/* GStreamer RTP DTMF source * * gstrtpdtmfsrc.c: * * 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-rtpdtmfsrc * @short_description: Generates RTP DTMF packets * * * * * The RTPDTMFSrc element generates RTP DTMF (RFC 2833) event 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 named events. * * * * 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 1. * * * * * * * * For example, the following code informs the pipeline (and in turn, the * RTPDTMFSrc element inside the pipeline) about the start of an RTP 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); * * * * */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #include "gstrtpdtmfsrc.h" #define GST_RTP_DTMF_TYPE_EVENT 1 #define DEFAULT_PACKET_INTERVAL 50 /* ms */ #define MIN_PACKET_INTERVAL 10 /* ms */ #define MAX_PACKET_INTERVAL 50 /* ms */ #define DEFAULT_SSRC -1 #define DEFAULT_PT 96 #define DEFAULT_TIMESTAMP_OFFSET -1 #define DEFAULT_SEQNUM_OFFSET -1 #define DEFAULT_CLOCK_RATE 8000 #define MIN_EVENT 0 #define MAX_EVENT 16 #define MIN_EVENT_STRING "0" #define MAX_EVENT_STRING "16" #define MIN_VOLUME 0 #define MAX_VOLUME 36 #define MIN_EVENT_DURATION 50 #define MIN_INTER_DIGIT_INTERVAL 50 #define MIN_PULSE_DURATION 70 #define MIN_DUTY_CYCLE (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION) #define DEFAULT_PACKET_REDUNDANCY 1 #define MIN_PACKET_REDUNDANCY 1 #define MAX_PACKET_REDUNDANCY 5 /* elementfactory information */ static const GstElementDetails gst_rtp_dtmf_src_details = GST_ELEMENT_DETAILS ("RTP DTMF packet generator", "Source/Network", "Generates RTP DTMF packets", "Zeeshan Ali "); GST_DEBUG_CATEGORY_STATIC (gst_rtp_dtmf_src_debug); #define GST_CAT_DEFAULT gst_rtp_dtmf_src_debug /* signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; enum { PROP_0, PROP_SSRC, PROP_TIMESTAMP_OFFSET, PROP_SEQNUM_OFFSET, PROP_PT, PROP_CLOCK_RATE, PROP_TIMESTAMP, PROP_SEQNUM, PROP_INTERVAL, PROP_REDUNDANCY }; static GstStaticPadTemplate gst_rtp_dtmf_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"audio\", " "payload = (int) [ 96, 127 ], " "clock-rate = (int) [ 0, MAX ], " "ssrc = (int) [ 0, MAX ], " "events = (int) [ " MIN_EVENT_STRING ", " MAX_EVENT_STRING " ], " "encoding-name = (string) \"telephone-event\"") ); static GstElementClass *parent_class = NULL; static void gst_rtp_dtmf_src_base_init (gpointer g_class); static void gst_rtp_dtmf_src_class_init (GstRTPDTMFSrcClass * klass); static void gst_rtp_dtmf_src_init (GstRTPDTMFSrc * dtmfsrc, gpointer g_class); static void gst_rtp_dtmf_src_finalize (GObject * object); GType gst_rtp_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 (GstRTPDTMFSrcClass), (GBaseInitFunc) gst_rtp_dtmf_src_base_init, NULL, (GClassInitFunc) gst_rtp_dtmf_src_class_init, NULL, NULL, sizeof (GstRTPDTMFSrc), 0, (GInstanceInitFunc) gst_rtp_dtmf_src_init, }; base_src_type = g_type_register_static (GST_TYPE_ELEMENT, "GstRTPDTMFSrc", &base_src_info, 0); } return base_src_type; } static void gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static gboolean gst_rtp_dtmf_src_handle_event (GstPad * pad, GstEvent * event); static GstStateChangeReturn gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition); static void gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc); static void gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc); static void gst_rtp_dtmf_src_stop (GstRTPDTMFSrc *dtmfsrc); static void gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc *dtmfsrc, gint event_number, gint event_volume); static void gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc *dtmfsrc); static void gst_rtp_dtmf_src_set_caps (GstRTPDTMFSrc *dtmfsrc); static void gst_rtp_dtmf_src_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_src_debug, "rtpdtmfsrc", 0, "rtpdtmfsrc