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Simplify fix15().

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Turn it into 2 macros, and use av_clip_int16() and lrintf().
This matches the int16 to float sample conversion in audioconvert.c.
The regression test output is different due to lrintf() rounding.

Originally committed as revision 25956 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Justin Ruggles 2010-12-14 14:51:02 +00:00
parent 228315bbd3
commit 918cd2255c
2 changed files with 8 additions and 15 deletions
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21
libavcodec/ac3enc.c
View file

@ -33,6 +33,8 @@
#include "ac3.h"
#include "audioconvert.h"
#define SCALE_FLOAT(a, bits) lrintf((a) * (float)(1 << (bits)))
typedef struct AC3EncodeContext {
PutBitContext pb; ///< bitstream writer context
@ -86,16 +88,7 @@ static int16_t xsin1[128];
/* new exponents are sent if their Norm 1 exceed this number */
#define EXP_DIFF_THRESHOLD 1000
static inline int16_t fix15(float a)
{
int v;
v = (int)(a * (float)(1 << 15));
if (v < -32767)
v = -32767;
else if (v > 32767)
v = 32767;
return v;
}
#define FIX15(a) av_clip_int16(SCALE_FLOAT(a, 15))
typedef struct IComplex {
int16_t re,im;
@ -110,8 +103,8 @@ static av_cold void fft_init(int ln)
for(i=0;i<(n/2);i++) {
alpha = 2 * M_PI * (float)i / (float)n;
costab[i] = fix15(cos(alpha));
sintab[i] = fix15(sin(alpha));
costab[i] = FIX15(cos(alpha));
sintab[i] = FIX15(sin(alpha));
}
}
@ -126,8 +119,8 @@ static av_cold void mdct_init(int nbits)
for(i=0;i<n4;i++) {
alpha = 2 * M_PI * (i + 1.0 / 8.0) / n;
xcos1[i] = fix15(-cos(alpha));
xsin1[i] = fix15(-sin(alpha));
xcos1[i] = FIX15(-cos(alpha));
xsin1[i] = FIX15(-sin(alpha));
}
}

2
tests/ref/acodec/ac3
View file

@ -1,2 +1,2 @@
081bfd5e683b13cccf1b07c43c1c5005 *./tests/data/acodec/ac3.rm
464093173530736050e338caf152044c *./tests/data/acodec/ac3.rm
98751 ./tests/data/acodec/ac3.rm