[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

gst/dtmf/gstrtpdtmfsrc.c

@@ -127,6 +127,8 @@
 #include <stdlib.h>
 #include <string.h>
 
+#include <glib.h>
+
 #include "gstrtpdtmfsrc.h"
 
 #define GST_RTP_DTMF_TYPE_EVENT  1
@@ -146,6 +148,10 @@
 #define MAX_VOLUME               36
 #define MIN_EVENT_DURATION       50
 
+#define MIN_INTER_DIGIT_INTERVAL 50
+#define MIN_PULSE_DURATION       70
+#define MIN_DUTY_CYCLE           (MIN_INTER_DIGIT_INTERVAL + MIN_PULSE_DURATION)
+
 #define DEFAULT_PACKET_REDUNDANCY 1
 #define MIN_PACKET_REDUNDANCY 1
 #define MAX_PACKET_REDUNDANCY 5
@@ -233,11 +239,14 @@ static gboolean gst_rtp_dtmf_src_handle_event (GstPad * pad, GstEvent * event);
 static GstStateChangeReturn gst_rtp_dtmf_src_change_state (GstElement * element,
     GstStateChange transition);
 static void gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc);
-static void gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc, gint event_number,
+static void gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc);
-    gint event_volume);
 static void gst_rtp_dtmf_src_stop (GstRTPDTMFSrc *dtmfsrc);
+static void gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc *dtmfsrc,
+    gint event_number, gint event_volume);
+static void gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc *dtmfsrc);
 static void gst_rtp_dtmf_src_set_caps (GstRTPDTMFSrc *dtmfsrc);
 
+
 static void
 gst_rtp_dtmf_src_base_init (gpointer g_class)
 {
@@ -327,10 +336,13 @@ gst_rtp_dtmf_src_init (GstRTPDTMFSrc * dtmfsrc, gpointer g_class)
   dtmfsrc->ts_offset = DEFAULT_TIMESTAMP_OFFSET;
   dtmfsrc->pt = DEFAULT_PT;
   dtmfsrc->clock_rate = DEFAULT_CLOCK_RATE;
-  dtmfsrc->payload = NULL;
   dtmfsrc->interval = DEFAULT_PACKET_INTERVAL;
   dtmfsrc->packet_redundancy = DEFAULT_PACKET_REDUNDANCY;
 
+
+  dtmfsrc->event_queue = g_async_queue_new ();
+  dtmfsrc->last_event = NULL;
+
   GST_DEBUG_OBJECT (dtmfsrc, "init done");
 }
 
@@ -339,6 +351,12 @@ gst_rtp_dtmf_src_finalize (GObject * object)
 {
   GstRTPDTMFSrc *dtmfsrc;
 
+
+  if (dtmfsrc->event_queue) {
+    g_async_queue_unref (dtmfsrc->event_queue);
+    dtmfsrc->event_queue = NULL;
+  }
+
   dtmfsrc = GST_RTP_DTMF_SRC (object);
 
   G_OBJECT_CLASS (parent_class)->finalize (object);
@@ -373,12 +391,12 @@ gst_rtp_dtmf_src_handle_dtmf_event (GstRTPDTMFSrc *dtmfsrc,
 
     GST_DEBUG_OBJECT (dtmfsrc, "Received start event %d with volume %d",
             event_number, event_volume);
-    gst_rtp_dtmf_src_start (dtmfsrc, event_number, event_volume);
+    gst_rtp_dtmf_src_add_start_event (dtmfsrc, event_number, event_volume);
   }
 
   else {
     GST_DEBUG_OBJECT (dtmfsrc, "Received stop event");
-    gst_rtp_dtmf_src_stop (dtmfsrc);
+    gst_rtp_dtmf_src_add_stop_event (dtmfsrc);
   }
 
   return TRUE;
@@ -391,6 +409,7 @@ gst_rtp_dtmf_src_handle_custom_upstream (GstRTPDTMFSrc *dtmfsrc,
     GstEvent * event)
 {
   gboolean result = FALSE;
+  gchar *struct_str;
   const GstStructure *structure;
 
   if (GST_STATE (dtmfsrc) != GST_STATE_PLAYING) {
@@ -400,6 +419,9 @@ gst_rtp_dtmf_src_handle_custom_upstream (GstRTPDTMFSrc *dtmfsrc,
 
   GST_DEBUG_OBJECT (dtmfsrc, "Received event is of our interest");
   structure = gst_event_get_structure (event);
+  struct_str = gst_structure_to_string (structure);
+  GST_DEBUG_OBJECT (dtmfsrc, "Event has structure %s", struct_str);
+  g_free (struct_str);
   if (structure && gst_structure_has_name (structure, "dtmf-event"))
     result = gst_rtp_dtmf_src_handle_dtmf_event (dtmfsrc, structure);
 
