Hi,
Le 15 févr. 2014 18:31, "Janne Grunau" <[email protected]> a écrit :
>
> On 2014-02-14 16:03:13 +0000, Christophe Gisquet wrote:
> > The vector dequantization has a test in a loop preventing effective SIMD
> > implementation. By moving it out of the loop, this loop can be DSPized.
> >
> > Therefore, modify the current DSP implementation. In particular, the
> > DSP implementation no longer has to handle null loop sizes.
> > ---
> > libavcodec/arm/dca.h | 23 ----------------------
> > libavcodec/arm/dcadsp_init_arm.c | 41
++++++++++++++++++++++++++++++++++++++++
> > libavcodec/dcadec.c | 30 +++++++----------------------
> > libavcodec/dcadsp.c | 21 ++++++++++++++------
> > libavcodec/dcadsp.h | 8 +++++++-
> > 5 files changed, 70 insertions(+), 53 deletions(-)
> >
> > diff --git a/libavcodec/arm/dca.h b/libavcodec/arm/dca.h
> > index 580bd75..4aed576 100644
> > --- a/libavcodec/arm/dca.h
> > +++ b/libavcodec/arm/dca.h
> > @@ -81,27 +81,4 @@ static inline int decode_blockcodes(int code1, int
code2, int levels,
> >
> > #endif
> >
> > -#if HAVE_NEON_INLINE && HAVE_ASM_MOD_Y
> > -
> > -#define int8x8_fmul_int32 int8x8_fmul_int32
> > -static inline void int8x8_fmul_int32(av_unused DCADSPContext *dsp,
> > - float *dst, const int8_t *src,
int scale)
> > -{
> > - __asm__ ("vcvt.f32.s32 %2, %2, #4 \n"
> > - "vld1.8 {d0}, [%1,:64] \n"
> > - "vmovl.s8 q0, d0 \n"
> > - "vmovl.s16 q1, d1 \n"
> > - "vmovl.s16 q0, d0 \n"
> > - "vcvt.f32.s32 q0, q0 \n"
> > - "vcvt.f32.s32 q1, q1 \n"
> > - "vmul.f32 q0, q0, %y2 \n"
> > - "vmul.f32 q1, q1, %y2 \n"
> > - "vst1.32 {q0-q1}, [%m0,:128] \n"
> > - : "=Um"(*(float (*)[8])dst)
> > - : "r"(src), "x"(scale)
> > - : "d0", "d1", "d2", "d3");
> > -}
> > -
> > -#endif
> > -
> > #endif /* AVCODEC_ARM_DCA_H */
> > diff --git a/libavcodec/arm/dcadsp_init_arm.c
b/libavcodec/arm/dcadsp_init_arm.c
> > index 2ea1289..9942581 100644
> > --- a/libavcodec/arm/dcadsp_init_arm.c
> > +++ b/libavcodec/arm/dcadsp_init_arm.c
> > @@ -53,6 +53,44 @@ void ff_synth_filter_float_neon(FFTContext *imdct,
> > float out[32], const float in[32],
> > float scale);
> >
> > +#if HAVE_NEON_INLINE && HAVE_ASM_MOD_Y
> > +
> > +static inline void int8x8_fmul_int32(av_unused DCADSPContext *dsp,
> > + float *dst, const int8_t *src,
int scale)
> > +{
> > + __asm__ ("vcvt.f32.s32 %2, %2, #4 \n"
> > + "vld1.8 {d0}, [%1,:64] \n"
> > + "vmovl.s8 q0, d0 \n"
> > + "vmovl.s16 q1, d1 \n"
> > + "vmovl.s16 q0, d0 \n"
> > + "vcvt.f32.s32 q0, q0 \n"
> > + "vcvt.f32.s32 q1, q1 \n"
> > + "vmul.f32 q0, q0, %y2 \n"
> > + "vmul.f32 q1, q1, %y2 \n"
> > + "vst1.32 {q0-q1}, [%m0,:128] \n"
> > + : "=Um"(*(float (*)[8])dst)
> > + : "r"(src), "x"(scale)
> > + : "d0", "d1", "d2", "d3");
> > +}
> > +
> > +static void decode_hf_neon(float dst[DCA_SUBBANDS][8],
> > + const int32_t vq_num[DCA_SUBBANDS],
> > + const int8_t hf_vq[1024][32], intptr_t
vq_offset,
> > + int32_t scale[DCA_SUBBANDS][2],
> > + intptr_t start, intptr_t end)
> > +{
> > + int l;
> > + for (l = start; l < end; l++) {
> > + /* 1 vector -> 32 samples but we only need the 8 samples
> > + * for this subsubframe. */
> > + int hfvq = vq_num[l];
> > +
> > + int8x8_fmul_int32(dst[l], hf_vq[hfvq] + vq_offset,
scale[l][0]);
> > + }
> > +}
> > +
> > +#endif /* HAVE_NEON_INLINE && HAVE_ASM_MOD_Y */
> > +
> > av_cold void ff_dcadsp_init_arm(DCADSPContext *s)
> > {
> > int cpu_flags = av_get_cpu_flags();
