mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-01-11 09:55:36 +00:00
[MOVED FROM GST-P-FARSIGHT] Complete port to basesrc
20070917003654-3e2dc-db0f84dabd9dd1ac929a0461865b8aaa8ef91a77.gz
This commit is contained in:
Olivier Crete
Edward Hervey
parent
f135a43de2
commit
b55bec0a96
2 changed files with 75 additions and 70 deletions
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@ -147,8 +147,8 @@
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#define MIN_VOLUME 0
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#define MAX_VOLUME 36
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#define MIN_INTER_DIGIT_INTERVAL (50 * GST_MSECOND)
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#define MIN_PULSE_DURATION (70 * GST_MSECOND)
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#define MIN_INTER_DIGIT_INTERVAL 50 /* ms */
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#define MIN_PULSE_DURATION 70 /* ms */
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#define DEFAULT_PACKET_REDUNDANCY 1
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#define MIN_PACKET_REDUNDANCY 1
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@ -330,7 +330,7 @@ gst_rtp_dtmf_src_init (GstRTPDTMFSrc * object, GstRTPDTMFSrcClass * g_class)
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object->packet_redundancy = DEFAULT_PACKET_REDUNDANCY;
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object->event_queue = g_async_queue_new ();
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object->last_event = NULL;
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object->payload = NULL;
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GST_DEBUG_OBJECT (object, "init done");
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}
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@ -528,7 +528,7 @@ gst_rtp_dtmf_src_set_stream_lock (GstRTPDTMFSrc *dtmfsrc, gboolean lock)
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"lock", G_TYPE_BOOLEAN, lock, NULL);
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event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_OOB, structure);
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if (!gst_pad_push_event (dtmfsrc->srcpad, event)) {
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if (!gst_pad_push_event (GST_BASE_SRC_PAD (dtmfsrc), event)) {
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GST_WARNING_OBJECT (dtmfsrc, "stream-lock event not handled");
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}
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@ -545,7 +545,7 @@ gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc *dtmfsrc)
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clock = gst_element_get_clock (GST_ELEMENT (dtmfsrc));
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if (clock != NULL) {
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dtmfsrc->timestamp = gst_clock_get_time (clock)
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+ MIN_INTER_DIGIT_INTERVAL - base_time;
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+ (MIN_INTER_DIGIT_INTERVAL * GST_MSECOND) - base_time;
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dtmfsrc->start_timestamp = dtmfsrc->timestamp;
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gst_object_unref (clock);
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} else {
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@ -557,8 +557,8 @@ gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc *dtmfsrc)
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dtmfsrc->rtp_timestamp = dtmfsrc->ts_base +
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gst_util_uint64_scale_int (
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gst_segment_to_running_time (&dtmfsrc->segment, GST_FORMAT_TIME,
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dtmfsrc->timestamp),
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gst_segment_to_running_time (&GST_BASE_SRC (dtmfsrc)->segment,
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GST_FORMAT_TIME, dtmfsrc->timestamp),
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dtmfsrc->clock_rate, GST_SECOND);
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}
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@ -574,6 +574,7 @@ gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc *dtmfsrc, gint event_number,
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event->payload = g_new0 (GstRTPDTMFPayload, 1);
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event->payload->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
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event->payload->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
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event->payload->duration = dtmfsrc->interval * dtmfsrc->clock_rate / 1000;
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g_async_queue_push (dtmfsrc->event_queue, event);
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}
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@ -584,26 +585,21 @@ gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc *dtmfsrc)
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GstRTPDTMFSrcEvent * event = g_malloc (sizeof(GstRTPDTMFSrcEvent));
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event->event_type = RTP_DTMF_EVENT_TYPE_STOP;
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event->payload = g_new0 (GstRTPDTMFPayload, 1);
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event->payload->event = 0;
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event->payload->volume = 0;
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g_async_queue_push (dtmfsrc->event_queue, event);
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}
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static void
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gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc,
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GstRTPDTMFSrcEvent *event, GstBuffer *buf)
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gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
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{
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gst_rtp_buffer_set_ssrc (buf, dtmfsrc->current_ssrc);
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gst_rtp_buffer_set_payload_type (buf, dtmfsrc->pt);
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/* Only the very first packet gets a marker */
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if (dtmfsrc->first_packet) {
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gst_rtp_buffer_set_marker (buf, TRUE);
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dtmfsrc->first_packet = FALSE;
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} else if (dtmfsrc->last_packet) {
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event->payload->e = 1;
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dtmfsrc->last_packet = FALSE;
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dtmfsrc->payload->e = 1;
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}
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dtmfsrc->seqnum++;
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@ -614,54 +610,60 @@ gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc,
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}
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static void
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gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc,
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GstRTPDTMFSrcEvent *event,GstBuffer *buf)
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gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
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{
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GstRTPDTMFPayload *payload;
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gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc,event, buf);
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/* duration of DTMF payload */
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event->payload->duration +=
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dtmfsrc->interval * dtmfsrc->clock_rate / 1000;
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gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc, buf);
