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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-28 04:31:06 +00:00
[MOVED FROM GST-P-FARSIGHT] Fix overly long lines and tabs
20070827195610-3e2dc-396a3fa01e16f184e4109c71fe2deb6e516bdf0d.gz
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58ec497deb
commit
a0beb104de
2 changed files with 49 additions and 35 deletions
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@ -269,13 +269,13 @@ static void gst_dtmf_src_get_property (GObject * object, guint prop_id,
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static gboolean gst_dtmf_src_handle_event (GstPad * pad, GstEvent * event);
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static GstStateChangeReturn gst_dtmf_src_change_state (GstElement * element,
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GstStateChange transition);
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static void gst_dtmf_src_generate_tone(GstDTMFSrcEvent *event, DTMF_KEY key, float duration,
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GstBuffer * buffer);
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static void gst_dtmf_src_generate_tone(GstDTMFSrcEvent *event, DTMF_KEY key,
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float duration, GstBuffer * buffer);
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static void gst_dtmf_src_push_next_tone_packet (GstDTMFSrc *dtmfsrc);
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static void gst_dtmf_src_start (GstDTMFSrc *dtmfsrc);
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static void gst_dtmf_src_stop (GstDTMFSrc *dtmfsrc);
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static void gst_dtmf_src_add_start_event (GstDTMFSrc *dtmfsrc, gint event_number,
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gint event_volume);
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static void gst_dtmf_src_add_start_event (GstDTMFSrc *dtmfsrc,
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gint event_number, gint event_volume);
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static void gst_dtmf_src_add_stop_event (GstDTMFSrc *dtmfsrc);
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static void
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@ -615,7 +615,8 @@ gst_dtmf_src_generate_silence(GstBuffer * buffer, float duration)
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}
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static void
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gst_dtmf_src_generate_tone(GstDTMFSrcEvent *event, DTMF_KEY key, float duration, GstBuffer * buffer)
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gst_dtmf_src_generate_tone(GstDTMFSrcEvent *event, DTMF_KEY key, float duration,
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GstBuffer * buffer)
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{
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gint16 *p;
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gint tone_size;
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@ -686,12 +687,14 @@ gst_dtmf_src_wait_for_buffer_ts (GstDTMFSrc *dtmfsrc, GstBuffer * buf)
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static GstBuffer *
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gst_dtmf_src_create_next_tone_packet (GstDTMFSrc *dtmfsrc, GstDTMFSrcEvent *event)
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gst_dtmf_src_create_next_tone_packet (GstDTMFSrc *dtmfsrc,
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GstDTMFSrcEvent *event)
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{
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GstBuffer *buf = NULL;
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gboolean send_silence = FALSE;
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GST_DEBUG_OBJECT (dtmfsrc, "Creating buffer for tone %s", DTMF_KEYS[event->event_number].event_name);
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GST_DEBUG_OBJECT (dtmfsrc, "Creating buffer for tone %s",
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DTMF_KEYS[event->event_number].event_name);
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/* create buffer to hold the tone */
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buf = gst_buffer_new ();
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@ -738,7 +741,8 @@ gst_dtmf_src_push_next_tone_packet (GstDTMFSrc *dtmfsrc)
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event = g_async_queue_pop (dtmfsrc->event_queue);
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if (event->event_type == DTMF_EVENT_TYPE_STOP) {
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GST_WARNING_OBJECT (dtmfsrc, "Received a DTMF stop event when already stopped");
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GST_WARNING_OBJECT (dtmfsrc,
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"Received a DTMF stop event when already stopped");
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} else if (event->event_type == DTMF_EVENT_TYPE_START) {
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gst_dtmf_prepare_timestamps (dtmfsrc);
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@ -748,12 +752,14 @@ gst_dtmf_src_push_next_tone_packet (GstDTMFSrc *dtmfsrc)
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event->packet_count = 0;
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dtmfsrc->last_event = event;
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}
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} else if (dtmfsrc->last_event->packet_count * dtmfsrc->interval >= MIN_DUTY_CYCLE) {
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} else if (dtmfsrc->last_event->packet_count * dtmfsrc->interval >=
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MIN_DUTY_CYCLE) {
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event = g_async_queue_try_pop (dtmfsrc->event_queue);
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if (event != NULL) {
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if (event->event_type == DTMF_EVENT_TYPE_START) {
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GST_WARNING_OBJECT (dtmfsrc, "Received two consecutive DTMF start events");
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GST_WARNING_OBJECT (dtmfsrc,
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"Received two consecutive DTMF start events");
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} else if (event->event_type == DTMF_EVENT_TYPE_STOP) {
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gst_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
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g_free (dtmfsrc->last_event);
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@ -794,8 +800,9 @@ gst_dtmf_src_change_state (GstElement * element, GstStateChange transition)
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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gst_segment_init (&dtmfsrc->segment, GST_FORMAT_TIME);
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gst_pad_push_event (dtmfsrc->srcpad, gst_event_new_new_segment (FALSE,
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dtmfsrc->segment.rate, dtmfsrc->segment.format,
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dtmfsrc->segment.start, dtmfsrc->segment.stop, dtmfsrc->segment.time));
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dtmfsrc->segment.rate, dtmfsrc->segment.format,
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dtmfsrc->segment.start, dtmfsrc->segment.stop,
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dtmfsrc->segment.time));
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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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@ -90,8 +90,8 @@
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* <entry>method</entry>
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* <entry>G_TYPE_INT</entry>
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* <entry>1</entry>
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* <entry>The method used for sending event, this element will react if this field
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* is absent or 1.
