44 #define PSY_3GPP_THR_SPREAD_HI 1.5f
45 #define PSY_3GPP_THR_SPREAD_LOW 3.0f
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;
243 for (
i = 1;
i < 13;
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);
315 if (!
ctx->model_priv_data)
317 pctx =
ctx->model_priv_data;
322 chan_bitrate = (
int)(chan_bitrate / 120.0 * (
ctx->avctx->global_quality ?
ctx->avctx->global_quality : 120));
330 ctx->bitres.size -=
ctx->bitres.size % 8;
333 for (j = 0; j < 2; j++) {
336 float line_to_frequency =
ctx->avctx->sample_rate / (j ? 256.f : 2048.0f);
337 float avg_chan_bits = chan_bitrate * (j ? 128.0f : 1024.0f) /
ctx->avctx->sample_rate;
346 for (
g = 0;
g <
ctx->num_bands[j];
g++) {
349 coeffs[
g].
barks = (bark + prev) / 2.0;
352 for (
g = 0;
g <
ctx->num_bands[j] - 1;
g++) {
354 float bark_width = coeffs[
g+1].
barks - coeffs->
barks;
357 coeff->spread_low[1] =
ff_exp10(-bark_width * en_spread_low);
359 pe_min = bark_pe * bark_width;
360 minsnr =
exp2(pe_min / band_sizes[
g]) - 1.5f;
364 for (
g = 0;
g <
ctx->num_bands[j];
g++) {
365 minscale =
ath(start * line_to_frequency,
ATH_ADD);
366 for (
i = 1;
i < band_sizes[
g];
i++)
368 coeffs[
g].
ath = minscale - minath;
369 start += band_sizes[
g];
401 0xB6, 0x6C, 0xD8, 0xB2, 0x66, 0xC6, 0x96, 0x36, 0x36
409 const int16_t *audio,
415 int attack_ratio = br <= 16000 ? 18 : 10;
419 int next_type = pch->next_window_seq;
424 int switch_to_eight = 0;
425 float sum = 0.0, sum2 = 0.0;
428 for (
i = 0;
i < 8;
i++) {
429 for (j = 0; j < 128; j++) {
436 for (
i = 0;
i < 8;
i++) {
437 if (
s[
i] > pch->win_energy * attack_ratio) {
443 pch->win_energy = pch->win_energy*7/8 + sum2/64;
445 wi.window_type[1] = prev_type;
453 grouping = pch->next_grouping;
469 pch->next_window_seq = next_type;
471 for (
i = 0;
i < 3;
i++)
472 wi.window_type[
i] = prev_type;
483 for (
i = 0;
i < 8;
i++) {
484 if (!((grouping >>
i) & 1))
486 wi.grouping[lastgrp]++;
503 float clipped_pe, bit_save, bit_spend, bit_factor, fill_level, forgetful_min_pe;
507 fill_level =
av_clipf((
float)
ctx->fill_level /
size, clip_low, clip_high);
509 bit_save = (fill_level + bitsave_add) * bitsave_slope;
510 assert(bit_save <= 0.3f && bit_save >= -0.05000001f);
511 bit_spend = (fill_level + bitspend_add) * bitspend_slope;
512 assert(bit_spend <= 0.5f && bit_spend >= -0.1f);
519 bit_factor = 1.0f - bit_save + ((bit_spend - bit_save) / (
ctx->pe.max -
ctx->pe.min)) * (clipped_pe -
ctx->pe.min);
527 ctx->pe.min =
FFMIN(pe, forgetful_min_pe);
533 ctx->frame_bits * bit_factor,
563 float thr_avg, reduction;
565 if(active_lines == 0.0)
568 thr_avg =
exp2f((
a - pe) / (4.0f * active_lines));
569 reduction =
exp2f((
a - desired_pe) / (4.0f * active_lines)) - thr_avg;
571 return FFMAX(reduction, 0.0f);
577 float thr = band->
thr;
581 thr = sqrtf(thr) + reduction;
599 #ifndef calc_thr_3gpp
601 const uint8_t *band_sizes,
const float *coefs,
const int cutoff)
604 int start = 0, wstart = 0;
607 for (
g = 0;
g < num_bands;
g++) {
610 float form_factor = 0.0f;
613 if (wstart < cutoff) {
614 for (
i = 0;
i < band_sizes[
g];
i++) {
615 band->
energy += coefs[start+
i] * coefs[start+
i];
