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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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[MOVED FROM GST-P-FARSIGHT] Ported the event queue work from dtmfsrc to rtpdtmfsrc
Added a queue based system for the rtpdtmfsrc. Now it waits for start/stop messages on the queue, and makes sure that the minimum duty cycle (120ms) is respected between each tone, including inter-digit silence. 20070822175533-4f0f6-f27414c406f1f7b00c9a9084a988cf3a7930fe5c.gz
This commit is contained in:
Youness Alaoui
Edward Hervey
parent
01f5b39eb2
commit
26a8292d83
2 changed files with 215 additions and 103 deletions
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@ -45,7 +45,7 @@
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* <colspec colname='Type' />
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* <colspec colname='Possible values' />
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* <colspec colname='Purpose' />
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*
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*
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* <thead>
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* <row>
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* <entry>Name</entry>
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@ -54,7 +54,7 @@
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* <entry>Purpose</entry>
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* </row>
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* </thead>
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*
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*
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* <tbody>
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* <row>
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* <entry>type</entry>
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@ -98,12 +98,12 @@
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* </tgroup>
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* </informaltable>
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* </para>
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*
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*
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* <para>For example, the following code informs the pipeline (and in turn, the
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* RTPDTMFSrc element inside the pipeline) about the start of an RTP DTMF named
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* event '1' of volume -25 dBm0:
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* </para>
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*
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*
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* <para>
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* <programlisting>
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* structure = gst_structure_new ("dtmf-event",
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@ -111,7 +111,7 @@
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* "number", G_TYPE_INT, 1,
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* "volume", G_TYPE_INT, 25,
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* "start", G_TYPE_BOOLEAN, TRUE, NULL);
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*
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*
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* event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure);
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* gst_element_send_event (pipeline, event);
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* </programlisting>
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@ -127,6 +127,8 @@
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#include <stdlib.h>
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#include <string.h>
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#include <glib.h>
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#include "gstrtpdtmfsrc.h"
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#define GST_RTP_DTMF_TYPE_EVENT 1
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@ -146,6 +148,10 @@
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#define MAX_VOLUME 36
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#define MIN_EVENT_DURATION 50
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#define MIN_INTER_DIGIT_INTERVAL 50
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#define MIN_PULSE_DURATION 70
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#define MIN_DUTY_CYCLE (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION)
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#define DEFAULT_PACKET_REDUNDANCY 1
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#define MIN_PACKET_REDUNDANCY 1
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#define MAX_PACKET_REDUNDANCY 5
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@ -188,8 +194,8 @@ GST_STATIC_PAD_TEMPLATE ("src",
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GST_STATIC_CAPS ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) [ 96, 127 ], "
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"clock-rate = (int) [ 0, MAX ], "
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"ssrc = (int) [ 0, MAX ], "
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"clock-rate = (int) [ 0, MAX ], "
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"ssrc = (int) [ 0, MAX ], "
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"events = (int) [ " MIN_EVENT_STRING ", " MAX_EVENT_STRING " ], "
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"encoding-name = (string) \"telephone-event\"")
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);
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@ -233,22 +239,25 @@ static gboolean gst_rtp_dtmf_src_handle_event (GstPad * pad, GstEvent * event);
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static GstStateChangeReturn gst_rtp_dtmf_src_change_state (GstElement * element,
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GstStateChange transition);
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static void gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc);
