#define MIN_VOLUME 0
#define MAX_VOLUME 36
-#define MIN_INTER_DIGIT_INTERVAL (50 * GST_MSECOND)
-#define MIN_PULSE_DURATION (70 * GST_MSECOND)
+#define MIN_INTER_DIGIT_INTERVAL 50 /* ms */
+#define MIN_PULSE_DURATION 70 /* ms */
#define DEFAULT_PACKET_REDUNDANCY 1
#define MIN_PACKET_REDUNDANCY 1
object->packet_redundancy = DEFAULT_PACKET_REDUNDANCY;
object->event_queue = g_async_queue_new ();
- object->last_event = NULL;
+ object->payload = NULL;
GST_DEBUG_OBJECT (object, "init done");
}
"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)) {
+ if (!gst_pad_push_event (GST_BASE_SRC_PAD (dtmfsrc), event)) {
GST_WARNING_OBJECT (dtmfsrc, "stream-lock event not handled");
}
clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
if (clock != NULL) {
dtmfsrc->timestamp = gst_clock_get_time (clock)
- + MIN_INTER_DIGIT_INTERVAL - base_time;
+ + (MIN_INTER_DIGIT_INTERVAL * GST_MSECOND) - base_time;
dtmfsrc->start_timestamp = dtmfsrc->timestamp;
gst_object_unref (clock);
} else {
dtmfsrc->rtp_timestamp = dtmfsrc->ts_base +
gst_util_uint64_scale_int (
- gst_segment_to_running_time (&dtmfsrc->segment, GST_FORMAT_TIME,
- dtmfsrc->timestamp),
+ gst_segment_to_running_time (&GST_BASE_SRC (dtmfsrc)->segment,
+ GST_FORMAT_TIME, dtmfsrc->timestamp),
dtmfsrc->clock_rate, GST_SECOND);
}
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);
+ event->payload->duration = dtmfsrc->interval * dtmfsrc->clock_rate / 1000;
g_async_queue_push (dtmfsrc->event_queue, event);
}
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_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc,
- GstRTPDTMFSrcEvent *event, GstBuffer *buf)
+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);
+ /* Only the very first packet gets a marker */
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->payload->e = 1;
}
dtmfsrc->seqnum++;
}
static void
-gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc,
- GstRTPDTMFSrcEvent *event,GstBuffer *buf)
+gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, 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;
+ gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc, buf);
/* timestamp and duration of GstBuffer */
- GST_BUFFER_DURATION (buf) = dtmfsrc->interval * GST_MSECOND;
+ /* Redundant buffer have no duration ... */
+ if (dtmfsrc->redundancy_count > 1)
+ GST_BUFFER_DURATION (buf) = 0;
+ else
+ 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));
+ 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_PULSE_DURATION * dtmfsrc->clock_rate / GST_MSECOND)
- payload->duration = MIN_PULSE_DURATION * dtmfsrc->clock_rate / GST_MSECOND;
+ payload->duration < MIN_PULSE_DURATION * dtmfsrc->clock_rate / 1000 )
+ payload->duration = MIN_PULSE_DURATION * dtmfsrc->clock_rate / 1000;
payload->duration = g_htons (payload->duration);
+
+
+ /* duration of DTMF payloadfor the NEXT packet */
+ /* not updated for redundant packets */
+ if (dtmfsrc->redundancy_count == 0)
+ dtmfsrc->payload->duration +=
+ dtmfsrc->interval * dtmfsrc->clock_rate / 1000;
+
}
static GstBuffer *
-gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc,
- GstRTPDTMFSrcEvent *event)
+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, event, buf);
+ gst_rtp_dtmf_prepare_buffer_data (dtmfsrc, buf);
/* Set caps on the buffer before pushing it */
- gst_buffer_set_caps (buf, GST_PAD_CAPS (dtmfsrc->srcpad));
+ gst_buffer_set_caps (buf, GST_PAD_CAPS (GST_BASE_SRC_PAD (dtmfsrc)));
return buf;
}
-
static GstFlowReturn
gst_rtp_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
guint length, GstBuffer ** buffer)
do {
- if (dtmfsrc->last_event == NULL) {
+ if (dtmfsrc->payload == NULL) {
+ GST_DEBUG_OBJECT (dtmfsrc, "popping");
event = g_async_queue_pop (dtmfsrc->event_queue);