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/* timestamp and duration of GstBuffer */
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GST_BUFFER_DURATION (buf) = dtmfsrc->interval * GST_MSECOND;
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/* Redundant buffer have no duration ... */
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if (dtmfsrc->redundancy_count > 1)
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GST_BUFFER_DURATION (buf) = 0;
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else
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GST_BUFFER_DURATION (buf) = dtmfsrc->interval * GST_MSECOND;
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GST_BUFFER_TIMESTAMP (buf) = dtmfsrc->timestamp;
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dtmfsrc->timestamp += GST_BUFFER_DURATION (buf);
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payload = (GstRTPDTMFPayload *) gst_rtp_buffer_get_payload (buf);
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/* copy payload and convert to network-byte order */
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g_memmove (payload, event->payload, sizeof (GstRTPDTMFPayload));
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g_memmove (payload, dtmfsrc->payload, sizeof (GstRTPDTMFPayload));
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/* Force the packet duration to a certain minumum
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* if its the end of the event
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*/
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if (payload->e &&
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payload->duration < MIN_PULSE_DURATION * dtmfsrc->clock_rate / GST_MSECOND)
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payload->duration = MIN_PULSE_DURATION * dtmfsrc->clock_rate / GST_MSECOND;
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payload->duration < MIN_PULSE_DURATION * dtmfsrc->clock_rate / 1000 )
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payload->duration = MIN_PULSE_DURATION * dtmfsrc->clock_rate / 1000;
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payload->duration = g_htons (payload->duration);
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/* duration of DTMF payloadfor the NEXT packet */
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/* not updated for redundant packets */
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if (dtmfsrc->redundancy_count == 0)
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dtmfsrc->payload->duration +=
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dtmfsrc->interval * dtmfsrc->clock_rate / 1000;
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}
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static GstBuffer *
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gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc,
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GstRTPDTMFSrcEvent *event)
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gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
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{
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GstBuffer *buf = NULL;
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/* create buffer to hold the payload */
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buf = gst_rtp_buffer_new_allocate (sizeof (GstRTPDTMFPayload), 0, 0);
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gst_rtp_dtmf_prepare_buffer_data (dtmfsrc, event, buf);
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gst_rtp_dtmf_prepare_buffer_data (dtmfsrc, buf);
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/* Set caps on the buffer before pushing it */
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gst_buffer_set_caps (buf, GST_PAD_CAPS (dtmfsrc->srcpad));
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gst_buffer_set_caps (buf, GST_PAD_CAPS (GST_BASE_SRC_PAD (dtmfsrc)));
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return buf;
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}
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static GstFlowReturn
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gst_rtp_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
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guint length, GstBuffer ** buffer)
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@ -676,9 +678,12 @@ gst_rtp_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
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do {
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if (dtmfsrc->last_event == NULL) {
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if (dtmfsrc->payload == NULL) {
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GST_DEBUG_OBJECT (dtmfsrc, "popping");
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event = g_async_queue_pop (dtmfsrc->event_queue);
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GST_DEBUG_OBJECT (dtmfsrc, "popped %d", event->event_type);
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switch (event->event_type) {
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case RTP_DTMF_EVENT_TYPE_STOP:
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GST_WARNING_OBJECT (dtmfsrc,
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@ -688,12 +693,14 @@ gst_rtp_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
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case RTP_DTMF_EVENT_TYPE_START:
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dtmfsrc->first_packet = TRUE;
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dtmfsrc->last_packet = FALSE;
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/* Set the redundanc on the first packet */
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dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
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gst_rtp_dtmf_prepare_timestamps (dtmfsrc);
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/* Don't forget to get exclusive access to the stream */
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gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
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dtmfsrc->last_event = event;
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dtmfsrc->payload = event->payload;
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break;
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case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
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@ -709,11 +716,18 @@ gst_rtp_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
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GST_OBJECT_UNLOCK (dtmfsrc);
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break;
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}
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} else if (dtmfsrc->timestamp - dtmfsrc->start_timestamp >=
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g_free (event);
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} else if (!dtmfsrc->first_packet && !dtmfsrc->last_packet &&
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(dtmfsrc->timestamp - dtmfsrc->start_timestamp)/GST_MSECOND >=
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MIN_PULSE_DURATION) {
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GST_DEBUG_OBJECT (dtmfsrc, "try popping");
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event = g_async_queue_try_pop (dtmfsrc->event_queue);
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if (event != NULL) {
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GST_DEBUG_OBJECT (dtmfsrc, "try popped %d", event->event_type);
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switch (event->event_type) {
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case RTP_DTMF_EVENT_TYPE_START:
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GST_WARNING_OBJECT (dtmfsrc,
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@ -723,6 +737,8 @@ gst_rtp_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
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case RTP_DTMF_EVENT_TYPE_STOP:
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dtmfsrc->first_packet = FALSE;
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dtmfsrc->last_packet = TRUE;
