/* * AAC encoder psychoacoustic model * Copyright (C) 2008 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 encoder psychoacoustic model */ #include "libavutil/attributes.h" #include "libavutil/ffmath.h" #include "avcodec.h" #include "aactab.h" #include "psymodel.h" /*********************************** * TODOs: * try other bitrate controlling mechanism (maybe use ratecontrol.c?) * control quality for quality-based output **********************************/ /** * constants for 3GPP AAC psychoacoustic model * @{ */ #define PSY_3GPP_THR_SPREAD_HI 1.5f // spreading factor for low-to-hi threshold spreading (15 dB/Bark) #define PSY_3GPP_THR_SPREAD_LOW 3.0f // spreading factor for hi-to-low threshold spreading (30 dB/Bark) /* spreading factor for low-to-hi energy spreading, long block, > 22kbps/channel (20dB/Bark) */ #define PSY_3GPP_EN_SPREAD_HI_L1 2.0f /* spreading factor for low-to-hi energy spreading, long block, <= 22kbps/channel (15dB/Bark) */ #define PSY_3GPP_EN_SPREAD_HI_L2 1.5f /* spreading factor for low-to-hi energy spreading, short block (15 dB/Bark) */ #define PSY_3GPP_EN_SPREAD_HI_S 1.5f /* spreading factor for hi-to-low energy spreading, long block (30dB/Bark) */ #define PSY_3GPP_EN_SPREAD_LOW_L 3.0f /* spreading factor for hi-to-low energy spreading, short block (20dB/Bark) */ #define PSY_3GPP_EN_SPREAD_LOW_S 2.0f #define PSY_3GPP_RPEMIN 0.01f #define PSY_3GPP_RPELEV 2.0f #define PSY_3GPP_C1 3.0f /* log2(8) */ #define PSY_3GPP_C2 1.3219281f /* log2(2.5) */ #define PSY_3GPP_C3 0.55935729f /* 1 - C2 / C1 */ #define PSY_SNR_1DB 7.9432821e-1f /* -1dB */ #define PSY_SNR_25DB 3.1622776e-3f /* -25dB */ #define PSY_3GPP_SAVE_SLOPE_L -0.46666667f #define PSY_3GPP_SAVE_SLOPE_S -0.36363637f #define PSY_3GPP_SAVE_ADD_L -0.84285712f #define PSY_3GPP_SAVE_ADD_S -0.75f #define PSY_3GPP_SPEND_SLOPE_L 0.66666669f #define PSY_3GPP_SPEND_SLOPE_S 0.81818181f #define PSY_3GPP_SPEND_ADD_L -0.35f #define PSY_3GPP_SPEND_ADD_S -0.26111111f #define PSY_3GPP_CLIP_LO_L 0.2f #define PSY_3GPP_CLIP_LO_S 0.2f #define PSY_3GPP_CLIP_HI_L 0.95f #define PSY_3GPP_CLIP_HI_S 0.75f #define PSY_3GPP_AH_THR_LONG 0.5f #define PSY_3GPP_AH_THR_SHORT 0.63f #define PSY_PE_FORGET_SLOPE 511 enum { PSY_3GPP_AH_NONE, PSY_3GPP_AH_INACTIVE, PSY_3GPP_AH_ACTIVE }; #define PSY_3GPP_BITS_TO_PE(bits) ((bits) * 1.18f) #define PSY_3GPP_PE_TO_BITS(bits) ((bits) / 1.18f) /* LAME psy model constants */ #define PSY_LAME_FIR_LEN 21 ///< LAME psy model FIR order #define AAC_BLOCK_SIZE_LONG 1024 ///< long block size #define AAC_BLOCK_SIZE_SHORT 128 ///< short block size #define AAC_NUM_BLOCKS_SHORT 8 ///< number of blocks in a short sequence #define PSY_LAME_NUM_SUBBLOCKS 3 ///< Number of sub-blocks in each short block /** * @} */ /** * information for single band used by 3GPP TS26.403-inspired psychoacoustic model */ typedef struct AacPsyBand{ float energy; ///< band energy float thr; ///< energy threshold float thr_quiet; ///< threshold in quiet float nz_lines; ///< number of non-zero spectral lines float active_lines; ///< number of active spectral lines float pe; ///< perceptual entropy float pe_const; ///< constant part of the PE calculation float norm_fac; ///< normalization factor for linearization int avoid_holes; ///< hole avoidance flag }AacPsyBand; /** * single/pair channel context for psychoacoustic model */ typedef struct AacPsyChannel{ AacPsyBand band[128]; ///< bands information AacPsyBand prev_band[128]; ///< bands information from the previous frame float win_energy; ///< sliding average of channel energy float iir_state[2]; ///< hi-pass IIR filter state uint8_t next_grouping; ///< stored grouping scheme for the next frame (in case of 8 short window sequence) enum WindowSequence next_window_seq; ///< window sequence to be used in the next frame /* LAME psy model specific members */ float attack_threshold; ///< attack threshold for this channel float prev_energy_subshort[AAC_NUM_BLOCKS_SHORT * PSY_LAME_NUM_SUBBLOCKS]; int prev_attack; ///< attack