); GST_DEBUG_OBJECT (dtmfsrc, "end event check"); if (dtmfsrc->last_event) { buf = gst_dtmf_src_create_next_tone_packet (dtmfsrc, dtmfsrc->last_event); GST_DEBUG_OBJECT (dtmfsrc, "Created buffer of size %d", GST_BUFFER_SIZE (buf)); *buffer = buf; ret = GST_FLOW_OK; } else { *buffer = NULL; ret = GST_FLOW_WRONG_STATE; } GST_DEBUG_OBJECT (dtmfsrc, "returning"); return ret; } 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: case GST_STATE_CHANGE_PAUSED_TO_PLAYING: /* Flushing the event queue */ event = g_async_queue_try_pop (dtmfsrc->event_queue); while (event != NULL) { g_free (event); event = g_async_queue_try_pop (dtmfsrc->event_queue); } 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: if (dtmfsrc->last_event) { /* Don't forget to release the stream lock */ gst_dtmf_src_set_stream_lock (dtmfsrc, FALSE); g_free (dtmfsrc->last_event); dtmfsrc->last_event = NULL; } /* Flushing the event queue */ event = g_async_queue_try_pop (dtmfsrc->event_queue); while (event != NULL) { g_free (event); event = g_async_queue_try_pop (dtmfsrc->event_queue); } /* Indicate that we don't do PRE_ROLL */ no_preroll = TRUE; break; case GST_STATE_CHANGE_PAUSED_TO_READY: event = g_malloc (sizeof(GstDTMFSrcEvent)); event->event_type = DTMF_EVENT_TYPE_PAUSE_TASK; g_async_queue_push (dtmfsrc->event_queue, event); event = NULL; 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); }