@@ -553,7 +575,8 @@ gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc *dtmfsrc)
 
   clock = GST_ELEMENT_CLOCK (dtmfsrc);
   if (clock != NULL)
-    dtmfsrc->timestamp = gst_clock_get_time (GST_ELEMENT_CLOCK (dtmfsrc));
+    dtmfsrc->timestamp = gst_clock_get_time (GST_ELEMENT_CLOCK (dtmfsrc))
+	+ (MIN_INTER_DIGIT_INTERVAL * GST_MSECOND);
 
   else {
     GST_ERROR_OBJECT (dtmfsrc, "No clock set for element %s",
@@ -569,23 +592,10 @@ gst_rtp_dtmf_prepare_timestamps (GstRTPDTMFSrc *dtmfsrc)
 }
 
 static void
-gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc,
+gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc)
-        gint event_number, gint event_volume)
 {
-  g_return_if_fail (dtmfsrc->payload == NULL);
-
-  dtmfsrc->payload = g_new0 (GstRTPDTMFPayload, 1);
-  dtmfsrc->payload->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
-  dtmfsrc->payload->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
-  dtmfsrc->first_packet = TRUE;
-  dtmfsrc->last_packet = FALSE;
-
-  gst_rtp_dtmf_prepare_timestamps (dtmfsrc);
   gst_rtp_dtmf_src_set_caps (dtmfsrc);
 
-  /* Don't forget to get exclusive access to the stream */
-  gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, TRUE);
-  
   if (!gst_pad_start_task (dtmfsrc->srcpad,
       (GstTaskFunction) gst_rtp_dtmf_src_push_next_rtp_packet, dtmfsrc)) {
     GST_ERROR_OBJECT (dtmfsrc, "Failed to start task on src pad");
@@ -595,14 +605,60 @@ gst_rtp_dtmf_src_start (GstRTPDTMFSrc *dtmfsrc,
 static void
 gst_rtp_dtmf_src_stop (GstRTPDTMFSrc *dtmfsrc)
 {
-  g_return_if_fail (dtmfsrc->payload != NULL);
+  /* Don't forget to release the stream lock */
+  gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
 
-  /* Push the last packet with e-bit set */
-  /* Next packet sent will be the last */
-  dtmfsrc->last_packet = TRUE;
 
+  /* Flushing the event queue */
+  GstRTPDTMFSrcEvent *event = g_async_queue_try_pop (dtmfsrc->event_queue);
+
+  while (event != NULL) {
+    g_free (event);
+    event = g_async_queue_try_pop (dtmfsrc->event_queue);
   }
 
+  if (dtmfsrc->last_event) {
+    g_free (dtmfsrc->last_event);
+    dtmfsrc->last_event = NULL;
+  }
+
+  if (!gst_pad_pause_task (dtmfsrc->srcpad)) {
+    GST_ERROR_OBJECT (dtmfsrc, "Failed to pause task on src pad");
+    return;
+  }
+}
+
+
+
+static void
+gst_rtp_dtmf_src_add_start_event (GstRTPDTMFSrc *dtmfsrc, gint event_number,
+    gint event_volume)
+{
+
+  GstRTPDTMFSrcEvent * event = g_malloc (sizeof(GstRTPDTMFSrcEvent));
+  event->event_type = RTP_DTMF_EVENT_TYPE_START;
+
+  event->payload = g_new0 (GstRTPDTMFPayload, 1);
+  event->payload->event = CLAMP (event_number, MIN_EVENT, MAX_EVENT);
+  event->payload->volume = CLAMP (event_volume, MIN_VOLUME, MAX_VOLUME);
+
+  g_async_queue_push (dtmfsrc->event_queue, event);
+}
+
+static void
+gst_rtp_dtmf_src_add_stop_event (GstRTPDTMFSrc *dtmfsrc)
+{
+
+  GstRTPDTMFSrcEvent * event = g_malloc (sizeof(GstRTPDTMFSrcEvent));
+  event->event_type = RTP_DTMF_EVENT_TYPE_STOP;
+  event->payload = g_new0 (GstRTPDTMFPayload, 1);
+  event->payload->event = 0;
+  event->payload->volume = 0;
+
+  g_async_queue_push (dtmfsrc->event_queue, event);
+}
+
+
 static void
 gst_rtp_dtmf_src_wait_for_buffer_ts (GstRTPDTMFSrc *dtmfsrc, GstBuffer * buf)
 {
@@ -629,7 +685,7 @@ gst_rtp_dtmf_src_wait_for_buffer_ts (GstRTPDTMFSrc *dtmfsrc, GstBuffer * buf)
 }
 
 static void
-gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
+gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc, GstRTPDTMFSrcEvent *event, GstBuffer *buf)
 {
   gst_rtp_buffer_set_ssrc (buf, dtmfsrc->current_ssrc);
   gst_rtp_buffer_set_payload_type (buf, dtmfsrc->pt);
@@ -637,7 +693,7 @@ gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
     gst_rtp_buffer_set_marker (buf, TRUE);
     dtmfsrc->first_packet = FALSE;
   } else if (dtmfsrc->last_packet) {
-    dtmfsrc->payload->e = 1;
+    event->payload->e = 1;
     dtmfsrc->last_packet = FALSE;
   }
 