616 form_factor += sqrtf(
fabs(coefs[start+
i]));
619 Temp = band->
energy > 0 ? sqrtf((
float)band_sizes[
g] / band->
energy) : 0;
621 band->
nz_lines = form_factor * sqrtf(Temp);
623 start += band_sizes[
g];
624 wstart += band_sizes[
g];
630 #ifndef psy_hp_filter
644 hpfsmpl[
i] = (sum1 + sum2) * 32768.0f;
658 float desired_bits, desired_pe, delta_pe, reduction=
NAN, spread_en[128] = {0};
659 float a = 0.0f, active_lines = 0.0f, norm_fac = 0.0f;
660 float pe = pctx->chan_bitrate > 32000 ? 0.0f :
FFMAX(50.0f, 100.0f - pctx->chan_bitrate * 100.0f / 32000.0f);
661 const int num_bands =
ctx->num_bands[wi->num_windows == 8];
662 const uint8_t *band_sizes =
ctx->bands[wi->num_windows == 8];
663 AacPsyCoeffs *coeffs = pctx->psy_coef[wi->num_windows == 8];
666 const int cutoff = bandwidth * 2048 / wi->num_windows /
ctx->avctx->sample_rate;
669 calc_thr_3gpp(wi, num_bands, pch, band_sizes, coefs, cutoff);
672 for (
w = 0;
w < wi->num_windows*16;
w += 16) {
676 spread_en[0] =
bands[0].energy;
677 for (
g = 1;
g < num_bands;
g++) {
679 spread_en[
w+
g] =
FFMAX(
bands[
g].energy, spread_en[
w+
g-1] * coeffs[
g].spread_hi[1]);
681 for (
g = num_bands - 2;
g >= 0;
g--) {
683 spread_en[
w+
g] =
FFMAX(spread_en[
w+
g], spread_en[
w+
g+1] * coeffs[
g].spread_low[1]);
686 for (
g = 0;
g < num_bands;
g++) {
701 if (spread_en[
w+
g] * avoid_hole_thr > band->
energy || coeffs[
g].min_snr > 1.0f)
714 desired_pe = pe * (
ctx->avctx->global_quality ?
ctx->avctx->global_quality : 120) / (2 * 2.5f * 120.0f);
719 if (
ctx->bitres.bits > 0) {
724 pctx->pe.max =
FFMAX(pe, pctx->pe.max);
725 pctx->pe.min =
FFMIN(pe, pctx->pe.min);
734 if (
ctx->bitres.bits > 0)
739 ctx->bitres.alloc = desired_bits;
741 if (desired_pe < pe) {
743 for (
w = 0;
w < wi->num_windows*16;
w += 16) {
748 for (
g = 0;
g < num_bands;
g++) {
760 for (
i = 0;
i < 2;
i++) {
761 float pe_no_ah = 0.0f, desired_pe_no_ah;
762 active_lines =
a = 0.0f;
763 for (
w = 0;
w < wi->num_windows*16;
w += 16) {
764 for (
g = 0;
g < num_bands;
g++) {
768 pe_no_ah += band->
pe;
774 desired_pe_no_ah =
FFMAX(desired_pe - (pe - pe_no_ah), 0.0f);
775 if (active_lines > 0.0f)
779 for (
w = 0;
w < wi->num_windows*16;
w += 16) {
780 for (
g = 0;
g < num_bands;
g++) {
783 if (active_lines > 0.0f)
786 if (band->
thr > 0.0f)
793 delta_pe = desired_pe - pe;
794 if (
fabs(delta_pe) > 0.05f * desired_pe)
798 if (pe < 1.15f * desired_pe) {
800 norm_fac = norm_fac ? 1.0f / norm_fac : 0;
801 for (
w = 0;
w < wi->num_windows*16;
w += 16) {
802 for (
g = 0;
g < num_bands;
g++) {
806 float delta_sfb_pe = band->
norm_fac * norm_fac * delta_pe;
807 float thr = band->
thr;
819 while (pe > desired_pe &&
g--) {
820 for (
w = 0;
w < wi->num_windows*16;
w+= 16) {
833 for (
w = 0;
w < wi->num_windows*16;
w += 16) {
834 for (
g = 0;
g < num_bands;
g++) {
845 memcpy(pch->prev_band, pch->band,
sizeof(pch->band));
854 for (ch = 0; ch < group->
num_ch; ch++)
880 ctx->next_window_seq = blocktype;
884 const float *la,
int channel,
int prev_type)
889 int uselongblock = 1;
896 const float *pf = hpfsmpl;
911 energy_short[0] += energy_subshort[
i];
917 for (; pf < pfe; pf++)
928 if (p > energy_subshort[