value for the last short block in the previous sequence }AacPsyChannel; /** * psychoacoustic model frame type-dependent coefficients */ typedef struct AacPsyCoeffs{ float ath; ///< absolute threshold of hearing per bands float barks; ///< Bark value for each spectral band in long frame float spread_low[2]; ///< spreading factor for low-to-high threshold spreading in long frame float spread_hi [2]; ///< spreading factor for high-to-low threshold spreading in long frame float min_snr; ///< minimal SNR }AacPsyCoeffs; /** * 3GPP TS26.403-inspired psychoacoustic model specific data */ typedef struct AacPsyContext{ int chan_bitrate; ///< bitrate per channel int frame_bits; ///< average bits per frame int fill_level; ///< bit reservoir fill level struct { float min; ///< minimum allowed PE for bit factor calculation float max; ///< maximum allowed PE for bit factor calculation float previous; ///< allowed PE of the previous frame float correction; ///< PE correction factor } pe; AacPsyCoeffs psy_coef[2][64]; AacPsyChannel *ch; float global_quality; ///< normalized global quality taken from avctx }AacPsyContext; /** * LAME psy model preset struct */ typedef struct PsyLamePreset { int quality; ///< Quality to map the rest of the vaules to. /* This is overloaded to be both kbps per channel in ABR mode, and * requested quality in constant quality mode. */ float st_lrm; ///< short threshold for L, R, and M channels } PsyLamePreset; /** * LAME psy model preset table for ABR */ static const PsyLamePreset psy_abr_map[] = { /* TODO: Tuning. These were taken from LAME. */ /* kbps/ch st_lrm */ { 8, 6.60}, { 16, 6.60}, { 24, 6.60}, { 32, 6.60}, { 40, 6.60}, { 48, 6.60}, { 56, 6.60}, { 64, 6.40}, { 80, 6.00}, { 96, 5.60}, {112, 5.20}, {128, 5.20}, {160, 5.20} }; /** * LAME psy model preset table for constant quality */ static const PsyLamePreset psy_vbr_map[] = { /* vbr_q st_lrm */ { 0, 4.20}, { 1, 4.20}, { 2, 4.20}, { 3, 4.20}, { 4, 4.20}, { 5, 4.20}, { 6, 4.20}, { 7, 4.20}, { 8, 4.20}, { 9, 4.20}, {10, 4.20} }; /** * LAME psy model FIR coefficient table */ static const float psy_fir_coeffs[] = { -8.65163e-18 * 2, -0.00851586 * 2, -6.74764e-18 * 2, 0.0209036 * 2, -3.36639e-17 * 2, -0.0438162 * 2, -1.54175e-17 * 2, 0.0931738 * 2, -5.52212e-17 * 2, -0.313819 * 2 }; #if ARCH_MIPS # include "mips/aacpsy_mips.h" #endif /* ARCH_MIPS */ /** * Calculate the ABR attack threshold from the above LAME psymodel table. */ static float lame_calc_attack_threshold(int bitrate) { /* Assume max bitrate to start with */ int lower_range = 12, upper_range = 12; int lower_range_kbps = psy_abr_map[12].quality; int upper_range_kbps = psy_abr_map[12].quality; int i; /* Determine which bitrates the value specified falls between. * If the loop ends without breaking our above assumption of 320kbps was correct. */ for (i = 1; i < 13; i++) { if (FFMAX(bitrate, psy_abr_map[i].quality) != bitrate) { upper_range = i; upper_range_kbps = psy_abr_map[i ].quality; lower_range = i - 1; lower_range_kbps = psy_abr_map[i - 1].quality; break; /* Upper range found */ } } /* Determine which range the value specified is closer to */ if ((upper_range_kbps - bitrate) > (bitrate - lower_range_kbps)) return psy_abr_map[lower_range].st_lrm; return psy_abr_map[upper_range].st_lrm; } /** * LAME psy model specific initialization */ static av_cold void lame_window_init(AacPsyContext *ctx, AVCodecContext *avctx) { int i, j; for (i = 0; i < avctx->channels; i++) { AacPsyChannel *pch = &ctx->ch[i]; if (avctx->flags & AV_CODEC_FLAG_QSCALE) pch->attack_threshold = psy_vbr_map[avctx->global_quality / FF_QP2LAMBDA].st_lrm; else pch->attack_threshold = lame_calc_attack_threshold(avctx->bit_rate / avctx->channels / 1000); for (j = 0; j < AAC_NUM_BLOCKS_SHORT * PSY_LAME_NUM_SUBBLOCKS; j++) pch->prev_energy_subshort[j] = 10.0f; } } /** * Calculate Bark value for given line. */ static av_cold float calc_bark(float f) { return 13.3f * atanf(0.00076f * f) + 3.5f * atanf((f / 7500.0f) * (f / 7500.0f)); } #define ATH_ADD 4 /** * Calculate ATH value for given frequency. * Borrowed from Lame. */ static av_cold float ath(float f, float add) { f /= 1000.0f; return 3.64 * pow(f, -0.8) - 6.8 * exp(-0.6 * (f - 3.4) * (f - 3.4)) + 6.0 * exp(-0.15 * (f - 8.7) * (f - 8.7)) + (0.6 + 0.04 * add) * 0.001 * f * f * f * f; } static av_cold int psy_3gpp_init(FFPsyContext *ctx) { AacPsyContext *pctx; float bark; int i, j, g, start; float prev, minscale, minath, minsnr, pe_min; int chan_bitrate = ctx->avctx->bit_rate / ((ctx->avctx->flags & AV_CODEC_FLAG_QSCALE) ? 