diff options
Diffstat (limited to 'gst/rtpdtmf')
| -rw-r--r-- | gst/rtpdtmf/Makefile.am | 9 | ||||
| -rw-r--r-- | gst/rtpdtmf/gstrtpdtmfsrc.c | 872 | ||||
| -rw-r--r-- | gst/rtpdtmf/gstrtpdtmfsrc.h | 101 |
3 files changed, 0 insertions, 982 deletions
diff --git a/gst/rtpdtmf/Makefile.am b/gst/rtpdtmf/Makefile.am deleted file mode 100644 index 603b32c2..00000000 --- a/gst/rtpdtmf/Makefile.am +++ /dev/null @@ -1,9 +0,0 @@ -plugin_LTLIBRARIES = libgstrtpdtmf.la - -libgstrtpdtmf_la_SOURCES = gstrtpdtmfsrc.c - -libgstrtpdtmf_la_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) $(ERROR_CFLAGS) -DEXTERN_BUF -DRTP_SUPPORT -libgstrtpdtmf_la_LIBADD = $(GST_LIBS_LIBS) -libgstrtpdtmf_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstrtp-@GST_MAJORMINOR@ - -noinst_HEADERS = gstrtpdtmfsrc.h diff --git a/gst/rtpdtmf/gstrtpdtmfsrc.c b/gst/rtpdtmf/gstrtpdtmfsrc.c deleted file mode 100644 index 8e4f31b5..00000000 --- a/gst/rtpdtmf/gstrtpdtmfsrc.c +++ /dev/null @@ -1,872 +0,0 @@ -/* GStreamer RTP DTMF source - * - * gstrtpdtmfsrc.c: - * - * 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-rtpdtmfsrc - * @short_description: Generates RTP DTMF packets - * - * <refsect2> - * - * <para> - * 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: - * </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 named events. - * </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 1. - * </entry> - * </row> - * </tbody> - * </tgroup> - * </informaltable> - * </para> - * - * <para>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: - * </para> - * - * <para> - * <programlisting> - * 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); - * </programlisting> - * </para> - * - * </refsect2> - */ - -#ifdef HAVE_CONFIG_H -# include "config.h" -#endif - -#include <stdlib.h> -#include <string.h> - -#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 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 <zeeshan.ali@nokia.com>"); - -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, gint event_number, - gint event_volume); -static void gst_rtp_dtmf_src_stop (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, - "dtmfsrc", 0, "dtmfsrc 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->payload = NULL; - dtmfsrc->interval = DEFAULT_PACKET_INTERVAL; - dtmfsrc->packet_redundancy = DEFAULT_PACKET_REDUNDANCY; - - GST_DEBUG_OBJECT (dtmfsrc, "init done"); -} - -static void -gst_rtp_dtmf_src_finalize (GObject * object) -{ - GstRTPDTMFSrc *dtmfsrc; - - dtmfsrc = GST_RTP_DTMF_SRC (object); - - 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_start (dtmfsrc, event_number, event_volume); - } - - else { - GST_DEBUG_OBJECT (dtmfsrc, "Received stop event"); - gst_rtp_dtmf_src_stop (dtmfsrc); - } - - return TRUE; -failure: - return FALSE; -} - -static gboolean -gst_rtp_dtmf_src_handle_custom_upstream (GstRTPDTMFSrc *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_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; - - dtmfsrc = GST_RTP_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_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; - 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_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); - gst_pad_push_event (dtmfsrc->srcpad, event); -} - -static void -gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc *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; - } - - 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, - gint event_number, gint event_volume) -{ - g_return_if_fail (dtmfsrc->payload == NULL); - - dtmfsrc->payload = g_new0 (GstRTPDTMFPayload, 1); - dtmfsrc->payload->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT); - dtmfsrc->payload->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME); - dtmfsrc->first_packet = TRUE; - dtmfsrc->last_packet = FALSE; - - gst_rtp_dtmf_prepare_timestamps (dtmfsrc); - gst_rtp_dtmf_src_set_caps (dtmfsrc); - - /* Don't forget to get exclusive access to the stream */ - gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, TRUE); - - 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) -{ - g_return_if_fail (dtmfsrc->payload != NULL); - - /* Push the last packet with e-bit set */ - /* Next packet sent will be the last */ - dtmfsrc->last_packet = TRUE; - -} - -static void -gst_rtp_dtmf_src_wait_for_buffer_ts (GstRTPDTMFSrc *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 void -gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc, 