44 #define PSY_3GPP_THR_SPREAD_HI   1.5f // spreading factor for low-to-hi threshold spreading  (15 dB/Bark) 
   45 #define PSY_3GPP_THR_SPREAD_LOW  3.0f // spreading factor for hi-to-low threshold spreading  (30 dB/Bark) 
   47 #define PSY_3GPP_EN_SPREAD_HI_L1 2.0f 
   49 #define PSY_3GPP_EN_SPREAD_HI_L2 1.5f 
   51 #define PSY_3GPP_EN_SPREAD_HI_S  1.5f 
   53 #define PSY_3GPP_EN_SPREAD_LOW_L 3.0f 
   55 #define PSY_3GPP_EN_SPREAD_LOW_S 2.0f 
   57 #define PSY_3GPP_RPEMIN      0.01f 
   58 #define PSY_3GPP_RPELEV      2.0f 
   60 #define PSY_3GPP_C1          3.0f            
   61 #define PSY_3GPP_C2          1.3219281f      
   62 #define PSY_3GPP_C3          0.55935729f     
   64 #define PSY_SNR_1DB          7.9432821e-1f   
   65 #define PSY_SNR_25DB         3.1622776e-3f   
   67 #define PSY_3GPP_SAVE_SLOPE_L  -0.46666667f 
   68 #define PSY_3GPP_SAVE_SLOPE_S  -0.36363637f 
   69 #define PSY_3GPP_SAVE_ADD_L    -0.84285712f 
   70 #define PSY_3GPP_SAVE_ADD_S    -0.75f 
   71 #define PSY_3GPP_SPEND_SLOPE_L  0.66666669f 
   72 #define PSY_3GPP_SPEND_SLOPE_S  0.81818181f 
   73 #define PSY_3GPP_SPEND_ADD_L   -0.35f 
   74 #define PSY_3GPP_SPEND_ADD_S   -0.26111111f 
   75 #define PSY_3GPP_CLIP_LO_L      0.2f 
   76 #define PSY_3GPP_CLIP_LO_S      0.2f 
   77 #define PSY_3GPP_CLIP_HI_L      0.95f 
   78 #define PSY_3GPP_CLIP_HI_S      0.75f 
   80 #define PSY_3GPP_AH_THR_LONG    0.5f 
   81 #define PSY_3GPP_AH_THR_SHORT   0.63f 
   83 #define PSY_PE_FORGET_SLOPE  511 
   91 #define PSY_3GPP_BITS_TO_PE(bits) ((bits) * 1.18f) 
   92 #define PSY_3GPP_PE_TO_BITS(bits) ((bits) / 1.18f) 
   95 #define PSY_LAME_FIR_LEN 21          
   96 #define AAC_BLOCK_SIZE_LONG 1024    
 
   97 #define AAC_BLOCK_SIZE_SHORT 128    
 
   98 #define AAC_NUM_BLOCKS_SHORT 8      
 
   99 #define PSY_LAME_NUM_SUBBLOCKS 3    
 
  220     -8.65163e-18 * 2, -0.00851586 * 2, -6.74764e-18 * 2, 0.0209036 * 2,
 
  221     -3.36639e-17 * 2, -0.0438162 * 2,  -1.54175e-17 * 2, 0.0931738 * 2,
 
  222     -5.52212e-17 * 2, -0.313819 * 2
 
  235     int lower_range = 12, upper_range = 12;
 
  236     int lower_range_kbps = psy_abr_map[12].
quality;
 
  237     int upper_range_kbps = psy_abr_map[12].
quality;
 
  243     for (i = 1; i < 13; i++) {
 
  244         if (
FFMAX(bitrate, psy_abr_map[i].quality) != bitrate) {
 
  246             upper_range_kbps = psy_abr_map[i    ].
quality;
 
  248             lower_range_kbps = psy_abr_map[i - 1].
quality;
 
  254     if ((upper_range_kbps - bitrate) > (bitrate - lower_range_kbps))
 
  255         return psy_abr_map[lower_range].
st_lrm;
 
  256     return psy_abr_map[upper_range].
st_lrm;
 
