Commit d56668bd authored by Ronald S. Bultje's avatar Ronald S. Bultje
Browse files

floatdsp: move scalarproduct_float from dsputil to avfloatdsp.

This makes the aac decoder and all voice codecs independent of dsputil.
parent 5959bfac
......@@ -291,7 +291,6 @@ typedef struct AACContext {
FFTContext mdct;
FFTContext mdct_small;
FFTContext mdct_ltp;
DSPContext dsp;
FmtConvertContext fmt_conv;
AVFloatDSPContext fdsp;
int random_state;
......
......@@ -895,7 +895,6 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
ff_aac_sbr_init();
ff_dsputil_init(&ac->dsp, avctx);
ff_fmt_convert_init(&ac->fmt_conv, avctx);
avpriv_float_dsp_init(&ac->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
......@@ -1358,7 +1357,7 @@ static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
cfo[k] = ac->random_state;
}
band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len);
band_energy = ac->fdsp.scalarproduct_float(cfo, cfo, off_len);
scale = sf[idx] / sqrtf(band_energy);
ac->fdsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
}
......
......@@ -21,9 +21,9 @@
*/
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/mathematics.h"
#include "avcodec.h"
#include "dsputil.h"
#include "acelp_pitch_delay.h"
#include "celp_math.h"
......@@ -120,7 +120,7 @@ float ff_amr_set_fixed_gain(float fixed_gain_factor, float fixed_mean_energy,
// Note 10^(0.05 * -10log(average x2)) = 1/sqrt((average x2)).
float val = fixed_gain_factor *
exp2f(M_LOG2_10 * 0.05 *
(ff_scalarproduct_float_c(pred_table, prediction_error, 4) +
(avpriv_scalarproduct_float_c(pred_table, prediction_error, 4) +
energy_mean)) /
sqrtf(fixed_mean_energy);
......
......@@ -23,8 +23,8 @@
#include <inttypes.h>
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "avcodec.h"
#include "dsputil.h"
#include "acelp_vectors.h"
const uint8_t ff_fc_2pulses_9bits_track1[16] =
......@@ -183,7 +183,7 @@ void ff_adaptive_gain_control(float *out, const float *in, float speech_energ,
int size, float alpha, float *gain_mem)
{
int i;
float postfilter_energ = ff_scalarproduct_float_c(in, in, size);
float postfilter_energ = avpriv_scalarproduct_float_c(in, in, size);
float gain_scale_factor = 1.0;
float mem = *gain_mem;
......@@ -204,7 +204,7 @@ void ff_scale_vector_to_given_sum_of_squares(float *out, const float *in,
float sum_of_squares, const int n)
{
int i;
float scalefactor = ff_scalarproduct_float_c(in, in, n);
float scalefactor = avpriv_scalarproduct_float_c(in, in, n);
if (scalefactor)
scalefactor = sqrt(sum_of_squares / scalefactor);
for (i = 0; i < n; i++)
......
......@@ -44,8 +44,8 @@
#include <math.h>
#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "avcodec.h"
#include "dsputil.h"
#include "libavutil/common.h"
#include "celp_filters.h"
#include "acelp_filters.h"
......@@ -794,8 +794,8 @@ static int synthesis(AMRContext *p, float *lpc,
// emphasize pitch vector contribution
if (p->pitch_gain[4] > 0.5 && !overflow) {
float energy = ff_scalarproduct_float_c(excitation, excitation,
AMR_SUBFRAME_SIZE);
float energy = avpriv_scalarproduct_float_c(excitation, excitation,
AMR_SUBFRAME_SIZE);
float pitch_factor =
p->pitch_gain[4] *
(p->cur_frame_mode == MODE_12k2 ?
......@@ -871,8 +871,8 @@ static float tilt_factor(float *lpc_n, float *lpc_d)
ff_celp_lp_synthesis_filterf(hf, lpc_d, hf, AMR_TILT_RESPONSE,
LP_FILTER_ORDER);
rh0 = ff_scalarproduct_float_c(hf, hf, AMR_TILT_RESPONSE);
rh1 = ff_scalarproduct_float_c(hf, hf + 1, AMR_TILT_RESPONSE - 1);
rh0 = avpriv_scalarproduct_float_c(hf, hf, AMR_TILT_RESPONSE);
rh1 = avpriv_scalarproduct_float_c(hf, hf + 1, AMR_TILT_RESPONSE - 1);
// The spec only specifies this check for 12.2 and 10.2 kbit/s
// modes. But in the ref source the tilt is always non-negative.
