* <colspec colname='Type' />
* <colspec colname='Possible values' />
* <colspec colname='Purpose' />
- *
+ *
* <thead>
* <row>
* <entry>Name</entry>
* <entry>Purpose</entry>
* </row>
* </thead>
- *
+ *
* <tbody>
* <row>
* <entry>type</entry>
* </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",
* "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>
#include <stdlib.h>
#include <string.h>
+#include <glib.h>
+
#include "gstrtpdtmfsrc.h"
#define GST_RTP_DTMF_TYPE_EVENT 1
#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
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
"payload = (int) [ 96, 127 ], "
- "clock-rate = (int) [ 0, MAX ], "
- "ssrc = (int) [ 0, MAX ], "
+ "clock-rate = (int) [ 0, MAX ], "
+ "ssrc = (int) [ 0, MAX ], "
"events = (int) [ " MIN_EVENT_STRING ", " MAX_EVENT_STRING " ], "
"encoding-name = (string) \"telephone-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_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);
}
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_finalize);
- gobject_class->set_property =
+ 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);
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;
-
+
+
+ dtmfsrc->event_queue = g_async_queue_new ();
+ dtmfsrc->last_event = NULL;
+
GST_DEBUG_OBJECT (dtmfsrc, "init done");
}
{
GstRTPDTMFSrc *dtmfsrc;
+
+ if (dtmfsrc->event_queue) {
+ g_async_queue_unref (dtmfsrc->event_queue);
+ dtmfsrc->event_queue = NULL;
+ }
+
dtmfsrc = GST_RTP_DTMF_SRC (object);
G_OBJECT_CLASS (parent_class)->finalize (object);
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);
+ gst_rtp_dtmf_src_add_start_event (dtmfsrc, event_number, event_volume);
}
else {
GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
- gst_rtp_dtmf_src_stop (dtmfsrc);
+ gst_rtp_dtmf_src_add_stop_event (dtmfsrc);
}
return TRUE;
GstEvent * event)
{
gboolean result = FALSE;
+ gchar *struct_str;
const GstStructure *structure;
if (GST_STATE (dtmfsrc) != GST_STATE_PLAYING) {
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);
clock = GST_ELEMENT_CLOCK (dtmfsrc);
if (clock != NULL)
- dtmfsrc->timestamp = gst_clock_get_time (GST_ELEMENT_CLOCK (dtmfsrc));
+ dtmfsrc->timestamp = gst_clock_get_time (GST_ELEMENT_CLOCK (dtmfsrc))
+ + (MIN_INTER_DIGIT_INTERVAL * GST_MSECOND);
else {
GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s",
}
static void
-gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc,
- gint event_number, gint event_volume)
+gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc)
{
- 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);
+ /* Don't forget to release the stream lock */
+ gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
+
+
+ /* Flushing the event queue */
+ GstRTPDTMFSrcEvent *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;
+ }
+
+ 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_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)
+{
- /* Push the last packet with e-bit set */
- /* Next packet sent will be the last */
- dtmfsrc->last_packet = TRUE;
+ 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_CLOCK (dtmfsrc);
if (clock != NULL) {
GstClockID clock_id;
}
static void
-gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
+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);
gst_rtp_buffer_set_marker (buf, TRUE);
dtmfsrc->first_packet = FALSE;
} else if (dtmfsrc->last_packet) {
- dtmfsrc->payload->e = 1;
+ 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, GstBuffer *buf)
+gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, GstRTPDTMFSrcEvent *event,GstBuffer *buf)
{
GstRTPDTMFPayload *payload;
-
- gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc, buf);
+
+ gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc,event, buf);
/* duration of DTMF payload */
- dtmfsrc->payload->duration +=
+ event->payload->duration +=
dtmfsrc->interval * dtmfsrc->clock_rate / 1000;
- /* timestamp and duration of GstBuffer */
+ /* 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));
+ g_memmove (payload, event->payload, sizeof (GstRTPDTMFPayload));
/* Force the packet duration to a certain minumum
* if its the end of the event
*/
}
static GstBuffer *
-gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
+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, buf);
+ 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));
GstBuffer *buf = NULL;
GstFlowReturn ret;
gint redundancy_count = 1;
+ GstRTPDTMFSrcEvent *event;
- if (dtmfsrc->first_packet == TRUE || dtmfsrc->last_packet == TRUE) {
- redundancy_count = dtmfsrc->packet_redundancy;
+ g_async_queue_ref (dtmfsrc->event_queue);
- 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);
- }
+ 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) {
- /* create buffer to hold the payload */
- buf = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc);
+ dtmfsrc->first_packet = TRUE;
+ dtmfsrc->last_packet = FALSE;
+ gst_rtp_dtmf_prepare_timestamps (dtmfsrc);
- while ( redundancy_count-- ) {
- gst_buffer_ref(buf);
+ /* Don't forget to get exclusive access to the stream */
+ gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
- 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);
+ 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);
- gst_buffer_unref(buf);
- GST_DEBUG_OBJECT (dtmfsrc,
- "pushed DTMF event '%d' on src pad", dtmfsrc->payload->event);
+ if (dtmfsrc->last_event) {
- if (dtmfsrc->payload->e) {
- /* Don't forget to release the stream lock */
- gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
+ if (dtmfsrc->first_packet == TRUE || dtmfsrc->last_packet == TRUE) {
+ redundancy_count = dtmfsrc->packet_redundancy;
- g_free (dtmfsrc->payload);
- dtmfsrc->payload = NULL;
+ 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);
+ }
- if (!gst_pad_pause_task (dtmfsrc->srcpad)) {
- GST_ERROR_OBJECT (dtmfsrc, "Failed to pause task on src pad");
- return;
}
- }
+ /* 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", 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);
+ event->payload = NULL;
+
+ g_free (dtmfsrc->last_event);
+ dtmfsrc->last_event = NULL;
+
+ }
+ }
}
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
else
dtmfsrc->seqnum_base = dtmfsrc->seqnum_offset;
dtmfsrc->seqnum = dtmfsrc->seqnum_base;
-
+
if (dtmfsrc->ts_offset == -1)
dtmfsrc->ts_base = g_random_int ();
else
/* 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;
}
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;
projects / platform / upstream / gst-plugins-good.git / commitdiff