Merge commit '1cd432e167b1a80853760c89a33606e2b5f229c2'

* commit '1cd432e167b1a80853760c89a33606e2b5f229c2':
  configure: fix libcdio check
  rtsp: Allow setting the reordering buffer size via an AVOption
  rtsp: Vertically align a constant definition
  rtp: Update the check for distinguishing between RTP and RTCP
  aac: fix build with hardcoded tables
  fate: dependencies for screen codec tests
  riff: Move functions around to be covered by appropriate #ifdefs

Conflicts:
	configure
	tests/fate/screen.mak

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 14 insertions, 2 deletions

diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 80f1047c5d..0328d28989 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -75,6 +75,9 @@
     { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
     { "data", "Data", 0, AV_OPT_TYPE_CONST, {.i64 = 1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
 
+#define RTSP_REORDERING_OPTS() \
+    { "reorder_queue_size", "Number of packets to buffer for handling of reordered packets", OFFSET(reordering_queue_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, DEC }
+
 const AVOption ff_rtsp_options[] = {
     { "initial_pause",  "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, DEC },
     FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
@@ -88,17 +91,20 @@ const AVOption ff_rtsp_options[] = {
     { "min_port", "Minimum local UDP port", OFFSET(rtp_port_min), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MIN}, 0, 65535, DEC|ENC },
     { "max_port", "Maximum local UDP port", OFFSET(rtp_port_max), AV_OPT_TYPE_INT, {.i64 = RTSP_RTP_PORT_MAX}, 0, 65535, DEC|ENC },
     { "timeout", "Maximum timeout (in seconds) to wait for incoming connections. -1 is infinite. Implies flag listen", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = -1}, INT_MIN, INT_MAX, DEC },
+    RTSP_REORDERING_OPTS(),
     { NULL },
 };
 
 static const AVOption sdp_options[] = {
     RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
     RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
+    RTSP_REORDERING_OPTS(),
     { NULL },
 };
 
 static const AVOption rtp_options[] = {
     RTSP_FLAG_OPTS("rtp_flags", "RTP flags"),
+    RTSP_REORDERING_OPTS(),
     { NULL },
 };
 
@@ -608,6 +614,13 @@ int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
 {
     RTSPState *rt = s->priv_data;
     AVStream *st = NULL;
+    int reordering_queue_size = rt->reordering_queue_size;
+    if (reordering_queue_size < 0) {
+        if (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
+            reordering_queue_size = 0;
+        else
+            reordering_queue_size = RTP_REORDER_QUEUE_DEFAULT_SIZE;
+    }
 
     /* open the RTP context */
     if (rtsp_st->stream_index >= 0)
@@ -632,8 +645,7 @@ int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
     else if (CONFIG_RTPDEC)
         rtsp_st->transport_priv = ff_rtp_parse_open(s, st, rtsp_st->rtp_handle,
                                          rtsp_st->sdp_payload_type,
-            (rt->lower_transport == RTSP_LOWER_TRANSPORT_TCP || !s->max_delay)
-            ? 0 : RTP_REORDER_QUEUE_DEFAULT_SIZE);
+                                         reordering_queue_size);
 
     if (!rtsp_st->transport_priv) {
          return AVERROR(ENOMEM);