element"); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_rtp_dtmf_src_template)); gst_element_class_set_details (element_class, &gst_rtp_dtmf_src_details); } static void gst_rtp_dtmf_src_class_init (GstRTPDTMFSrcClass * 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_rtp_dtmf_src_finalize); gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_set_property); gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_get_property); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_TIMESTAMP, g_param_spec_uint ("timestamp", "Timestamp", "The RTP timestamp of the last processed packet", 0, G_MAXUINT, 0, G_PARAM_READABLE)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM, g_param_spec_uint ("seqnum", "Sequence number", "The RTP sequence number of the last processed packet", 0, G_MAXUINT, 0, G_PARAM_READABLE)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_TIMESTAMP_OFFSET, g_param_spec_int ("timestamp-offset", "Timestamp Offset", "Offset to add to all outgoing timestamps (-1 = random)", -1, G_MAXINT, DEFAULT_TIMESTAMP_OFFSET, G_PARAM_READWRITE)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SEQNUM_OFFSET, g_param_spec_int ("seqnum-offset", "Sequence number Offset", "Offset to add to all outgoing seqnum (-1 = random)", -1, G_MAXINT, DEFAULT_SEQNUM_OFFSET, G_PARAM_READWRITE)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_CLOCK_RATE, g_param_spec_uint ("clock-rate", "clockrate", "The clock-rate at which to generate the dtmf packets", 0, G_MAXUINT, DEFAULT_CLOCK_RATE, G_PARAM_READWRITE)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_SSRC, g_param_spec_uint ("ssrc", "SSRC", "The SSRC of the packets (-1 == random)", 0, G_MAXUINT, DEFAULT_SSRC, G_PARAM_READWRITE)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_PT, g_param_spec_uint ("pt", "payload type", "The payload type of the packets", 0, 0x80, DEFAULT_PT, G_PARAM_READWRITE)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_INTERVAL, g_param_spec_int ("interval", "Interval between rtp packets", "Interval in ms between two rtp packets", MIN_PACKET_INTERVAL, MAX_PACKET_INTERVAL, DEFAULT_PACKET_INTERVAL, G_PARAM_READWRITE)); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_REDUNDANCY, g_param_spec_int ("packet-redundancy", "Packet Redundancy", "Number of packets to send to indicate start and stop dtmf events", MIN_PACKET_REDUNDANCY, MAX_PACKET_REDUNDANCY, DEFAULT_PACKET_REDUNDANCY, G_PARAM_READWRITE)); gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_change_state); } static void gst_rtp_dtmf_src_init (GstRTPDTMFSrc * dtmfsrc, gpointer g_class) { dtmfsrc->srcpad = gst_pad_new_from_static_template (&gst_rtp_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_rtp_dtmf_src_handle_event); dtmfsrc->ssrc = DEFAULT_SSRC; dtmfsrc->seqnum_offset = DEFAULT_SEQNUM_OFFSET; dtmfsrc->ts_offset = DEFAULT_TIMESTAMP_OFFSET; dtmfsrc->pt = DEFAULT_PT; dtmfsrc->clock_rate = DEFAULT_CLOCK_RATE; dtmfsrc->interval = DEFAULT_PACKET_INTERVAL; dtmfsrc->packet_redundancy = DEFAULT_PACKET_REDUNDANCY; dtmfsrc->event_queue = g_async_queue_new (); dtmfsrc->last_event = NULL; dtmfsrc->clock_id = NULL; GST_DEBUG_OBJECT (dtmfsrc, "init done"); } static void gst_rtp_dtmf_src_finalize (GObject * object) { GstRTPDTMFSrc *dtmfsrc; dtmfsrc = GST_RTP_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_rtp_dtmf_src_handle_dtmf_event (GstRTPDTMFSrc *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) || event_type != GST_RTP_DTMF_TYPE_EVENT) goto failure; if (gst_structure_get_int (event_structure, "method", &method)) { if (method != 1) { 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_rtp_dtmf_src_add_start_event (dtmfsrc, event_number, event_volume); } else { GST_DEBUG_OBJECT (dtmfsrc, "Received stop event"); gst_rtp_dtmf_src_add_stop_event (dtmfsrc); } return TRUE; failure: return FALSE; } static gboolean gst_rtp_dtmf_src_handle_custom_upstream (GstRTPDTMFSrc *dtmfsrc, GstEvent * event) { gboolean result = FALSE; gchar *struct_str; 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); struct_str = gst_structure_to_string (structure); GST_DEBUG_OBJECT (dtmfsrc, "Event has structure %s", struct_str); g_free (struct_str); if (structure && gst_structure_has_name (structure, "dtmf-event")) result = gst_rtp_dtmf_src_handle_dtmf_event (dtmfsrc, structure); ret: return result; } static gboolean gst_rtp_dtmf_src_handle_event (GstPad * pad, GstEvent * event) { GstRTPDTMFSrc *dtmfsrc; gboolean result = FALSE; GstElement *parent = gst_pad_get_parent_element (pad); dtmfsrc = GST_RTP_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_rtp_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_rtp_dtmf_src_stop (dtmfsrc); result = TRUE; break; case GST_EVENT_FLUSH_STOP: gst_segment_init (&dtmfsrc->segment, GST_FORMAT_UNDEFINED); break; default: result = gst_pad_event_default (pad, event); break; } gst_object_unref (parent); gst_event_unref (event); return result; } static void gst_rtp_dtmf_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstRTPDTMFSrc *dtmfsrc; dtmfsrc = GST_RTP_DTMF_SRC (object); switch (prop_id) { case PROP_TIMESTAMP_OFFSET: dtmfsrc->ts_offset = g_value_get_int (value); break; case PROP_SEQNUM_OFFSET: dtmfsrc->seqnum_offset = g_value_get_int (value); break; case PROP_CLOCK_RATE: dtmfsrc->clock_rate = g_value_get_uint (value); gst_rtp_dtmf_src_set_caps (dtmfsrc); break; case PROP_SSRC: dtmfsrc->ssrc = g_value_get_uint (value); break; case PROP_PT: dtmfsrc->pt = g_value_get_uint (value); gst_rtp_dtmf_src_set_caps (dtmfsrc); break; case PROP_INTERVAL: dtmfsrc->interval = g_value_get_int (value); break; case PROP_REDUNDANCY: dtmfsrc->packet_redundancy = g_value_get_int (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtp_dtmf_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRTPDTMFSrc *dtmfsrc; dtmfsrc = GST_RTP_DTMF_SRC (object); switch (prop_id) { case PROP_TIMESTAMP_OFFSET: g_value_set_int (value, dtmfsrc->ts_offset); break; case PROP_SEQNUM_OFFSET: g_value_set_int (value, dtmfsrc->seqnum_offset); break; case PROP_CLOCK_RATE: g_value_set_uint (value, dtmfsrc->clock_rate); break; case PROP_SSRC: g_value_set_uint (value, dtmfsrc->ssrc); break; case PROP_PT: g_value_set_uint (value, dtmfsrc->pt); break; case PROP_TIMESTAMP: g_value_set_uint (value, dtmfsrc->rtp_timestamp); break; case PROP_SEQNUM: g_value_set_uint (value, dtmfsrc->seqnum); break; case PROP_INTERVAL: g_value_set_uint (value, dtmfsrc->interval); break; case PROP_REDUNDANCY: g_value_set_uint (value, dtmfsrc->packet_redundancy); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtp_dtmf_src_set_stream_lock (GstRTPDTMFSrc *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_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc *dtmfsrc) { GstClock *clock; clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc)); if (clock != NULL) { dtmfsrc->timestamp = gst_clock_get_time (clock) + (MIN_INTER_DIGIT_INTERVAL * GST_MSECOND); 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); } dtmfsrc->rtp_timestamp = dtmfsrc->ts_base + gst_util_uint64_scale_int ( gst_segment_to_running_time (&dtmfsrc->segment, GST_FORMAT_TIME, dtmfsrc->timestamp), dtmfsrc->clock_rate, GST_SECOND); } static void gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc) { gst_rtp_dtmf_src_set_caps (dtmfsrc); if (!gst_pad_start_task (dtmfsrc->srcpad, (GstTaskFunction) gst_rtp_dtmf_src_push_next_rtp_packet, dtmfsrc)) { GST_ERROR_OBJECT (dtmfsrc, "Failed to start task on src pad"); } } static void gst_rtp_dtmf_src_stop (GstRTPDTMFSrc *dtmfsrc) { GstRTPDTMFSrcEvent *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; } 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_rtp_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); } if (dtmfsrc->last_event) { g_free (dtmfsrc->last_event); dtmfsrc->last_event = NULL; } } static void gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc *dtmfsrc, gint event_number, gint event_volume) { GstRTPDTMFSrcEvent * event = g_malloc (sizeof(GstRTPDTMFSrcEvent)); event->event_type = RTP_DTMF_EVENT_TYPE_START; event->payload = g_new0 (GstRTPDTMFPayload, 1); event->payload->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT); event->payload->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME); g_async_queue_push (dtmfsrc->event_queue, event); } static void gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc *dtmfsrc) { GstRTPDTMFSrcEvent * event = g_malloc (sizeof(GstRTPDTMFSrcEvent)); event->event_type = RTP_DTMF_EVENT_TYPE_STOP; event->payload = g_new0 (GstRTPDTMFPayload, 1); event->payload->event = 0; event->payload->volume = 0; g_async_queue_push (dtmfsrc->event_queue, event); } static void gst_rtp_dtmf_src_wait_for_buffer_ts (GstRTPDTMFSrc *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); } } 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 void gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc, GstRTPDTMFSrcEvent *event, GstBuffer *buf) { gst_rtp_buffer_set_ssrc (buf, dtmfsrc->current_ssrc); gst_rtp_buffer_set_payload_type (buf, dtmfsrc->pt); if (dtmfsrc->first_packet) { gst_rtp_buffer_set_marker (buf, TRUE); dtmfsrc->first_packet = FALSE; } else if (dtmfsrc->last_packet) { event->payload->e = 1; dtmfsrc->last_packet = FALSE; } dtmfsrc->seqnum++; gst_rtp_buffer_set_seq (buf, dtmfsrc->seqnum); /* timestamp of RTP header */ gst_rtp_buffer_set_timestamp (buf, dtmfsrc->rtp_timestamp); } static void gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, GstRTPDTMFSrcEvent *event,GstBuffer *buf) { GstRTPDTMFPayload *payload; gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc,event, buf); /* duration of DTMF payload */ event->payload->duration += dtmfsrc->interval * dtmfsrc->clock_rate / 1000; /* 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); payload = (GstRTPDTMFPayload *) gst_rtp_buffer_get_payload (buf); /* copy payload and convert to network-byte order */ g_memmove (payload, event->payload, sizeof (GstRTPDTMFPayload)); /* Force the packet duration to a certain minumum * if its the end of the event */ if (payload->e && payload->duration < MIN_EVENT_DURATION * dtmfsrc->clock_rate / 1000) payload->duration = MIN_EVENT_DURATION * dtmfsrc->clock_rate / 1000; payload->duration = g_htons (payload->duration); } static GstBuffer * gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc, GstRTPDTMFSrcEvent *event) { GstBuffer *buf = NULL; /* create buffer to hold the payload */ buf = gst_rtp_buffer_new_allocate (sizeof (GstRTPDTMFPayload), 0, 0); gst_rtp_dtmf_prepare_buffer_data (dtmfsrc, event, buf); /* FIXME: Should we sync to clock ourselves or leave it to sink */ gst_rtp_dtmf_src_wait_for_buffer_ts (dtmfsrc, buf); event->sent_packets++; /* Set caps on the buffer before pushing it */ gst_buffer_set_caps (buf, GST_PAD_CAPS (dtmfsrc->srcpad)); return buf; } static void gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc) { GstBuffer *buf = NULL; GstFlowReturn ret; gint redundancy_count = 1; GstRTPDTMFSrcEvent *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 == RTP_DTMF_EVENT_TYPE_STOP) { GST_WARNING_OBJECT (dtmfsrc, "Received a DTMF stop event when already stopped"); } else if (event->event_type == RTP_DTMF_EVENT_TYPE_START) { dtmfsrc->first_packet = TRUE; dtmfsrc->last_packet = FALSE; gst_rtp_dtmf_prepare_timestamps (dtmfsrc); /* Don't forget to get exclusive