i + 1])
929 p = p / energy_subshort[
i + 1];
930 else if (energy_subshort[
i + 1] > p * 10.0f)
931 p = energy_subshort[
i + 1] / (p * 10.0f);
940 if (attack_intensity[
i] > pch->attack_threshold)
948 const float u = energy_short[
i - 1];
949 const float v = energy_short[
i];
950 const float m =
FFMAX(
u, v);
952 if (
u < 1.7f * v && v < 1.7f *
u) {
953 if (
i == 1 && attacks[0] < attacks[
i])
958 att_sum += attacks[
i];
961 if (attacks[0] <= pch->prev_attack)
964 att_sum += attacks[0];
966 if (pch->prev_attack == 3 || att_sum) {
970 if (attacks[
i] && attacks[
i-1])
995 for (
i = 0;
i < 8;
i++) {
996 if (!((pch->next_grouping >>
i) & 1))
1008 for (
i = 0;
i < 9;
i++) {
1016 pch->prev_attack = attacks[8];
1023 .
name =
"3GPP TS 26.403-inspired model",
static float lame_calc_attack_threshold(int bitrate)
Calculate the ABR attack threshold from the above LAME psymodel table.
static av_cold void lame_window_init(AacPsyContext *ctx, AVCodecContext *avctx)
LAME psy model specific initialization.
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.
#define PSY_3GPP_SPEND_SLOPE_L
static void lame_apply_block_type(AacPsyChannel *ctx, FFPsyWindowInfo *wi, int uselongblock)
#define PSY_3GPP_CLIP_HI_L
#define PSY_3GPP_EN_SPREAD_HI_L1
static const PsyLamePreset psy_abr_map[]
LAME psy model preset table for ABR.
#define PSY_3GPP_CLIP_LO_S
#define PSY_3GPP_AH_THR_LONG
static const float psy_fir_coeffs[]
LAME psy model FIR coefficient table.
static av_cold float calc_bark(float f)
Calculate Bark value for given line.
static float calc_reduced_thr_3gpp(AacPsyBand *band, float min_snr, float reduction)
#define PSY_3GPP_PE_TO_BITS(bits)
#define PSY_3GPP_THR_SPREAD_HI
constants for 3GPP AAC psychoacoustic model
static void psy_3gpp_analyze(FFPsyContext *ctx, int channel, const float **coeffs, const FFPsyWindowInfo *wi)
static float calc_reduction_3gpp(float a, float desired_pe, float pe, float active_lines)
static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio, const float *la, int channel, int prev_type)
#define PSY_3GPP_EN_SPREAD_HI_S
static void psy_hp_filter(const float *firbuf, float *hpfsmpl, const float *psy_fir_coeffs)
static float iir_filter(int in, float state[2])
IIR filter used in block switching decision.
static const uint8_t window_grouping[9]
window grouping information stored as bits (0 - new group, 1 - group continues)
#define PSY_3GPP_THR_SPREAD_LOW
#define PSY_3GPP_EN_SPREAD_LOW_L
#define PSY_3GPP_SAVE_SLOPE_L
#define PSY_PE_FORGET_SLOPE
#define AAC_BLOCK_SIZE_LONG
long block size
#define PSY_3GPP_CLIP_LO_L
#define PSY_3GPP_BITS_TO_PE(bits)
#define PSY_3GPP_SAVE_ADD_L
#define AAC_BLOCK_SIZE_SHORT
short block size
static float calc_pe_3gpp(AacPsyBand *band)
#define PSY_3GPP_AH_THR_SHORT
static const PsyLamePreset psy_vbr_map[]
LAME psy model preset table for constant quality.
#define PSY_LAME_NUM_SUBBLOCKS
Number of sub-blocks in each short block.
static av_cold float ath(float f, float add)
Calculate ATH value for given frequency.
#define AAC_NUM_BLOCKS_SHORT
number of blocks in a short sequence
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.
