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celp_math: Replace duplicate ff_dot_productf() by ff_scalarproduct_c()

tags/n1.0
Diego Biurrun 12 years ago
parent
5549854335
commit
dafcbfe443
15 changed files with 77 additions and 83 deletions
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  1. +6
    -6
      libavcodec/Makefile
  2. +1
    -2
      libavcodec/acelp_pitch_delay.c
  3. +3
    -3
      libavcodec/acelp_vectors.c
  4. +11
    -9
      libavcodec/amrnbdec.c
  5. +19
    -14
      libavcodec/amrwbdec.c
  6. +0
    -11
      libavcodec/celp_math.c
  7. +0
    -10
      libavcodec/celp_math.h
  8. +2
    -2
      libavcodec/dsputil.c
  9. +11
    -0
      libavcodec/dsputil.h
  10. +0
    -1
      libavcodec/lsp.c
  11. +6
    -7
      libavcodec/qcelpdec.c
  12. +2
    -3
      libavcodec/ra288.c
  13. +4
    -5
      libavcodec/sipr.c
  14. +4
    -3
      libavcodec/sipr16k.c
  15. +8
    -7
      libavcodec/wmavoice.c

+ 6
- 6
libavcodec/Makefile View File

@@ -84,11 +84,11 @@ OBJS-$(CONFIG_ALAC_DECODER) += alac.o
OBJS-$(CONFIG_ALAC_ENCODER) += alacenc.o
OBJS-$(CONFIG_ALS_DECODER) += alsdec.o bgmc.o mpeg4audio.o
OBJS-$(CONFIG_AMRNB_DECODER) += amrnbdec.o celp_filters.o \
celp_math.o acelp_filters.o \
acelp_filters.o \
acelp_vectors.o \
acelp_pitch_delay.o
OBJS-$(CONFIG_AMRWB_DECODER) += amrwbdec.o celp_filters.o \
celp_math.o acelp_filters.o \
acelp_filters.o \
acelp_vectors.o \
acelp_pitch_delay.o
OBJS-$(CONFIG_AMV_DECODER) += sp5xdec.o mjpegdec.o mjpeg.o
@@ -298,7 +298,7 @@ OBJS-$(CONFIG_PPM_ENCODER) += pnmenc.o pnm.o
OBJS-$(CONFIG_PRORES_DECODER) += proresdec.o proresdata.o proresdsp.o
OBJS-$(CONFIG_PRORES_ENCODER) += proresenc.o proresdata.o proresdsp.o
OBJS-$(CONFIG_PTX_DECODER) += ptx.o
OBJS-$(CONFIG_QCELP_DECODER) += qcelpdec.o celp_math.o \
OBJS-$(CONFIG_QCELP_DECODER) += qcelpdec.o \
celp_filters.o acelp_vectors.o \
acelp_filters.o
OBJS-$(CONFIG_QDM2_DECODER) += qdm2.o
@@ -311,7 +311,7 @@ OBJS-$(CONFIG_R210_DECODER) += r210dec.o
OBJS-$(CONFIG_RA_144_DECODER) += ra144dec.o ra144.o celp_filters.o
OBJS-$(CONFIG_RA_144_ENCODER) += ra144enc.o ra144.o celp_filters.o \
audio_frame_queue.o
OBJS-$(CONFIG_RA_288_DECODER) += ra288.o celp_math.o celp_filters.o
OBJS-$(CONFIG_RA_288_DECODER) += ra288.o celp_filters.o
OBJS-$(CONFIG_RALF_DECODER) += ralf.o
OBJS-$(CONFIG_RAWVIDEO_DECODER) += rawdec.o
OBJS-$(CONFIG_RAWVIDEO_ENCODER) += rawenc.o
@@ -332,7 +332,7 @@ OBJS-$(CONFIG_SGI_DECODER) += sgidec.o
OBJS-$(CONFIG_SGI_ENCODER) += sgienc.o rle.o
OBJS-$(CONFIG_SHORTEN_DECODER) += shorten.o
OBJS-$(CONFIG_SIPR_DECODER) += sipr.o acelp_pitch_delay.o \
celp_math.o acelp_vectors.o \
acelp_vectors.o \
acelp_filters.o celp_filters.o \
sipr16k.o
OBJS-$(CONFIG_SMACKAUD_DECODER) += smacker.o
@@ -408,7 +408,7 @@ OBJS-$(CONFIG_WMAV1_ENCODER) += wmaenc.o wma.o wma_common.o aactab.o
OBJS-$(CONFIG_WMAV2_DECODER) += wmadec.o wma.o wma_common.o aactab.o
OBJS-$(CONFIG_WMAV2_ENCODER) += wmaenc.o wma.o wma_common.o aactab.o
OBJS-$(CONFIG_WMAVOICE_DECODER) += wmavoice.o \
celp_math.o celp_filters.o \
celp_filters.o \
acelp_vectors.o acelp_filters.o
OBJS-$(CONFIG_WMV1_DECODER) += msmpeg4.o msmpeg4data.o
OBJS-$(CONFIG_WMV2_DECODER) += wmv2dec.o wmv2.o \


