/* * AAC coefficients encoder * Copyright (C) 2008-2009 Konstantin Shishkov * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * AAC coefficients encoder */ /*********************************** * TODOs: * speedup quantizer selection * add sane pulse detection ***********************************/ #include "libavutil/libm.h" // brought forward to work around cygwin header breakage #include <float.h> #include "libavutil/mathematics.h"#include "mathops.h"#include "avcodec.h"#include "put_bits.h"#include "aac.h"#include "aacenc.h"#include "aactab.h"#include "aacenctab.h"#include "aacenc_utils.h"#include "aacenc_quantization.h" #include "aacenc_is.h"#include "aacenc_tns.h" #include "libavcodec/aaccoder_twoloop.h" /* Parameter of f(x) = a*(lambda/100), defines the maximum fourier spread * beyond which no PNS is used (since the SFBs contain tone rather than noise) */#define NOISE_SPREAD_THRESHOLD 0.9f /* Parameter of f(x) = a*(100/lambda), defines how much PNS is allowed to * replace low energy non zero bands */#define NOISE_LAMBDA_REPLACE 1.948f #include "libavcodec/aaccoder_trellis.h" typedef float (*quantize_and_encode_band_func)(struct AACEncContext *s, PutBitContext *pb, const float *in, float *quant, const float *scaled, int size, int scale_idx, int cb, const float lambda, const float uplim, int *bits, float *energy); /** * Calculate rate distortion cost for quantizing with given codebook * * @return quantization distortion */static av_always_inline float quantize_and_encode_band_cost_template( struct AACEncContext *s, PutBitContext *pb, const float *in, float *out, const float *scaled, int size, int scale_idx, int cb, const float lambda, const float uplim, int *bits, float *energy, int BT_ZERO, int BT_UNSIGNED, int BT_PAIR, int BT_ESC, int BT_NOISE, int BT_STEREO, const float ROUNDING){ const int q_idx = POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512; const float Q = ff_aac_pow2sf_tab [q_idx]; const float Q34 = ff_aac_pow34sf_tab[q_idx]; const float IQ = ff_aac_pow2sf_tab [POW_SF2_ZERO + scale_idx - SCALE_ONE_POS + SCALE_DIV_512]; const float CLIPPED_ESCAPE = 165140.0f*IQ; float cost = 0; float qenergy = 0; const int dim = BT_PAIR ? 2 : 4; int resbits = 0; int off; if (BT_ZERO || BT_NOISE || BT_STEREO) { for (int i = 0; i < size; i++) cost += in[i]*in[i]; if (bits) *bits = 0; if (energy) *energy = qenergy; if (out) { for (int i = 0; i < size; i += dim) for (int j = 0; j < dim; j++) out[i+j] = 0.0f; } return cost * lambda; } if (!scaled) { s->aacdsp.abs_pow34(s->scoefs, in, size); scaled = s->scoefs; } s->aacdsp.quant_bands(s->qcoefs, in, scaled, size, !BT_UNSIGNED, aac_cb_maxval[cb], Q34, ROUNDING); if (BT_UNSIGNED) { off = 0; } else { off = aac_cb_maxval[cb]; } for (int i = 0; i < size; i += dim) { const float *vec; int *quants = s->qcoefs + i; int curidx = 0; int curbits; float quantized, rd = 0.0f; for (int j = 0; j < dim; j++) { curidx *= aac_cb_range[cb]; curidx += quants[j] + off; } curbits = ff_aac_spectral_bits[cb-1][curidx]; vec = &ff_aac_codebook_vectors[cb-1][curidx*dim]; if (BT_UNSIGNED) { for (int j = 0; j < dim; j++) { float t = fabsf(in[i+j]); float di; if (BT_ESC && vec[j] == 64.0f) { //FIXME: slow if (t >= CLIPPED_ESCAPE) { quantized = CLIPPED_ESCAPE; curbits += 21; } else { int c = av_clip_uintp2(quant(t, Q, ROUNDING), 13); quantized = c*cbrtf(c)*IQ; curbits += av_log2(c)*2 - 4 + 1; } } else { quantized = vec[j]*IQ; } di = t - quantized; if (out) out[i+j] = in[i+j] >= 0 ? quantized : -quantized; if (vec[j] != 0.0f) curbits++; qenergy += quantized*quantized; rd += di*di; } } else { for (int