FFmpeg  4.0
aacpsy.c
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1 /*
2  * AAC encoder psychoacoustic model
3  * Copyright (C) 2008 Konstantin Shishkov
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * AAC encoder psychoacoustic model
25  */
26 
27 #include "libavutil/attributes.h"
28 #include "libavutil/ffmath.h"
29 
30 #include "avcodec.h"
31 #include "aactab.h"
32 #include "psymodel.h"
33 
34 /***********************************
35  * TODOs:
36  * try other bitrate controlling mechanism (maybe use ratecontrol.c?)
37  * control quality for quality-based output
38  **********************************/
39 
40 /**
41  * constants for 3GPP AAC psychoacoustic model
42  * @{
43  */
44 #define PSY_3GPP_THR_SPREAD_HI 1.5f // spreading factor for low-to-hi threshold spreading (15 dB/Bark)
45 #define PSY_3GPP_THR_SPREAD_LOW 3.0f // spreading factor for hi-to-low threshold spreading (30 dB/Bark)
46 /* spreading factor for low-to-hi energy spreading, long block, > 22kbps/channel (20dB/Bark) */
47 #define PSY_3GPP_EN_SPREAD_HI_L1 2.0f
48 /* spreading factor for low-to-hi energy spreading, long block, <= 22kbps/channel (15dB/Bark) */
49 #define PSY_3GPP_EN_SPREAD_HI_L2 1.5f
50 /* spreading factor for low-to-hi energy spreading, short block (15 dB/Bark) */
51 #define PSY_3GPP_EN_SPREAD_HI_S 1.5f
52 /* spreading factor for hi-to-low energy spreading, long block (30dB/Bark) */
53 #define PSY_3GPP_EN_SPREAD_LOW_L 3.0f
54 /* spreading factor for hi-to-low energy spreading, short block (20dB/Bark) */
55 #define PSY_3GPP_EN_SPREAD_LOW_S 2.0f
56 
57 #define PSY_3GPP_RPEMIN 0.01f
58 #define PSY_3GPP_RPELEV 2.0f
59 
60 #define PSY_3GPP_C1 3.0f /* log2(8) */
61 #define PSY_3GPP_C2 1.3219281f /* log2(2.5) */
62 #define PSY_3GPP_C3 0.55935729f /* 1 - C2 / C1 */
63 
64 #define PSY_SNR_1DB 7.9432821e-1f /* -1dB */
65 #define PSY_SNR_25DB 3.1622776e-3f /* -25dB */
66 
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
79 
80 #define PSY_3GPP_AH_THR_LONG 0.5f
81 #define PSY_3GPP_AH_THR_SHORT 0.63f
82 
83 #define PSY_PE_FORGET_SLOPE 511
84 
85 enum {
89 };
90 
91 #define PSY_3GPP_BITS_TO_PE(bits) ((bits) * 1.18f)
92 #define PSY_3GPP_PE_TO_BITS(bits) ((bits) / 1.18f)
93 
94 /* LAME psy model constants */
95 #define PSY_LAME_FIR_LEN 21 ///< LAME psy model FIR order
96 #define AAC_BLOCK_SIZE_LONG 1024 ///< long block size
97 #define AAC_BLOCK_SIZE_SHORT 128 ///< short block size
98 #define AAC_NUM_BLOCKS_SHORT 8 ///< number of blocks in a short sequence
99 #define PSY_LAME_NUM_SUBBLOCKS 3 ///< Number of sub-blocks in each short block
100 
101 /**
102  * @}
103  */
104 
105 /**
106  * information for single band used by 3GPP TS26.403-inspired psychoacoustic model
107  */
108 typedef struct AacPsyBand{
109  float energy; ///< band energy
110  float thr; ///< energy threshold
111  float thr_quiet; ///< threshold in quiet
112  float nz_lines; ///< number of non-zero spectral lines
113  float active_lines; ///< number of active spectral lines
114  float pe; ///< perceptual entropy
115  float pe_const; ///< constant part of the PE calculation
116  float norm_fac; ///< normalization factor for linearization
117  int avoid_holes; ///< hole avoidance flag
118 }AacPsyBand;
119 
120 /**
121  * single/pair channel context for psychoacoustic model
122  */
123 typedef struct AacPsyChannel{
