mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-25 09:40:37 +00:00
[MOVED FROM GST-P-FARSIGHT] Clean-up and refactorize dtmfsrc code
20070402124635-65035-3d13244461c1dd1fcc96b74124ad7a74d2ff0144.gz
This commit is contained in:
zeeshan.ali@nokia.com
Edward Hervey
parent
f7d6d695aa
commit
50dbdcc4e1
1 changed files with 161 additions and 115 deletions
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@ -315,6 +315,61 @@ gst_rtp_dtmf_src_finalize (GObject * object)
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_rtp_dtmf_src_handle_dtmf_event (GstRTPDTMFSrc *dtmfsrc,
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const GstStructure * event_structure)
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{
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gint event_type;
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gboolean start;
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if (!gst_structure_get_int (event_structure, "type", &event_type) ||
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!gst_structure_get_boolean (event_structure, "start", &start) ||
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event_type != GST_RTP_DTMF_TYPE_EVENT)
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goto failure;
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if (start) {
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gint event_number;
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gint event_volume;
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if (!gst_structure_get_int (event_structure, "number", &event_number) ||
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!gst_structure_get_int (event_structure, "volume", &event_volume))
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goto failure;
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GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d",
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event_number, event_volume);
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gst_rtp_dtmf_src_start (dtmfsrc, event_number, event_volume);
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}
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else {
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GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
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gst_rtp_dtmf_src_stop (dtmfsrc);
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}
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return TRUE;
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failure:
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return FALSE;
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}
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static gboolean
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gst_rtp_dtmf_src_handle_custom_upstream (GstRTPDTMFSrc *dtmfsrc, GstEvent * event)
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{
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gboolean result = FALSE;
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const GstStructure *structure;
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if (GST_STATE (dtmfsrc) != GST_STATE_PLAYING) {
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GST_DEBUG_OBJECT (dtmfsrc, "Received event while not in PLAYING state");
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goto ret;
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}
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GST_DEBUG_OBJECT (dtmfsrc, "Received event is of our interest");
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structure = gst_event_get_structure (event);
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if (structure && gst_structure_has_name (structure, "dtmf-event"))
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result = gst_rtp_dtmf_src_handle_dtmf_event (dtmfsrc, structure);
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ret:
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return result;
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}
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static gboolean
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gst_rtp_dtmf_src_handle_event (GstPad * pad, GstEvent * event)
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{
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@ -327,44 +382,7 @@ gst_rtp_dtmf_src_handle_event (GstPad * pad, GstEvent * event)
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_CUSTOM_UPSTREAM:
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{
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const GstStructure *structure;
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if (GST_STATE (dtmfsrc) != GST_STATE_PLAYING) {
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GST_DEBUG_OBJECT (dtmfsrc, "Received event while not in PLAYING state");
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break;
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}
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GST_DEBUG_OBJECT (dtmfsrc, "Received event is of our interest");
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structure = gst_event_get_structure (event);
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if (structure && gst_structure_has_name (structure, "dtmf-event")) {
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gint event_type;
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gboolean start;
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if (!gst_structure_get_int (structure, "type", &event_type) ||
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!gst_structure_get_boolean (structure, "start", &start) ||
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event_type != GST_RTP_DTMF_TYPE_EVENT)
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break;
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if (start) {
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gint event_number;
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gint event_volume;
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if (!gst_structure_get_int (structure, "number", &event_number) ||
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!gst_structure_get_int (structure, "volume", &event_volume))
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break;
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GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d",
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event_number, event_volume);
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gst_rtp_dtmf_src_start (dtmfsrc, event_number, event_volume);
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}
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else {
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GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
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gst_rtp_dtmf_src_stop (dtmfsrc);
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}
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}
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result = TRUE;
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result = gst_rtp_dtmf_src_handle_custom_upstream (dtmfsrc, event);
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break;
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}
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/* Ideally this element should not be flushed but let's handle the event
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@ -464,18 +482,10 @@ gst_rtp_dtmf_src_set_stream_lock (GstRTPDTMFSrc *dtmfsrc, gboolean lock)
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}
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static void
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gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc,
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gint event_number, gint event_volume)
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gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc *dtmfsrc)
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{
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GstClock *clock;
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g_return_if_fail (dtmfsrc->payload == NULL);
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dtmfsrc->payload = g_new0 (GstRTPDTMFPayload, 1);
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dtmfsrc->payload->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
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dtmfsrc->payload->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
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dtmfsrc->first_packet = TRUE;
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clock = GST_ELEMENT_CLOCK (dtmfsrc);
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if (clock != NULL)
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dtmfsrc->timestamp = gst_clock_get_time (GST_ELEMENT_CLOCK (dtmfsrc));
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@ -489,8 +499,20 @@ gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc,
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gst_util_uint64_scale_int (
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dtmfsrc->timestamp - gst_element_get_base_time (GST_ELEMENT (dtmfsrc)),
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dtmfsrc->clock_rate, GST_SECOND);
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}
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static void
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gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc,
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gint event_number, gint event_volume)
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{
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g_return_if_fail (dtmfsrc->payload == NULL);
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dtmfsrc->payload = g_new0 (GstRTPDTMFPayload, 1);
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dtmfsrc->payload->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
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dtmfsrc->payload->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
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dtmfsrc->first_packet = TRUE;
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gst_rtp_dtmf_prepare_timestamps (dtmfsrc);
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gst_rtp_dtmf_src_set_caps (dtmfsrc);
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/* Don't forget to get exclusive access to the stream */
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@ -524,47 +546,10 @@ gst_rtp_dtmf_src_stop (GstRTPDTMFSrc *dtmfsrc)