  266     for (i = 0; i < avctx->
channels; i++) {
 
  284     return 13.3f * 
atanf(0.00076f * f) + 3.5f * 
atanf((f / 7500.0f) * (f / 7500.0f));
 
  295     return    3.64 * pow(f, -0.8)
 
  296             - 6.8  * 
exp(-0.6  * (f - 3.4) * (f - 3.4))
 
  297             + 6.0  * 
exp(-0.15 * (f - 8.7) * (f - 8.7))
 
  298             + (0.6 + 0.04 * add) * 0.001 * f * f * f * f;
 
  305     float prev, minscale, minath, minsnr, pe_min;
 
  309     const float num_bark   = 
calc_bark((
float)bandwidth);
 
  330     for (j = 0; j < 2; j++) {
 
  334         float avg_chan_bits = chan_bitrate * (j ? 128.0f : 1024.0f) / ctx->
avctx->
sample_rate;
 
  343         for (g = 0; g < ctx->
num_bands[j]; g++) {
 
  345             bark = 
calc_bark((i-1) * line_to_frequency);
 
  346             coeffs[
g].
barks = (bark + prev) / 2.0;
 
  349         for (g = 0; g < ctx->
num_bands[j] - 1; g++) {
 
  351             float bark_width = coeffs[g+1].
barks - coeffs->
barks;
 
  356             pe_min = bark_pe * bark_width;
 
  357             minsnr = 
exp2(pe_min / band_sizes[g]) - 1.5f;
 
  361         for (g = 0; g < ctx->
num_bands[j]; g++) {
 
  362             minscale = 
ath(start * line_to_frequency, 
ATH_ADD);
 
  363             for (i = 1; i < band_sizes[
g]; i++)
 
  364                 minscale = 
FFMIN(minscale, 
ath((start + i) * line_to_frequency, 
ATH_ADD));
 
  365             coeffs[
g].
ath = minscale - minath;
 
  366             start += band_sizes[
g];
 
  388     ret = 0.7548f * (in - state[0]) + 0.5095f * state[1];
 
  398     0xB6, 0x6C, 0xD8, 0xB2, 0x66, 0xC6, 0x96, 0x36, 0x36
 
  406                                                  const int16_t *audio,
 
  412     int attack_ratio     = br <= 16000 ? 18 : 10;
 
  416     int next_type        = pch->next_window_seq;
 
  421         int switch_to_eight = 0;
 
  422         float sum = 0.0, sum2 = 0.0;
 
  425         for (i = 0; i < 8; i++) {
 
  426             for (j = 0; j < 128; j++) {
 
  433         for (i = 0; i < 8; i++) {
 
  434             if (s[i] > pch->win_energy * attack_ratio) {
 
  440         pch->win_energy = pch->win_energy*7/8 + sum2/64;
 
  442         wi.window_type[1] = prev_type;
 
  450             grouping = pch->next_grouping;
 
  466         pch->next_window_seq = next_type;
 
  468         for (i = 0; i < 3; i++)
 
  469             wi.window_type[i] = prev_type;
 
  480         for (i = 0; i < 8; i++) {
 
  481             if (!((grouping >> i) & 1))
 
  483             wi.grouping[lastgrp]++;
 
  500     float clipped_pe, bit_save, bit_spend, bit_factor, fill_level, forgetful_min_pe;
 
  504     fill_level = av_clipf((
float)ctx->
fill_level / size, clip_low, clip_high);
 
  505     clipped_pe = av_clipf(pe, ctx->
pe.
min, ctx->
pe.
max);
 
  506     bit_save   = (fill_level + bitsave_add) * bitsave_slope;
 
  507     assert(bit_save <= 0.3f && bit_save >= -0.05000001f);
 
  508     bit_spend  = (fill_level + bitspend_add) * bitspend_slope;
 
  509     assert(bit_spend <= 0.5f && bit_spend >= -0.1f);
 
  516     bit_factor = 1.0f - bit_save + ((bit_spend - bit_save) / (ctx->
pe.
max - ctx->
pe.
min)) * (clipped_pe - ctx->
pe.
min);
 
  560     float thr_avg, reduction;
 