......@@ -892,8 +892,8 @@ static void postfilter(AMRContext *p, float *lpc, float *buf_out)
int i;
float *samples = p->samples_in + LP_FILTER_ORDER; // Start of input
float speech_gain = ff_scalarproduct_float_c(samples, samples,
AMR_SUBFRAME_SIZE);
float speech_gain = avpriv_scalarproduct_float_c(samples, samples,
AMR_SUBFRAME_SIZE);
float pole_out[AMR_SUBFRAME_SIZE + LP_FILTER_ORDER]; // Output of pole filter
const float *gamma_n, *gamma_d; // Formant filter factor table
......@@ -998,9 +998,9 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data,
p->fixed_gain[4] =
ff_amr_set_fixed_gain(fixed_gain_factor,
ff_scalarproduct_float_c(p->fixed_vector,
p->fixed_vector,
AMR_SUBFRAME_SIZE) /
avpriv_scalarproduct_float_c(p->fixed_vector,
p->fixed_vector,
AMR_SUBFRAME_SIZE) /
AMR_SUBFRAME_SIZE,
p->prediction_error,
energy_mean[p->cur_frame_mode], energy_pred_fac);
......
......@@ -26,10 +26,10 @@
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/lfg.h"
#include "avcodec.h"
#include "dsputil.h"
#include "lsp.h"
#include "celp_filters.h"
#include "acelp_filters.h"
......@@ -595,11 +595,11 @@ static void pitch_sharpening(AMRWBContext *ctx, float *fixed_vector)
static float voice_factor(float *p_vector, float p_gain,
float *f_vector, float f_gain)
{
double p_ener = (double) ff_scalarproduct_float_c(p_vector, p_vector,
AMRWB_SFR_SIZE) *
double p_ener = (double) avpriv_scalarproduct_float_c(p_vector, p_vector,
AMRWB_SFR_SIZE) *
p_gain * p_gain;
double f_ener = (double) ff_scalarproduct_float_c(f_vector, f_vector,
AMRWB_SFR_SIZE) *
double f_ener = (double) avpriv_scalarproduct_float_c(f_vector, f_vector,
AMRWB_SFR_SIZE) *
f_gain * f_gain;
return (p_ener - f_ener) / (p_ener + f_ener);
......@@ -768,8 +768,8 @@ static void synthesis(AMRWBContext *ctx, float *lpc, float *excitation,
/* emphasize pitch vector contribution in low bitrate modes */
if (ctx->pitch_gain[0] > 0.5 && ctx->fr_cur_mode <= MODE_8k85) {
int i;
float energy = ff_scalarproduct_float_c(excitation, excitation,
AMRWB_SFR_SIZE);
float energy = avpriv_scalarproduct_float_c(excitation, excitation,
AMRWB_SFR_SIZE);
// XXX: Weird part in both ref code and spec. A unknown parameter
// {beta} seems to be identical to the current pitch gain
......@@ -828,9 +828,9 @@ static void upsample_5_4(float *out, const float *in, int o_size)
i++;
for (k = 1; k < 5; k++) {
out[i] = ff_scalarproduct_float_c(in0 + int_part,
upsample_fir[4 - frac_part],
UPS_MEM_SIZE);
out[i] = avpriv_scalarproduct_float_c(in0 + int_part,
upsample_fir[4 - frac_part],
UPS_MEM_SIZE);
int_part++;
frac_part--;
i++;
......@@ -856,8 +856,8 @@ static float find_hb_gain(AMRWBContext *ctx, const float *synth,
if (ctx->fr_cur_mode == MODE_23k85)
return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));
tilt = ff_scalarproduct_float_c(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
ff_scalarproduct_float_c(synth, synth, AMRWB_SFR_SIZE);
tilt = avpriv_scalarproduct_float_c(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
avpriv_scalarproduct_float_c(synth, synth, AMRWB_SFR_SIZE);
/* return gain bounded by [0.1, 1.0] */
return av_clipf((1.0 - FFMAX(0.0, tilt)) * (1.25 - 0.25 * wsp), 0.1, 1.0);
......@@ -876,7 +876,8 @@ static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
const float *synth_exc, float hb_gain)
{
int i;
float energy = ff_scalarproduct_float_c(synth_exc, synth_exc, AMRWB_SFR_SIZE);
float energy = avpriv_scalarproduct_float_c(synth_exc, synth_exc,
AMRWB_SFR_SIZE);
/* Generate a white-noise excitation */
for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
......@@ -1168,9 +1169,9 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data,
ctx->fixed_gain[0] =
ff_amr_set_fixed_gain(fixed_gain_factor,
ff_scalarproduct_float_c(ctx->fixed_vector,
ctx->fixed_vector,
AMRWB_SFR_SIZE) /
avpriv_scalarproduct_float_c(ctx->fixed_vector,
ctx->fixed_vector,
AMRWB_SFR_SIZE) /
AMRWB_SFR_SIZE,
ctx->prediction_error,
ENERGY_MEAN, energy_pred_fac);
......