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* <entry>The method used for sending event, this element will react if this
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* field is absent or 1.
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* </entry>
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* </row>
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* </tbody>
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@ -576,7 +576,7 @@ gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc *dtmfsrc)
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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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+ (MIN_INTER_DIGIT_INTERVAL * GST_MSECOND);
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+ (MIN_INTER_DIGIT_INTERVAL * GST_MSECOND);
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else {
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GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s",
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@ -685,7 +685,8 @@ gst_rtp_dtmf_src_wait_for_buffer_ts (GstRTPDTMFSrc *dtmfsrc, GstBuffer * buf)
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}
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static void
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gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc, GstRTPDTMFSrcEvent *event, GstBuffer *buf)
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gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc,
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GstRTPDTMFSrcEvent *event, 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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@ -705,7 +706,8 @@ gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc, GstRTPDTMFSrcEvent *ev
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}
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static void
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gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, GstRTPDTMFSrcEvent *event,GstBuffer *buf)
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gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc,
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GstRTPDTMFSrcEvent *event,GstBuffer *buf)
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{
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GstRTPDTMFPayload *payload;
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@ -735,7 +737,8 @@ gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, GstRTPDTMFSrcEvent *ev
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}
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static GstBuffer *
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gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc, GstRTPDTMFSrcEvent *event)
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gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc,
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GstRTPDTMFSrcEvent *event)
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{
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GstBuffer *buf = NULL;
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@ -769,7 +772,8 @@ gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
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event = g_async_queue_pop (dtmfsrc->event_queue);
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if (event->event_type == RTP_DTMF_EVENT_TYPE_STOP) {
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GST_WARNING_OBJECT (dtmfsrc, "Received a DTMF stop event when already stopped");
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GST_WARNING_OBJECT (dtmfsrc,
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"Received a DTMF stop event when already stopped");
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} else if (event->event_type == RTP_DTMF_EVENT_TYPE_START) {
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dtmfsrc->first_packet = TRUE;
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@ -783,15 +787,17 @@ gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
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dtmfsrc->last_event = event;
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}
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} else if (dtmfsrc->last_event->sent_packets * dtmfsrc->interval >= MIN_PULSE_DURATION){
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} else if (dtmfsrc->last_event->sent_packets * dtmfsrc->interval >=
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MIN_PULSE_DURATION){
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event = g_async_queue_try_pop (dtmfsrc->event_queue);
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if (event != NULL) {
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if (event->event_type == RTP_DTMF_EVENT_TYPE_START) {
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GST_WARNING_OBJECT (dtmfsrc, "Received two consecutive DTMF start events");
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GST_WARNING_OBJECT (dtmfsrc,
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"Received two consecutive DTMF start events");
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} else if (event->event_type == 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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dtmfsrc->first_packet = FALSE;
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dtmfsrc->last_packet = TRUE;
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}
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}
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}
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@ -803,30 +809,31 @@ gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
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redundancy_count = dtmfsrc->packet_redundancy;
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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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redundancy_count);
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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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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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redundancy_count);
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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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redundancy_count);
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}
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}
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/* create buffer to hold the payload */
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buf = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc, dtmfsrc->last_event);
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buf = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc,
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dtmfsrc->last_event);
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while ( redundancy_count-- ) {
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gst_buffer_ref(buf);
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GST_DEBUG_OBJECT (dtmfsrc,
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"pushing buffer on src pad of size %d with redundancy count %d",
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GST_BUFFER_SIZE (buf), redundancy_count);
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"pushing buffer on src pad of size %d with redundancy count %d",
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GST_BUFFER_SIZE (buf), redundancy_count);
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ret = gst_pad_push (dtmfsrc->srcpad, buf);
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if (ret != GST_FLOW_OK)
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GST_ERROR_OBJECT (dtmfsrc,
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"Failed to push buffer on src pad");
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GST_ERROR_OBJECT (dtmfsrc,
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"Failed to push buffer on src pad");
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/* Make sure only the first packet sent has the marker set */
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gst_rtp_buffer_set_marker (buf, FALSE);
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@ -834,7 +841,7 @@ gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
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gst_buffer_unref(buf);
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GST_DEBUG_OBJECT (dtmfsrc,
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"pushed DTMF event '%d' on src pad", event->payload->event);
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"pushed DTMF event '%d' on src pad", event->payload->event);
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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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