  562     if(active_lines == 0.0)
 
  565     thr_avg   = 
exp2f((a - pe) / (4.0f * active_lines));
 
  566     reduction = 
exp2f((a - desired_pe) / (4.0f * active_lines)) - thr_avg;
 
  568     return FFMAX(reduction, 0.0f);
 
  574     float thr = band->
thr;
 
  578         thr = sqrtf(thr) + reduction;
 
  596 #ifndef calc_thr_3gpp 
  598                           const uint8_t *band_sizes, 
const float *coefs, 
const int cutoff)
 
  601     int start = 0, wstart = 0;
 
  604         for (g = 0; g < num_bands; g++) {
 
  607             float form_factor = 0.0f;
 
  610             if (wstart < cutoff) {
 
  611                 for (i = 0; i < band_sizes[
g]; i++) {
 
  612                     band->
energy += coefs[start+i] * coefs[start+i];
 
  613                     form_factor  += sqrtf(fabs(coefs[start+i]));
 
  616             Temp = band->
energy > 0 ? sqrtf((
float)band_sizes[g] / band->
energy) : 0;
 
  618             band->
nz_lines = form_factor * sqrtf(Temp);
 
  620             start += band_sizes[
g];
 
  621             wstart += band_sizes[
g];
 
  627 #ifndef psy_hp_filter 
  636             sum1 += psy_fir_coeffs[j] * (firbuf[i + j] + firbuf[i + 
PSY_LAME_FIR_LEN - j]);
 
  637             sum2 += psy_fir_coeffs[j + 1] * (firbuf[i + j + 1] + firbuf[i + 
PSY_LAME_FIR_LEN - j - 1]);
 
  641         hpfsmpl[i] = (sum1 + sum2) * 32768.0f;
 
  655     float desired_bits, desired_pe, delta_pe, reduction= 
NAN, spread_en[128] = {0};
 
  656     float a = 0.0f, active_lines = 0.0f, norm_fac = 0.0f;
 
  657     float pe = pctx->chan_bitrate > 32000 ? 0.0f : 
FFMAX(50.0f, 100.0f - pctx->chan_bitrate * 100.0f / 32000.0f);
 
  658     const int      num_bands   = 
ctx->num_bands[wi->num_windows == 8];
 
  659     const uint8_t *band_sizes  = 
ctx->bands[wi->num_windows == 8];
 
  663     const int cutoff           = bandwidth * 2048 / wi->num_windows / 
ctx->avctx->sample_rate;
 
  666     calc_thr_3gpp(wi, num_bands, pch, band_sizes, coefs, cutoff);
 
  669     for (w = 0; w < wi->num_windows*16; w += 16) {
 
  673         spread_en[0] = bands[0].
energy;
 
  674         for (
g = 1; 
g < num_bands; 
g++) {
 
  675             bands[
g].
thr   = 
FFMAX(bands[
g].thr,    bands[
g-1].thr * coeffs[
g].spread_hi[0]);
 
  676             spread_en[w+
g] = 
FFMAX(bands[
g].energy, spread_en[w+
g-1] * coeffs[
g].spread_hi[1]);
 
  678         for (
g = num_bands - 2; 
g >= 0; 
g--) {
 
  679             bands[
g].
thr   = 
FFMAX(bands[
g].thr,   bands[
g+1].thr * coeffs[
g].spread_low[0]);
 
  680             spread_en[w+
g] = 
FFMAX(spread_en[w+
g], spread_en[w+
g+1] * coeffs[
g].spread_low[1]);
 
  683         for (
g = 0; 
g < num_bands; 
g++) {
 
  698             if (spread_en[w+
g] * avoid_hole_thr > band->
energy || coeffs[
g].
min_snr > 1.0f)
 
  711         desired_pe = pe * (
ctx->avctx->global_quality ? 
ctx->avctx->global_quality : 120) / (2 * 2.5f * 120.0f);
 
  716         if (
ctx->bitres.bits > 0) {
 
  721         pctx->pe.max = 
FFMAX(pe, pctx->pe.max);
 
  722         pctx->pe.min = 
FFMIN(pe, pctx->pe.min);
 
