--- /dev/null
+/* GStreamer RTP DTMF source
+ * Copyright (C) 2007 Zeeshan Ali <zeenix@gstreamer.net>
+ *
+ * gstrtpdtmfsrc.c:
+ *
+ * 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.
+ */
+
+#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 ((guint16) 50) /* ms */
+#define DEFAULT_SSRC -1
+#define DEFAULT_PT 96
+#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
+
+/* 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_PT,
+ PROP_CLOCK_RATE,
+};
+
+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 ], "
+ "event = (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_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));
+
+ 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->pt = DEFAULT_PT;
+ dtmfsrc->clock_rate = DEFAULT_CLOCK_RATE;
+
+ gst_rtp_dtmf_src_set_caps (dtmfsrc);
+
+ 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_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:
+ {
+ const GstStructure *structure;
+
+ if (GST_STATE (dtmfsrc) != GST_STATE_PLAYING) {
+ GST_DEBUG_OBJECT (dtmfsrc, "Received event while not in PLAYING state");
+ break;
+ }
+
+ 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")) {
+ gint event_type;
+ gboolean start;
+
+ if (!gst_structure_get_int (structure, "type", &event_type) ||
+ !gst_structure_get_boolean (structure, "start", &start) ||
+ event_type != GST_RTP_DTMF_TYPE_EVENT)
+ break;
+
+ if (start) {
+ gint event_number;
+ gint event_volume;
+
+ if (!gst_structure_get_int (structure, "number", &event_number) ||
+ !gst_structure_get_int (structure, "volume", &event_volume))
+ break;
+
+ gst_rtp_dtmf_src_start (dtmfsrc, event_number, event_volume);
+ }
+
+ else {
+ gst_rtp_dtmf_src_stop (dtmfsrc);
+ }
+ }
+ break;
+ }
+ default:
+ 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_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;
+ 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_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;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+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->next_ts = gst_clock_get_time (GST_ELEMENT_CLOCK (dtmfsrc));
+ dtmfsrc->first_packet = TRUE;
+
+ if (dtmfsrc->ssrc == -1)
+ dtmfsrc->current_ssrc = g_random_int ();
+ else
+ dtmfsrc->current_ssrc = dtmfsrc->ssrc;
+
+ 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);
+
+ if (!gst_pad_stop_task (dtmfsrc->srcpad)) {
+ GST_ERROR_OBJECT (dtmfsrc, "Failed to stop task on src pad");
+ return;
+ }
+
+ /* Push the last packet with e-bit set */
+ dtmfsrc->payload->e = 1;
+ gst_rtp_dtmf_src_push_next_rtp_packet (dtmfsrc);
+
+ g_free (dtmfsrc->payload);
+ dtmfsrc->payload = NULL;
+}
+
+static void
+gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
+{
+ GstBuffer *buf = NULL;
+ GstFlowReturn ret;
+ GstRTPDTMFPayload *payload;
+ GstClock * clock;
+ guint32 rtp_ts;
+
+ /* create buffer to hold the payload */
+ buf = gst_rtp_buffer_new_allocate (sizeof (GstRTPDTMFPayload), 0, 0);
+
+ 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;
+ }
+
+ /* timestamp and duration of GstBuffer */
+ GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->next_ts;
+ GST_BUFFER_DURATION (buf) = DEFAULT_PACKET_INTERVAL * GST_MSECOND;
+
+ /* duration of DTMF payload */
+ dtmfsrc->payload->duration +=
+ DEFAULT_PACKET_INTERVAL * dtmfsrc->clock_rate / 1000;
+
+ payload = (GstRTPDTMFPayload *) gst_rtp_buffer_get_payload (buf);
+ /* timestamp of RTP header */
+ rtp_ts = dtmfsrc->next_ts * dtmfsrc->clock_rate / GST_SECOND;
+ gst_rtp_buffer_set_timestamp (buf, rtp_ts);
+
+ /* copy payload and convert to network-byte order */
+ g_memmove (payload, dtmfsrc->payload, sizeof (GstRTPDTMFPayload));
+ payload->duration = g_htons (payload->duration);
+
+ /* FIXME: Should we sync to clock ourselves or leave it to sink */
+ clock = GST_ELEMENT_CLOCK (dtmfsrc);
+ if (clock != NULL) {
+ GstClockID clock_id;
+ GstClockReturn clock_ret;
+
+ clock_id = gst_clock_new_single_shot_id (clock, dtmfsrc->next_ts);
+ 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));
+ }
+
+ dtmfsrc->next_ts += GST_BUFFER_DURATION (buf);
+
+ GST_DEBUG_OBJECT (dtmfsrc,
+ "pushing buffer on src pad of size %d", GST_BUFFER_SIZE (buf));
+ ret = gst_pad_push (dtmfsrc->srcpad, buf);
+ if (ret != GST_FLOW_OK)
+ GST_ERROR_OBJECT (dtmfsrc,
+ "Failed to push buffer on src pad", GST_BUFFER_SIZE (buf));
+
+ GST_DEBUG_OBJECT (dtmfsrc,
+ "pushed DTMF event '%d' on src pad", dtmfsrc->payload->event);
+}
+
+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, NULL);
+
+ if (!gst_pad_set_caps (dtmfsrc->srcpad, caps)) {
+ GST_ERROR_OBJECT (dtmfsrc, "Failed to set caps % on src pad",
+ GST_PTR_FORMAT, caps);
+ }
+
+ gst_caps_unref (caps);
+}
+
+static GstStateChangeReturn
+gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition)
+{
+ GstRTPDTMFSrc *dtmfsrc;
+ GstStateChangeReturn result;
+ gboolean no_preroll = TRUE;
+
+ dtmfsrc = GST_RTP_DTMF_SRC (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_NULL_TO_READY:
+ break;
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ /* Indicate that we don't do PRE_ROLL */
+ no_preroll = TRUE;
+ break;
+ case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+ 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;
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ break;
+ case GST_STATE_CHANGE_READY_TO_NULL:
+ 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;
+}
+
+#define PACKAGE "gstreamer"
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+ GST_VERSION_MINOR,
+ "dtmf",
+ "DTMF plugins",
+ plugin_init, "0.1" , "LGPL", "DTMF", "");
projects / platform / upstream / gstreamer.git / commitdiff