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}
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static void
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gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
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gst_rtp_dtmf_src_wait_for_buffer_ts (GstRTPDTMFSrc *dtmfsrc, GstBuffer * buf)
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{
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GstBuffer *buf = NULL;
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GstFlowReturn ret;
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GstRTPDTMFPayload *payload;
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GstClock * clock;
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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_buffer_set_ssrc (buf, dtmfsrc->current_ssrc);
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gst_rtp_buffer_set_payload_type (buf, dtmfsrc->pt);
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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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}
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dtmfsrc->seqnum++;
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gst_rtp_buffer_set_seq (buf, dtmfsrc->seqnum);
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GstClock *clock;
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/* timestamp of RTP header */
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gst_rtp_buffer_set_timestamp (buf, dtmfsrc->rtp_timestamp);
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dtmfsrc->rtp_timestamp +=
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DEFAULT_PACKET_INTERVAL * dtmfsrc->clock_rate / 1000;
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/* duration of DTMF payload */
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dtmfsrc->payload->duration +=
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DEFAULT_PACKET_INTERVAL * dtmfsrc->clock_rate / 1000;
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/* timestamp and duration of GstBuffer */
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GST_BUFFER_DURATION (buf) = DEFAULT_PACKET_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, dtmfsrc->payload, sizeof (GstRTPDTMFPayload));
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payload->duration = g_htons (payload->duration);
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/* FIXME: Should we sync to clock ourselves or leave it to sink */
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clock = GST_ELEMENT_CLOCK (dtmfsrc);
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if (clock != NULL) {
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GstClockID clock_id;
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@ -582,9 +567,77 @@ gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
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else {
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GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s", GST_ELEMENT_NAME (dtmfsrc));
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}
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}
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static void
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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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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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}
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dtmfsrc->seqnum++;
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gst_rtp_buffer_set_seq (buf, dtmfsrc->seqnum);
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/* timestamp of RTP header */
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gst_rtp_buffer_set_timestamp (buf, dtmfsrc->rtp_timestamp);
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dtmfsrc->rtp_timestamp +=
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DEFAULT_PACKET_INTERVAL * dtmfsrc->clock_rate / 1000;
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}
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static void
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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, buf);
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/* duration of DTMF payload */
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dtmfsrc->payload->duration +=
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DEFAULT_PACKET_INTERVAL * dtmfsrc->clock_rate / 1000;
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/* timestamp and duration of GstBuffer */
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GST_BUFFER_DURATION (buf) = DEFAULT_PACKET_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, dtmfsrc->payload, sizeof (GstRTPDTMFPayload));
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payload->duration = g_htons (payload->duration);
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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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{
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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, buf);
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/* FIXME: Should we sync to clock ourselves or leave it to sink */
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gst_rtp_dtmf_src_wait_for_buffer_ts (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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return buf;
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}
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static void
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gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
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{
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GstBuffer *buf = NULL;
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GstFlowReturn ret;
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/* create buffer to hold the payload */
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buf = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc);
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GST_DEBUG_OBJECT (dtmfsrc,
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"pushing buffer on src pad of size %d", GST_BUFFER_SIZE (buf));
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ret = gst_pad_push (dtmfsrc->srcpad, buf);
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@ -620,6 +673,26 @@ gst_rtp_dtmf_src_set_caps (GstRTPDTMFSrc *dtmfsrc)
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gst_caps_unref (caps);
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}
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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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if (dtmfsrc->ssrc == -1)
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dtmfsrc->current_ssrc = g_random_int ();
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else
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dtmfsrc->current_ssrc = dtmfsrc->ssrc;
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if (dtmfsrc->seqnum_offset == -1)
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dtmfsrc->seqnum_base = g_random_int_range (0, G_MAXUINT16);
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else
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dtmfsrc->seqnum_base = dtmfsrc->seqnum_offset;
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dtmfsrc->seqnum = dtmfsrc->seqnum_base;
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if (dtmfsrc->ts_offset == -1)
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dtmfsrc->ts_base = g_random_int ();
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else
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dtmfsrc->ts_base = dtmfsrc->ts_offset;
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}
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static GstStateChangeReturn
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gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition)
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{
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@ -630,33 +703,10 @@ gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition)
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dtmfsrc = GST_RTP_DTMF_SRC (element);
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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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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dtmfsrc->current_ssrc = dtmfsrc->ssrc;
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if (dtmfsrc->seqnum_offset == -1)
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dtmfsrc->seqnum_base = g_random_int_range (0, G_MAXUINT16);
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else
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dtmfsrc->seqnum_base = dtmfsrc->seqnum_offset;
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dtmfsrc->seqnum = dtmfsrc->seqnum_base;
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if (dtmfsrc->ts_offset == -1)
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dtmfsrc->ts_base = g_random_int ();
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else
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dtmfsrc->ts_base = dtmfsrc->ts_offset;
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gst_rtp_dtmf_src_ready_to_paused (dtmfsrc);
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/* Indicate that we don't do PRE_ROLL */
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no_preroll = TRUE;
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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break;
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default:
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break;
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@ -672,10 +722,6 @@ gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition)
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/* Indicate that we don't do PRE_ROLL */
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no_preroll = TRUE;
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break;
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_NULL:
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break;
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default:
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break;
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}
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