+ GST_DEBUG_OBJECT (dtmfsrc, "popped %d", event->event_type);
+
switch (event->event_type) {
case RTP_DTMF_EVENT_TYPE_STOP:
GST_WARNING_OBJECT (dtmfsrc,
case RTP_DTMF_EVENT_TYPE_START:
dtmfsrc->first_packet = TRUE;
dtmfsrc->last_packet = FALSE;
+ /* Set the redundanc on the first packet */
+ dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
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);
- dtmfsrc->last_event = event;
+ dtmfsrc->payload = event->payload;
break;
case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
GST_OBJECT_UNLOCK (dtmfsrc);
break;
}
- } else if (dtmfsrc->timestamp - dtmfsrc->start_timestamp >=
+
+ g_free (event);
+ } else if (!dtmfsrc->first_packet && !dtmfsrc->last_packet &&
+ (dtmfsrc->timestamp - dtmfsrc->start_timestamp)/GST_MSECOND >=
MIN_PULSE_DURATION) {
+ GST_DEBUG_OBJECT (dtmfsrc, "try popping");
event = g_async_queue_try_pop (dtmfsrc->event_queue);
+
if (event != NULL) {
+ GST_DEBUG_OBJECT (dtmfsrc, "try popped %d", event->event_type);
+
switch (event->event_type) {
case RTP_DTMF_EVENT_TYPE_START:
GST_WARNING_OBJECT (dtmfsrc,
case RTP_DTMF_EVENT_TYPE_STOP:
dtmfsrc->first_packet = FALSE;
dtmfsrc->last_packet = TRUE;
+ /* Set the redundanc on the last packet */
+ dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
break;
case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
GST_OBJECT_UNLOCK (dtmfsrc);
break;
}
+ g_free (event);
}
}
- } while (dtmfsrc->last_event == NULL);
+ } while (dtmfsrc->payload == NULL);
GST_DEBUG_OBJECT (dtmfsrc, "Processed events, now lets wait on the clock");
clock = gst_element_get_clock (GST_ELEMENT (basesrc));
- clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp -
+ clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
gst_object_unref (clock);
send_last:
- if (dtmfsrc->last_event) {
-
- if (dtmfsrc->first_packet == TRUE || dtmfsrc->last_packet == TRUE) {
- dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
+ if (dtmfsrc->dirty)
+ if (!gst_rtp_dtmf_src_negotiate (basesrc))
+ return GST_FLOW_NOT_NEGOTIATED;
- if(dtmfsrc->first_packet == TRUE) {
- GST_DEBUG_OBJECT (dtmfsrc,
- "redundancy count set to %d due to dtmf start",
- dtmfsrc->redundancy_count);
- } else if(dtmfsrc->last_packet == TRUE) {
- GST_DEBUG_OBJECT (dtmfsrc,
- "redundancy count set to %d due to dtmf stop",
- dtmfsrc->redundancy_count);
- }
- }
+ /* create buffer to hold the payload */
+ *buffer = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc);
+ if (dtmfsrc->redundancy_count)
+ dtmfsrc->redundancy_count--;
- /* create buffer to hold the payload */
- *buffer = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc,
- dtmfsrc->last_event);
+ /* Only the very first one has a marker */
+ dtmfsrc->first_packet = FALSE;
- if (dtmfsrc->last_event->payload->e) {
- /* Don't forget to release the stream lock */
- gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
+ /* This is the end of the event */
+ if (dtmfsrc->last_packet == TRUE && dtmfsrc->redundancy_count == 0) {
- g_free (dtmfsrc->last_event->payload);
- dtmfsrc->last_event->payload = NULL;
+ /* 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;
+ g_free (dtmfsrc->payload);
+ dtmfsrc->payload = NULL;
- }
+ dtmfsrc->last_packet = FALSE;
}
-
return GST_FLOW_OK;
paused_locked:
paused:
- if (dtmfsrc->last_event) {
+ if (dtmfsrc->payload) {
dtmfsrc->first_packet = FALSE;
dtmfsrc->last_packet = TRUE;
+ /* Set the redundanc on the last packet */
+ dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
goto send_last;
+ } else {
+ return GST_FLOW_WRONG_STATE;
}
-
- return GST_FLOW_WRONG_STATE;
}
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
projects / platform / upstream / gstreamer.git / commitdiff