RTP: apply de-jitter from last dequeuing rather than last queuing

Previously, if the absolute delay dropped by a longer interval than the
packetization time, we could end up repeatedly waiting into the future,
and never dequeue anything.
(cherry picked from commit 4f49f2aa)

Conflicts:

	modules/access/rtp/session.c

modules/access/rtp/session.c

@@ -146,6 +146,7 @@ struct rtp_source_t
 
     uint16_t last_seq; /* sequence of the next dequeued packet */
     block_t *blocks; /* re-ordered blocks queue */
+    mtime_t  last_dq; /* last dequeued (decoded) packet local timestamp */
     void    *opaque[0]; /* Per-source private payload data */
 };
 
@@ -314,6 +315,7 @@ rtp_queue (demux_t *demux, rtp_session_t *session, block_t *block)
         }
     }
     src->last_rx = now;
+    block->i_pts = now;
     src->last_ts = rtp_timestamp (block);
 
     /* Check sequence number */
@@ -410,6 +412,7 @@ 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->last_dq = block->i_pts;
     block->i_pts = UINT64_C(1) * CLOCK_FREQ * timestamp / pt->frequency;
 
     /* CSRC count */
@@ -487,7 +490,7 @@ bool rtp_dequeue (demux_t *demux, const rtp_session_t *session,
              * match for random gaussian jitter). Additionnaly, we implicitly
              * wait for misordering times the packetization time.
              */
-            mtime_t deadline = src->last_rx;
+            mtime_t deadline = src->last_dq;
             const rtp_pt_t *pt = rtp_find_ptype (session, src, block, NULL);
             if (pt)
                 deadline += UINT64_C(3) * CLOCK_FREQ * src->jitter