access to the stream */ gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, TRUE); event->sent_packets = 0; dtmfsrc->last_event = event; } } else if (dtmfsrc->last_event->sent_packets * dtmfsrc->interval >= MIN_PULSE_DURATION){ event = g_async_queue_try_pop (dtmfsrc->event_queue); if (event != NULL) { if (event->event_type == RTP_DTMF_EVENT_TYPE_START) { GST_WARNING_OBJECT (dtmfsrc, "Received two consecutive DTMF start events"); } else if (event->event_type == RTP_DTMF_EVENT_TYPE_STOP) { dtmfsrc->first_packet = FALSE; dtmfsrc->last_packet = TRUE; } } } g_async_queue_unref (dtmfsrc->event_queue); if (dtmfsrc->last_event) { if (dtmfsrc->first_packet == TRUE || dtmfsrc->last_packet == TRUE) { redundancy_count = dtmfsrc->packet_redundancy; if(dtmfsrc->first_packet == TRUE) { GST_DEBUG_OBJECT (dtmfsrc, "redundancy count set to %d due to dtmf start", redundancy_count); } else if(dtmfsrc->last_packet == TRUE) { GST_DEBUG_OBJECT (dtmfsrc, "redundancy count set to %d due to dtmf stop", redundancy_count); } } /* create buffer to hold the payload */ buf = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc, dtmfsrc->last_event); while ( redundancy_count-- ) { gst_buffer_ref(buf); GST_DEBUG_OBJECT (dtmfsrc, "pushing buffer on src pad of size %d with redundancy count %d", GST_BUFFER_SIZE (buf), redundancy_count); ret = gst_pad_push (dtmfsrc->srcpad, buf); if (ret != GST_FLOW_OK) GST_ERROR_OBJECT (dtmfsrc, "Failed to push buffer on src pad"); /* Make sure only the first packet sent has the marker set */ gst_rtp_buffer_set_marker (buf, FALSE); } gst_buffer_unref(buf); GST_DEBUG_OBJECT (dtmfsrc, "pushed DTMF event '%d' on src pad", dtmfsrc->last_event->payload->event); if (dtmfsrc->last_event->payload->e) { /* Don't forget to release the stream lock */ gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE); g_free (dtmfsrc->last_event->payload); dtmfsrc->last_event->payload = NULL; g_free (dtmfsrc->last_event); dtmfsrc->last_event = NULL; } } } static void gst_rtp_dtmf_src_set_caps (GstRTPDTMFSrc *dtmfsrc) { GstCaps *caps; caps = gst_caps_new_simple ("application/x-rtp", "media", G_TYPE_STRING, "audio", "payload", G_TYPE_INT, dtmfsrc->pt, "clock-rate", G_TYPE_INT, dtmfsrc->clock_rate, "encoding-name", G_TYPE_STRING, "telephone-event", "ssrc", G_TYPE_UINT, dtmfsrc->current_ssrc, "clock-base", G_TYPE_UINT, dtmfsrc->ts_base, "seqnum-base", G_TYPE_UINT, dtmfsrc->seqnum_base, NULL); 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); gst_caps_unref (caps); } static void gst_rtp_dtmf_src_ready_to_paused (GstRTPDTMFSrc *dtmfsrc) { gst_segment_init (&dtmfsrc->segment, GST_FORMAT_UNDEFINED); if (dtmfsrc->ssrc == -1) dtmfsrc->current_ssrc = g_random_int (); else dtmfsrc->current_ssrc = dtmfsrc->ssrc; if (dtmfsrc->seqnum_offset == -1) dtmfsrc->seqnum_base = g_random_int_range (0, G_MAXUINT16); else dtmfsrc->seqnum_base = dtmfsrc->seqnum_offset; dtmfsrc->seqnum = dtmfsrc->seqnum_base; if (dtmfsrc->ts_offset == -1) dtmfsrc->ts_base = g_random_int (); else dtmfsrc->ts_base = dtmfsrc->ts_offset; } static GstStateChangeReturn gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition) { GstRTPDTMFSrc *dtmfsrc; GstStateChangeReturn result; gboolean no_preroll = FALSE; dtmfsrc = GST_RTP_DTMF_SRC (element); switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: gst_rtp_dtmf_src_ready_to_paused (dtmfsrc); /* Indicate that we don't do PRE_ROLL */ no_preroll = TRUE; break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: gst_rtp_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 */ no_preroll = TRUE; gst_rtp_dtmf_src_stop (dtmfsrc); 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_rtp_dtmf_src_plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "rtpdtmfsrc", GST_RANK_NONE, GST_TYPE_RTP_DTMF_SRC); }