j = 0; j < dim; j++) { quantized = vec[j]*IQ; qenergy += quantized*quantized; if (out) out[i+j] = quantized; rd += (in[i+j] - quantized)*(in[i+j] - quantized); } } cost += rd * lambda + curbits; resbits += curbits; if (cost >= uplim) return uplim; if (pb) { put_bits(pb, ff_aac_spectral_bits[cb-1][curidx], ff_aac_spectral_codes[cb-1][curidx]); if (BT_UNSIGNED) for (int j = 0; j < dim; j++) if (ff_aac_codebook_vectors[cb-1][curidx*dim+j] != 0.0f) put_bits(pb, 1, in[i+j] < 0.0f); if (BT_ESC) { for (int j = 0; j < 2; j++) { if (ff_aac_codebook_vectors[cb-1][curidx*2+j] == 64.0f) { int coef = av_clip_uintp2(quant(fabsf(in[i+j]), Q, ROUNDING), 13); int len = av_log2(coef); put_bits(pb, len - 4 + 1, (1 << (len - 4 + 1)) - 2); put_sbits(pb, len, coef); } } } } } if (bits) *bits = resbits; if (energy) *energy = qenergy; return cost;} static inline float quantize_and_encode_band_cost_NONE(struct AACEncContext *s, PutBitContext *pb, const float *in, float *quant, const float *scaled, int size, int scale_idx, int cb, const float lambda, const float uplim, int *bits, float *energy) { av_assert0(0); return 0.0f;} #define QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NAME, BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE, BT_STEREO, ROUNDING) \static float quantize_and_encode_band_cost_ ## NAME( \ struct AACEncContext *s, \ PutBitContext *pb, const float *in, float *quant, \ const float *scaled, int size, int scale_idx, \ int cb, const float lambda, const float uplim, \ int *bits, float *energy) { \ return quantize_and_encode_band_cost_template( \ s, pb, in, quant, scaled, size, scale_idx, \ BT_ESC ? ESC_BT : cb, lambda, uplim, bits, energy, \ BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE, BT_STEREO, \ ROUNDING); \} QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ZERO, 1, 0, 0, 0, 0, 0, ROUND_STANDARD)QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SQUAD, 0, 0, 0, 0, 0, 0, ROUND_STANDARD)QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UQUAD, 0, 1, 0, 0, 0, 0, ROUND_STANDARD)QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SPAIR, 0, 0, 1, 0, 0, 0, ROUND_STANDARD)QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UPAIR, 0, 1, 1, 0, 0, 0, ROUND_STANDARD)QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC, 0, 1, 1, 1, 0, 0, ROUND_STANDARD)QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC_RTZ, 0, 1, 1, 1, 0, 0, ROUND_TO_ZERO)QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NOISE, 0, 0, 0, 0, 1, 0, ROUND_STANDARD)QUANTIZE_AND_ENCODE_BAND_COST_FUNC(STEREO,0, 0, 0, 0, 0, 1, ROUND_STANDARD) static const quantize_and_encode_band_func quantize_and_encode_band_cost_arr[] ={ quantize_and_encode_band_cost_ZERO, quantize_and_encode_band_cost_SQUAD, quantize_and_encode_band_cost_SQUAD, quantize_and_encode_band_cost_UQUAD, quantize_and_encode_band_cost_UQUAD, quantize_and_encode_band_cost_SPAIR, quantize_and_encode_band_cost_SPAIR, quantize_and_encode_band_cost_UPAIR, quantize_and_encode_band_cost_UPAIR, quantize_and_encode_band_cost_UPAIR, quantize_and_encode_band_cost_UPAIR, quantize_and_encode_band_cost_ESC, quantize_and_encode_band_cost_NONE, /* CB 12 doesn't exist */ quantize_and_encode_band_cost_NOISE, quantize_and_encode_band_cost_STEREO, quantize_and_encode_band_cost_STEREO,}; static const quantize_and_encode_band_func quantize_and_encode_band_cost_rtz_arr[] ={ quantize_and_encode_band_cost_ZERO, quantize_and_encode_band_cost_SQUAD, quantize_and_encode_band_cost_SQUAD, quantize_and_encode_band_cost_UQUAD, quantize_and_encode_band_cost_UQUAD, quantize_and_encode_band_cost_SPAIR, quantize_and_encode_band_cost_SPAIR, quantize_and_encode_band_cost_UPAIR, quantize_and_encode_band_cost_UPAIR, quantize_and_encode_band_cost_UPAIR, quantize_and_encode_band_cost_UPAIR, quantize_and_encode_band_cost_ESC_RTZ, quantize_and_encode_band_cost_NONE, /* CB 12 doesn't exist */ quantize_and_encode_band_cost_NOISE, quantize_and_encode_band_cost_STEREO, quantize_and_encode_band_cost_STEREO,}; float ff_quantize_and_encode_band_cost(struct AACEncContext *s, PutBitContext *pb, const float *in, float *quant, const float *scaled, int size, int scale_idx, int cb, const float lambda, const float uplim, int *bits, float *energy){ return quantize_and_encode_band_cost_arr[cb](s, pb, in, quant, scaled, size, scale_idx, cb, lambda, uplim, bits, energy);} static inline void quantize_and_encode_band(struct AACEncContext *s, PutBitContext *pb, const float *in, float *out, int size, int scale_idx, int cb, const float lambda, int rtz){ (rtz ? quantize_and_encode_band_cost_rtz_arr : quantize_and_encode_band_cost_arr)[cb](s, pb, in, out, NULL, size, scale_idx, cb, lambda, INFINITY, NULL, NULL);} /** * structure used in optimal codebook search */typedef struct BandCodingPath { int prev_idx; ///< pointer to the previous path point float cost; ///< path cost int run;} BandCodingPath; typedef struct TrellisPath { float cost; int prev;} TrellisPath; #define TRELLIS_STAGES 121#define TRELLIS_STATES (SCALE_MAX_DIFF+1) static void set_special_band_scalefactors(AACEncContext *s, SingleChannelElement *sce){ int w, g; int prevscaler_n = -255, prevscaler_i = 0; int bands = 0; for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { for (g = 0; g < sce->ics.num_swb; g++) { if (sce->zeroes[w*16+g]) continue; if (sce->band_type[w*16+g] == INTENSITY_BT || sce->band_type[w*16+g] == INTENSITY_BT2) { sce->sf_idx[w*16+g] = av_clip(roundf(log2f(sce->is_ener[w*16+g])*2), -155, 100); bands++; } else if (sce->band_type[w*16+g] == NOISE_BT) { sce->sf_idx[w*16+g] = av_clip(3+ceilf(log2f(sce->pns_ener[w*16+g])*2), -100, 155); if (prevscaler_n == -255) prevscaler_n = sce->sf_idx[w*16+g]; bands++; } } } if (!bands) return; /* Clip the scalefactor indices */ for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { for (g = 0; g < sce->ics.num_swb; g++) { if (sce->zeroes[w*16+g]) continue; if (sce->band_type[w*16+g] == INTENSITY_BT || sce->band_type[w*16+g] == INTENSITY_BT2) { sce->sf_idx[w*16+g] = prevscaler_i = av_clip(sce->sf_idx[w*16+g], prevscaler_i - SCALE_MAX_DIFF, prevscaler_i + SCALE_MAX_DIFF); } else if (sce->band_type[w*16+g] == NOISE_BT) { sce->sf_idx[w*16+g] = prevscaler_n = av_clip(sce->sf_idx[w*16+g], prevscaler_n - SCALE_MAX_DIFF, prevscaler_n + SCALE_MAX_DIFF); } } }} static void search_for_quantizers_fast(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce, const float lambda){ int start = 0, i, w, w2, g; int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->ch_layout.nb_channels * (lambda / 120.f); float dists[128] = { 0 }, uplims[128] = { 0 }; float maxvals[128]; int fflag, minscaler; int its = 0; int allz = 0; float minthr = INFINITY; // for values above this the decoder might end up in an endless loop // due to always having more bits than what can be encoded. destbits = FFMIN(destbits, 5800); //some heuristic to determine initial quantizers will reduce search time //determine zero bands and upper limits for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { start = 0; for (g = 0; g < sce->ics.num_swb; g++) { int nz = 0; float uplim = 0.0f; for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) { FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g]; uplim += band->threshold; if (band->energy <= band->threshold || band->threshold == 0.0f) { sce->zeroes[(w+w2)*16+g] = 1; continue; } nz = 1; } uplims[w*16+g] = uplim *512; sce->band_type[w*16+g] = 0; sce->zeroes[w*16+g] = !nz; if (nz) minthr = FFMIN(minthr, uplim); allz |= nz; start += sce->ics.swb_sizes[g]; } } for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { for (g = 0; g < sce->ics.num_swb; g++) { if (sce->zeroes[w*16+g]) { sce->sf_idx[w*16+g] = SCALE_ONE_POS; continue; } sce->sf_idx[w*16+g] = SCALE_ONE_POS + FFMIN(log2f(uplims[w*16+g]/minthr)*4,59); } } if (!allz) return; s->aacdsp.abs_pow34(s->scoefs, sce->coeffs, 1024); ff_quantize_band_cost_cache_init(s); for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { start = w*128; for (g = 0; g < sce->ics.num_swb; g++) { const float *scaled = s->scoefs + start; maxvals[w*16+g] = find_max_val(sce->ics.group_len[w], sce->ics.swb_sizes[g], scaled); start += sce->ics.swb_sizes[g]; } } //perform two-loop search //outer loop - improve quality do { int tbits, qstep; minscaler = sce->sf_idx[0]; //inner loop - quantize spectrum to fit into given number of bits qstep = its ? 1 : 32; do { int prev = -1; tbits = 0; for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { start = w*128; for (g = 0; g < sce->ics.num_swb; g++) { const float *coefs = sce->coeffs + start; const float *scaled = s->scoefs + start; int bits = 0; int cb; float dist = 0.0f; if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) { start += sce->ics.swb_sizes[g]; continue; } minscaler = FFMIN(minscaler, sce->sf_idx[w*16+g]); cb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]); for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) { int b; dist += quantize_band_cost_cached(s, w + w2, g, coefs + w2*128, scaled + w2*128, sce->ics.swb_sizes[g], sce->sf_idx[w*16+g], cb, 1.0f, INFINITY, &b, NULL, 0); bits += b; } dists[w*16+g] = dist - bits; if (prev != -1) { bits += ff_aac_scalefactor_bits[sce->sf_idx[w*16+g] - prev + SCALE_DIFF_ZERO]; } tbits += bits; start += sce->ics.swb_sizes[g]; prev = sce->sf_idx[w*16+g]; } } if (tbits > destbits) { for (i = 0; i < 128; i++) if (sce->sf_idx[i] < 218 - qstep) sce->sf_idx[i] += qstep; } else { for (i = 0; i < 128; i++) if (sce->sf_idx[i] > 60 - qstep) sce->sf_idx[i] -= qstep; } qstep >>= 1; if (!qstep && tbits > destbits*1.02 && sce->sf_idx[0] < 217) qstep = 1; } while (qstep); fflag = 0; minscaler = av_clip(minscaler, 60, 255 - SCALE_MAX_DIFF); for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { for (g = 0; g < sce->ics.num_swb; g++) { int prevsc = sce->sf_idx[w*16+g]; if (dists[w*16+g] > uplims[w*16+g] && sce->sf_idx[w*16+g] > 60) { if (find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]-1)) sce->sf_idx[w*16+g]--; else //Try to make sure there is some energy in every band sce->sf_idx[w*16+g]-=2; } sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler, minscaler + SCALE_MAX_DIFF); sce->sf_idx[w*16+g] = FFMIN(sce->sf_idx[w*16+g], 219); if (sce->sf_idx[w*16+g] != prevsc) fflag = 1; sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]); } } its++; } while (fflag && its < 10);} static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce){ FFPsyBand *band; int w, g, w2, i; int wlen = 1024 / sce->ics.num_windows; int bandwidth, cutoff; float *PNS = &s->scoefs[0*128], *PNS34 = &s->scoefs[1*128]; float *NOR34 = &s->scoefs[3*128]; uint8_t nextband[128]; const float lambda = s->lambda; const float freq_mult = avctx->sample_rate*0.5f/wlen; const float thr_mult = NOISE_LAMBDA_REPLACE*(100.0f/lambda);Lines 1–500 of 870. 