avformat/hlsenc: set default http method to PUT when method is null

[ffmpeg] / libavformat / 4xm.c

diff --git a/libavformat/4xm.c b/libavformat/4xm.c
index 8a50778686f871f3142d32118b407fdb0840496b..ead6d2b4243a83d80c3931c03605f27a5bb3fab0 100644 (file)
--- a/libavformat/4xm.c
+++ b/libavformat/4xm.c
@@ -29,6 +29,7 @@
 
 #include "libavutil/intreadwrite.h"
 #include "libavutil/intfloat.h"
+#include "libavcodec/internal.h"
 #include "avformat.h"
 #include "internal.h"
 
@@ -153,8 +154,10 @@ static int parse_strk(AVFormatContext *s,
     fourxm->tracks[track].audio_pts   = 0;
 
     if (fourxm->tracks[track].channels    <= 0 ||
+        fourxm->tracks[track].channels     > FF_SANE_NB_CHANNELS ||
         fourxm->tracks[track].sample_rate <= 0 ||
-        fourxm->tracks[track].bits        <= 0) {
+        fourxm->tracks[track].bits        <= 0 ||
+        fourxm->tracks[track].bits         > INT_MAX / FF_SANE_NB_CHANNELS) {
         av_log(s, AV_LOG_ERROR, "audio header invalid\n");
         return AVERROR_INVALIDDATA;
     }
@@ -163,6 +166,12 @@ static int parse_strk(AVFormatContext *s,
         return AVERROR_INVALIDDATA;
     }
 
+    if (fourxm->tracks[track].sample_rate > INT64_MAX / fourxm->tracks[track].bits / fourxm->tracks[track].channels) {
+        av_log(s, AV_LOG_ERROR, "Overflow during bit rate calculation %d * %d * %d\n",
+               fourxm->tracks[track].sample_rate, fourxm->tracks[track].bits, fourxm->tracks[track].channels);
+        return AVERROR_INVALIDDATA;
+    }
+
     /* allocate a new AVStream */
     st = avformat_new_stream(s, NULL);
     if (!st)
@@ -178,7 +187,7 @@ static int parse_strk(AVFormatContext *s,
     st->codecpar->channels              = fourxm->tracks[track].channels;
     st->codecpar->sample_rate           = fourxm->tracks[track].sample_rate;
     st->codecpar->bits_per_coded_sample = fourxm->tracks[track].bits;
-    st->codecpar->bit_rate              = st->codecpar->channels *
+    st->codecpar->bit_rate              = (int64_t)st->codecpar->channels *
                                           st->codecpar->sample_rate *
                                           st->codecpar->bits_per_coded_sample;
     st->codecpar->block_align           = st->codecpar->channels *