On 9/12/2022 3:39 PM, James Almer wrote:
From 55eb5a18b4bf029f52f9d9108a750c576ba780ee Mon Sep 17 00:00:00 2001
From: Paul B Mahol <one...@gmail.com>
Date: Mon, 12 Sep 2022 18:53:31 +0200
Subject: [PATCH] avcodec/x86/audiodsp: add scalarproduct avx2

Signed-off-by: Paul B Mahol <one...@gmail.com>
---
 libavcodec/x86/audiodsp.asm    | 24 ++++++++++++++++++++++++
 libavcodec/x86/audiodsp_init.c |  6 ++++++
 2 files changed, 30 insertions(+)

diff --git a/libavcodec/x86/audiodsp.asm b/libavcodec/x86/audiodsp.asm
index b604b0443c..55051f6aa7 100644
--- a/libavcodec/x86/audiodsp.asm
+++ b/libavcodec/x86/audiodsp.asm
@@ -44,6 +44,30 @@ cglobal scalarproduct_int16, 3,3,3, v1, v2, order
     movd   eax, m2
     RET

+INIT_YMM avx2
+cglobal scalarproduct_int16, 3,4,3, v1, v2, order, offset
+    xor offsetq, offsetq
+    add orderd, orderd
+    pxor    m1, m1
+    cmp orderd, 32

This parameter needs to be multiple of 16. What will happen below if it's for example 48? Are both buffers padded enough to handle 16 bytes of overread?

Nevermind, it's int16_t* buffers.

You can simplify this as:

INIT_YMM avx2
cglobal scalarproduct_int16, 3,3,3, v1, v2, order
    add orderd, orderd
    add v1q, orderq
    add v2q, orderq
    neg orderq
    pxor    m1, m1
.loop:
    movu    m0, [v1q + orderq]
    pmaddwd m0, [v2q + orderq]
    paddd   m1, m0
    add     orderq, mmsize
    jl .loop
    HADDD   m1, m0
    movd   eax, xm1
    RET


+    jl   .l16
+.loop:
+    movu    m0, [v1q + offsetq]
+    pmaddwd m0, [v2q + offsetq]
+    paddd   m1, m0
+    add     offsetq, mmsize
+    cmp     offsetq, orderq

You should use the neg trick from the sse2 version so you can remove the cmp from this loop.

+    jl .loop
+    HADDD   m1, m0
+    movd   eax, xm1
+    RET
+.l16:
+    movu    xm0, [v1q + offsetq]
+    pmaddwd xm0, [v2q + offsetq]
+    paddd   xm1, xm0
+    HADDD  xm1, xm0
+    movd   eax, xm1
+    RET

 ;-----------------------------------------------------------------------------  ; void ff_vector_clip_int32(int32_t *dst, const int32_t *src, int32_t min, diff --git a/libavcodec/x86/audiodsp_init.c b/libavcodec/x86/audiodsp_init.c
index aa5e43e570..77d5948442 100644
--- a/libavcodec/x86/audiodsp_init.c
+++ b/libavcodec/x86/audiodsp_init.c
@@ -24,6 +24,9 @@
 #include "libavutil/x86/cpu.h"
 #include "libavcodec/audiodsp.h"

+int32_t ff_scalarproduct_int16_avx2(const int16_t *v1, const int16_t *v2,
+                                    int order);
+
 int32_t ff_scalarproduct_int16_sse2(const int16_t *v1, const int16_t *v2,
                                     int order);

@@ -53,4 +56,7 @@ av_cold void ff_audiodsp_init_x86(AudioDSPContext *c)

     if (EXTERNAL_SSE4(cpu_flags))
         c->vector_clip_int32 = ff_vector_clip_int32_sse4;
+
+    if (EXTERNAL_AVX2(cpu_flags))
+        c->scalarproduct_int16 = ff_scalarproduct_int16_avx2;
 }
--
2.37.2

_______________________________________________
ffmpeg-devel mailing list
ffmpeg-devel@ffmpeg.org
https://ffmpeg.org/mailman/listinfo/ffmpeg-devel

To unsubscribe, visit link above, or email
ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".

Reply via email to