/* 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") ); 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; 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); 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; 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 (GstPad * pad, GstEvent * event) { GstDTMFSrc *dtmfsrc; gboolean result = FALSE; GstElement *parent = gst_pad_get_parent_element (pad); dtmfsrc = GST_DTMF_SRC (parent); 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_TIME); break; default: result = gst_pad_event_default (pad, event); break; } gst_object_unref (parent); 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); if (!gst_pad_push_event (dtmfsrc->srcpad, event)) { GST_WARNING_OBJECT (dtmfsrc, "stream-lock event not handled"); } } static void gst_dtmf_prepare_timestamps (GstDTMFSrc *dtmfsrc) { GstClock *clock; clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc)); if (clock != NULL) { dtmfsrc->timestamp = gst_clock_get_time (clock); 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_start (GstDTMFSrc *dtmfsrc) { const GstCaps * caps = gst_pad_get_pad_template_caps (dtmfsrc->srcpad); if (!gst_pad_set_caps (dtmfsrc->srcpad, (GstCaps *)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) { GstDTMFSrcEvent *event = NULL; if (dtmfsrc->clock_id != NULL) { gst_clock_id_unschedule(dtmfsrc->clock_id); gst_clock_id_unref (dtmfsrc->clock_id); dtmfsrc->clock_id = NULL; } g_async_queue_lock (dtmfsrc->event_queue); event = g_malloc (sizeof(GstRTPDTMFSrcEvent)); event->event_type = RTP_DTMF_EVENT_TYPE_PAUSE_TASK; g_async_queue_push_unlocked (dtmfsrc->event_queue, event); g_async_queue_unlock (dtmfsrc->event_queue); event = NULL; if (!gst_pad_pause_task (dtmfsrc->srcpad)) { GST_ERROR_OBJECT (dtmfsrc, "Failed to pause task on src pad"); return; } 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); } } 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_get_clock (GST_ELEMENT (dtmfsrc)); if (clock != NULL) { GstClockReturn clock_ret; dtmfsrc->clock_id = gst_clock_new_single_shot_id (clock, GST_BUFFER_TIMESTAMP (buf)); gst_object_unref (clock); clock_ret = gst_clock_id_wait (dtmfsrc->clock_id, NULL); if (clock_ret == GST_CLOCK_UNSCHEDULED) { GST_DEBUG_OBJECT (dtmfsrc, "Clock wait unscheduled"); /* we don't free anything in case of an unscheduled, because it would be unscheduled * by the stop function which will do the free itself. We can't handle it here * in case we stop the task before the unref is done */ } else { if (clock_ret != GST_CLOCK_OK && clock_ret != GST_CLOCK_EARLY) { gchar *clock_name = NULL; clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc)); clock_name = gst_element_get_name (clock); gst_object_unref (clock); GST_ERROR_OBJECT (dtmfsrc, "Failed to wait on clock %s", clock_name); g_free (clock_name); } gst_clock_id_unref (dtmfsrc->clock_id); dtmfsrc->clock_id = NULL; } } else { gchar *dtmf_name = gst_element_get_name (dtmfsrc); GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s", dtmf_name); g_free (dtmf_name); } } static GstBuffer * gst_dtmf_src_create_next_tone_packet (GstDTMFSrc *dtmfsrc, GstDTMFSrcEvent *event) { GstBuffer *buf = NULL; gboolean send_silence = FALSE; 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); /* 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 (event->event_type == RTP_DTMF_EVENT_TYPE_PAUSE_TASK) { g_free (event); g_async_queue_unref (dtmfsrc->event_queue); return; } } 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; } } } 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_TIME); gst_pad_push_event (dtmfsrc->srcpad, gst_event_new_new_segment (FALSE, dtmfsrc->segment.rate, dtmfsrc->segment.format, dtmfsrc->segment.start, dtmfsrc->segment.stop, dtmfsrc->segment.time)); /* 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); }