RTP: compute deadline from RX time of oldest packet in queue

Jitter computation is, by design, relative to the time of arrival of
packets. Normally, it should be the estimated time of arrival (ETA) of
the missing packet in the sequence, plus a tolerance value computed
from the delay variance (jitter). Unfortunately, we do not know that
ETA since we do not know the RTP timestamp of missing packets.

diff --git a/modules/access/rtp/session.c b/modules/access/rtp/session.c
index fc8d6759155ae1a9a054c36beaf6bc7abbb6cd88..22f127ea3385a12559cb7f085508cc956831100c 100644 (file)
--- a/modules/access/rtp/session.c
+++ b/modules/access/rtp/session.c
@@ -146,7 +146,6 @@ struct rtp_source_t
 
     uint16_t last_seq; /* sequence of the next dequeued packet */
     block_t *blocks; /* re-ordered blocks queue */
-    mtime_t  ref_ts; /* reference timestamp for reordering */
     void    *opaque[0]; /* Per-source private payload data */
 };
 
@@ -165,7 +164,6 @@ rtp_source_create (demux_t *demux, const rtp_session_t *session,
 
     source->ssrc = ssrc;
     source->jitter = 0;
-    source->ref_ts = 0;
     source->max_seq = source->bad_seq = init_seq;
     source->last_seq = init_seq - 1;
     source->blocks = NULL;
@@ -316,6 +314,7 @@ rtp_queue (demux_t *demux, rtp_session_t *session, block_t *block)
         }
     }
     src->last_rx = now;
+    block->i_pts = now; /* store reception time until dequeued */
     src->last_ts = rtp_timestamp (block);
 
     /* Check sequence number */
@@ -412,7 +411,6 @@ rtp_decode (demux_t *demux, const rtp_session_t *session, rtp_source_t *src)
     /* FIXME: handle timestamp wrap properly */
     /* TODO: inter-medias/sessions sync (using RTCP-SR) */
     const uint32_t timestamp = rtp_timestamp (block);
-    src->ref_ts = 0;
     block->i_pts = CLOCK_FREQ * timestamp / pt->frequency;
 
     /* CSRC count */
@@ -489,19 +487,25 @@ bool rtp_dequeue (demux_t *demux, const rtp_session_t *session,
             }
 
             /* Wait for 3 times the inter-arrival delay variance (about 99.7%
-             * match for random gaussian jitter). Additionnaly, we implicitly
-             * wait for misordering times the packetization time.
+             * match for random gaussian jitter).
              */
-            mtime_t deadline = src->ref_ts;
+            mtime_t deadline;
             const rtp_pt_t *pt = rtp_find_ptype (session, src, block, NULL);
-            if (!deadline)
-                deadline = src->ref_ts = now;
             if (pt)
-                deadline += CLOCK_FREQ * 3 * src->jitter / pt->frequency;
+                deadline = CLOCK_FREQ * 3 * src->jitter / pt->frequency;
+            else
+                deadline = 0; /* no jitter estimate with no frequency :( */
 
             /* Make sure we wait at least for 25 msec */
-            deadline = __MAX(deadline, src->ref_ts + CLOCK_FREQ / 40);
-
+            if (deadline < (CLOCK_FREQ / 40))
+                deadline = CLOCK_FREQ / 40;
+
+            /* Additionnaly, we implicitly wait for the packetization time
+             * multiplied by the number of missing packets. block is the first
+             * non-missing packet (lowest sequence number). We have no better
+             * estimated time of arrival, as we do not know the RTP timestamp
+             * of not yet received packets. */
+            deadline += block->i_pts;
             if (now >= deadline)
             {
                 rtp_decode (demux, session, src);