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static void gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc, gint event_number,
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gint event_volume);
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static void gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc);
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static void gst_rtp_dtmf_src_stop (GstRTPDTMFSrc *dtmfsrc);
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static void gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc *dtmfsrc,
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gint event_number, gint event_volume);
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static void gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc *dtmfsrc);
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static void gst_rtp_dtmf_src_set_caps (GstRTPDTMFSrc *dtmfsrc);
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static void
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gst_rtp_dtmf_src_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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GST_DEBUG_CATEGORY_INIT (gst_rtp_dtmf_src_debug,
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"rtpdtmfsrc", 0, "rtpdtmfsrc element");
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_dtmf_src_template));
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gst_element_class_set_details (element_class, &gst_rtp_dtmf_src_details);
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}
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@ -264,7 +273,7 @@ gst_rtp_dtmf_src_class_init (GstRTPDTMFSrcClass * klass)
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_finalize);
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gobject_class->set_property =
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gobject_class->set_property =
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GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_set_property);
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gobject_class->get_property =
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GST_DEBUG_FUNCPTR (gst_rtp_dtmf_src_get_property);
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@ -321,16 +330,19 @@ gst_rtp_dtmf_src_init (GstRTPDTMFSrc * dtmfsrc, gpointer g_class)
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gst_element_add_pad (GST_ELEMENT (dtmfsrc), dtmfsrc->srcpad);
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gst_pad_set_event_function (dtmfsrc->srcpad, gst_rtp_dtmf_src_handle_event);
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dtmfsrc->ssrc = DEFAULT_SSRC;
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dtmfsrc->seqnum_offset = DEFAULT_SEQNUM_OFFSET;
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dtmfsrc->ts_offset = DEFAULT_TIMESTAMP_OFFSET;
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dtmfsrc->pt = DEFAULT_PT;
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dtmfsrc->clock_rate = DEFAULT_CLOCK_RATE;
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dtmfsrc->payload = NULL;
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dtmfsrc->interval = DEFAULT_PACKET_INTERVAL;
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dtmfsrc->packet_redundancy = DEFAULT_PACKET_REDUNDANCY;
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dtmfsrc->event_queue = g_async_queue_new ();
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dtmfsrc->last_event = NULL;
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GST_DEBUG_OBJECT (dtmfsrc, "init done");
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}
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@ -339,6 +351,12 @@ gst_rtp_dtmf_src_finalize (GObject * object)
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{
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GstRTPDTMFSrc *dtmfsrc;
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if (dtmfsrc->event_queue) {
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g_async_queue_unref (dtmfsrc->event_queue);
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dtmfsrc->event_queue = NULL;
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}
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dtmfsrc = GST_RTP_DTMF_SRC (object);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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@ -373,12 +391,12 @@ gst_rtp_dtmf_src_handle_dtmf_event (GstRTPDTMFSrc *dtmfsrc,
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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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gst_rtp_dtmf_src_add_start_event (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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gst_rtp_dtmf_src_add_stop_event (dtmfsrc);
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}
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return TRUE;
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@ -391,6 +409,7 @@ gst_rtp_dtmf_src_handle_custom_upstream (GstRTPDTMFSrc *dtmfsrc,
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GstEvent * event)
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{
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gboolean result = FALSE;
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gchar *struct_str;
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const GstStructure *structure;
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if (GST_STATE (dtmfsrc) != GST_STATE_PLAYING) {
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@ -400,6 +419,9 @@ gst_rtp_dtmf_src_handle_custom_upstream (GstRTPDTMFSrc *dtmfsrc,
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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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struct_str = gst_structure_to_string (structure);
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GST_DEBUG_OBJECT (dtmfsrc, "Event has structure %s", struct_str);
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g_free (struct_str);
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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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@ -553,7 +575,8 @@ 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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dtmfsrc->timestamp = gst_clock_get_time (GST_ELEMENT_CLOCK (dtmfsrc))