......@@ -142,8 +142,6 @@ void ff_avg_h264_chroma_mc8_neon(uint8_t *, uint8_t *, int, int, int, int);
void ff_avg_h264_chroma_mc4_neon(uint8_t *, uint8_t *, int, int, int, int);
void ff_avg_h264_chroma_mc2_neon(uint8_t *, uint8_t *, int, int, int, int);
float ff_scalarproduct_float_neon(const float *v1, const float *v2, int len);
void ff_vector_clipf_neon(float *dst, const float *src, float min, float max,
int len);
void ff_vector_clip_int32_neon(int32_t *dst, const int32_t *src, int32_t min,
......@@ -293,7 +291,6 @@ void ff_dsputil_init_neon(DSPContext *c, AVCodecContext *avctx)
c->avg_h264_qpel_pixels_tab[1][15] = ff_avg_h264_qpel8_mc33_neon;
}
c->scalarproduct_float = ff_scalarproduct_float_neon;
c->vector_clipf = ff_vector_clipf_neon;
c->vector_clip_int32 = ff_vector_clip_int32_neon;
......
......@@ -531,19 +531,6 @@ function ff_add_pixels_clamped_neon, export=1
bx lr
endfunc
function ff_scalarproduct_float_neon, export=1
vmov.f32 q2, #0.0
1: vld1.32 {q0},[r0,:128]!
vld1.32 {q1},[r1,:128]!
vmla.f32 q2, q0, q1
subs r2, r2, #4
bgt 1b
vadd.f32 d0, d4, d5
vpadd.f32 d0, d0, d0
NOVFP vmov.32 r0, d0[0]
bx lr
endfunc
function ff_vector_clipf_neon, export=1
VFP vdup.32 q1, d0[1]
VFP vdup.32 q0, d0[0]
......
......@@ -2353,17 +2353,6 @@ WRAPPER8_16_SQ(quant_psnr8x8_c, quant_psnr16_c)
WRAPPER8_16_SQ(rd8x8_c, rd16_c)
WRAPPER8_16_SQ(bit8x8_c, bit16_c)
float ff_scalarproduct_float_c(const float *v1, const float *v2, int len)
{
float p = 0.0;
int i;
for (i = 0; i < len; i++)
p += v1[i] * v2[i];
return p;
}
static inline uint32_t clipf_c_one(uint32_t a, uint32_t mini,
uint32_t maxi, uint32_t maxisign)
{
......@@ -2694,7 +2683,6 @@ av_cold void ff_dsputil_init(DSPContext* c, AVCodecContext *avctx)
c->scalarproduct_and_madd_int16 = scalarproduct_and_madd_int16_c;
c->apply_window_int16 = apply_window_int16_c;
c->vector_clip_int32 = vector_clip_int32_c;
c->scalarproduct_float = ff_scalarproduct_float_c;
c->shrink[0]= av_image_copy_plane;
c->shrink[1]= ff_shrink22;
......
......@@ -342,13 +342,6 @@ typedef struct DSPContext {
/* assume len is a multiple of 8, and arrays are 16-byte aligned */
void (*vector_clipf)(float *dst /* align 16 */, const float *src /* align 16 */, float min, float max, int len /* align 16 */);
/**
* Calculate the scalar product of two vectors of floats.