124  AacPsyBand band[128]; ///< bands information
125  AacPsyBand prev_band[128]; ///< bands information from the previous frame
126 
127  float win_energy; ///< sliding average of channel energy
128  float iir_state[2]; ///< hi-pass IIR filter state
129  uint8_t next_grouping; ///< stored grouping scheme for the next frame (in case of 8 short window sequence)
130  enum WindowSequence next_window_seq; ///< window sequence to be used in the next frame
131  /* LAME psy model specific members */
132  float attack_threshold; ///< attack threshold for this channel
133  float prev_energy_subshort[AAC_NUM_BLOCKS_SHORT * PSY_LAME_NUM_SUBBLOCKS];
134  int prev_attack; ///< attack value for the last short block in the previous sequence
136 
137 /**
138  * psychoacoustic model frame type-dependent coefficients
139  */
140 typedef struct AacPsyCoeffs{
141  float ath; ///< absolute threshold of hearing per bands
142  float barks; ///< Bark value for each spectral band in long frame
143  float spread_low[2]; ///< spreading factor for low-to-high threshold spreading in long frame
144  float spread_hi [2]; ///< spreading factor for high-to-low threshold spreading in long frame
145  float min_snr; ///< minimal SNR
146 }AacPsyCoeffs;
147 
148 /**
149  * 3GPP TS26.403-inspired psychoacoustic model specific data
150  */
151 typedef struct AacPsyContext{
152  int chan_bitrate; ///< bitrate per channel
153  int frame_bits; ///< average bits per frame
154  int fill_level; ///< bit reservoir fill level
155  struct {
156  float min; ///< minimum allowed PE for bit factor calculation
157  float max; ///< maximum allowed PE for bit factor calculation
158  float previous; ///< allowed PE of the previous frame
159  float correction; ///< PE correction factor
160  } pe;
161  AacPsyCoeffs psy_coef[2][64];
163  float global_quality; ///< normalized global quality taken from avctx
165 
166 /**
167  * LAME psy model preset struct
168  */
169 typedef struct PsyLamePreset {
170  int quality; ///< Quality to map the rest of the vaules to.
171  /* This is overloaded to be both kbps per channel in ABR mode, and
172  * requested quality in constant quality mode.
173  */
174  float st_lrm; ///< short threshold for L, R, and M channels
175 } PsyLamePreset;
176 
177 /**
178  * LAME psy model preset table for ABR
179  */
180 static const PsyLamePreset psy_abr_map[] = {
181 /* TODO: Tuning. These were taken from LAME. */
182 /* kbps/ch st_lrm */
183  { 8, 6.60},
184  { 16, 6.60},
185  { 24, 6.60},
186  { 32, 6.60},
187  { 40, 6.60},
188  { 48, 6.60},
189  { 56, 6.60},
190  { 64, 6.40},
191  { 80, 6.00},
192  { 96, 5.60},
193  {112, 5.20},
194  {128, 5.20},
195  {160, 5.20}
196 };
197 
198 /**
199 * LAME psy model preset table for constant quality
200 */
201 static const PsyLamePreset psy_vbr_map[] = {
202 /* vbr_q st_lrm */
203  { 0, 4.20},
204  { 1, 4.20},
205  { 2, 4.20},
206  { 3, 4.20},
207  { 4, 4.20},
208  { 5, 4.20},
209  { 6, 4.20},
210  { 7, 4.20},
211  { 8, 4.20},
212  { 9, 4.20},
213  {10, 4.20}
214 };
215 
216 /**
217  * LAME psy model FIR coefficient table
218  */
219 static const float psy_fir_coeffs[] = {
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
223 };
224 
225 #if ARCH_MIPS
226 # include "mips/aacpsy_mips.h"
227 #endif /* ARCH_MIPS */
228 
229 /**
230  * Calculate the ABR attack threshold from the above LAME psymodel table.
