* AAC encoder psychoacoustic model
* Copyright (C) 2008 Konstantin Shishkov
*
- * This file is part of FFmpeg.
+ * This file is part of Libav.
*
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav 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,
+ * Libav 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
+ * License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
- * @file libavcodec/aacpsy.c
+ * @file
* AAC encoder psychoacoustic model
*/
#include "avcodec.h"
-#include "aacpsy.h"
#include "aactab.h"
+#include "psymodel.h"
/***********************************
* TODOs:
- * General:
- * better audio preprocessing (add DC highpass filter?)
- * more psy models
- * maybe improve coefficient quantization function in some way
- *
- * 3GPP-based psy model:
* thresholds linearization after their modifications for attaining given bitrate
* try other bitrate controlling mechanism (maybe use ratecontrol.c?)
* control quality for quality-based output
**********************************/
-/**
- * Quantize one coefficient.
- * @return absolute value of the quantized coefficient
- * @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
- */
-static av_always_inline int quant(float coef, const float Q)
-{
- return av_clip((int)(pow(fabsf(coef) * Q, 0.75) + 0.4054), 0, 8191);
-}
-
-static inline float get_approximate_quant_error(float *c, int size, int scale_idx)
-{
- int i;
- int q;
- float coef, unquant, sum = 0.0f;
- const float Q = ff_aac_pow2sf_tab[200 - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
- const float IQ = ff_aac_pow2sf_tab[200 + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
- for(i = 0; i < size; i++){
- coef = fabs(c[i]);
- q = quant(c[i], Q);
- unquant = (q * cbrt(q)) * IQ;
- sum += (coef - unquant) * (coef - unquant);
- }
- return sum;
-}
-
/**
* constants for 3GPP AAC psychoacoustic model
* @{
*/
-#define PSY_3GPP_SPREAD_LOW 1.5f // spreading factor for ascending threshold spreading (15 dB/Bark)
-#define PSY_3GPP_SPREAD_HI 3.0f // spreading factor for descending threshold spreading (30 dB/Bark)
+#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)
+
+#define PSY_3GPP_RPEMIN 0.01f
+#define PSY_3GPP_RPELEV 2.0f
+
+/* 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 Psy3gppBand{
+typedef struct AacPsyBand{
float energy; ///< band energy
- float ffac; ///< form factor
-}Psy3gppBand;
+ float thr; ///< energy threshold
+ float thr_quiet; ///< threshold in quiet
+}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 Psy3gppCoeffs{
- float ath [64]; ///< absolute threshold of hearing per bands
- float barks [64]; ///< Bark value for each spectral band in long frame
- float spread_low[64]; ///< spreading factor for low-to-high threshold spreading in long frame
- float spread_hi [64]; ///< spreading factor for high-to-low threshold spreading in long frame
-}Psy3gppCoeffs;
+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{
+ AacPsyCoeffs psy_coef[2][64];
+ AacPsyChannel *ch;
+}AacPsyContext;
+
+/**
+ * LAME psy model preset struct
+ */
+typedef struct {
+ 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
+};
+
+/**
+ * calculates the attack threshold for ABR from the above table for the LAME psy model
+ */
+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 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 & 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 inline float calc_bark(float f)
+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;
+
+ ctx->model_priv_data = av_mallocz(sizeof(AacPsyContext));
+ pctx = (AacPsyContext*) ctx->model_priv_data;
+
+ minath = ath(3410, 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);
+ 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] = pow(10.0, -bark_width * PSY_3GPP_THR_SPREAD_LOW);
+ coeff->spread_hi [0] = pow(10.0, -bark_width * PSY_3GPP_THR_SPREAD_HI);
+ }
