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>

9 files changed, 42 insertions, 14 deletions

diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 440609928f..b7240a2d51 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -385,7 +385,7 @@ RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext
             av_free(s);
             return NULL;
         }
-    } else {
+    } else if (st) {
         switch(st->codec->codec_id) {
         case CODEC_ID_MPEG1VIDEO:
         case CODEC_ID_MPEG2VIDEO:
diff --git a/libavformat/rtpdec_amr.c b/libavformat/rtpdec_amr.c
index de824904aa..86033cf8d2 100644
--- a/libavformat/rtpdec_amr.c
+++ b/libavformat/rtpdec_amr.c
@@ -169,6 +169,9 @@ static int amr_parse_sdp_line(AVFormatContext *s, int st_index,
     const char *p;
     int ret;
 
+    if (st_index < 0)
+        return 0;
+
     /* Parse an fmtp line this one:
      * a=fmtp:97 octet-align=1; interleaving=0
      * That is, a normal fmtp: line followed by semicolon & space
diff --git a/libavformat/rtpdec_asf.c b/libavformat/rtpdec_asf.c
index 65b466669d..84503f1749 100644
--- a/libavformat/rtpdec_asf.c
+++ b/libavformat/rtpdec_asf.c
@@ -130,6 +130,8 @@ int ff_wms_parse_sdp_a_line(AVFormatContext *s, const char *p)
 static int asfrtp_parse_sdp_line(AVFormatContext *s, int stream_index,
                                  PayloadContext *asf, const char *line)
 {
+    if (stream_index < 0)
+        return 0;
     if (av_strstart(line, "stream:", &line)) {
         RTSPState *rt = s->priv_data;
 
diff --git a/libavformat/rtpdec_h264.c b/libavformat/rtpdec_h264.c
index 01514c1bef..0366ba0205 100644
--- a/libavformat/rtpdec_h264.c
+++ b/libavformat/rtpdec_h264.c
@@ -357,10 +357,15 @@ static void h264_free_context(PayloadContext *data)
 static int parse_h264_sdp_line(AVFormatContext *s, int st_index,
                                PayloadContext *h264_data, const char *line)
 {
-    AVStream *stream = s->streams[st_index];
-    AVCodecContext *codec = stream->codec;
+    AVStream *stream;
+    AVCodecContext *codec;
     const char *p = line;
 
+    if (st_index < 0)
+        return 0;
+
+    stream = s->streams[st_index];
+    codec = stream->codec;
     assert(h264_data->cookie == MAGIC_COOKIE);
 
     if (av_strstart(p, "framesize:", &p)) {
diff --git a/libavformat/rtpdec_latm.c b/libavformat/rtpdec_latm.c
index a642b5a079..cc24c82c50 100644
--- a/libavformat/rtpdec_latm.c
+++ b/libavformat/rtpdec_latm.c
@@ -168,6 +168,9 @@ static int latm_parse_sdp_line(AVFormatContext *s, int st_index,
 {
     const char *p;
 
+    if (st_index < 0)
+        return 0;
+
     if (av_strstart(line, "fmtp:", &p))
         return ff_parse_fmtp(s->streams[st_index], data, p, parse_fmtp);
 
diff --git a/libavformat/rtpdec_mpeg4.c b/libavformat/rtpdec_mpeg4.c
index 9b388782b3..b11cc79842 100644
--- a/libavformat/rtpdec_mpeg4.c
+++ b/libavformat/rtpdec_mpeg4.c
@@ -223,6 +223,9 @@ static int parse_sdp_line(AVFormatContext *s, int st_index,
 {
     const char *p;
 
+    if (st_index < 0)
+        return 0;
+
     if (av_strstart(line, "fmtp:", &p))
         return ff_parse_fmtp(s->streams[st_index], data, p, parse_fmtp);
 
diff --git a/libavformat/rtpdec_xiph.c b/libavformat/rtpdec_xiph.c
index 6aeea2fd3c..771a13d9d2 100644
--- a/libavformat/rtpdec_xiph.c
+++ b/libavformat/rtpdec_xiph.c
@@ -372,10 +372,13 @@ static int xiph_parse_fmtp_pair(AVStream* stream,
 }
 
 static int xiph_parse_sdp_line(AVFormatContext *s, int st_index,
-                                 PayloadContext *data, const char *line)
+                               PayloadContext *data, const char *line)
 {
     const char *p;
 
+    if (st_index < 0)
+        return 0;
+
     if (av_strstart(line, "fmtp:", &p)) {
         return ff_parse_fmtp(s->streams[st_index], data, p,
                              xiph_parse_fmtp_pair);
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)
diff --git a/libavformat/rtspdec.c b/libavformat/rtspdec.c
index ea2d517e3c..9c7df96a8f 100644
--- a/libavformat/rtspdec.c
+++ b/libavformat/rtspdec.c
@@ -159,8 +159,9 @@ static int rtsp_read_header(AVFormatContext *s)
     if (ret)
         return ret;
 
-    rt->real_setup_cache = av_mallocz(2 * s->nb_streams * sizeof(*rt->real_setup_cache));
-    if (!rt->real_setup_cache)
+    rt->real_setup_cache = !s->nb_streams ? NULL :
+                           av_mallocz(2 * s->nb_streams * sizeof(*rt->real_setup_cache));
+    if (!rt->real_setup_cache && s->nb_streams)
         return AVERROR(ENOMEM);
     rt->real_setup = rt->real_setup_cache + s->nb_streams;