vqa: use 1/sample_rate as the audio stream time base

[ffmpeg] / libavformat / nsvdec.c

diff --git a/libavformat/nsvdec.c b/libavformat/nsvdec.c
index d0fac9afb03c9ec327b8784f68585e97a8da4498..18dfde2867ee42f140192039b029d63fcaffab78 100644 (file)
--- a/libavformat/nsvdec.c
+++ b/libavformat/nsvdec.c
@@ -21,6 +21,7 @@
 
 #include "libavutil/mathematics.h"
 #include "avformat.h"
+#include "internal.h"
 #include "riff.h"
 #include "libavutil/dict.h"
 
@@ -457,7 +458,7 @@ static int nsv_parse_NSVs_header(AVFormatContext *s, AVFormatParameters *ap)
             st->codec->height = vheight;
             st->codec->bits_per_coded_sample = 24; /* depth XXX */
 
-            av_set_pts_info(st, 64, framerate.den, framerate.num);
+            avpriv_set_pts_info(st, 64, framerate.den, framerate.num);
             st->start_time = 0;
             st->duration = av_rescale(nsv->duration, framerate.num, 1000*framerate.den);
 
@@ -489,7 +490,7 @@ static int nsv_parse_NSVs_header(AVFormatContext *s, AVFormatParameters *ap)
             st->need_parsing = AVSTREAM_PARSE_FULL; /* for PCM we will read a chunk later and put correct info */
 
             /* set timebase to common denominator of ms and framerate */
-            av_set_pts_info(st, 64, 1, framerate.num*1000);
+            avpriv_set_pts_info(st, 64, 1, framerate.num*1000);
             st->start_time = 0;
             st->duration = (int64_t)nsv->duration * framerate.num;
 #endif