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) { - dtmfsrc->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, GstBuffer *buf) -{ - GstRTPDTMFPayload *payload; - - gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc, buf); - - /* duration of DTMF payload */ - dtmfsrc->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, dtmfsrc->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) -{ - 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, buf); - - /* FIXME: Should we sync to clock ourselves or leave it to sink */ - gst_rtp_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_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc) -{ - GstBuffer *buf = NULL; - GstFlowReturn ret; - gint redundancy_count = 1; - - 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); - - 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", GST_BUFFER_SIZE (buf)); - - /* 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->payload->event); - - if (dtmfsrc->payload->e) { - /* Don't forget to release the stream lock */ - gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE); - - g_free (dtmfsrc->payload); - dtmfsrc->payload = NULL; - - if (!gst_pad_pause_task (dtmfsrc->srcpad)) { - GST_ERROR_OBJECT (dtmfsrc, "Failed to pause task on src pad"); - return; - } - - } - -} - -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; - 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; - 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; - } -} - -static gboolean -gst_rtp_dtmf_src_plugin_init (GstPlugin * plugin) -{ - return gst_element_register (plugin, "rtpdtmfsrc", - GST_RANK_NONE, GST_TYPE_RTP_DTMF_SRC); -} - -static gboolean -plugin_init (GstPlugin * plugin) -{ - if (!gst_rtp_dtmf_src_plugin_init (plugin)) - return FALSE; - - return TRUE; -} - -GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, - GST_VERSION_MINOR, - "dtmf", - "DTMF plugins", - plugin_init, "0.1" , "LGPL", "DTMF", ""); diff --git a/gst/rtpdtmf/gstrtpdtmfsrc.h b/gst/rtpdtmf/gstrtpdtmfsrc.h deleted file mode 100644 index 797526ea..00000000 --- a/gst/rtpdtmf/gstrtpdtmfsrc.h +++ /dev/null @@ -1,101 +0,0 @@ -/* GStreamer RTP DTMF source - * - * gstrtpdtmfsrc.h: - * - * Copyright (C) <2007> Nokia Corporation. - * Contact: Zeeshan Ali <zeeshan.ali@nokia.com> - * Copyright (C) <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. - */ - -#ifndef __GST_RTP_DTMF_SRC_H__ -#define __GST_RTP_DTMF_SRC_H__ - -#include <gst/gst.h> -#include <gst/rtp/gstrtpbuffer.h> - -G_BEGIN_DECLS - -#define GST_TYPE_RTP_DTMF_SRC (gst_rtp_dtmf_src_get_type()) -#define GST_RTP_DTMF_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_DTMF_SRC,GstRTPDTMFSrc)) -#define GST_RTP_DTMF_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_DTMF_SRC,GstRTPDTMFSrcClass)) -#define GST_RTP_DTMF_SRC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTP_DTMF_SRC, GstRTPDTMFSrcClass)) -#define GST_IS_RTP_DTMF_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_DTMF_SRC)) -#define GST_IS_RTP_DTMF_SRC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_DTMF_SRC)) -#define GST_RTP_DTMF_SRC_CAST(obj) ((GstRTPDTMFSrc *)(obj)) - -typedef struct { - unsigned event:8; /* Current DTMF event */ -#if G_BYTE_ORDER == G_LITTLE_ENDIAN - unsigned volume:6; /* power level of the tone, in dBm0 */ - unsigned r:1; /* Reserved-bit */ - unsigned e:1; /* End-bit */ -#elif G_BYTE_ORDER == G_BIG_ENDIAN - unsigned e:1; /* End-bit */ - unsigned r:1; /* Reserved-bit */ - unsigned volume:6; /* power level of the tone, in dBm0 */ -#else -#error "G_BYTE_ORDER should be big or little endian." -#endif - unsigned duration:16; /* Duration of digit, in timestamp units */ -} GstRTPDTMFPayload; - -typedef struct _GstRTPDTMFSrc GstRTPDTMFSrc; -typedef struct _GstRTPDTMFSrcClass GstRTPDTMFSrcClass; - -/** - * GstRTPDTMFSrc: - * @element: the parent element. - * - * The opaque #GstRTPDTMFSrc data structure. - */ -struct _GstRTPDTMFSrc { - GstElement element; - - GstPad *srcpad; - GstRTPDTMFPayload *payload; - - guint32 ts_base; - guint16 seqnum_base; - - gint16 seqnum_offset; - guint16 seqnum; - gint32 ts_offset; - guint32 rtp_timestamp; - guint32 clock_rate; - guint pt; - guint ssrc; - guint current_ssrc; - gboolean first_packet; - gboolean last_packet; - - GstClockTime timestamp; - GstSegment segment; - - guint16 interval; - guint16 packet_redundancy; -}; - -struct _GstRTPDTMFSrcClass { - GstElementClass parent_class; -}; - -GType gst_rtp_dtmf_src_get_type (void); - -G_END_DECLS - -#endif /* __GST_RTP_DTMF_SRC_H__ */ |