Merge remote-tracking branch 'qatar/master'

* qatar/master:
  rtsp: Don't use av_malloc(0) if there are no streams
  rtsp: Don't use uninitialized data if there are no streams
  vaapi: mpeg2: fix slice_vertical_position calculation.
  hwaccel: mpeg2: decode first field, if requested.
  cosmetics: Fix indentation
  rtsp: Don't expose the MS-RTSP RTX data stream to the caller

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

1 files changed, 16 insertions, 8 deletions

diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 9961b5a33c..fb6df692ac 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -374,6 +374,10 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
 
         if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
             /* no corresponding stream */
+        } else if (rt->server_type == RTSP_SERVER_WMS &&
+                   codec_type == AVMEDIA_TYPE_DATA) {
+            /* RTX stream, a stream that carries all the other actual
+             * audio/video streams. Don't expose this to the callers. */
         } else {
             st = avformat_new_stream(s, NULL);
             if (!st)
@@ -430,9 +434,11 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
             /* NOTE: rtpmap is only supported AFTER the 'm=' tag */
             get_word(buf1, sizeof(buf1), &p);
             payload_type = atoi(buf1);
-            st = s->streams[s->nb_streams - 1];
             rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
-            sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
+            if (rtsp_st->stream_index >= 0) {
+                st = s->streams[rtsp_st->stream_index];
+                sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
+            }
         } else if (av_strstart(p, "fmtp:", &p) ||
                    av_strstart(p, "framesize:", &p)) {
             /* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
@@ -467,14 +473,15 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
             if (rt->server_type == RTSP_SERVER_WMS)
                 ff_wms_parse_sdp_a_line(s, p);
             if (s->nb_streams > 0) {
+                rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
+
                 if (rt->server_type == RTSP_SERVER_REAL)
-                    ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
+                    ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
 
-                rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
                 if (rtsp_st->dynamic_handler &&
                     rtsp_st->dynamic_handler->parse_sdp_a_line)
                     rtsp_st->dynamic_handler->parse_sdp_a_line(s,
-                        s->nb_streams - 1,
+                        rtsp_st->stream_index,
                         rtsp_st->dynamic_protocol_context, buf);
             }
         }
@@ -1245,8 +1252,9 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
              * UDP. When trying to set it up for TCP streams, the server
              * will return an error. Therefore, we skip those streams. */
             if (rt->server_type == RTSP_SERVER_WMS &&
-                s->streams[rtsp_st->stream_index]->codec->codec_type ==
-                    AVMEDIA_TYPE_DATA)
+                (rtsp_st->stream_index < 0 ||
+                 s->streams[rtsp_st->stream_index]->codec->codec_type ==
+                    AVMEDIA_TYPE_DATA))
                 continue;
             snprintf(transport, sizeof(transport) - 1,
                      "%s/TCP;", trans_pref);
@@ -1378,7 +1386,7 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
             goto fail;
     }
 
-    if (reply->timeout > 0)
+    if (rt->nb_rtsp_streams && reply->timeout > 0)
         rt->timeout = reply->timeout;
 
     if (rt->server_type == RTSP_SERVER_REAL)