+ 1
- 2
libavcodec/acelp_pitch_delay.c View File

@@ -25,7 +25,6 @@
#include "avcodec.h"
#include "dsputil.h"
#include "acelp_pitch_delay.h"
#include "celp_math.h"

int ff_acelp_decode_8bit_to_1st_delay3(int ac_index)
{
@@ -120,7 +119,7 @@ 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_dot_productf(pred_table, prediction_error, 4) +
(ff_scalarproduct_float_c(pred_table, prediction_error, 4) +
energy_mean)) /
sqrtf(fixed_mean_energy);



+ 3
- 3
libavcodec/acelp_vectors.c View File

@@ -24,8 +24,8 @@

#include "libavutil/common.h"
#include "avcodec.h"
#include "dsputil.h"
#include "acelp_vectors.h"
#include "celp_math.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_dot_productf(in, in, size);
float postfilter_energ = ff_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_dot_productf(in, in, n);
float scalefactor = ff_scalarproduct_float_c(in, in, n);
if (scalefactor)
scalefactor = sqrt(sum_of_squares / scalefactor);
for (i = 0; i < n; i++)


+ 11
- 9
libavcodec/amrnbdec.c View File

@@ -44,8 +44,8 @@
#include <math.h>

#include "avcodec.h"
#include "dsputil.h"
#include "libavutil/common.h"
#include "celp_math.h"
#include "celp_filters.h"
#include "acelp_filters.h"
#include "acelp_vectors.h"
@@ -784,8 +784,8 @@ static int synthesis(AMRContext *p, float *lpc,

// emphasize pitch vector contribution
if (p->pitch_gain[4] > 0.5 && !overflow) {
float energy = ff_dot_productf(excitation, excitation,
AMR_SUBFRAME_SIZE);
float energy = ff_scalarproduct_float_c(excitation, excitation,
AMR_SUBFRAME_SIZE);
float pitch_factor =
p->pitch_gain[4] *
(p->cur_frame_mode == MODE_12k2 ?
@@ -861,8 +861,8 @@ 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_dot_productf(hf, hf, AMR_TILT_RESPONSE);
rh1 = ff_dot_productf(hf, hf + 1, AMR_TILT_RESPONSE - 1);
rh0 = ff_scalarproduct_float_c(hf, hf, AMR_TILT_RESPONSE);
rh1 = ff_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.
@@ -882,8 +882,8 @@ 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_dot_productf(samples, samples,
AMR_SUBFRAME_SIZE);
float speech_gain = ff_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
@@ -988,8 +988,10 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data,

p->fixed_gain[4] =
ff_amr_set_fixed_gain(fixed_gain_factor,
ff_dot_productf(p->fixed_vector, p->fixed_vector,
AMR_SUBFRAME_SIZE)/AMR_SUBFRAME_SIZE,
ff_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);



+ 19
- 14
libavcodec/amrwbdec.c View File

@@ -28,8 +28,8 @@
#include "libavutil/lfg.h"

#include "avcodec.h"
#include "dsputil.h"
#include "lsp.h"
#include "celp_math.h"
#include "celp_filters.h"
#include "acelp_filters.h"
#include "acelp_vectors.h"
@@ -585,10 +585,12 @@ 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_dot_productf(p_vector, p_vector,
AMRWB_SFR_SIZE) * p_gain * p_gain;
double f_ener = (double) ff_dot_productf(f_vector, f_vector,
AMRWB_SFR_SIZE) * f_gain * f_gain;
double p_ener = (double) ff_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) *
f_gain * f_gain;

return (p_ener - f_ener) / (p_ener + f_ener);
}
@@ -756,8 +758,8 @@ 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_dot_productf(excitation, excitation,
AMRWB_SFR_SIZE);
float energy = ff_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
@@ -816,8 +818,9 @@ static void upsample_5_4(float *out, const float *in, int o_size)
i++;

for (k = 1; k < 5; k++) {
out[i] = ff_dot_productf(in0 + int_part, upsample_fir[4 - frac_part],
UPS_MEM_SIZE);
out[i] = ff_scalarproduct_float_c(in0 + int_part,
upsample_fir[4 - frac_part],
UPS_MEM_SIZE);
int_part++;
frac_part--;
i++;
@@ -843,8 +846,8 @@ 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_dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
ff_dot_productf(synth, synth, AMRWB_SFR_SIZE);
tilt = ff_scalarproduct_float_c(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
ff_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);
@@ -863,7 +866,7 @@ static void scaled_hb_excitation(AMRWBContext *ctx, float *hb_exc,
const float *synth_exc, float hb_gain)
{
int i;
float energy = ff_dot_productf(synth_exc, synth_exc, AMRWB_SFR_SIZE);
float energy = ff_scalarproduct_float_c(synth_exc, synth_exc, AMRWB_SFR_SIZE);