2.0f : ctx->avctx->channels); const int bandwidth = ctx->cutoff ? ctx->cutoff : AAC_CUTOFF(ctx->avctx); const float num_bark = calc_bark((float)bandwidth); if (bandwidth <= 0) return AVERROR(EINVAL); ctx->model_priv_data = av_mallocz(sizeof(AacPsyContext)); if (!ctx->model_priv_data) return AVERROR(ENOMEM); pctx = ctx->model_priv_data; pctx->global_quality = (ctx->avctx->global_quality ? ctx->avctx->global_quality : 120) * 0.01f; if (ctx->avctx->flags & AV_CODEC_FLAG_QSCALE) { /* Use the target average bitrate to compute spread parameters */ chan_bitrate = (int)(chan_bitrate / 120.0 * (ctx->avctx->global_quality ? ctx->avctx->global_quality : 120)); } pctx->chan_bitrate = chan_bitrate; pctx->frame_bits = FFMIN(2560, chan_bitrate * AAC_BLOCK_SIZE_LONG / ctx->avctx->sample_rate); pctx->pe.min = 8.0f * AAC_BLOCK_SIZE_LONG * bandwidth / (ctx->avctx->sample_rate * 2.0f); pctx->pe.max = 12.0f * AAC_BLOCK_SIZE_LONG * bandwidth / (ctx->avctx->sample_rate * 2.0f); ctx->bitres.size = 6144 - pctx->frame_bits; ctx->bitres.size -= ctx->bitres.size % 8; pctx->fill_level = ctx->bitres.size; minath = ath(3410 - 0.733 * ATH_ADD, ATH_ADD); for (j = 0; j < 2; j++) { AacPsyCoeffs *coeffs = pctx->psy_coef[j]; const uint8_t *band_sizes = ctx->bands[j]; float line_to_frequency = ctx->avctx->sample_rate / (j ? 256.f : 2048.0f); float avg_chan_bits = chan_bitrate * (j ? 128.0f : 1024.0f) / ctx->avctx->sample_rate; /* reference encoder uses 2.4% here instead of 60% like the spec says */ float bark_pe = 0.024f * PSY_3GPP_BITS_TO_PE(avg_chan_bits) / num_bark; float en_spread_low = j ? PSY_3GPP_EN_SPREAD_LOW_S : PSY_3GPP_EN_SPREAD_LOW_L; /* High energy spreading for long blocks <= 22kbps/channel and short blocks are the same. */ float en_spread_hi = (j || (chan_bitrate <= 22.0f)) ? PSY_3GPP_EN_SPREAD_HI_S : PSY_3GPP_EN_SPREAD_HI_L1; i = 0; prev = 0.0; for (g = 0; g < ctx->num_bands[j]; g++) { i += band_sizes[g]; bark = calc_bark((i-1) * line_to_frequency); coeffs[g].barks = (bark + prev) / 2.0; prev = bark; } for (g = 0; g < ctx->num_bands[j] - 1; g++) { AacPsyCoeffs *coeff = &coeffs[g]; float bark_width = coeffs[g+1].barks - coeffs->barks; coeff->spread_low[0] = ff_exp10(-bark_width * PSY_3GPP_THR_SPREAD_LOW); coeff->spread_hi [0] = ff_exp10(-bark_width * PSY_3GPP_THR_SPREAD_HI); coeff->spread_low[1] = ff_exp10(-bark_width * en_spread_low); coeff->spread_hi [1] = ff_exp10(-bark_width * en_spread_hi); pe_min = bark_pe * bark_width; minsnr = exp2(pe_min / band_sizes[g]) - 1.5f; coeff->min_snr = av_clipf(1.0f / minsnr, PSY_SNR_25DB, PSY_SNR_1DB); } start = 0; for (g = 0; g < ctx->num_bands[j]; g++) { minscale = ath(start * line_to_frequency, ATH_ADD); for (i = 1; i < band_sizes[g]; i++) minscale = FFMIN(minscale, ath((start + i) * line_to_frequency, ATH_ADD)); coeffs[g].ath = minscale - minath; start += band_sizes[g]; } } pctx->ch = av_calloc(ctx->avctx->channels, sizeof(*pctx->ch)); if (!pctx->ch) { av_freep(&ctx->model_priv_data); return AVERROR(ENOMEM); } lame_window_init(pctx, ctx->avctx); return 0; } /** * IIR filter used in block switching decision */ static float iir_filter(int in, float state[2]) { float ret; ret = 0.7548f * (in - state[0]) + 0.5095f * state[1]; state[0] = in; state[1] = ret; return ret; } /** * window grouping information stored as bits (0 - new group, 1 - group continues) */ static const uint8_t window_grouping[9] = { 0xB6, 0x6C, 0xD8, 0xB2, 0x66, 0xC6, 0x96, 0x36, 0x36 }; /** * Tell encoder which window types to use. * @see 3GPP TS26.403 5.4.1 "Blockswitching" */ static av_unused FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx, const int16_t *audio, const int16_t *la, int channel, int prev_type) { int i, j; int br = ((AacPsyContext*)ctx->model_priv_data)->chan_bitrate; int attack_ratio = br <= 16000 ? 