  731         if (
ctx->bitres.bits > 0)
 
  736     ctx->bitres.alloc = desired_bits;
 
  738     if (desired_pe < pe) {
 
  740         for (w = 0; w < wi->num_windows*16; w += 16) {
 
  745             for (
g = 0; 
g < num_bands; 
g++) {
 
  757         for (i = 0; i < 2; i++) {
 
  758             float pe_no_ah = 0.0f, desired_pe_no_ah;
 
  759             active_lines = a = 0.0f;
 
  760             for (w = 0; w < wi->num_windows*16; w += 16) {
 
  761                 for (
g = 0; 
g < num_bands; 
g++) {
 
  765                         pe_no_ah += band->
pe;
 
  771             desired_pe_no_ah = 
FFMAX(desired_pe - (pe - pe_no_ah), 0.0f);
 
  772             if (active_lines > 0.0f)
 
  776             for (w = 0; w < wi->num_windows*16; w += 16) {
 
  777                 for (
g = 0; 
g < num_bands; 
g++) {
 
  780                     if (active_lines > 0.0f)
 
  783                     if (band->
thr > 0.0f)
 
  790             delta_pe = desired_pe - pe;
 
  791             if (fabs(delta_pe) > 0.05f * desired_pe)
 
  795         if (pe < 1.15f * desired_pe) {
 
  797             norm_fac = 1.0f / norm_fac;
 
  798             for (w = 0; w < wi->num_windows*16; w += 16) {
 
  799                 for (
g = 0; 
g < num_bands; 
g++) {
 
  803                         float delta_sfb_pe = band->
norm_fac * norm_fac * delta_pe;
 
  804                         float thr = band->
thr;
 
  816             while (pe > desired_pe && 
g--) {
 
  817                 for (w = 0; w < wi->num_windows*16; w+= 16) {
 
  830     for (w = 0; w < wi->num_windows*16; w += 16) {
 
  831         for (
g = 0; 
g < num_bands; 
g++) {
 
  842     memcpy(pch->prev_band, pch->band, 
sizeof(pch->band));
 
  851     for (ch = 0; ch < group->
num_ch; ch++)
 
  881                                        const float *la, 
int channel, 
int prev_type)
 
  886     int uselongblock = 1;
 
  893         const float *pf = hpfsmpl;
 
  905             energy_subshort[i] = pch->prev_energy_subshort[i + ((
AAC_NUM_BLOCKS_SHORT - 1) * PSY_LAME_NUM_SUBBLOCKS)];
 
  906             assert(pch->prev_energy_subshort[i + ((
AAC_NUM_BLOCKS_SHORT - 2) * PSY_LAME_NUM_SUBBLOCKS + 1)] > 0);
 
  907             attack_intensity[i] = energy_subshort[i] / pch->prev_energy_subshort[i + ((
AAC_NUM_BLOCKS_SHORT - 2) * PSY_LAME_NUM_SUBBLOCKS + 1)];
 
  908             energy_short[0] += energy_subshort[i];
 
  914             for (; pf < pfe; pf++)
 
  915                 p = 
FFMAX(p, fabsf(*pf));
 
  925             if (p > energy_subshort[i + 1])
 
  926                 p = p / energy_subshort[i + 1];
 
  927             else if (energy_subshort[i + 1] > p * 10.0f)
 
  928                 p = energy_subshort[i + 1] / (p * 10.0f);
 
  936             if (!attacks[i / PSY_LAME_NUM_SUBBLOCKS])
 
  937                 if (attack_intensity[i] > pch->attack_threshold)
 
  945             const float u = energy_short[i - 1];
 
  946             const float v = energy_short[i];
 
  947             const float m = 
FFMAX(u, v);
 
  949                 if (u < 1.7f * v && v < 1.7f * u) {   
 
  950                     if (i == 1 && attacks[0] < attacks[i])
 
  955             att_sum += attacks[i];
 
  958         if (attacks[0] <= pch->prev_attack)
 
  961         att_sum += attacks[0];
 
  963         if (pch->prev_attack == 3 || att_sum) {
 
  966             for (i = 1; i < AAC_NUM_BLOCKS_SHORT + 1; i++)
 