/* Generate a white-noise excitation */
for (i = 0; i < AMRWB_SFR_SIZE_16k; i++)
@@ -1156,8 +1159,10 @@ static int amrwb_decode_frame(AVCodecContext *avctx, void *data,

ctx->fixed_gain[0] =
ff_amr_set_fixed_gain(fixed_gain_factor,
ff_dot_productf(ctx->fixed_vector, ctx->fixed_vector,
AMRWB_SFR_SIZE) / AMRWB_SFR_SIZE,
ff_scalarproduct_float_c(ctx->fixed_vector,
ctx->fixed_vector,
AMRWB_SFR_SIZE) /
AMRWB_SFR_SIZE,
ctx->prediction_error,
ENERGY_MEAN, energy_pred_fac);



+ 0
- 11
libavcodec/celp_math.c View File

@@ -86,14 +86,3 @@ int ff_log2(uint32_t value)

return (power_int << 15) + value;
}

float ff_dot_productf(const float* a, const float* b, int length)
{
float sum = 0;
int i;

for(i=0; i<length; i++)
sum += a[i] * b[i];

return sum;
}

+ 0
- 10
libavcodec/celp_math.h View File

@@ -55,14 +55,4 @@ static inline int bidir_sal(int value, int offset)
else return value << offset;
}

/**
* Return the dot product.
* @param a input data array
* @param b input data array
* @param length number of elements
*
* @return dot product = sum of elementwise products
*/
float ff_dot_productf(const float* a, const float* b, int length);

#endif /* AVCODEC_CELP_MATH_H */

+ 2
- 2
libavcodec/dsputil.c View File

@@ -2424,7 +2424,7 @@ static void butterflies_float_interleave_c(float *dst, const float *src0,
}
}

static float scalarproduct_float_c(const float *v1, const float *v2, int len)
float ff_scalarproduct_float_c(const float *v1, const float *v2, int len)
{
float p = 0.0;
int i;
@@ -2877,7 +2877,7 @@ 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 = scalarproduct_float_c;
c->scalarproduct_float = ff_scalarproduct_float_c;
c->butterflies_float = butterflies_float_c;
c->butterflies_float_interleave = butterflies_float_interleave_c;
c->vector_fmul_scalar = vector_fmul_scalar_c;


+ 11
- 0
libavcodec/dsputil.h View File

@@ -549,6 +549,17 @@ 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


+ 0
- 1
libavcodec/lsp.c View File

@@ -27,7 +27,6 @@
#define FRAC_BITS 14
#include "mathops.h"
#include "lsp.h"
#include "celp_math.h"

void ff_acelp_reorder_lsf(int16_t* lsfq, int lsfq_min_distance, int lsfq_min, int lsfq_max, int lp_order)
{


+ 6
- 7
libavcodec/qcelpdec.c View File

@@ -32,10 +32,8 @@
#include "avcodec.h"
#include "internal.h"
#include "get_bits.h"
#include "dsputil.h"
#include "qcelpdata.h"

#include "celp_math.h"
#include "celp_filters.h"
#include "acelp_filters.h"
#include "acelp_vectors.h"
@@ -401,8 +399,9 @@ static void apply_gain_ctrl(float *v_out, const float *v_ref, const float *v_in)

for (i = 0; i < 160; i += 40)
ff_scale_vector_to_given_sum_of_squares(v_out + i, v_in + i,
ff_dot_productf(v_ref + i,
v_ref + i, 40),
ff_scalarproduct_float_c(v_ref + i,
v_ref + i,
40),
40);
}

@@ -678,8 +677,8 @@ 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_dot_productf(q->formant_mem + 10,
q->formant_mem + 10, 160),
ff_scalarproduct_float_c(q->formant_mem + 10,
q->formant_mem + 10, 160),
160, 0.9375, &q->postfilter_agc_mem);
}



+ 2
- 3
libavcodec/ra288.c View File

@@ -25,7 +25,6 @@
#include "get_bits.h"
#include "ra288.h"
#include "lpc.h"
#include "celp_math.h"
#include "celp_filters.h"

#define MAX_BACKWARD_FILTER_ORDER 36
@@ -74,7 +73,7 @@ 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_dot_productf(src, src - n, len);
tgt[n] = ff_scalarproduct_float_c(src, src - n, len);