231  */
232 static float lame_calc_attack_threshold(int bitrate)
233 {
234  /* Assume max bitrate to start with */
235  int lower_range = 12, upper_range = 12;
236  int lower_range_kbps = psy_abr_map[12].quality;
237  int upper_range_kbps = psy_abr_map[12].quality;
238  int i;
239 
240  /* Determine which bitrates the value specified falls between.
241  * If the loop ends without breaking our above assumption of 320kbps was correct.
242  */
243  for (i = 1; i < 13; i++) {
244  if (FFMAX(bitrate, psy_abr_map[i].quality) != bitrate) {
245  upper_range = i;
246  upper_range_kbps = psy_abr_map[i ].quality;
247  lower_range = i - 1;
248  lower_range_kbps = psy_abr_map[i - 1].quality;
249  break; /* Upper range found */
250  }
251  }
252 
253  /* Determine which range the value specified is closer to */
254  if ((upper_range_kbps - bitrate) > (bitrate - lower_range_kbps))
255  return psy_abr_map[lower_range].st_lrm;
256  return psy_abr_map[upper_range].st_lrm;
257 }
258 
259 /**
260  * LAME psy model specific initialization
261  */
263 {
264  int i, j;
265 
266  for (i = 0; i < avctx->channels; i++) {
267  AacPsyChannel *pch = &ctx->ch[i];
268 
269  if (avctx->flags & AV_CODEC_FLAG_QSCALE)
270  pch->attack_threshold = psy_vbr_map[avctx->global_quality / FF_QP2LAMBDA].st_lrm;
271  else
272  pch->attack_threshold = lame_calc_attack_threshold(avctx->bit_rate / avctx->channels / 1000);
273 
274  for (j = 0; j < AAC_NUM_BLOCKS_SHORT * PSY_LAME_NUM_SUBBLOCKS; j++)
275  pch->prev_energy_subshort[j] = 10.0f;
276  }
277 }
278 
279 /**
280  * Calculate Bark value for given line.
281  */
282 static av_cold float calc_bark(float f)
283 {
284  return 13.3f * atanf(0.00076f * f) + 3.5f * atanf((f / 7500.0f) * (f / 7500.0f));
285 }
286 
287 #define ATH_ADD 4
288 /**
289  * Calculate ATH value for given frequency.
290  * Borrowed from Lame.
291  */
292 static av_cold float ath(float f, float add)
293 {
294  f /= 1000.0f;
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;
299 }
300 
302  AacPsyContext *pctx;
303  float bark;
304  int i, j, g, start;
305  float prev, minscale, minath, minsnr, pe_min;
306  int chan_bitrate = ctx->avctx->bit_rate / ((ctx->avctx->flags & AV_CODEC_FLAG_QSCALE) ? 2.0f : ctx->avctx->channels);
307 
308  const int bandwidth = ctx->cutoff ? ctx->cutoff : AAC_CUTOFF(ctx->avctx);
309  const float num_bark = calc_bark((float)bandwidth);
310 
311  ctx->model_priv_data = av_mallocz(sizeof(AacPsyContext));
312  if (!ctx->model_priv_data)
313  return AVERROR(ENOMEM);
314  pctx = ctx->model_priv_data;
315  pctx->global_quality = (ctx->avctx->global_quality ? ctx->avctx->global_quality : 120) * 0.01f;
316 
317  if (ctx->avctx->flags & AV_CODEC_FLAG_QSCALE) {
318  /* Use the target average bitrate to compute spread parameters */
319  chan_bitrate = (int)(chan_bitrate / 120.0 * (ctx->avctx->global_quality ? ctx->avctx->global_quality : 120));
320  }
321 
322  pctx->chan_bitrate = chan_bitrate;
323  pctx->frame_bits = FFMIN(2560, chan_bitrate * AAC_BLOCK_SIZE_LONG / ctx->avctx->sample_rate);