> > @@ -65,6 +103,9 @@ av_cold void ff_dcadsp_init_arm(DCADSPContext *s)
> > if (have_neon(cpu_flags)) {
> > s->lfe_fir[0] = ff_dca_lfe_fir0_neon;
> > s->lfe_fir[1] = ff_dca_lfe_fir1_neon;
> > +#if HAVE_NEON_INLINE && HAVE_ASM_MOD_Y
> > + s->decode_hf = decode_hf_neon;
> > +#endif
>
> ok but not optimal, I'll rewrite it as external asm, doesn't block this
patch
>
> > }
> > }
> >
> > diff --git a/libavcodec/dcadec.c b/libavcodec/dcadec.c
> > index 2d88cb4..371c838 100644
> > --- a/libavcodec/dcadec.c
> > +++ b/libavcodec/dcadec.c
> > @@ -50,14 +50,10 @@
> > #if ARCH_ARM
> > # include "arm/dca.h"
> > #endif
> > -#if ARCH_X86
> > -# include "x86/dca.h"
> > -#endif
> >
> > //#define TRACE
> >
> > #define DCA_PRIM_CHANNELS_MAX (7)
> > -#define DCA_SUBBANDS (32)
> > #define DCA_ABITS_MAX (32) /* Should be 28 */
> > #define DCA_SUBSUBFRAMES_MAX (4)
> > #define DCA_SUBFRAMES_MAX (16)
> > @@ -340,7 +336,7 @@ typedef struct {
> > int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///<
prediction VQ coefs
> > int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///<
bit allocation index
> > int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///<
transition mode (transients)
> > - int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2]; ///<
scale factors (2 if transient)
> > + int32_t scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2];///<
scale factors (2 if transient)
>
> Don't we assume that int is 32 bit wide? I guess much more would break
> if that assumption doesn't hold. change of course ok when using it in asm.
I think it's dangerous because:
- natural size of int on x86_64 is 64b
- win64 has some freedom to leave garbage in 32msbs of 64b gprs and thus
x86 sometimes uses intptr_t
But some audio DSP had this potential issue without consequences.
>
> > int joint_huff[DCA_PRIM_CHANNELS_MAX]; ///<
joint subband scale factors codebook
> > int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///<
joint subband scale factors
> > float downmix_coef[DCA_PRIM_CHANNELS_MAX + 1][2]; ///<
stereo downmix coefficients
> > @@ -353,7 +349,7 @@ typedef struct {
> > uint8_t core_downmix_amode; ///<
audio channel arrangement of embedded downmix
> > uint16_t core_downmix_codes[DCA_PRIM_CHANNELS_MAX + 1][4]; ///<
embedded downmix coefficients (9-bit codes)
> >
> > - int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///<
VQ encoded high frequency subbands
> > + int32_t high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///<
VQ encoded high frequency subbands
> >
> > float lfe_data[2 * DCA_LFE_MAX * (DCA_BLOCKS_MAX + 4)]; ///<
Low frequency effect data
> > int lfe_scale_factor;
> > @@ -1088,14 +1084,6 @@ static int decode_blockcodes(int code1, int
code2, int levels, int32_t *values)
> > static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
> > static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
> >
> > -#ifndef int8x8_fmul_int32
> > -static inline void int8x8_fmul_int32(DCADSPContext *dsp, float *dst,
> > - const int8_t *src, int scale)
> > -{
> > - dsp->int8x8_fmul_int32(dst, src, scale);
> > -}
> > -#endif
> > -
> > static int dca_subsubframe(DCAContext *s, int base_channel, int
block_index)
> > {
> > int k, l;
> > @@ -1215,20 +1203,16 @@ static int dca_subsubframe(DCAContext *s, int
base_channel, int block_index)
> > /*
> > * Decode VQ encoded high frequencies
> > */
> > - for (l = s->vq_start_subband[k]; l < s->subband_activity[k];