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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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@ -569,23 +592,10 @@ gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc *dtmfsrc)
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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_src_start (GstRTPDTMFSrc *dtmfsrc)
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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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dtmfsrc->last_packet = FALSE;
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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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gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
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if (!gst_pad_start_task (dtmfsrc->srcpad,
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(GstTaskFunction) gst_rtp_dtmf_src_push_next_rtp_packet, dtmfsrc)) {
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GST_ERROR_OBJECT (dtmfsrc, "Failed to start task on src pad");
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@ -595,19 +605,65 @@ gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc,
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static void
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gst_rtp_dtmf_src_stop (GstRTPDTMFSrc *dtmfsrc)
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{
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g_return_if_fail (dtmfsrc->payload != NULL);
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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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/* Push the last packet with e-bit set */
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/* Next packet sent will be the last */
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dtmfsrc->last_packet = TRUE;
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/* Flushing the event queue */
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GstRTPDTMFSrcEvent *event = g_async_queue_try_pop (dtmfsrc->event_queue);
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while (event != NULL) {
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g_free (event);
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event = g_async_queue_try_pop (dtmfsrc->event_queue);
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}
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if (dtmfsrc->last_event) {
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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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if (!gst_pad_pause_task (dtmfsrc->srcpad)) {
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GST_ERROR_OBJECT (dtmfsrc, "Failed to pause task on src pad");
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return;
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}
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}
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static void
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gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc *dtmfsrc, gint event_number,
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gint event_volume)
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{
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GstRTPDTMFSrcEvent * event = g_malloc (sizeof(GstRTPDTMFSrcEvent));
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event->event_type = RTP_DTMF_EVENT_TYPE_START;
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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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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_src_add_stop_event (GstRTPDTMFSrc *dtmfsrc)
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{
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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_src_wait_for_buffer_ts (GstRTPDTMFSrc *dtmfsrc, GstBuffer * buf)
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{
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GstClock *clock;
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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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@ -629,7 +685,7 @@ 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, GstBuffer *buf)
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gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc, 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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@ -637,37 +693,37 @@ gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
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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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dtmfsrc->payload->e = 1;
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event->payload->e = 1;
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dtmfsrc->last_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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}
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static void
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gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
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gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, GstRTPDTMFSrcEvent *event,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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gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc,event, buf);
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/* duration of DTMF payload */
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dtmfsrc->payload->duration +=
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event->payload->duration +=
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dtmfsrc->interval * dtmfsrc->clock_rate / 1000;
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/* timestamp and duration of GstBuffer */
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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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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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g_memmove (payload, event->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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@ -679,18 +735,20 @@ gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