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/* * AAC coefficients encoder * Copyright (C) 2008-2009 Konstantin Shishkov * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file * AAC coefficients encoder */ /*********************************** * TODOs: * speedup quantizer selection * add sane pulse detection ***********************************/ #include "libavutil/libm.h" // brought forward to work around cygwin header breakage #include <float.h> #include "libavutil/mathematics.h" #include "mathops.h" #include "avcodec.h" #include "put_bits.h" #include "aac.h" #include "aacenc.h" #include "aactab.h" #include "aacenctab.h" #include "aacenc_utils.h" #include "aacenc_quantization.h" #include "aacenc_is.h" #include "aacenc_tns.h" #include "libavcodec/aaccoder_twoloop.h" /* Parameter of f(x) = a*(lambda/100), defines the maximum fourier spread * beyond which no PNS is used (since the SFBs contain tone rather than noise) */ #define NOISE_SPREAD_THRESHOLD 0.9f /* Parameter of f(x) = a*(100/lambda), defines how much PNS is allowed to * replace low energy non zero bands */ #define NOISE_LAMBDA_REPLACE 1.948f #include "libavcodec/aaccoder_trellis.h" typedef float (*quantize_and_encode_band_func)(struct AACEncContext *s, PutBitContext *pb, const float *in, float *quant, const float *scaled, int size, int scale_idx, int cb, const float lambda, const float uplim, int *bits, float *energy); /** * Calculate rate distortion cost for quantizing with given codebook * * @return quantization distortion */ static av_always_inline float quantize_and_encode_band_cost_template( struct AACEncContext *s, PutBitContext *pb, const float *in, float *out, const float *scaled, int size, int scale_idx, int cb, const float lambda, const float uplim, int *bits, float *energy, int BT_ZERO, int BT_UNSIGNED, int BT_PAIR, int BT_ESC, int BT_NOISE, int BT_STEREO, const float ROUNDING) { const int q_idx = POW_SF2_ZERO - scale_idx + SCALE_ONE_POS - SCALE_DIV_512; const float Q = ff_aac_pow2sf_tab [q_idx]; const float Q34 = ff_aac_pow34sf_tab[q_idx]; const float IQ = ff_aac_pow2sf_tab [POW_SF2_ZERO + scale_idx - SCALE_ONE_POS + SCALE_DIV_512]; const float CLIPPED_ESCAPE = 165140.0f*IQ; float cost = 0; float qenergy = 0; const int dim = BT_PAIR ? 2 : 4; int resbits = 0; int off; if (BT_ZERO || BT_NOISE || BT_STEREO) { for (int i = 0; i < size; i++) cost += in[i]*in[i]; if (bits) *bits = 0; if (energy) *energy = qenergy; if (out) { for (int i = 0; i < size; i += dim) for (int j = 0; j < dim; j++) out[i+j] = 0.0f; } return cost * lambda; } if (!scaled) { s->aacdsp.abs_pow34(s->scoefs, in, size); scaled = s->scoefs; } s->aacdsp.quant_bands(s->qcoefs, in, scaled, size, !BT_UNSIGNED, aac_cb_maxval[cb], Q34, ROUNDING); if (BT_UNSIGNED) { off = 0; } else { off = aac_cb_maxval[cb]; } for (int i = 0; i < size; i += dim) { const float *vec; int *quants = s->qcoefs + i; int curidx = 0; int curbits; float quantized, rd = 0.0f; for (int j = 0; j < dim; j++) { curidx *= aac_cb_range[cb]; curidx += quants[j] + off; } curbits = ff_aac_spectral_bits[cb-1][curidx]; vec = &ff_aac_codebook_vectors[cb-1][curidx*dim]; if (BT_UNSIGNED) { for (int j = 0; j < dim; j++) { float t = fabsf(in[i+j]); float di; if (BT_ESC && vec[j] == 64.0f) { //FIXME: slow if (t >= CLIPPED_ESCAPE) { quantized = CLIPPED_ESCAPE; curbits += 21; } else { int c = av_clip_uintp2(quant(t, Q, ROUNDING), 13); quantized = c*cbrtf(c)*IQ; curbits += av_log2(c)*2 - 4 + 1; } } else { quantized = vec[j]*IQ; } di = t - quantized; if (out) out[i+j] = in[i+j] >= 0 ? quantized : -quantized; if (vec[j] != 0.0f) curbits++; qenergy += quantized*quantized; rd += di*di; } } else { for (int j = 0; j < dim; j++) { quantized = vec[j]*IQ; qenergy += quantized*quantized; if (out) out[i+j] = quantized; rd += (in[i+j] - quantized)*(in[i+j] - quantized); } } cost += rd * lambda + curbits; resbits += curbits; if (cost >= uplim) return uplim; if (pb) { put_bits(pb, ff_aac_spectral_bits[cb-1][curidx], ff_aac_spectral_codes[cb-1][curidx]); if (BT_UNSIGNED) for (int j = 0; j < dim; j++) if (ff_aac_codebook_vectors[cb-1][curidx*dim+j] != 0.0f) put_bits(pb, 1, in[i+j] < 0.0f); if (BT_ESC) { for (int j = 0; j < 2; j++) { if (ff_aac_codebook_vectors[cb-1][curidx*2+j] == 64.0f) { int coef = av_clip_uintp2(quant(fabsf(in[i+j]), Q, ROUNDING), 13); int len = av_log2(coef); put_bits(pb, len - 4 + 1, (1 << (len - 4 + 1)) - 2); put_sbits(pb, len, coef); } } } } } if (bits) *bits = resbits; if (energy) *energy = qenergy; return cost; } static inline float quantize_and_encode_band_cost_NONE(struct AACEncContext *s, PutBitContext *pb, const float *in, float *quant, const float *scaled, int size, int scale_idx, int cb, const float lambda, const float uplim, int *bits, float *energy) { av_assert0(0); return 0.0f; } #define QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NAME, BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE, BT_STEREO, ROUNDING) \ static float quantize_and_encode_band_cost_ ## NAME( \ struct AACEncContext *s, \ PutBitContext *pb, const float *in, float *quant, \ const float *scaled, int size, int scale_idx, \ int cb, const float lambda, const float uplim, \ int *bits, float *energy) { \ return quantize_and_encode_band_cost_template( \ s, pb, in, quant, scaled, size, scale_idx, \ BT_ESC ? ESC_BT : cb, lambda, uplim, bits, energy, \ BT_ZERO, BT_UNSIGNED, BT_PAIR, BT_ESC, BT_NOISE, BT_STEREO, \ ROUNDING); \ } QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ZERO, 1, 0, 0, 0, 0, 0, ROUND_STANDARD) QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SQUAD, 0, 0, 0, 0, 0, 0, ROUND_STANDARD) QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UQUAD, 0, 1, 0, 0, 0, 0, ROUND_STANDARD) QUANTIZE_AND_ENCODE_BAND_COST_FUNC(SPAIR, 0, 0, 1, 0, 0, 0, ROUND_STANDARD) QUANTIZE_AND_ENCODE_BAND_COST_FUNC(UPAIR, 0, 1, 1, 0, 0, 0, ROUND_STANDARD) QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC, 0, 1, 1, 1, 0, 0, ROUND_STANDARD) QUANTIZE_AND_ENCODE_BAND_COST_FUNC(ESC_RTZ, 0, 1, 1, 1, 0, 0, ROUND_TO_ZERO) QUANTIZE_AND_ENCODE_BAND_COST_FUNC(NOISE, 0, 0, 0, 0, 1, 0, ROUND_STANDARD) QUANTIZE_AND_ENCODE_BAND_COST_FUNC(STEREO,0, 0, 0, 0, 0, 1, ROUND_STANDARD) static const quantize_and_encode_band_func quantize_and_encode_band_cost_arr[] = { quantize_and_encode_band_cost_ZERO, quantize_and_encode_band_cost_SQUAD, quantize_and_encode_band_cost_SQUAD, quantize_and_encode_band_cost_UQUAD, quantize_and_encode_band_cost_UQUAD, quantize_and_encode_band_cost_SPAIR, quantize_and_encode_band_cost_SPAIR, quantize_and_encode_band_cost_UPAIR, quantize_and_encode_band_cost_UPAIR, quantize_and_encode_band_cost_UPAIR, quantize_and_encode_band_cost_UPAIR, quantize_and_encode_band_cost_ESC, quantize_and_encode_band_cost_NONE, /* CB 12 doesn't exist */ quantize_and_encode_band_cost_NOISE, quantize_and_encode_band_cost_STEREO, quantize_and_encode_band_cost_STEREO, }; static const quantize_and_encode_band_func