Merge remote-tracking branch 'qatar/master'

* qatar/master:
  dnxhddec: optimise dnxhd_decode_dct_block()
  rtp: remove disabled code
  eac3enc: use different numbers of blocks per frame to allow higher bitrates
  dnxhd: add regression test for 10-bit
  dnxhd: 10-bit support
  dsputil: update per-arch init funcs for non-h264 high bit depth
  dsputil: template get_pixels() for different bit depths
  dsputil: create 16/32-bit dctcoef versions of some functions
  jfdctint: add 10-bit version
  mov: add clcp type track as Subtitle stream.
  mpeg4: add Mpeg4 Profiles names.
  mpeg4: decode Level Profile for MPEG4 Part 2.
  ffprobe: display bitstream level.
  imgconvert: remove unused glue and xglue macros

Conflicts:
	libavcodec/dsputil_template.c

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

3 files changed, 1 insertions, 40 deletions

diff --git a/libavformat/mov.c b/libavformat/mov.c
index f26162da4f..fdb29a01bf 100644
--- a/libavformat/mov.c
+++ b/libavformat/mov.c
@@ -456,7 +456,7 @@ static int mov_read_hdlr(MOVContext *c, AVIOContext *pb, MOVAtom atom)
         st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
     else if(type == MKTAG('m','1','a',' '))
         st->codec->codec_id = CODEC_ID_MP2;
-    else if(type == MKTAG('s','u','b','p'))
+    else if((type == MKTAG('s','u','b','p')) || (type == MKTAG('c','l','c','p')))
         st->codec->codec_type = AVMEDIA_TYPE_SUBTITLE;
 
     avio_rb32(pb); /* component  manufacture */
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 995f3bffd4..f9252a9bb3 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -218,22 +218,6 @@ static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
     return 1;
 }
 
-#if 0
-/**
-* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
-* difference between the arrival and sent timestamp.  As a result, the jitter and transit statistics values
-* never change.  I left this in in case someone else can see a way. (rdm)
-*/
-static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
-{
-    uint32_t transit= arrival_timestamp - sent_timestamp;
-    int d;
-    s->transit= transit;
-    d= FFABS(transit - s->transit);
-    s->jitter += d - ((s->jitter + 8)>>4);
-}
-#endif
-
 int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
 {
     AVIOContext *pb;
diff --git a/libavformat/rtpproto.c b/libavformat/rtpproto.c
index d4d7f3cb85..6e76825d24 100644
--- a/libavformat/rtpproto.c
+++ b/libavformat/rtpproto.c
@@ -225,20 +225,6 @@ static int rtp_read(URLContext *h, uint8_t *buf, int size)
     int len, n;
     struct pollfd p[2] = {{s->rtp_fd, POLLIN, 0}, {s->rtcp_fd, POLLIN, 0}};
 
-#if 0
-    for(;;) {
-        from_len = sizeof(from);
-        len = recvfrom (s->rtp_fd, buf, size, 0,
-                        (struct sockaddr *)&from, &from_len);
-        if (len < 0) {
-            if (ff_neterrno() == AVERROR(EAGAIN) ||
-                ff_neterrno() == AVERROR(EINTR))
-                continue;
-            return AVERROR(EIO);
-        }
-        break;
-    }
-#else
     for(;;) {
         if (url_interrupt_cb())
             return AVERROR_EXIT;
@@ -277,7 +263,6 @@ static int rtp_read(URLContext *h, uint8_t *buf, int size)
             return AVERROR(EIO);
         }
     }
-#endif
     return len;
 }
 
@@ -296,14 +281,6 @@ static int rtp_write(URLContext *h, const uint8_t *buf, int size)
     }
 
     ret = ffurl_write(hd, buf, size);
-#if 0
-    {
-        struct timespec ts;
-        ts.tv_sec = 0;
-        ts.tv_nsec = 10 * 1000000;
-        nanosleep(&ts, NULL);
-    }
-#endif
     return ret;
 }