18 : 10; AacPsyContext *pctx = (AacPsyContext*) ctx->model_priv_data; AacPsyChannel *pch = &pctx->ch[channel]; uint8_t grouping = 0; int next_type = pch->next_window_seq; FFPsyWindowInfo wi = { { 0 } }; if (la) { float s[8], v; int switch_to_eight = 0; float sum = 0.0, sum2 = 0.0; int attack_n = 0; int stay_short = 0; for (i = 0; i < 8; i++) { for (j = 0; j < 128; j++) { v = iir_filter(la[i*128+j], pch->iir_state); sum += v*v; } s[i] = sum; sum2 += sum; } for (i = 0; i < 8; i++) { if (s[i] > pch->win_energy * attack_ratio) { attack_n = i + 1; switch_to_eight = 1; break; } } pch->win_energy = pch->win_energy*7/8 + sum2/64; wi.window_type[1] = prev_type; switch (prev_type) { case ONLY_LONG_SEQUENCE: wi.window_type[0] = switch_to_eight ? LONG_START_SEQUENCE : ONLY_LONG_SEQUENCE; next_type = switch_to_eight ? EIGHT_SHORT_SEQUENCE : ONLY_LONG_SEQUENCE; break; case LONG_START_SEQUENCE: wi.window_type[0] = EIGHT_SHORT_SEQUENCE; grouping = pch->next_grouping; next_type = switch_to_eight ? EIGHT_SHORT_SEQUENCE : LONG_STOP_SEQUENCE; break; case LONG_STOP_SEQUENCE: wi.window_type[0] = switch_to_eight ? LONG_START_SEQUENCE : ONLY_LONG_SEQUENCE; next_type = switch_to_eight ? EIGHT_SHORT_SEQUENCE : ONLY_LONG_SEQUENCE; break; case EIGHT_SHORT_SEQUENCE: stay_short = next_type == EIGHT_SHORT_SEQUENCE || switch_to_eight; wi.window_type[0] = stay_short ? EIGHT_SHORT_SEQUENCE : LONG_STOP_SEQUENCE; grouping = next_type == EIGHT_SHORT_SEQUENCE ? pch->next_grouping : 0; next_type = switch_to_eight ? EIGHT_SHORT_SEQUENCE : LONG_STOP_SEQUENCE; break; } pch->next_grouping = window_grouping[attack_n]; pch->next_window_seq = next_type; } else { for (i = 0; i < 3; i++) wi.window_type[i] = prev_type; grouping = (prev_type == EIGHT_SHORT_SEQUENCE) ? window_grouping[0] : 0; } wi.window_shape = 1; if (wi.window_type[0] != EIGHT_SHORT_SEQUENCE) { wi.num_windows = 1; wi.grouping[0] = 1; } else { int lastgrp = 0; wi.num_windows = 8; for (i = 0; i < 8; i++) { if (!((grouping >> i) & 1)) lastgrp = i; wi.grouping[lastgrp]++; } } return wi; } /* 5.6.1.2 "Calculation of Bit Demand" */ static int calc_bit_demand(AacPsyContext *ctx, float pe, int bits, int size, int short_window) { const float bitsave_slope = short_window ? PSY_3GPP_SAVE_SLOPE_S : PSY_3GPP_SAVE_SLOPE_L; const float bitsave_add = short_window ? PSY_3GPP_SAVE_ADD_S : PSY_3GPP_SAVE_ADD_L; const float bitspend_slope = short_window ? PSY_3GPP_SPEND_SLOPE_S : PSY_3GPP_SPEND_SLOPE_L; const float bitspend_add = short_window ? PSY_3GPP_SPEND_ADD_S : PSY_3GPP_SPEND_ADD_L; const float clip_low = short_window ? PSY_3GPP_CLIP_LO_S : PSY_3GPP_CLIP_LO_L; const float clip_high = short_window ? PSY_3GPP_CLIP_HI_S : PSY_3GPP_CLIP_HI_L; float clipped_pe, bit_save, bit_spend, bit_factor, fill_level, forgetful_min_pe; ctx->fill_level += ctx->frame_bits - bits; ctx->fill_level = av_clip(ctx->fill_level, 0, size); fill_level = av_clipf((float)ctx->fill_level / size, clip_low, clip_high); clipped_pe = av_clipf(pe, ctx->pe.min, ctx->pe.max); bit_save = (fill_level + bitsave_add) * bitsave_slope; assert(bit_save <= 0.3f && bit_save >= -0.05000001f); bit_spend = (fill_level + bitspend_add) * bitspend_slope; assert(bit_spend <= 0.5f && bit_spend >= -0.1f); /* The bit factor graph in the spec is obviously incorrect. * bit_spend + ((bit_spend - bit_spend))... * The reference encoder subtracts everything from 1, but also seems incorrect. * 1 - bit_save + ((bit_spend + bit_save))... * Hopefully below is correct. */ bit_factor = 1.0f - bit_save + ((bit_spend - bit_save) / (ctx->pe.max - ctx->pe.min)) * (clipped_pe - ctx->pe.min); /* NOTE: The reference encoder attempts to