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/* Set the redundanc on the last packet */
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dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
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break;
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case RTP_DTMF_EVENT_TYPE_PAUSE_TASK:
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@ -739,16 +755,17 @@ gst_rtp_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
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GST_OBJECT_UNLOCK (dtmfsrc);
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break;
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}
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g_free (event);
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}
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}
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} while (dtmfsrc->last_event == NULL);
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} while (dtmfsrc->payload == NULL);
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GST_DEBUG_OBJECT (dtmfsrc, "Processed events, now lets wait on the clock");
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clock = gst_element_get_clock (GST_ELEMENT (basesrc));
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clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp -
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clockid = gst_clock_new_single_shot_id (clock, dtmfsrc->timestamp +
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gst_element_get_base_time (GST_ELEMENT (dtmfsrc)));
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gst_object_unref (clock);
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@ -775,41 +792,31 @@ gst_rtp_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
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send_last:
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if (dtmfsrc->last_event) {
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if (dtmfsrc->dirty)
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if (!gst_rtp_dtmf_src_negotiate (basesrc))
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return GST_FLOW_NOT_NEGOTIATED;
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if (dtmfsrc->first_packet == TRUE || dtmfsrc->last_packet == TRUE) {
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dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
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/* create buffer to hold the payload */
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*buffer = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc);
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if(dtmfsrc->first_packet == TRUE) {
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GST_DEBUG_OBJECT (dtmfsrc,
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"redundancy count set to %d due to dtmf start",
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dtmfsrc->redundancy_count);
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} else if(dtmfsrc->last_packet == TRUE) {
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GST_DEBUG_OBJECT (dtmfsrc,
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"redundancy count set to %d due to dtmf stop",
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dtmfsrc->redundancy_count);
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}
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}
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if (dtmfsrc->redundancy_count)
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dtmfsrc->redundancy_count--;
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/* Only the very first one has a marker */
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dtmfsrc->first_packet = FALSE;
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/* create buffer to hold the payload */
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*buffer = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc,
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dtmfsrc->last_event);
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/* This is the end of the event */
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if (dtmfsrc->last_packet == TRUE && dtmfsrc->redundancy_count == 0) {
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if (dtmfsrc->last_event->payload->e) {
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/* Don't forget to release the stream lock */
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gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
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/* Don't forget to release the stream lock */
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gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
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g_free (dtmfsrc->last_event->payload);
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dtmfsrc->last_event->payload = NULL;
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g_free (dtmfsrc->payload);
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dtmfsrc->payload = NULL;
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g_free (dtmfsrc->last_event);
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dtmfsrc->last_event = NULL;
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}
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dtmfsrc->last_packet = FALSE;
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}
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return GST_FLOW_OK;
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paused_locked:
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@ -818,13 +825,15 @@ gst_rtp_dtmf_src_create (GstBaseSrc * basesrc, guint64 offset,
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paused:
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if (dtmfsrc->last_event) {
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if (dtmfsrc->payload) {
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dtmfsrc->first_packet = FALSE;
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dtmfsrc->last_packet = TRUE;
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/* Set the redundanc on the last packet */
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dtmfsrc->redundancy_count = dtmfsrc->packet_redundancy;
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goto send_last;
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} else {
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return GST_FLOW_WRONG_STATE;
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}
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return GST_FLOW_WRONG_STATE;
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}
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@ -950,8 +959,6 @@ gst_rtp_dtmf_src_negotiate (GstBaseSrc * basesrc)
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static void
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gst_rtp_dtmf_src_ready_to_paused (GstRTPDTMFSrc *dtmfsrc)
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{
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gst_segment_init (&dtmfsrc->segment, GST_FORMAT_UNDEFINED);
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if (dtmfsrc->ssrc == -1)
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dtmfsrc->current_ssrc = g_random_int ();
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else
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@ -84,12 +84,10 @@ typedef struct _GstRTPDTMFSrcEvent GstRTPDTMFSrcEvent;
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struct _GstRTPDTMFSrc {
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GstBaseSrc basesrc;
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GstPad* srcpad;
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GstSegment segment;
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GAsyncQueue* event_queue;
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GstRTPDTMFSrcEvent* last_event;
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GstClockID clockid;
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gboolean paused;
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GstRTPDTMFPayload* payload;
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GstClockTime timestamp;
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GstClockTime start_timestamp;
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