324  pctx->pe.min = 8.0f * AAC_BLOCK_SIZE_LONG * bandwidth / (ctx->avctx->sample_rate * 2.0f);
325  pctx->pe.max = 12.0f * AAC_BLOCK_SIZE_LONG * bandwidth / (ctx->avctx->sample_rate * 2.0f);
326  ctx->bitres.size = 6144 - pctx->frame_bits;
327  ctx->bitres.size -= ctx->bitres.size % 8;
328  pctx->fill_level = ctx->bitres.size;
329  minath = ath(3410 - 0.733 * ATH_ADD, ATH_ADD);
330  for (j = 0; j < 2; j++) {
331  AacPsyCoeffs *coeffs = pctx->psy_coef[j];
332  const uint8_t *band_sizes = ctx->bands[j];
333  float line_to_frequency = ctx->avctx->sample_rate / (j ? 256.f : 2048.0f);
334  float avg_chan_bits = chan_bitrate * (j ? 128.0f : 1024.0f) / ctx->avctx->sample_rate;
335  /* reference encoder uses 2.4% here instead of 60% like the spec says */
336  float bark_pe = 0.024f * PSY_3GPP_BITS_TO_PE(avg_chan_bits) / num_bark;
337  float en_spread_low = j ? PSY_3GPP_EN_SPREAD_LOW_S : PSY_3GPP_EN_SPREAD_LOW_L;
338  /* High energy spreading for long blocks <= 22kbps/channel and short blocks are the same. */
339  float en_spread_hi = (j || (chan_bitrate <= 22.0f)) ? PSY_3GPP_EN_SPREAD_HI_S : PSY_3GPP_EN_SPREAD_HI_L1;
340 
341  i = 0;
342  prev = 0.0;
343  for (g = 0; g < ctx->num_bands[j]; g++) {
344  i += band_sizes[g];
345  bark = calc_bark((i-1) * line_to_frequency);
346  coeffs[g].barks = (bark + prev) / 2.0;
347  prev = bark;
348  }
349  for (g = 0; g < ctx->num_bands[j] - 1; g++) {
350  AacPsyCoeffs *coeff = &coeffs[g];
351  float bark_width = coeffs[g+1].barks - coeffs->barks;
352  coeff->spread_low[0] = ff_exp10(-bark_width * PSY_3GPP_THR_SPREAD_LOW);
353  coeff->spread_hi [0] = ff_exp10(-bark_width * PSY_3GPP_THR_SPREAD_HI);
354  coeff->spread_low[1] = ff_exp10(-bark_width * en_spread_low);
355  coeff->spread_hi [1] = ff_exp10(-bark_width * en_spread_hi);
356  pe_min = bark_pe * bark_width;
357  minsnr = exp2(pe_min / band_sizes[g]) - 1.5f;
358  coeff->min_snr = av_clipf(1.0f / minsnr, PSY_SNR_25DB, PSY_SNR_1DB);
359  }
360  start = 0;
361  for (g = 0; g < ctx->num_bands[j]; g++) {
362  minscale = ath(start * line_to_frequency, ATH_ADD);
363  for (i = 1; i < band_sizes[g]; i++)
364  minscale = FFMIN(minscale, ath((start + i) * line_to_frequency, ATH_ADD));
365  coeffs[g].ath = minscale - minath;
366  start += band_sizes[g];
367  }
368  }
369 
370  pctx->ch = av_mallocz_array(ctx->avctx->channels, sizeof(AacPsyChannel));
371  if (!pctx->ch) {
372  av_freep(&ctx->model_priv_data);
373  return AVERROR(ENOMEM);
374  }
375 
376  lame_window_init(pctx, ctx->avctx);
377 
378  return 0;
379 }
380 
381 /**
382  * IIR filter used in block switching decision
383  */
384 static float iir_filter(int in, float state[2])
385 {
386  float ret;
387 
388  ret = 0.7548f * (in - state[0]) + 0.5095f * state[1];
389  state[0] = in;
390  state[1] = ret;
391  return ret;
392 }
393 
394 /**
395  * window grouping information stored as bits (0 - new group, 1 - group continues)
396  */
397 static const uint8_t window_grouping[9] = {
398  0xB6, 0x6C, 0xD8, 0xB2, 0x66, 0xC6, 0x96, 0x36, 0x36
399 };
400 
401 /**
402  * Tell encoder which window types to use.