center pe max/min around the current pe. * Here we do that by slowly forgetting pe.min when pe stays in a range that makes * it unlikely (ie: above the mean) */ ctx->pe.max = FFMAX(pe, ctx->pe.max); forgetful_min_pe = ((ctx->pe.min * PSY_PE_FORGET_SLOPE) + FFMAX(ctx->pe.min, pe * (pe / ctx->pe.max))) / (PSY_PE_FORGET_SLOPE + 1); ctx->pe.min = FFMIN(pe, forgetful_min_pe); /* NOTE: allocate a minimum of 1/8th average frame bits, to avoid * reservoir starvation from producing zero-bit frames */ return FFMIN( ctx->frame_bits * bit_factor, FFMAX(ctx->frame_bits + size - bits, ctx->frame_bits / 8)); } static float calc_pe_3gpp(AacPsyBand *band) { float pe, a; band->pe = 0.0f; band->pe_const = 0.0f; band->active_lines = 0.0f; if (band->energy > band->thr) { a = log2f(band->energy); pe = a - log2f(band->thr); band->active_lines = band->nz_lines; if (pe < PSY_3GPP_C1) { pe = pe * PSY_3GPP_C3 + PSY_3GPP_C2; a = a * PSY_3GPP_C3 + PSY_3GPP_C2; band->active_lines *= PSY_3GPP_C3; } band->pe = pe * band->nz_lines; band->pe_const = a * band->nz_lines; } return band->pe; } static float calc_reduction_3gpp(float a, float desired_pe, float pe, float active_lines) { float thr_avg, reduction; if(active_lines == 0.0) return 0; thr_avg = exp2f((a - pe) / (4.0f * active_lines)); reduction = exp2f((a - desired_pe) / (4.0f * active_lines)) - thr_avg; return FFMAX(reduction, 0.0f); } static float calc_reduced_thr_3gpp(AacPsyBand *band, float min_snr, float reduction) { float thr = band->thr; if (band->energy > thr) { thr = sqrtf(thr); thr = sqrtf(thr) + reduction; thr *= thr; thr *= thr; /* This deviates from the 3GPP spec to match the reference encoder. * It performs min(thr_reduced, max(thr, energy/min_snr)) only for bands * that have hole avoidance on (active or inactive). It always reduces the * threshold of bands with hole avoidance off. */ if (thr > band->energy * min_snr && band->avoid_holes != PSY_3GPP_AH_NONE) { thr = FFMAX(band->thr, band->energy * min_snr); band->avoid_holes = PSY_3GPP_AH_ACTIVE; } } return thr; } #ifndef calc_thr_3gpp 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) { int i, w, g; int start = 0, wstart = 0; for (w = 0; w < wi->num_windows*16; w += 16) { wstart = 0; for (g = 0; g < num_bands; g++) { AacPsyBand *band = &pch->band[w+g]; float form_factor = 0.0f; float Temp; band->energy = 0.0f; if (wstart < cutoff) { for (i = 0; i < band_sizes[g]; i++) { band->energy += coefs[start+i] * coefs[start+i]; form_factor += sqrtf(fabs(coefs[start+i])); } } Temp = band->energy > 0 ? sqrtf((float)band_sizes[g] / band->energy) : 0; band->thr = band->energy * 0.001258925f; band->nz_lines = form_factor * sqrtf(Temp); start += band_sizes[g]; wstart += band_sizes[g]; } } } #endif /* calc_thr_3gpp */ #ifndef psy_hp_filter static void psy_hp_filter(const float *firbuf, float *hpfsmpl, const float *psy_fir_coeffs) { int i, j; for (i = 0; i < AAC_BLOCK_SIZE_LONG; i++) { float sum1, sum2; sum1 = firbuf[i + (PSY_LAME_FIR_LEN - 1) / 2]; sum2 = 0.0; for (j = 0; j < ((PSY_LAME_FIR_LEN - 1) / 2) - 1; j += 2) { sum1 += psy_fir_coeffs[j] * (firbuf[i + j] + firbuf[i + PSY_LAME_FIR_LEN - j]); sum2 += psy_fir_coeffs[j + 1] * (firbuf[i + j + 1] + firbuf[i + PSY_LAME_FIR_LEN - j - 1]); } /* NOTE: The LAME psymodel expects it's input in the range -32768 to 32768. * Tuning this for normalized floats would be difficult. */ hpfsmpl[i] = (sum1 + sum2) * 32768.0f; } } #endif /* psy_hp_filter */ /** * Calculate band thresholds as suggested in 3GPP TS26.403 */ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel, const float *coefs, const FFPsyWindowInfo *wi) { AacPsyContext *pctx = (AacPsyContext*) ctx->model_priv_data; AacPsyChannel *pch = &pctx->ch[channel]; int i, w, g; float desired_bits, desired_pe, delta_pe, reduction= NAN, spread_en[128] = {0}; float a = 0.0f, active_lines = 0.0f, norm_fac = 0.0f; float pe = pctx->chan_bitrate > 32000 ? 