  967                 if (attacks[i] && attacks[i-1])
 
  992         for (i = 0; i < 8; i++) {
 
  993             if (!((pch->next_grouping >> i) & 1))
 
 1005     for (i = 0; i < 9; i++) {
 
 1013     pch->prev_attack = attacks[8];
 
 1020     .
name    = 
"3GPP TS 26.403-inspired model",
 
int quality
Quality to map the rest of the vaules to. 
float global_quality
normalized global quality taken from avctx 
static const uint8_t window_grouping[9]
window grouping information stored as bits (0 - new group, 1 - group continues) 
int grouping[8]
window grouping (for e.g. AAC) 
#define AAC_BLOCK_SIZE_SHORT
short block size 
static int calc_bit_demand(AacPsyContext *ctx, float pe, int bits, int size, int short_window)
uint8_t ** bands
scalefactor band sizes for possible frame sizes 
#define PSY_3GPP_AH_THR_SHORT
float iir_state[2]
hi-pass IIR filter state 
int64_t bit_rate
the average bitrate 
static const PsyLamePreset psy_vbr_map[]
LAME psy model preset table for constant quality. 
psychoacoustic information for an arbitrary group of channels 
static float calc_reduction_3gpp(float a, float desired_pe, float pe, float active_lines)
float ath
absolute threshold of hearing per bands 
#define PSY_3GPP_EN_SPREAD_HI_L1
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
static av_cold float ath(float f, float add)
Calculate ATH value for given frequency. 
float prev_energy_subshort[AAC_NUM_BLOCKS_SHORT *PSY_LAME_NUM_SUBBLOCKS]
enum WindowSequence next_window_seq
window sequence to be used in the next frame 
#define AAC_BLOCK_SIZE_LONG
long block size 
int * num_bands
number of scalefactor bands for possible frame sizes 
Macro definitions for various function/variable attributes. 
LAME psy model preset struct. 
float thr
energy threshold 
float correction
PE correction factor. 
static av_cold void psy_3gpp_end(FFPsyContext *apc)
float attack_threshold
attack threshold for this channel 
#define PSY_3GPP_EN_SPREAD_LOW_L
float nz_lines
number of non-zero spectral lines 
psychoacoustic model frame type-dependent coefficients 
int size
size of the bitresevoir in bits 
static float calc_reduced_thr_3gpp(AacPsyBand *band, float min_snr, float reduction)
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
#define PSY_LAME_FIR_LEN
LAME psy model FIR order. 
#define PSY_3GPP_CLIP_LO_L
#define PSY_3GPP_SPEND_SLOPE_S
#define PSY_3GPP_THR_SPREAD_LOW
context used by psychoacoustic model 
int flags
Flags modifying the (de)muxer behaviour. 
single band psychoacoustic information 
static float lame_calc_attack_threshold(int bitrate)
Calculate the ABR attack threshold from the above LAME psymodel table. 
uint8_t next_grouping
stored grouping scheme for the next frame (in case of 8 short window sequence) 
#define PSY_3GPP_SAVE_ADD_L
static av_cold float calc_bark(float f)
Calculate Bark value for given line. 
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values. 
#define PSY_3GPP_SPEND_ADD_S
AacPsyBand prev_band[128]
bands information from the previous frame 
3GPP TS26.403-inspired psychoacoustic model specific data 
single/pair channel context for psychoacoustic model 
static const float psy_fir_coeffs[]
LAME psy model FIR coefficient table. 
float barks
Bark value for each spectral band in long frame. 
int flags
AV_CODEC_FLAG_*. 
float pe_const
constant part of the PE calculation 
int num_windows
number of windows in a frame 
static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio, const float *la, int channel, int prev_type)
#define PSY_3GPP_SPEND_SLOPE_L
#define PSY_3GPP_THR_SPREAD_HI
constants for 3GPP AAC psychoacoustic model 
static void * av_mallocz_array(size_t nmemb, size_t size)
codec-specific psychoacoustic model implementation 
float thr_quiet
threshold in quiet 
static void psy_3gpp_analyze(FFPsyContext *ctx, int channel, const float **coeffs, const FFPsyWindowInfo *wi)