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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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gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc, GstRTPDTMFSrcEvent *event)
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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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gst_rtp_dtmf_prepare_buffer_data (dtmfsrc, event, 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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event->sent_packets++;
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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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@ -703,58 +761,93 @@ gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
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GstBuffer *buf = NULL;
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GstFlowReturn ret;
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gint redundancy_count = 1;
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GstRTPDTMFSrcEvent *event;
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if (dtmfsrc->first_packet == TRUE || dtmfsrc->last_packet == TRUE) {
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redundancy_count = dtmfsrc->packet_redundancy;
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g_async_queue_ref (dtmfsrc->event_queue);
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if (dtmfsrc->last_event == NULL) {
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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) {
|
||||
GST_WARNING_OBJECT (dtmfsrc, "Received a DTMF stop event when already stopped");
|
||||
} else if (event->event_type == RTP_DTMF_EVENT_TYPE_START) {
|
||||
|
||||
dtmfsrc->first_packet = TRUE;
|
||||
dtmfsrc->last_packet = FALSE;
|
||||
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);
|
||||
|
||||
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);
|
||||
|
||||
if (dtmfsrc->last_event) {
|
||||
|
||||
if (dtmfsrc->first_packet == TRUE || dtmfsrc->last_packet == TRUE) {
|
||||
redundancy_count = dtmfsrc->packet_redundancy;
|
||||
|
||||
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->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);
|
||||
}
|
||||
|
||||
}
|
||||
/* create buffer to hold the payload */
|
||||
buf = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc, dtmfsrc->last_event);
|
||||
|
||||
/* create buffer to hold the payload */
|
||||
buf = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc);
|
||||
while ( redundancy_count-- ) {
|
||||
gst_buffer_ref(buf);
|
||||
|
||||
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,
|
||||
"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));
|
||||
"pushed DTMF event '%d' on src pad", event->payload->event);
|
||||
|
||||
/* Make sure only the first packet sent has the marker set */
|
||||
gst_rtp_buffer_set_marker (buf, FALSE);
|
||||
}
|
||||
if (dtmfsrc->last_event->payload->e) {
|
||||
/* Don't forget to release the stream lock */
|
||||
gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
|
||||
|
||||
gst_buffer_unref(buf);
|
||||
GST_DEBUG_OBJECT (dtmfsrc,
|
||||
"pushed DTMF event '%d' on src pad", dtmfsrc->payload->event);
|
||||
g_free (dtmfsrc->last_event->payload);
|
||||
event->payload = NULL;
|
||||
|
||||
if (dtmfsrc->payload->e) {
|
||||
/* 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;
|
||||
|
||||
if (!gst_pad_pause_task (dtmfsrc->srcpad)) {
|
||||
GST_ERROR_OBJECT (dtmfsrc, "Failed to pause task on src pad");
|
||||
return;
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
static void
|
||||
|
|
@ -785,7 +878,7 @@ 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
|
||||
|
|
@ -796,7 +889,7 @@ gst_rtp_dtmf_src_ready_to_paused (GstRTPDTMFSrc *dtmfsrc)
|
|||
else
|
||||
dtmfsrc->seqnum_base = dtmfsrc->seqnum_offset;
|
||||
dtmfsrc->seqnum = dtmfsrc->seqnum_base;
|
||||
|
||||
|
||||
if (dtmfsrc->ts_offset == -1)
|
||||
dtmfsrc->ts_base = g_random_int ();
|
||||
else
|
||||
|
|
@ -818,6 +911,9 @@ gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition)
|
|||
/* 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;
|
||||
}
|
||||
|
|
@ -831,6 +927,7 @@ gst_rtp_dtmf_src_change_state (GstElement * element, GstStateChange transition)
|
|||
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;
|
||||
|
|
|
|||
|
|
@ -57,6 +57,23 @@ typedef struct {
|
|||
typedef struct _GstRTPDTMFSrc GstRTPDTMFSrc;
|
||||
typedef struct _GstRTPDTMFSrcClass GstRTPDTMFSrcClass;
|
||||
|
||||
|
||||
|
||||
static enum _GstRTPDTMFEventType {
|
||||
RTP_DTMF_EVENT_TYPE_START,
|
||||
RTP_DTMF_EVENT_TYPE_STOP
|
||||
};
|
||||
|
||||
typedef enum _GstRTPDTMFEventType GstRTPDTMFEventType;
|
||||
|
||||
struct _GstRTPDTMFSrcEvent {
|
||||
GstRTPDTMFEventType event_type;
|
||||
GstRTPDTMFPayload* payload;
|
||||
guint32 sent_packets;
|
||||
};
|
||||
|
||||
typedef struct _GstRTPDTMFSrcEvent GstRTPDTMFSrcEvent;
|
||||
|
||||
/**
|
||||
* GstRTPDTMFSrc:
|
||||
* @element: the parent element.
|
||||
|
|
@ -64,30 +81,28 @@ typedef struct _GstRTPDTMFSrcClass GstRTPDTMFSrcClass;
|
|||
* The opaque #GstRTPDTMFSrc data structure.
|
||||
*/
|
||||
struct _GstRTPDTMFSrc {
|
||||
GstElement element;
|
||||
GstElement element;
|
||||
|
||||
GstPad *srcpad;
|
||||
GstRTPDTMFPayload *payload;
|
||||
GstPad* srcpad;
|
||||
GstSegment segment;
|
||||
GAsyncQueue* event_queue;
|
||||
GstRTPDTMFSrcEvent* last_event;
|
||||
|
||||
GstClockTime timestamp;
|
||||
gboolean first_packet;
|
||||
gboolean last_packet;
|
||||
guint32 ts_base;
|
||||
guint16 seqnum_base;
|
||||
|
||||
gint16 seqnum_offset;
|
||||
guint16 seqnum;
|
||||
gint32 ts_offset;
|
||||
guint32 rtp_timestamp;
|
||||
guint32 clock_rate;
|
||||
guint pt;
|
||||
guint ssrc;
|
||||
guint current_ssrc;
|
||||
gboolean first_packet;
|
||||
gboolean last_packet;
|
||||
|
||||
GstClockTime timestamp;
|
||||
GstSegment segment;
|
||||
|
||||
guint16 interval;
|
||||
guint16 packet_redundancy;
|
||||
guint32 clock_rate;
|
||||
};
|
||||
|
||||
struct _GstRTPDTMFSrcClass {
|
||||
|
|
|
|||
Loading…
Reference in a new issue