l++) {
> > - /* 1 vector -> 32 samples but we only need the 8 samples
> > - * for this subsubframe. */
> > - int hfvq = s->high_freq_vq[k][l];
> > -
> > + if (s->subband_activity[k] > s->vq_start_subband[k]) {
> > if (!s->debug_flag & 0x01) {
> > av_log(s->avctx, AV_LOG_DEBUG,
> > "Stream with high frequencies VQ coding\n");
> > s->debug_flag |= 0x01;
> > }
> > -
> > - int8x8_fmul_int32(&s->dcadsp, subband_samples[k][l],
> > - &high_freq_vq[hfvq][subsubframe * 8],
> > - s->scale_factor[k][l][0]);
> > + s->dcadsp.decode_hf(subband_samples[k], s->high_freq_vq[k],
> > + high_freq_vq, subsubframe * 8,
> > + s->scale_factor[k],
s->vq_start_subband[k],
> > + s->subband_activity[k]);
> > }
> > }
> >
> > diff --git a/libavcodec/dcadsp.c b/libavcodec/dcadsp.c
> > index 30b732a..10fadd6 100644
> > --- a/libavcodec/dcadsp.c
> > +++ b/libavcodec/dcadsp.c
> > @@ -24,12 +24,21 @@
> > #include "libavutil/intreadwrite.h"
> > #include "dcadsp.h"
> >
> > -static void int8x8_fmul_int32_c(float *dst, const int8_t *src, int
scale)
> > +static void decode_hf_c(float dst[DCA_SUBBANDS][8],
> > + const int32_t vq_num[DCA_SUBBANDS],
> > + const int8_t hf_vq[1024][32], intptr_t
vq_offset,
> > + int32_t scale[DCA_SUBBANDS][2],
> > + intptr_t start, intptr_t end)
> > {
> > - float fscale = scale / 16.0;
> > - int i;
> > - for (i = 0; i < 8; i++)
> > - dst[i] = src[i] * fscale;
> > + int l;
> > + for (l = start; l < end; l++) {
> > + /* 1 vector -> 32 samples but we only need the 8 samples
> > + * for this subsubframe. */
> > + int i, hfvq = vq_num[l];
> > + float fscale = scale[l][0] / 16.0;
> > + for (i = 0; i < 8; i++)
> > + dst[l][i] = hf_vq[hfvq][vq_offset + i] * fscale;
> > + }
> > }
> >
> > static inline void
> > @@ -97,7 +106,7 @@ av_cold void ff_dcadsp_init(DCADSPContext *s)
> > s->lfe_fir[0] = dca_lfe_fir0_c;
> > s->lfe_fir[1] = dca_lfe_fir1_c;
> > s->qmf_32_subbands = dca_qmf_32_subbands;
> > - s->int8x8_fmul_int32 = int8x8_fmul_int32_c;
> > + s->decode_hf = decode_hf_c;
> > if (ARCH_ARM) ff_dcadsp_init_arm(s);
> > if (ARCH_X86) ff_dcadsp_init_x86(s);
> > }
> > diff --git a/libavcodec/dcadsp.h b/libavcodec/dcadsp.h
> > index e4c1bc7..0fa75a5 100644
> > --- a/libavcodec/dcadsp.h
> > +++ b/libavcodec/dcadsp.h
> > @@ -22,6 +22,8 @@
> > #include "avfft.h"
> > #include "synth_filter.h"
> >
> > +#define DCA_SUBBANDS 32
> > +
> > typedef struct DCADSPContext {
> > void (*lfe_fir[2])(float *out, const float *in, const float
*coefs);
> > void (*qmf_32_subbands)(float samples_in[32][8], int sb_act,
> > @@ -30,7 +32,11 @@ typedef struct DCADSPContext {
> > int *synth_buf_offset, float
synth_buf2[32],
> > const float window[512], float
*samples_out,
> > float raXin[32], float scale);
> > - void (*int8x8_fmul_int32)(float *dst, const int8_t *src, int
scale);
> > + void (*decode_hf)(float dst[DCA_SUBBANDS][8],
> > + const int32_t vq_num[DCA_SUBBANDS],
> > + const int8_t hf_vq[1024][32], intptr_t vq_offset,
> > + int32_t scale[DCA_SUBBANDS][2],
> > + intptr_t start, intptr_t end);
> > } DCADSPContext;
> >
> > void ff_dcadsp_init(DCADSPContext *s);
>
> ok and queued
>
> Janne
> _______________________________________________
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> [email protected]
> https://lists.libav.org/mailman/listinfo/libav-devel
Sent from a phone so please excuse terseness
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