* @param v1 first vector, 16-byte aligned
* @param v2 second vector, 16-byte aligned
* @param len length of vectors, multiple of 4
*/
float (*scalarproduct_float)(const float *v1, const float *v2, int len);
/* (I)DCT */
void (*fdct)(DCTELEM *block/* align 16*/);
......@@ -454,17 +447,6 @@ void ff_dsputil_init(DSPContext* p, AVCodecContext *avctx);
int ff_check_alignment(void);
/**
* Return the scalar product of two vectors.
*
* @param v1 first input vector
* @param v2 first input vector
* @param len number of elements
*
* @return sum of elementwise products
*/
float ff_scalarproduct_float_c(const float *v1, const float *v2, int len);
/**
* permute block according to permuatation.
* @param last last non zero element in scantable order
......
......@@ -30,10 +30,10 @@
#include <stddef.h>
#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "avcodec.h"
#include "internal.h"
#include "get_bits.h"
#include "dsputil.h"
#include "qcelpdata.h"
#include "celp_filters.h"
#include "acelp_filters.h"
......@@ -400,12 +400,10 @@ static void apply_gain_ctrl(float *v_out, const float *v_ref, const float *v_in)
{
int i;
for (i = 0; i < 160; i += 40)
ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i,
ff_scalarproduct_float_c(v_ref + i,
v_ref + i,
40),
40);
for (i = 0; i < 160; i += 40) {
float res = avpriv_scalarproduct_float_c(v_ref + i, v_ref + i, 40);
ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i, res, 40);
}
}
/**
......@@ -680,8 +678,9 @@ static void postfilter(QCELPContext *q, float *samples, float *lpc)
ff_tilt_compensation(&q->postfilter_tilt_mem, 0.3, pole_out + 10, 160);
ff_adaptive_gain_control(samples, pole_out + 10,
ff_scalarproduct_float_c(q->formant_mem + 10,
q->formant_mem + 10, 160),
avpriv_scalarproduct_float_c(q->formant_mem + 10,
q->formant_mem + 10,
160),
160, 0.9375, &q->postfilter_agc_mem);
}
......
......@@ -79,7 +79,7 @@ static av_cold int ra288_decode_init(AVCodecContext *avctx)
static void convolve(float *tgt, const float *src, int len, int n)
{
for (; n >= 0; n--)
tgt[n] = ff_scalarproduct_float_c(src, src - n, len);
tgt[n] = avpriv_scalarproduct_float_c(src, src - n, len);
}
......@@ -108,7 +108,7 @@ static void decode(RA288Context *ractx, float gain, int cb_coef)
for (i=0; i < 5; i++)
buffer[i] = codetable[cb_coef][i] * sumsum;
sum = ff_scalarproduct_float_c(buffer, buffer, 5) * ((1 << 24) / 5.);
sum = avpriv_scalarproduct_float_c(buffer, buffer, 5) * ((1 << 24) / 5.);
sum = FFMAX(sum, 1);
......
......@@ -26,11 +26,11 @@
#include <string.h>
#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "libavutil/mathematics.h"
#include "avcodec.h"
#define BITSTREAM_READER_LE
#include "get_bits.h"
#include "dsputil.h"
#include "internal.h"
#include "lsp.h"
......@@ -411,9 +411,10 @@ static void decode_frame(SiprContext *ctx, SiprParameters *params,
convolute_with_sparse(fixed_vector, &fixed_cb, impulse_response,
SUBFR_SIZE);
avg_energy =
(0.01 + ff_scalarproduct_float_c(fixed_vector, fixed_vector, SUBFR_SIZE)) /
SUBFR_SIZE;
avg_energy = (0.01 + avpriv_scalarproduct_float_c(fixed_vector,
fixed_vector,
SUBFR_SIZE)) /
SUBFR_SIZE;
ctx->past_pitch_gain = pitch_gain = gain_cb[params->gc_index[i]][0];
......@@ -454,9 +455,9 @@ static void decode_frame(SiprContext *ctx, SiprParameters *params,
if (ctx->mode == MODE_5k0) {
for (i = 0; i < subframe_count; i++) {
float energy = ff_scalarproduct_float_c(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i * SUBFR_SIZE,
ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i * SUBFR_SIZE,
SUBFR_SIZE);
float energy = avpriv_scalarproduct_float_c(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i * SUBFR_SIZE,
ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i * SUBFR_SIZE,
SUBFR_SIZE);
ff_adaptive_gain_control(&synth[i * SUBFR_SIZE],
&synth[i * SUBFR_SIZE], energy,
SUBFR_SIZE, 0.9, &ctx->postfilter_agc);
......