+ 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_mallocz(sizeof(AacPsyChannel) * ctx->avctx->channels);
+
+ 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 FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
+ const int16_t *audio, const int16_t *la,
+ int channel, int prev_type)
+{
+ int i, j;
+ int br = ctx->avctx->bit_rate / ctx->avctx->channels;
+ 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;
+
+ memset(&wi, 0, sizeof(wi));
+ 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)*ctx->avctx->channels], 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;
+}
+
+/**
+ * Calculate band thresholds as suggested in 3GPP TS26.403
+ */
+static void psy_3gpp_analyze(FFPsyContext *ctx, int channel,
+ const float *coefs, const FFPsyWindowInfo *wi)
+{
+ AacPsyContext *pctx = (AacPsyContext*) ctx->model_priv_data;
+ AacPsyChannel *pch = &pctx->ch[channel];
+ int start = 0;
+ int i, w, g;
+ 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];
+
+ //calculate energies, initial thresholds and related values - 5.4.2 "Threshold Calculation"
+ for (w = 0; w < wi->num_windows*16; w += 16) {
+ for (g = 0; g < num_bands; g++) {
+ AacPsyBand *band = &pch->band[w+g];
+ band->energy = 0.0f;
+ for (i = 0; i < band_sizes[g]; i++)
+ band->energy += coefs[start+i] * coefs[start+i];
+ band->thr = band->energy * 0.001258925f;
+ start += band_sizes[g];
+ }
+ }
+ //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 "Spreaded Energy Calculation"
+ for (g = 1; g < num_bands; g++)
+ bands[g].thr = FFMAX(bands[g].thr, bands[g-1].thr * coeffs[g].spread_hi[0]);
+ for (g = num_bands - 2; g >= 0; g--)
+ bands[g].thr = FFMAX(bands[g].thr, bands[g+1].thr * coeffs[g].spread_low[0]);
+ //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 || (wi->window_type[1] == LONG_START_SEQUENCE && !w)))
+ band->thr = FFMAX(PSY_3GPP_RPEMIN*band->thr, FFMIN(band->thr,
+ PSY_3GPP_RPELEV*pch->prev_band[w+g].thr_quiet));
+ }
+ }
+
+ 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->psy_bands[channel*PSY_MAX_BANDS+w+g];
+
+ psy_band->threshold = band->thr;
+ psy_band->energy = band->energy;
+ }
+ }
+
+ memcpy(pch->prev_band, pch->band, sizeof(pch->band));
+}
+
+static av_cold void psy_3gpp_end(FFPsyContext *apc)
+{
+ AacPsyContext *pctx = (AacPsyContext*) apc->model_priv_data;
+ 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 int16_t *audio, const int16_t *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;
+
+ memset(&wi, 0, sizeof(wi));
+ if (la) {
+ float hpfsmpl[AAC_BLOCK_SIZE_LONG];
+ float const *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 };
+ int chans = ctx->avctx->channels;
+ const int16_t *firbuf = la + (AAC_BLOCK_SIZE_SHORT/4 - PSY_LAME_FIR_LEN) * chans;
+ int j, att_sum = 0;
+
+ /* LAME comment: apply high pass filter of fs/4 */
+ for (i = 0; i < AAC_BLOCK_SIZE_LONG; i++) {
+ float sum1, sum2;
+ sum1 = firbuf[(i + ((PSY_LAME_FIR_LEN - 1) / 2)) * chans];
+ sum2 = 0.0;
+ for (j = 0; j < ((PSY_LAME_FIR_LEN - 1) / 2) - 1; j += 2) {
+ sum1 += psy_fir_coeffs[j] * (firbuf[(i + j) * chans] + firbuf[(i + PSY_LAME_FIR_LEN - j) * chans]);
+ sum2 += psy_fir_coeffs[j + 1] * (firbuf[(i + j + 1) * chans] + firbuf[(i + PSY_LAME_FIR_LEN - j - 1) * chans]);
+ }
+ hpfsmpl[i] = sum1 + sum2;
+ }
+
+ /* 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++) {
+ float const *const pfe = pf + AAC_BLOCK_SIZE_LONG / (AAC_NUM_BLOCKS_SHORT * PSY_LAME_NUM_SUBBLOCKS);
+ float p = 1.0f;
+ for (; pf < pfe; pf++)
+ if (p < fabsf(*pf))
+ 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;
+ /* FIXME: 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 ambigious, 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++) {
+ float const u = energy_short[i - 1];
+ float const v = energy_short[i];
+ float const 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,
+};
projects / ffmpeg / blobdiff