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| author | Michael Niedermayer <michaelni@gmx.at> | 2006-08-22 11:11:04 +0000 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2006-08-22 11:11:04 +0000 |
| commit | 355903f5a8ba254b439903a77d75f1d9c0d3f092 (patch) | |
| tree | 0ec30a95c92241d896d3b577c3786c698f5f9409 | |
| parent | 0bd2483a7553bfd60af3d8f137cab02795d59771 (diff) | |
make high precission synth filter 3 times faster on x86
Originally committed as revision 6046 to svn://svn.ffmpeg.org/ffmpeg/trunk
| -rw-r--r-- | libavcodec/mpegaudiodec.c | 7 |
1 files changed, 6 insertions, 1 deletions
diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c index 53a3890667..9593e03480 100644 --- a/libavcodec/mpegaudiodec.c +++ b/libavcodec/mpegaudiodec.c @@ -784,8 +784,13 @@ static inline int round_sample(int64_t *sum) return sum1; } +#ifdef ARCH_X86 +/* ask gcc devels why this is 3 times faster then the generic code below */ +#define MULS(ra, rb) \ + ({ int64_t rt; asm ("imull %2\n\t" : "=A"(rt) : "a" (ra), "g" (rb)); rt; }) +#else #define MULS(ra, rb) MUL64(ra, rb) - +#endif #endif #define SUM8(sum, op, w, p) \ |