403  * @see 3GPP TS26.403 5.4.1 "Blockswitching"
404  */
406  const int16_t *audio,
407  const int16_t *la,
408  int channel, int prev_type)
409 {
410  int i, j;
411  int br = ((AacPsyContext*)ctx->model_priv_data)->chan_bitrate;
412  int attack_ratio = br <= 16000 ? 18 : 10;
414  AacPsyChannel *pch = &pctx->ch[channel];
415  uint8_t grouping = 0;
416  int next_type = pch->next_window_seq;
417  FFPsyWindowInfo wi = { { 0 } };
418 
419  if (la) {
420  float s[8], v;
421  int switch_to_eight = 0;
422  float sum = 0.0, sum2 = 0.0;
423  int attack_n = 0;
424  int stay_short = 0;
425  for (i = 0; i < 8; i++) {
426  for (j = 0; j < 128; j++) {
427  v = iir_filter(la[i*128+j], pch->iir_state);
428  sum += v*v;
429  }
430  s[i] = sum;
431  sum2 += sum;
432  }
433  for (i = 0; i < 8; i++) {
434  if (s[i] > pch->win_energy * attack_ratio) {
435  attack_n = i + 1;
436  switch_to_eight = 1;
437  break;
438  }
439  }
440  pch->win_energy = pch->win_energy*7/8 + sum2/64;
441 
442  wi.window_type[1] = prev_type;
443  switch (prev_type) {
444  case ONLY_LONG_SEQUENCE:
445  wi.window_type[0] = switch_to_eight ? LONG_START_SEQUENCE : ONLY_LONG_SEQUENCE;
446  next_type = switch_to_eight ? EIGHT_SHORT_SEQUENCE : ONLY_LONG_SEQUENCE;
447  break;
448  case LONG_START_SEQUENCE:
449  wi.window_type[0] = EIGHT_SHORT_SEQUENCE;
450  grouping = pch->next_grouping;
451  next_type = switch_to_eight ? EIGHT_SHORT_SEQUENCE : LONG_STOP_SEQUENCE;
452  break;
453  case LONG_STOP_SEQUENCE:
454  wi.window_type[0] = switch_to_eight ? LONG_START_SEQUENCE : ONLY_LONG_SEQUENCE;
455  next_type = switch_to_eight ? EIGHT_SHORT_SEQUENCE : ONLY_LONG_SEQUENCE;
456  break;
458  stay_short = next_type == EIGHT_SHORT_SEQUENCE || switch_to_eight;
459  wi.window_type[0] = stay_short ? EIGHT_SHORT_SEQUENCE : LONG_STOP_SEQUENCE;
460  grouping = next_type == EIGHT_SHORT_SEQUENCE ? pch->next_grouping : 0;
461  next_type = switch_to_eight ? EIGHT_SHORT_SEQUENCE : LONG_STOP_SEQUENCE;
462  break;
463  }
464 
465  pch->next_grouping = window_grouping[attack_n];
466  pch->next_window_seq = next_type;
467  } else {
468  for (i = 0; i < 3; i++)
469  wi.window_type[i] = prev_type;
470  grouping = (prev_type == EIGHT_SHORT_SEQUENCE) ? window_grouping[0] : 0;
471  }
472 
473  wi.window_shape = 1;
474  if (wi.window_type[0] != EIGHT_SHORT_SEQUENCE) {
475  wi.num_windows = 1;
476  wi.grouping[0] = 1;
477  } else {
478  int lastgrp = 0;
479  wi.num_windows = 8;
480  for (i = 0; i < 8; i++) {
481  if (!((grouping >> i) & 1))
482  lastgrp = i;
483  wi.grouping[lastgrp]++;
484  }
485  }
486 
487  return wi;
488 }
489 
490 /* 5.6.1.2 "Calculation of Bit Demand" */
491 static int calc_bit_demand(AacPsyContext *ctx, float pe, int bits, int size,
492  int short_window)
493 {
494  const float bitsave_slope = short_window ? PSY_3GPP_SAVE_SLOPE_S : PSY_3GPP_SAVE_SLOPE_L;