#define PSY_3GPP_SPEND_SLOPE_S
#define PSY_3GPP_EN_SPREAD_LOW_S
static int calc_bit_demand(AacPsyContext *ctx, float pe, int bits, int size, int short_window)
#define PSY_3GPP_CLIP_HI_S
#define PSY_3GPP_SPEND_ADD_L
static av_cold int psy_3gpp_init(FFPsyContext *ctx)
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)
#define PSY_3GPP_SAVE_SLOPE_S
#define PSY_LAME_FIR_LEN
LAME psy model FIR order.
const FFPsyModel ff_aac_psy_model
#define PSY_3GPP_SPEND_ADD_S
#define PSY_3GPP_SAVE_ADD_S
static av_cold void psy_3gpp_end(FFPsyContext *apc)
Reference: libavcodec/aacpsy.c.
static const float bands[]
Macro definitions for various function/variable attributes.
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;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);return NULL;} return ac;} 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;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->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);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
Libavcodec external API header.
#define u(width, name, range_min, range_max)
static __device__ float fabsf(float a)
static __device__ float fabs(float a)
static float add(float src0, float src1)
channel
Use these values when setting the channel map with ebur128_set_channel().
internal math functions header
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
void * av_mallocz_array(size_t nmemb, size_t size)
Allocate a memory block for an array with av_mallocz().
FFPsyChannelGroup * ff_psy_find_group(FFPsyContext *ctx, int channel)
Determine what group a channel belongs to.
main external API structure.
int global_quality
Global quality for codecs which cannot change it per frame.
int64_t bit_rate
the average bitrate
int flags
AV_CODEC_FLAG_*.
int channels
number of audio channels
int flags
Flags modifying the (de)muxer behaviour.
int64_t bit_rate
Total stream bitrate in bit/s, 0 if not available.
information for single band used by 3GPP TS26.403-inspired psychoacoustic model
float thr_quiet
threshold in quiet
float norm_fac
normalization factor for linearization
float nz_lines
number of non-zero spectral lines
float active_lines
number of active spectral lines
float thr
energy threshold
int avoid_holes
hole avoidance flag
float pe
perceptual entropy
float pe_const
constant part of the PE calculation
single/pair channel context for psychoacoustic model
float attack_threshold
attack threshold for this channel
float iir_state[2]
hi-pass IIR filter state
float win_energy
sliding average of channel energy
enum WindowSequence next_window_seq
window sequence to be used in the next frame
float prev_energy_subshort[AAC_NUM_BLOCKS_SHORT *PSY_LAME_NUM_SUBBLOCKS]
AacPsyBand prev_band[128]
bands information from the previous frame
uint8_t next_grouping
stored grouping scheme for the next frame (in case of 8 short window sequence)
int prev_attack
attack value for the last short block in the previous sequence
AacPsyBand band[128]
bands information
psychoacoustic model frame type-dependent coefficients
float ath
absolute threshold of hearing per bands
float spread_low[2]
spreading factor for low-to-high threshold spreading in long frame
float barks
Bark value for each spectral band in long frame.
float spread_hi[2]
spreading factor for high-to-low threshold spreading in long frame
3GPP TS26.403-inspired psychoacoustic model specific data
float correction
PE correction factor.
int chan_bitrate
bitrate per channel
float min
minimum allowed PE for bit factor calculation
AacPsyCoeffs psy_coef[2][64]
float global_quality
normalized global quality taken from avctx
float previous
allowed PE of the previous frame
int fill_level
bit reservoir fill level
float max
maximum allowed PE for bit factor calculation
struct AacPsyContext::@9 pe
int frame_bits
average bits per frame
single band psychoacoustic information
psychoacoustic information for an arbitrary group of channels
uint8_t num_ch
number of channels in this group
context used by psychoacoustic model
void * model_priv_data
psychoacoustic model implementation private data
codec-specific psychoacoustic model implementation
windowing related information
int num_windows
number of windows in a frame
int grouping[8]
window grouping (for e.g. AAC)
int window_shape
window shape (sine/KBD/whatever)
int window_type[3]
window type (short/long/transitional, etc.) - current, previous and next
LAME psy model preset struct.
float st_lrm
short threshold for L, R, and M channels
int quality
Quality to map the rest of the vaules to.
static const double coeff[2][5]