0.0f : FFMAX(50.0f, 100.0f - pctx->chan_bitrate * 100.0f / 32000.0f); const int num_bands = ctx->num_bands[wi->num_windows == 8]; const uint8_t *band_sizes = ctx->bands[wi->num_windows == 8]; AacPsyCoeffs *coeffs = pctx->psy_coef[wi->num_windows == 8]; const float avoid_hole_thr = wi->num_windows == 8 ? PSY_3GPP_AH_THR_SHORT : PSY_3GPP_AH_THR_LONG; const int bandwidth = ctx->cutoff ? ctx->cutoff : AAC_CUTOFF(ctx->avctx); const int cutoff = bandwidth * 2048 / wi->num_windows / ctx->avctx->sample_rate; //calculate energies, initial thresholds and related values - 5.4.2 "Threshold Calculation" calc_thr_3gpp(wi, num_bands, pch, band_sizes, coefs, cutoff); //modify thresholds and energies - spread, threshold in quiet, pre-echo control for (w = 0; w < wi->num_windows*16; w += 16) { AacPsyBand *bands = &pch->band[w]; /* 5.4.2.3 "Spreading" & 5.4.3 "Spread Energy Calculation" */ spread_en[0] = bands[0].energy; for (g = 1; g < num_bands; g++) { bands[g].thr = FFMAX(bands[g].thr, bands[g-1].thr * coeffs[g].spread_hi[0]); spread_en[w+g] = FFMAX(bands[g].energy, spread_en[w+g-1] * coeffs[g].spread_hi[1]); } for (g = num_bands - 2; g >= 0; g--) { bands[g].thr = FFMAX(bands[g].thr, bands[g+1].thr * coeffs[g].spread_low[0]); spread_en[w+g] = FFMAX(spread_en[w+g], spread_en[w+g+1] * coeffs[g].spread_low[1]); } //5.4.2.4 "Threshold in quiet" for (g = 0; g < num_bands; g++) { AacPsyBand *band = &bands[g]; band->thr_quiet = band->thr = FFMAX(band->thr, coeffs[g].ath); //5.4.2.5 "Pre-echo control" if (!(wi->window_type[0] == LONG_STOP_SEQUENCE || (!w && wi->window_type[1] == LONG_START_SEQUENCE))) band->thr = FFMAX(PSY_3GPP_RPEMIN*band->thr, FFMIN(band->thr, PSY_3GPP_RPELEV*pch->prev_band[w+g].thr_quiet)); /* 5.6.1.3.1 "Preparatory steps of the perceptual entropy calculation" */ pe += calc_pe_3gpp(band); a += band->pe_const; active_lines += band->active_lines; /* 5.6.1.3.3 "Selection of the bands for avoidance of holes" */ if (spread_en[w+g] * avoid_hole_thr > band->energy || coeffs[g].min_snr > 1.0f) band->avoid_holes = PSY_3GPP_AH_NONE; else band->avoid_holes = PSY_3GPP_AH_INACTIVE; } } /* 5.6.1.3.2 "Calculation of the desired perceptual entropy" */ ctx->ch[channel].entropy = pe; if (ctx->avctx->flags & AV_CODEC_FLAG_QSCALE) { /* (2.5 * 120) achieves almost transparent rate, and we want to give * ample room downwards, so we make that equivalent to QSCALE=2.4 */ desired_pe = pe * (ctx->avctx->global_quality ? ctx->avctx->global_quality : 120) / (2 * 2.5f * 120.0f); desired_bits = FFMIN(2560, PSY_3GPP_PE_TO_BITS(desired_pe)); desired_pe = PSY_3GPP_BITS_TO_PE(desired_bits); // reflect clipping /* PE slope smoothing */ if (ctx->bitres.bits > 0) { desired_bits = FFMIN(2560, PSY_3GPP_PE_TO_BITS(desired_pe)); desired_pe = PSY_3GPP_BITS_TO_PE(desired_bits); // reflect clipping } pctx->pe.max = FFMAX(pe, pctx->pe.max); pctx->pe.min = FFMIN(pe, pctx->pe.min); } else { desired_bits = calc_bit_demand(pctx, pe, ctx->bitres.bits, ctx->bitres.size, wi->num_windows == 8); desired_pe = PSY_3GPP_BITS_TO_PE(desired_bits); /* NOTE: PE correction is kept simple. During initial testing it had very * little effect on the final bitrate. Probably a good idea to come * back and do more testing later. */ if (ctx->bitres.bits > 0) desired_pe *= av_clipf(pctx->pe.previous / PSY_3GPP_BITS_TO_PE(ctx->bitres.bits), 0.85f, 1.15f); } pctx->pe.previous = PSY_3GPP_BITS_TO_PE(desired_bits); ctx->bitres.alloc = desired_bits; if (desired_pe < pe) { /* 5.6.1.3.4 "First Estimation of the reduction value" */ for (w = 0; w < wi->num_windows*16; w += 16) { reduction = calc_reduction_3gpp(a, desired_pe, pe, active_lines); pe = 0.0f; a = 0.0f; active_lines = 0.0f; for (g = 0; g < num_bands; g++) { AacPsyBand *band = &pch->band[w+g]; band->thr = calc_reduced_thr_3gpp(band, coeffs[g].min_snr, reduction); /* recalculate PE */ pe += calc_pe_3gpp(band); a += band->pe_const; active_lines += band->active_lines; } } /* 5.6.1.3.5 "Second Estimation of the reduction value" */ for (i = 0; i < 2; i++) { float pe_no_ah = 0.0f, desired_pe_no_ah; active_lines = a = 0.0f; for (w = 0; w < wi->num_windows*16; w += 16) { for (g = 0; g < num_bands; g++) { AacPsyBand *band = &pch->band[w+g]; if (band->avoid_holes != PSY_3GPP_AH_ACTIVE) { pe_no_ah += band->pe; a += band->pe_const; active_lines += band->active_lines; } } } desired_pe_no_ah = FFMAX(desired_pe - (pe - pe_no_ah), 0.0f); if (active_lines > 0.0f) reduction = calc_reduction_3gpp(a, desired_pe_no_ah, pe_no_ah, active_lines); pe = 0.0f; for (w = 0; w < wi->num_windows*16; w += 16) { for (g = 0; g < num_bands; g++) { AacPsyBand *band = &pch->band[w+g]; if (active_lines > 0.0f) band->thr = calc_reduced_thr_3gpp(band, coeffs[g].min_snr, reduction); pe += calc_pe_3gpp(band); if (band->thr > 0.0f) band->norm_fac = band->active_lines / band->thr; else band->norm_fac = 0.0f; norm_fac += band->norm_fac; } } delta_pe = desired_pe - pe; if (fabs(delta_pe) > 0.05f * desired_pe) break; } if (pe < 1.15f * desired_pe) { /* 6.6.1.3.6 "Final threshold modification by linearization" */ norm_fac = norm_fac ? 1.0f / norm_fac : 0; for (w = 0; w < wi->num_windows*16; w += 16) { for (g = 0; g < num_bands; g++) { AacPsyBand *band = &pch->band[w+g]; if (band->active_lines > 0.5f) { float delta_sfb_pe = band->norm_fac * norm_fac * delta_pe; float thr = band->thr; thr *= exp2f(delta_sfb_pe / band->active_lines); if (thr > coeffs[g].min_snr * band->energy && band->avoid_holes == PSY_3GPP_AH_INACTIVE) thr = FFMAX(band->thr, coeffs[g].min_snr * band->energy); band->thr = thr; } } } } else { /* 5.6.1.3.7 "Further perceptual entropy reduction" */ g = num_bands; while (pe > desired_pe && g--) { for (w = 0; w < wi->num_windows*16; w+= 16) { AacPsyBand *band = &pch->band[w+g]; if (band->avoid_holes != PSY_3GPP_AH_NONE && coeffs[g].min_snr < PSY_SNR_1DB) { coeffs[g].min_snr = PSY_SNR_1DB; band->thr = band->energy * PSY_SNR_1DB; pe += band->active_lines * 1.5f - band->pe; } } } /* TODO: allow more holes (unused without mid/side) */ } } for (w = 0; w < wi->num_windows*16; w += 16) { for (g = 0; g < num_bands; g++) { AacPsyBand *band = &pch->band[w+g]; FFPsyBand *psy_band = &ctx->ch[channel].psy_bands[w+g]; psy_band->threshold = band->thr; psy_band->energy = band->energy; psy_band->spread = band->active_lines * 2.0f / band_sizes[g]; psy_band->bits = PSY_3GPP_PE_TO_BITS(band->pe); } } memcpy(pch->prev_band, pch->band, sizeof(pch->band)); } static void psy_3gpp_analyze(FFPsyContext *ctx, int channel, const float **coeffs, const FFPsyWindowInfo *wi) { int ch; FFPsyChannelGroup *group = ff_psy_find_group(ctx, channel); for (ch = 0; ch < group->num_ch; ch++) psy_3gpp_analyze_channel(ctx, channel + ch, coeffs[ch], &wi[ch]); } static av_cold void psy_3gpp_end(FFPsyContext *apc) { AacPsyContext *pctx = (AacPsyContext*) apc->model_priv_data; if (pctx) av_freep(&pctx->ch); av_freep(&apc->model_priv_data); } static void lame_apply_block_type(AacPsyChannel *ctx, FFPsyWindowInfo *wi, int uselongblock) { int blocktype = ONLY_LONG_SEQUENCE; if (uselongblock) { if (ctx->next_window_seq == EIGHT_SHORT_SEQUENCE) blocktype = LONG_STOP_SEQUENCE; } else { blocktype = EIGHT_SHORT_SEQUENCE; if (ctx->next_window_seq == ONLY_LONG_SEQUENCE) ctx->next_window_seq = LONG_START_SEQUENCE; if (ctx->next_window_seq == LONG_STOP_SEQUENCE) ctx->next_window_seq = EIGHT_SHORT_SEQUENCE; } wi->window_type[0] = ctx->next_window_seq; ctx->next_window_seq = blocktype; } static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio, const float *la, int channel, int prev_type) { AacPsyContext *pctx = (AacPsyContext*) ctx->model_priv_data; AacPsyChannel *pch = &pctx->ch[channel]; int grouping = 0; int uselongblock = 1; int attacks[AAC_NUM_BLOCKS_SHORT + 1] = { 0 }; int i; FFPsyWindowInfo wi = { { 0 } }; if (la) { float hpfsmpl[AAC_BLOCK_SIZE_LONG]; const float *pf = hpfsmpl; float attack_intensity[(AAC_NUM_BLOCKS_SHORT + 1) * PSY_LAME_NUM_SUBBLOCKS]; float energy_subshort[(AAC_NUM_BLOCKS_SHORT + 1) * PSY_LAME_NUM_SUBBLOCKS]; float energy_short[AAC_NUM_BLOCKS_SHORT + 1] = { 0 }; const float *firbuf = la + (AAC_BLOCK_SIZE_SHORT/4 - PSY_LAME_FIR_LEN); int att_sum = 0; /* LAME comment: apply high pass filter of fs/4 */ psy_hp_filter(firbuf, hpfsmpl, psy_fir_coeffs); /* Calculate the energies of each sub-shortblock */ for (i = 0; i < PSY_LAME_NUM_SUBBLOCKS; i++) { energy_subshort[i] = pch->prev_energy_subshort[i + ((AAC_NUM_BLOCKS_SHORT - 1) * PSY_LAME_NUM_SUBBLOCKS)]; assert(pch->prev_energy_subshort[i + ((AAC_NUM_BLOCKS_SHORT - 2) * PSY_LAME_NUM_SUBBLOCKS + 1)] > 0); attack_intensity[i] = energy_subshort[i] / pch->prev_energy_subshort[i + ((AAC_NUM_BLOCKS_SHORT - 2) * PSY_LAME_NUM_SUBBLOCKS + 1)]; energy_short[0] += energy_subshort[i]; } for (i = 0; i < AAC_NUM_BLOCKS_SHORT * PSY_LAME_NUM_SUBBLOCKS; i++) { const float *const pfe = pf + AAC_BLOCK_SIZE_LONG / (AAC_NUM_BLOCKS_SHORT * PSY_LAME_NUM_SUBBLOCKS); float p = 1.0f; for (; pf < pfe; pf++) p = FFMAX(p, fabsf(*pf)); pch->prev_energy_subshort[i] = energy_subshort[i + PSY_LAME_NUM_SUBBLOCKS] = p; energy_short[1 + i / PSY_LAME_NUM_SUBBLOCKS] += p; /* NOTE: The indexes below are [i + 3 - 2] in the LAME source. * Obviously the 3 and 2 have some significance, or this would be just [i + 1] * (which is what we use here). What the 3 stands for is ambiguous, as it is both * number of short blocks, and the number of sub-short blocks. * It seems that LAME is comparing each sub-block to sub-block + 1 in the * previous block. */ if (p > energy_subshort[i + 1]) p = p / energy_subshort[i + 1]; else if (energy_subshort[i + 1] > p * 10.0f) p = energy_subshort[i + 1] / (p * 10.0f); else p = 0.0; attack_intensity[i + PSY_LAME_NUM_SUBBLOCKS] = p; } /* compare energy between sub-short blocks */ for (i = 0; i < (AAC_NUM_BLOCKS_SHORT + 1) * PSY_LAME_NUM_SUBBLOCKS; i++) if (!attacks[i / PSY_LAME_NUM_SUBBLOCKS]) if (attack_intensity[i] > pch->attack_threshold) attacks[i / PSY_LAME_NUM_SUBBLOCKS] = (i % PSY_LAME_NUM_SUBBLOCKS) + 1; /* should have energy change between short blocks, in order to avoid periodic signals */ /* Good samples to show the effect are Trumpet test songs */ /* GB: tuned (1) to avoid too many short blocks for test sample TRUMPET */ /* RH: tuned (2) to let enough short blocks through for test sample FSOL and SNAPS */ for (i = 1; i < AAC_NUM_BLOCKS_SHORT + 1; i++) { const float u = energy_short[i - 1]; const float v = energy_short[i]; const float m = FFMAX(u, v); if (m < 40000) { /* (2) */ if (u < 1.7f * v && v < 1.7f * u) { /* (1) */ if (i == 1 && attacks[0] < attacks[i]) attacks[0] = 0; attacks[i] = 0; } } att_sum += attacks[i]; } if (attacks[0] <= pch->prev_attack) attacks[0] = 0; att_sum += attacks[0]; /* 3 below indicates the previous attack happened in the last sub-block of the previous sequence */ if (pch->prev_attack == 3 || att_sum) { uselongblock = 0; for (i = 1; i < AAC_NUM_BLOCKS_SHORT + 1; i++) if (attacks[i] && attacks[i-1]) attacks[i] = 0; } } else { /* We have no lookahead info, so just use same type as the previous sequence. */ uselongblock = !(prev_type == EIGHT_SHORT_SEQUENCE); } lame_apply_block_type(pch, &wi, uselongblock); wi.window_type[1] = prev_type; if (wi.window_type[0] != EIGHT_SHORT_SEQUENCE) { wi.num_windows = 1; wi.grouping[0] = 1; if (wi.window_type[0] == LONG_START_SEQUENCE) wi.window_shape = 0; else wi.window_shape = 1; } else { int lastgrp = 0; wi.num_windows = 8; wi.window_shape = 0; for (i = 0; i < 8; i++) { if (!((pch->next_grouping >> i) & 1)) lastgrp = i; wi.grouping[lastgrp]++; } } /* Determine grouping, based on the location of the first attack, and save for * the next frame. * FIXME: Move this to analysis. * TODO: Tune groupings depending on attack location * TODO: Handle more than one attack in a group */ for (i = 0; i < 9; i++) { if (attacks[i]) { grouping = i; break; } } pch->next_grouping = window_grouping[grouping]; pch->prev_attack = attacks[8]; return wi; } const FFPsyModel ff_aac_psy_model = { .name = "3GPP TS 26.403-inspired model", .init = psy_3gpp_init, .window = psy_lame_window, .analyze = psy_3gpp_analyze, .end = psy_3gpp_end, };