......@@ -25,8 +25,8 @@
#include "sipr.h"
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/mathematics.h"
#include "dsputil.h"
#include "lsp.h"
#include "celp_filters.h"
#include "acelp_vectors.h"
......@@ -163,11 +163,11 @@ static float acelp_decode_gain_codef(float gain_corr_factor, const float *fc_v,
const float *ma_prediction_coeff,
int subframe_size, int ma_pred_order)
{
mr_energy +=
ff_scalarproduct_float_c(quant_energy, ma_prediction_coeff, ma_pred_order);
mr_energy += avpriv_scalarproduct_float_c(quant_energy, ma_prediction_coeff,
ma_pred_order);
mr_energy = gain_corr_factor * exp(M_LN10 / 20. * mr_energy) /
sqrt((0.01 + ff_scalarproduct_float_c(fc_v, fc_v, subframe_size)));
sqrt((0.01 + avpriv_scalarproduct_float_c(fc_v, fc_v, subframe_size)));
return mr_energy;
}
......
......@@ -30,8 +30,8 @@
#include <math.h>
#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "libavutil/mem.h"
#include "dsputil.h"
#include "avcodec.h"
#include "internal.h"
#include "get_bits.h"
......@@ -523,7 +523,7 @@ static int kalman_smoothen(WMAVoiceContext *s, int pitch,
/* find best fitting point in history */
do {
dot = ff_scalarproduct_float_c(in, ptr, size);
dot = avpriv_scalarproduct_float_c(in, ptr, size);
if (dot > optimal_gain) {
optimal_gain = dot;
best_hist_ptr = ptr;
......@@ -532,7 +532,7 @@ static int kalman_smoothen(WMAVoiceContext *s, int pitch,
if (optimal_gain <= 0)
return -1;
dot = ff_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size);
dot = avpriv_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size);
if (dot <= 0) // would be 1.0
return -1;
......@@ -562,8 +562,8 @@ static float tilt_factor(const float *lpcs, int n_lpcs)
{
float rh0, rh1;
rh0 = 1.0 + ff_scalarproduct_float_c(lpcs, lpcs, n_lpcs);
rh1 = lpcs[0] + ff_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1);
rh0 = 1.0 + avpriv_scalarproduct_float_c(lpcs, lpcs, n_lpcs);
rh1 = lpcs[0] + avpriv_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1);
return rh1 / rh0;
}
......@@ -656,7 +656,8 @@ static void calc_input_response(WMAVoiceContext *s, float *lpcs,
-1.8 * tilt_factor(coeffs, remainder - 1),
coeffs, remainder);
}
sq = (1.0 / 64.0) * sqrtf(1 / ff_scalarproduct_float_c(coeffs, coeffs, remainder));
sq = (1.0 / 64.0) * sqrtf(1 / avpriv_scalarproduct_float_c(coeffs, coeffs,
remainder));
for (n = 0; n < remainder; n++)
coeffs[n] *= sq;
}
......@@ -1320,7 +1321,8 @@ static void synth_block_fcb_acb(WMAVoiceContext *s, GetBitContext *gb,
/* Calculate gain for adaptive & fixed codebook signal.
* see ff_amr_set_fixed_gain(). */
idx = get_bits(gb, 7);
fcb_gain = expf(ff_scalarproduct_float_c(s->gain_pred_err, gain_coeff, 6) -
fcb_gain = expf(avpriv_scalarproduct_float_c(s->gain_pred_err,
gain_coeff, 6) -
5.2409161640 + wmavoice_gain_codebook_fcb[idx]);
acb_gain = wmavoice_gain_codebook_acb[idx];
pred_err = av_clipf(wmavoice_gain_codebook_fcb[idx],
......