@@ -649,14 +705,14 @@ gst_rtp_dtmf_prepare_rtp_headers (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
 }
 
 static void
-gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
+gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, GstRTPDTMFSrcEvent *event,GstBuffer *buf)
 {
   GstRTPDTMFPayload *payload;
 
-  gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc, buf);
+  gst_rtp_dtmf_prepare_rtp_headers (dtmfsrc,event,  buf);
 
   /* duration of DTMF payload */
-  dtmfsrc->payload->duration +=
+  event->payload->duration +=
       dtmfsrc->interval * dtmfsrc->clock_rate / 1000;
 
   /* timestamp and duration of GstBuffer */
@@ -667,7 +723,7 @@ gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
   payload = (GstRTPDTMFPayload *) gst_rtp_buffer_get_payload (buf);
 
   /* copy payload and convert to network-byte order */
-  g_memmove (payload, dtmfsrc->payload, sizeof (GstRTPDTMFPayload));
+  g_memmove (payload, event->payload, sizeof (GstRTPDTMFPayload));
   /* Force the packet duration to a certain minumum
    * if its the end of the event
    */
@@ -679,18 +735,20 @@ gst_rtp_dtmf_prepare_buffer_data (GstRTPDTMFSrc *dtmfsrc, GstBuffer *buf)
 }
 
 static GstBuffer *
-gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
+gst_rtp_dtmf_src_create_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc, GstRTPDTMFSrcEvent *event)
 {
   GstBuffer *buf = NULL;
 
   /* create buffer to hold the payload */
   buf = gst_rtp_buffer_new_allocate (sizeof (GstRTPDTMFPayload), 0, 0);
 
-  gst_rtp_dtmf_prepare_buffer_data (dtmfsrc, buf);
+  gst_rtp_dtmf_prepare_buffer_data (dtmfsrc, event, buf);
 
   /* FIXME: Should we sync to clock ourselves or leave it to sink */
   gst_rtp_dtmf_src_wait_for_buffer_ts (dtmfsrc, buf);
 
+  event->sent_packets++;
+
   /* Set caps on the buffer before pushing it */
   gst_buffer_set_caps (buf, GST_PAD_CAPS (dtmfsrc->srcpad));
 
@@ -703,6 +761,43 @@ gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
   GstBuffer *buf = NULL;
   GstFlowReturn ret;
   gint redundancy_count = 1;
+  GstRTPDTMFSrcEvent *event;
+
+  g_async_queue_ref (dtmfsrc->event_queue);
+
+  if (dtmfsrc->last_event == NULL) {
+    event = g_async_queue_pop (dtmfsrc->event_queue);
+
+    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;
@@ -720,7 +815,7 @@ gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
     }
 
     /* create buffer to hold the payload */
-  buf = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc);
+    buf = gst_rtp_dtmf_src_create_next_rtp_packet (dtmfsrc, dtmfsrc->last_event);
 
     while ( redundancy_count-- ) {
       gst_buffer_ref(buf);
@@ -731,7 +826,7 @@ gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
       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));
+	    "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);
@@ -739,22 +834,20 @@ gst_rtp_dtmf_src_push_next_rtp_packet (GstRTPDTMFSrc *dtmfsrc)
 
     gst_buffer_unref(buf);
     GST_DEBUG_OBJECT (dtmfsrc,
-          "pushed DTMF event '%d' on src pad", dtmfsrc->payload->event);
+	"pushed DTMF event '%d' on src pad", event->payload->event);
 
-  if (dtmfsrc->payload->e) {
+    if (dtmfsrc->last_event->payload->e) {
       /* Don't forget to release the stream lock */
       gst_rtp_dtmf_src_set_stream_lock (dtmfsrc, FALSE);
 
-    g_free (dtmfsrc->payload);
+      g_free (dtmfsrc->last_event->payload);
-    dtmfsrc->payload = NULL;
+      event->payload = NULL;
 
-    if (!gst_pad_pause_task (dtmfsrc->srcpad)) {
+      g_free (dtmfsrc->last_event);
-      GST_ERROR_OBJECT (dtmfsrc, "Failed to pause task on src pad");
+      dtmfsrc->last_event = NULL;
-      return;
-    }
 
     }
-
+  }
 }
 
 static void
@@ -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;

gst/dtmf/gstrtpdtmfsrc.h

@@ -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.
@@ -67,27 +84,25 @@ struct _GstRTPDTMFSrc {
   GstElement           element;
 
   GstPad*	       srcpad;
-  GstRTPDTMFPayload *payload;
+  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 {