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale. 
int prev_attack
attack value for the last short block in the previous sequence 
#define PSY_3GPP_SAVE_SLOPE_S
uint8_t num_ch
number of channels in this group 
int frame_bits
average bits per frame 
int fill_level
bit reservoir fill level 
static void lame_apply_block_type(AacPsyChannel *ctx, FFPsyWindowInfo *wi, int uselongblock)
#define PSY_3GPP_SAVE_SLOPE_L
Reference: libavcodec/aacpsy.c. 
#define PSY_LAME_NUM_SUBBLOCKS
Number of sub-blocks in each short block. 
const FFPsyModel ff_aac_psy_model
static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel, const float *coefs, const FFPsyWindowInfo *wi)
Calculate band thresholds as suggested in 3GPP TS26.403. 
float st_lrm
short threshold for L, R, and M channels 
#define PSY_3GPP_EN_SPREAD_LOW_S
Libavcodec external API header. 
int sample_rate
samples per second 
FFPsyChannelGroup * ff_psy_find_group(FFPsyContext *ctx, int channel)
Determine what group a channel belongs to. 
main external API structure. 
float win_energy
sliding average of channel energy 
void * model_priv_data
psychoacoustic model implementation private data 
float active_lines
number of active spectral lines 
static const float bands[]
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
static float iir_filter(int in, float state[2])
IIR filter used in block switching decision. 
int avoid_holes
hole avoidance flag 
AacPsyBand band[128]
bands information 
struct AacPsyContext::@45 pe
#define PSY_3GPP_CLIP_HI_S
static const PsyLamePreset psy_abr_map[]
LAME psy model preset table for ABR. 
int window_shape
window shape (sine/KBD/whatever) 
#define PSY_PE_FORGET_SLOPE
#define PSY_3GPP_PE_TO_BITS(bits)
int cutoff
lowpass frequency cutoff for analysis 
float max
maximum allowed PE for bit factor calculation 
static void calc_thr_3gpp(const FFPsyWindowInfo *wi, const int num_bands, AacPsyChannel *pch, const uint8_t *band_sizes, const float *coefs, const int cutoff)
float previous
allowed PE of the previous frame 
AacPsyCoeffs psy_coef[2][64]
float min
minimum allowed PE for bit factor calculation 
int global_quality
Global quality for codecs which cannot change it per frame. 
static av_cold int psy_3gpp_init(FFPsyContext *ctx)
struct FFPsyContext::@104 bitres
static void psy_hp_filter(const float *firbuf, float *hpfsmpl, const float *psy_fir_coeffs)
float spread_hi[2]
spreading factor for high-to-low threshold spreading in long frame 
internal math functions header 
static av_unused FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx, const int16_t *audio, const int16_t *la, int channel, int prev_type)
Tell encoder which window types to use. 
static float calc_pe_3gpp(AacPsyBand *band)
windowing related information 
channel
Use these values when setting the channel map with ebur128_set_channel(). 
#define PSY_3GPP_BITS_TO_PE(bits)
float norm_fac
normalization factor for linearization 
int chan_bitrate
bitrate per channel 
#define PSY_3GPP_CLIP_LO_S
#define PSY_3GPP_AH_THR_LONG
static const int16_t coeffs[]
int channels
number of audio channels 
float pe
perceptual entropy 
#define PSY_3GPP_EN_SPREAD_HI_S
static const double coeff[2][5]
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda 
#define PSY_3GPP_SAVE_ADD_S
information for single band used by 3GPP TS26.403-inspired psychoacoustic model 
AVCodecContext * avctx
encoder context 
float spread_low[2]
spreading factor for low-to-high threshold spreading in long frame 
#define PSY_3GPP_CLIP_HI_L
int window_type[3]
window type (short/long/transitional, etc.) - current, previous and next 
#define AAC_NUM_BLOCKS_SHORT
number of blocks in a short sequence 
#define PSY_3GPP_SPEND_ADD_L
static av_cold void lame_window_init(AacPsyContext *ctx, AVCodecContext *avctx)
LAME psy model specific initialization.