......@@ -463,32 +463,6 @@ cglobal add_hfyu_left_prediction, 3,3,7, dst, src, w, left
.src_unaligned:
ADD_HFYU_LEFT_LOOP 0, 0
; float scalarproduct_float_sse(const float *v1, const float *v2, int len)
INIT_XMM sse
cglobal scalarproduct_float, 3,3,2, v1, v2, offset
neg offsetq
shl offsetq, 2
sub v1q, offsetq
sub v2q, offsetq
xorps xmm0, xmm0
.loop:
movaps xmm1, [v1q+offsetq]
mulps xmm1, [v2q+offsetq]
addps xmm0, xmm1
add offsetq, 16
js .loop
movhlps xmm1, xmm0
addps xmm0, xmm1
movss xmm1, xmm0
shufps xmm0, xmm0, 1
addss xmm0, xmm1
%if ARCH_X86_64 == 0
movss r0m, xmm0
fld dword r0m
%endif
RET
;-----------------------------------------------------------------------------
; void ff_vector_clip_int32(int32_t *dst, const int32_t *src, int32_t min,
; int32_t max, unsigned int len)
......
......@@ -1846,8 +1846,6 @@ int ff_add_hfyu_left_prediction_ssse3(uint8_t *dst, const uint8_t *src,
int ff_add_hfyu_left_prediction_sse4(uint8_t *dst, const uint8_t *src,
int w, int left);
float ff_scalarproduct_float_sse(const float *v1, const float *v2, int order);
void ff_vector_clip_int32_mmx (int32_t *dst, const int32_t *src,
int32_t min, int32_t max, unsigned int len);
void ff_vector_clip_int32_sse2 (int32_t *dst, const int32_t *src,
......@@ -2128,10 +2126,6 @@ static void dsputil_init_sse(DSPContext *c, AVCodecContext *avctx, int mm_flags)
c->vector_clipf = vector_clipf_sse;
#endif /* HAVE_INLINE_ASM */
#if HAVE_YASM
c->scalarproduct_float = ff_scalarproduct_float_sse;
#endif /* HAVE_YASM */
}
static void dsputil_init_sse2(DSPContext *c, AVCodecContext *avctx,
......
......@@ -43,6 +43,8 @@ void ff_vector_fmul_reverse_neon(float *dst, const float *src0,
void ff_butterflies_float_neon(float *v1, float *v2, int len);
float ff_scalarproduct_float_neon(const float *v1, const float *v2, int len);
void ff_float_dsp_init_neon(AVFloatDSPContext *fdsp)
{
fdsp->vector_fmul = ff_vector_fmul_neon;
......@@ -52,4 +54,5 @@ void ff_float_dsp_init_neon(AVFloatDSPContext *fdsp)
fdsp->vector_fmul_add = ff_vector_fmul_add_neon;
fdsp->vector_fmul_reverse = ff_vector_fmul_reverse_neon;
fdsp->butterflies_float = ff_butterflies_float_neon;
fdsp->scalarproduct_float = ff_scalarproduct_float_neon;
}
......@@ -256,3 +256,16 @@ function ff_butterflies_float_neon, export=1
bgt 1b
bx lr
endfunc
function ff_scalarproduct_float_neon, export=1
vmov.f32 q2, #0.0
1: vld1.32 {q0},[r0,:128]!
vld1.32 {q1},[r1,:128]!
vmla.f32 q2, q0, q1
subs r2, r2, #4
bgt 1b
vadd.f32 d0, d4, d5
vpadd.f32 d0, d0, d0
NOVFP vmov.32 r0, d0[0]
bx lr
endfunc
......@@ -101,6 +101,17 @@ static void butterflies_float_c(float *restrict v1, float *restrict v2,
}
}
float avpriv_scalarproduct_float_c(const float *v1, const float *v2, int len)
{
float p = 0.0;
int i;
for (i = 0; i < len; i++)
p += v1[i] * v2[i];
return p;
}
void avpriv_float_dsp_init(AVFloatDSPContext *fdsp, int bit_exact)
{
fdsp->vector_fmul = vector_fmul_c;
......@@ -111,6 +122,7 @@ void avpriv_float_dsp_init(AVFloatDSPContext *fdsp, int bit_exact)
fdsp->vector_fmul_add = vector_fmul_add_c;
fdsp->vector_fmul_reverse = vector_fmul_reverse_c;
fdsp->butterflies_float = butterflies_float_c;
fdsp->scalarproduct_float = avpriv_scalarproduct_float_c;
#if ARCH_ARM
ff_float_dsp_init_arm(fdsp);
......
......@@ -146,8 +146,30 @@ typedef struct AVFloatDSPContext {
* @param len length of vectors, multiple of 4
*/
void (*butterflies_float)(float *restrict v1, float *restrict v2, int len);