}

@@ -103,7 +102,7 @@ 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_dot_productf(buffer, buffer, 5) * ((1<<24)/5.);
sum = ff_scalarproduct_float_c(buffer, buffer, 5) * ((1 << 24) / 5.);

sum = FFMAX(sum, 1);



+ 4
- 5
libavcodec/sipr.c View File

@@ -32,7 +32,6 @@
#include "dsputil.h"

#include "lsp.h"
#include "celp_math.h"
#include "acelp_vectors.h"
#include "acelp_pitch_delay.h"
#include "acelp_filters.h"
@@ -411,7 +410,7 @@ static void decode_frame(SiprContext *ctx, SiprParameters *params,
SUBFR_SIZE);

avg_energy =
(0.01 + ff_dot_productf(fixed_vector, fixed_vector, SUBFR_SIZE))/
(0.01 + ff_scalarproduct_float_c(fixed_vector, fixed_vector, SUBFR_SIZE)) /
SUBFR_SIZE;

ctx->past_pitch_gain = pitch_gain = gain_cb[params->gc_index[i]][0];
@@ -453,9 +452,9 @@ static void decode_frame(SiprContext *ctx, SiprParameters *params,

if (ctx->mode == MODE_5k0) {
for (i = 0; i < subframe_count; i++) {
float energy = ff_dot_productf(ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
ctx->postfilter_syn5k0 + LP_FILTER_ORDER + i*SUBFR_SIZE,
SUBFR_SIZE);
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);
ff_adaptive_gain_control(&synth[i * SUBFR_SIZE],
&synth[i * SUBFR_SIZE], energy,
SUBFR_SIZE, 0.9, &ctx->postfilter_agc);


+ 4
- 3
libavcodec/sipr16k.c View File

@@ -26,8 +26,9 @@
#include "sipr.h"
#include "libavutil/common.h"
#include "libavutil/mathematics.h"
#include "dsputil.h"
#include "lsp.h"
#include "celp_math.h"
#include "celp_filters.h"
#include "acelp_vectors.h"
#include "acelp_pitch_delay.h"
#include "acelp_filters.h"
@@ -163,10 +164,10 @@ static float acelp_decode_gain_codef(float gain_corr_factor, const float *fc_v,
int subframe_size, int ma_pred_order)
{
mr_energy +=
ff_dot_productf(quant_energy, ma_prediction_coeff, ma_pred_order);
ff_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_dot_productf(fc_v, fc_v, subframe_size)));
sqrt((0.01 + ff_scalarproduct_float_c(fc_v, fc_v, subframe_size)));
return mr_energy;
}



+ 8
- 7
libavcodec/wmavoice.c View File

@@ -28,11 +28,12 @@
#define UNCHECKED_BITSTREAM_READER 1

#include <math.h>

#include "dsputil.h"
#include "avcodec.h"
#include "get_bits.h"
#include "put_bits.h"
#include "wmavoice_data.h"
#include "celp_math.h"
#include "celp_filters.h"
#include "acelp_vectors.h"
#include "acelp_filters.h"
@@ -518,7 +519,7 @@ static int kalman_smoothen(WMAVoiceContext *s, int pitch,

/* find best fitting point in history */
do {
dot = ff_dot_productf(in, ptr, size);
dot = ff_scalarproduct_float_c(in, ptr, size);
if (dot > optimal_gain) {
optimal_gain = dot;
best_hist_ptr = ptr;
@@ -527,7 +528,7 @@ static int kalman_smoothen(WMAVoiceContext *s, int pitch,

if (optimal_gain <= 0)
return -1;
dot = ff_dot_productf(best_hist_ptr, best_hist_ptr, size);
dot = ff_scalarproduct_float_c(best_hist_ptr, best_hist_ptr, size);
if (dot <= 0) // would be 1.0
return -1;

@@ -557,8 +558,8 @@ static float tilt_factor(const float *lpcs, int n_lpcs)
{
float rh0, rh1;

rh0 = 1.0 + ff_dot_productf(lpcs, lpcs, n_lpcs);
rh1 = lpcs[0] + ff_dot_productf(lpcs, &lpcs[1], n_lpcs - 1);
rh0 = 1.0 + ff_scalarproduct_float_c(lpcs, lpcs, n_lpcs);
rh1 = lpcs[0] + ff_scalarproduct_float_c(lpcs, &lpcs[1], n_lpcs - 1);

return rh1 / rh0;
}
@@ -651,7 +652,7 @@ 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_dot_productf(coeffs, coeffs, remainder));
sq = (1.0 / 64.0) * sqrtf(1 / ff_scalarproduct_float_c(coeffs, coeffs, remainder));
for (n = 0; n < remainder; n++)
coeffs[n] *= sq;
}
@@ -1315,7 +1316,7 @@ 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_dot_productf(s->gain_pred_err, gain_coeff, 6) -
fcb_gain = expf(ff_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],


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