quantize_and_encode_band_cost_rtz_arr[] = { quantize_and_encode_band_cost_ZERO, quantize_and_encode_band_cost_SQUAD, quantize_and_encode_band_cost_SQUAD, quantize_and_encode_band_cost_UQUAD, quantize_and_encode_band_cost_UQUAD, quantize_and_encode_band_cost_SPAIR, quantize_and_encode_band_cost_SPAIR, quantize_and_encode_band_cost_UPAIR, quantize_and_encode_band_cost_UPAIR, quantize_and_encode_band_cost_UPAIR, quantize_and_encode_band_cost_UPAIR, quantize_and_encode_band_cost_ESC_RTZ, quantize_and_encode_band_cost_NONE, /* CB 12 doesn't exist */ quantize_and_encode_band_cost_NOISE, quantize_and_encode_band_cost_STEREO, quantize_and_encode_band_cost_STEREO, }; float ff_quantize_and_encode_band_cost(struct AACEncContext *s, PutBitContext *pb, const float *in, float *quant, const float *scaled, int size, int scale_idx, int cb, const float lambda, const float uplim, int *bits, float *energy) { return quantize_and_encode_band_cost_arr[cb](s, pb, in, quant, scaled, size, scale_idx, cb, lambda, uplim, bits, energy); } static inline void quantize_and_encode_band(struct AACEncContext *s, PutBitContext *pb, const float *in, float *out, int size, int scale_idx, int cb, const float lambda, int rtz) { (rtz ? quantize_and_encode_band_cost_rtz_arr : quantize_and_encode_band_cost_arr)[cb](s, pb, in, out, NULL, size, scale_idx, cb, lambda, INFINITY, NULL, NULL); } /** * structure used in optimal codebook search */ typedef struct BandCodingPath { int prev_idx; ///< pointer to the previous path point float cost; ///< path cost int run; } BandCodingPath; typedef struct TrellisPath { float cost; int prev; } TrellisPath; #define TRELLIS_STAGES 121 #define TRELLIS_STATES (SCALE_MAX_DIFF+1) static void set_special_band_scalefactors(AACEncContext *s, SingleChannelElement *sce) { int w, g; int prevscaler_n = -255, prevscaler_i = 0; int bands = 0; for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { for (g = 0; g < sce->ics.num_swb; g++) { if (sce->zeroes[w*16+g]) continue; if (sce->band_type[w*16+g] == INTENSITY_BT || sce->band_type[w*16+g] == INTENSITY_BT2) { sce->sf_idx[w*16+g] = av_clip(roundf(log2f(sce->is_ener[w*16+g])*2), -155, 100); bands++; } else if (sce->band_type[w*16+g] == NOISE_BT) { sce->sf_idx[w*16+g] = av_clip(3+ceilf(log2f(sce->pns_ener[w*16+g])*2), -100, 155); if (prevscaler_n == -255) prevscaler_n = sce->sf_idx[w*16+g]; bands++; } } } if (!bands) return; /* Clip the scalefactor indices */ for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { for (g = 0; g < sce->ics.num_swb; g++) { if (sce->zeroes[w*16+g]) continue; if (sce->band_type[w*16+g] == INTENSITY_BT || sce->band_type[w*16+g] == INTENSITY_BT2) { sce->sf_idx[w*16+g] = prevscaler_i = av_clip(sce->sf_idx[w*16+g], prevscaler_i - SCALE_MAX_DIFF, prevscaler_i + SCALE_MAX_DIFF); } else if (sce->band_type[w*16+g] == NOISE_BT) { sce->sf_idx[w*16+g] = prevscaler_n = av_clip(sce->sf_idx[w*16+g], prevscaler_n - SCALE_MAX_DIFF, prevscaler_n + SCALE_MAX_DIFF); } } } } static void search_for_quantizers_fast(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce, const float lambda) { int start = 0, i, w, w2, g; int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / avctx->ch_layout.nb_channels * (lambda / 120.f); float dists[128] = { 0 }, uplims[128] = { 0 }; float maxvals[128]; int fflag, minscaler; int its = 0; int allz = 0; float minthr = INFINITY; // for values above this the decoder might end up in an endless loop // due to always having more bits than what can be encoded. destbits = FFMIN(destbits, 5800); //some heuristic to determine initial quantizers will reduce search time //determine zero bands and upper limits for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { start = 0; for (g = 0; g < sce->ics.num_swb; g++) { int nz = 0; float uplim = 0.0f; for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) { FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g]; uplim += band->threshold; if (band->energy <= band->threshold || band->threshold == 0.0f) { sce->zeroes[(w+w2)*16+g] = 1; continue; } nz = 1; } uplims[w*16+g] = uplim *512; sce->band_type[w*16+g] = 0; sce->zeroes[w*16+g] = !nz; if (nz) minthr = FFMIN(minthr, uplim); allz |= nz; start += sce->ics.swb_sizes[g]; } } for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { for (g = 0; g < sce->ics.num_swb; g++) { if (sce->zeroes[w*16+g]) { sce->sf_idx[w*16+g] = SCALE_ONE_POS; continue; } sce->sf_idx[w*16+g] = SCALE_ONE_POS + FFMIN(log2f(uplims[w*16+g]/minthr)*4,59); } } if (!allz) return; s->aacdsp.abs_pow34(s->scoefs, sce->coeffs, 1024); ff_quantize_band_cost_cache_init(s); for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { start = w*128; for (g = 0; g < sce->ics.num_swb; g++) { const float *scaled = s->scoefs + start; maxvals[w*16+g] = find_max_val(sce->ics.group_len[w], sce->ics.swb_sizes[g], scaled); start += sce->ics.swb_sizes[g]; } } //perform two-loop search //outer loop - improve quality do { int tbits, qstep; minscaler = sce->sf_idx[0]; //inner loop - quantize spectrum to fit into given number of bits qstep = its ? 1 : 32; do { int prev = -1; tbits = 0; for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { start = w*128; for (g = 0; g < sce->ics.num_swb; g++) { const float *coefs = sce->coeffs + start; const float *scaled = s->scoefs + start; int bits = 0; int cb; float dist = 0.0f; if (sce->zeroes[w*16+g] || sce->sf_idx[w*16+g] >= 218) { start += sce->ics.swb_sizes[g]; continue; } minscaler = FFMIN(minscaler, sce->sf_idx[w*16+g]); cb = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]); for (w2 = 0; w2 < sce->ics.group_len[w]; w2++) { int b; dist += quantize_band_cost_cached(s, w + w2, g, coefs + w2*128, scaled + w2*128, sce->ics.swb_sizes[g], sce->sf_idx[w*16+g], cb, 1.0f, INFINITY, &b, NULL, 0); bits += b; } dists[w*16+g] = dist - bits; if (prev != -1) { bits += ff_aac_scalefactor_bits[sce->sf_idx[w*16+g] - prev + SCALE_DIFF_ZERO]; } tbits += bits; start += sce->ics.swb_sizes[g]; prev = sce->sf_idx[w*16+g]; } } if (tbits > destbits) { for (i = 0; i < 128; i++) if (sce->sf_idx[i] < 218 - qstep) sce->sf_idx[i] += qstep; } else { for (i = 0; i < 128; i++) if (sce->sf_idx[i] > 60 - qstep) sce->sf_idx[i] -= qstep; } qstep >>= 1; if (!qstep && tbits > destbits*1.02 && sce->sf_idx[0] < 217) qstep = 1; } while (qstep); fflag = 0; minscaler = av_clip(minscaler, 60, 255 - SCALE_MAX_DIFF); for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { for (g = 0; g < sce->ics.num_swb; g++) { int prevsc = sce->sf_idx[w*16+g]; if (dists[w*16+g] > uplims[w*16+g] && sce->sf_idx[w*16+g] > 60) { if (find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]-1)) sce->sf_idx[w*16+g]--; else //Try to make sure there is some energy in every band sce->sf_idx[w*16+g]-=2; } sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], minscaler, minscaler + SCALE_MAX_DIFF); sce->sf_idx[w*16+g] = FFMIN(sce->sf_idx[w*16+g], 219); if (sce->sf_idx[w*16+g] != prevsc) fflag = 1; sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]); } } its++; } while (fflag && its < 10); } static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce) { FFPsyBand *band; int w, g, w2, i; int wlen = 1024 / sce->ics.num_windows; int bandwidth, cutoff; float *PNS = &s->scoefs[0*128], *PNS34 = &s->scoefs[1*128]; float *NOR34 = &s->scoefs[3*128]; uint8_t nextband[128]; const float lambda = s->lambda; const float freq_mult = avctx->sample_rate*0.5f/wlen; const float thr_mult = NOISE_LAMBDA_REPLACE*(100.0f/lambda);