[FFmpeg-cvslog] lavr: x86: optimized 2-channel flt to s16p conversion
Justin Ruggles
git at videolan.org
Fri Aug 24 14:44:31 CEST 2012
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Wed May 2 18:21:26 2012 -0400| [31d0d7181d26a3c3324f604e8b5b7548b59a2091] | committer: Justin Ruggles
lavr: x86: optimized 2-channel flt to s16p conversion
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=31d0d7181d26a3c3324f604e8b5b7548b59a2091
---
libavresample/x86/audio_convert.asm | 46 ++++++++++++++++++++++++++++++++
libavresample/x86/audio_convert_init.c | 9 +++++++
2 files changed, 55 insertions(+)
diff --git a/libavresample/x86/audio_convert.asm b/libavresample/x86/audio_convert.asm
index 12bd577..730f16a 100644
--- a/libavresample/x86/audio_convert.asm
+++ b/libavresample/x86/audio_convert.asm
@@ -1045,3 +1045,49 @@ CONV_S16_TO_FLTP_6CH
INIT_XMM avx
CONV_S16_TO_FLTP_6CH
%endif
+
+;------------------------------------------------------------------------------
+; void ff_conv_flt_to_s16p_2ch(int16_t *const *dst, float *src, int len,
+; int channels);
+;------------------------------------------------------------------------------
+
+%macro CONV_FLT_TO_S16P_2CH 0
+cglobal conv_flt_to_s16p_2ch, 3,4,6, dst0, src, len, dst1
+ lea lenq, [2*lend]
+ mov dst1q, [dst0q+gprsize]
+ mov dst0q, [dst0q ]
+ lea srcq, [srcq+4*lenq]
+ add dst0q, lenq
+ add dst1q, lenq
+ neg lenq
+ mova m5, [pf_s16_scale]
+.loop:
+ mova m0, [srcq+4*lenq ]
+ mova m1, [srcq+4*lenq+ mmsize]
+ mova m2, [srcq+4*lenq+2*mmsize]
+ mova m3, [srcq+4*lenq+3*mmsize]
+ DEINT2_PS 0, 1, 4
+ DEINT2_PS 2, 3, 4
+ mulps m0, m0, m5
+ mulps m1, m1, m5
+ mulps m2, m2, m5
+ mulps m3, m3, m5
+ cvtps2dq m0, m0
+ cvtps2dq m1, m1
+ cvtps2dq m2, m2
+ cvtps2dq m3, m3
+ packssdw m0, m2
+ packssdw m1, m3
+ mova [dst0q+lenq], m0
+ mova [dst1q+lenq], m1
+ add lenq, mmsize
+ jl .loop
+ REP_RET
+%endmacro
+
+INIT_XMM sse2
+CONV_FLT_TO_S16P_2CH
+%if HAVE_AVX
+INIT_XMM avx
+CONV_FLT_TO_S16P_2CH
+%endif
diff --git a/libavresample/x86/audio_convert_init.c b/libavresample/x86/audio_convert_init.c
index 8d15824..f32ca24 100644
--- a/libavresample/x86/audio_convert_init.c
+++ b/libavresample/x86/audio_convert_init.c
@@ -120,6 +120,11 @@ extern void ff_conv_s16_to_fltp_6ch_sse4 (float *const *dst, int16_t *src,
extern void ff_conv_s16_to_fltp_6ch_avx (float *const *dst, int16_t *src,
int len, int channels);
+extern void ff_conv_flt_to_s16p_2ch_sse2(int16_t *const *dst, float *src,
+ int len, int channels);
+extern void ff_conv_flt_to_s16p_2ch_avx (int16_t *const *dst, float *src,
+ int len, int channels);
+
av_cold void ff_audio_convert_init_x86(AudioConvert *ac)
{
#if HAVE_YASM
@@ -175,6 +180,8 @@ av_cold void ff_audio_convert_init_x86(AudioConvert *ac)
2, 16, 8, "SSE2", ff_conv_s16_to_fltp_2ch_sse2);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16,
6, 16, 4, "SSE2", ff_conv_s16_to_fltp_6ch_sse2);
+ ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLT,
+ 2, 16, 8, "SSE2", ff_conv_flt_to_s16p_2ch_sse2);
}
if (mm_flags & AV_CPU_FLAG_SSSE3 && HAVE_SSE) {
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16P,
@@ -219,6 +226,8 @@ av_cold void ff_audio_convert_init_x86(AudioConvert *ac)
2, 16, 8, "AVX", ff_conv_s16_to_fltp_2ch_avx);
ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16,
6, 16, 4, "AVX", ff_conv_s16_to_fltp_6ch_avx);
+ ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLT,
+ 2, 16, 8, "AVX", ff_conv_flt_to_s16p_2ch_avx);
}
#endif
}
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