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-rw-r--r--libhb/decavcodec.c10
1 files changed, 5 insertions, 5 deletions
diff --git a/libhb/decavcodec.c b/libhb/decavcodec.c
index dae2544c2..2b7ddcbaa 100644
--- a/libhb/decavcodec.c
+++ b/libhb/decavcodec.c
@@ -233,7 +233,7 @@ static void decavcodecClose( hb_work_object_t * w )
}
if ( pv->buffer )
{
- free( pv->buffer );
+ av_free( pv->buffer );
pv->buffer = NULL;
}
free( pv );
@@ -339,7 +339,7 @@ static int decavcodecWork( hb_work_object_t * w, hb_buffer_t ** buf_in,
pv->pts_next = cur;
- free( bufaligned );
+ av_free( bufaligned );
return HB_WORK_OK;
}
@@ -429,7 +429,7 @@ static int decavcodecBSInfo( hb_work_object_t *w, const hb_buffer_t *buf,
}
}
}
- free( buffer );
+ av_free( buffer );
av_parser_close( parser );
hb_avcodec_close( context );
return ret;
@@ -1160,7 +1160,7 @@ static void decodeAudio( hb_work_private_t *pv, uint8_t *data, int size )
NULL, 0 );
// get output buffer size (in 2-byte samples) then malloc a buffer
out_size = ( out_size * 2 ) / isamp;
- buffer = malloc( out_size );
+ buffer = av_malloc( out_size );
// we're doing straight sample format conversion which behaves as if
// there were only one channel.
@@ -1194,7 +1194,7 @@ static void decodeAudio( hb_work_private_t *pv, uint8_t *data, int size )
// if we allocated a buffer for sample format conversion, free it
if ( buffer != pv->buffer )
{
- free( buffer );
+ av_free( buffer );
}
}
}