495  const float bitsave_add = short_window ? PSY_3GPP_SAVE_ADD_S : PSY_3GPP_SAVE_ADD_L;
496  const float bitspend_slope = short_window ? PSY_3GPP_SPEND_SLOPE_S : PSY_3GPP_SPEND_SLOPE_L;
497  const float bitspend_add = short_window ? PSY_3GPP_SPEND_ADD_S : PSY_3GPP_SPEND_ADD_L;
498  const float clip_low = short_window ? PSY_3GPP_CLIP_LO_S : PSY_3GPP_CLIP_LO_L;
499  const float clip_high = short_window ? PSY_3GPP_CLIP_HI_S : PSY_3GPP_CLIP_HI_L;
500  float clipped_pe, bit_save, bit_spend, bit_factor, fill_level, forgetful_min_pe;
501 
502  ctx->fill_level += ctx->frame_bits - bits;
503  ctx->fill_level = av_clip(ctx->fill_level, 0, size);
504  fill_level = av_clipf((float)ctx->fill_level / size, clip_low, clip_high);
505  clipped_pe = av_clipf(pe, ctx->pe.min, ctx->pe.max);
506  bit_save = (fill_level + bitsave_add) * bitsave_slope;
507  assert(bit_save <= 0.3f && bit_save >= -0.05000001f);
508  bit_spend = (fill_level + bitspend_add) * bitspend_slope;
509  assert(bit_spend <= 0.5f && bit_spend >= -0.1f);
510  /* The bit factor graph in the spec is obviously incorrect.
511  * bit_spend + ((bit_spend - bit_spend))...
512  * The reference encoder subtracts everything from 1, but also seems incorrect.
513  * 1 - bit_save + ((bit_spend + bit_save))...
514  * Hopefully below is correct.
515  */
516  bit_factor = 1.0f - bit_save + ((bit_spend - bit_save) / (ctx->pe.max - ctx->pe.min)) * (clipped_pe - ctx->pe.min);
517  /* NOTE: The reference encoder attempts to center pe max/min around the current pe.
518  * Here we do that by slowly forgetting pe.min when pe stays in a range that makes
519  * it unlikely (ie: above the mean)
520  */
521  ctx->pe.max = FFMAX(pe, ctx->pe.max);
522  forgetful_min_pe = ((ctx->pe.min * PSY_PE_FORGET_SLOPE)
523  + FFMAX(ctx->pe.min, pe * (pe / ctx->pe.max))) / (PSY_PE_FORGET_SLOPE + 1);
524  ctx->pe.min = FFMIN(pe, forgetful_min_pe);
525 
526  /* NOTE: allocate a minimum of 1/8th average frame bits, to avoid
527  * reservoir starvation from producing zero-bit frames
528  */
529  return FFMIN(
530  ctx->frame_bits * bit_factor,
531  FFMAX(ctx->frame_bits + size - bits, ctx->frame_bits / 8));
532 }
533 
534 static float calc_pe_3gpp(AacPsyBand *band)
535 {
536  float pe, a;
537 
538  band->pe = 0.0f;
539  band->pe_const = 0.0f;
540  band->active_lines = 0.0f;
541  if (band->energy > band->thr) {
542  a = log2f(band->energy);
543  pe = a - log2f(band->thr);
544  band->active_lines = band->nz_lines;
545  if (pe < PSY_3GPP_C1) {
546  pe = pe * PSY_3GPP_C3 + PSY_3GPP_C2;
547  a = a * PSY_3GPP_C3 + PSY_3GPP_C2;
548  band->active_lines *= PSY_3GPP_C3;
549  }
550  band->pe = pe * band->nz_lines;
551  band->pe_const = a * band->nz_lines;
552  }
553 
554  return band->pe;
555 }
556 
557 static float calc_reduction_3gpp(float a, float desired_pe, float pe,
558  float active_lines)
559 {
560  float thr_avg, reduction;
561 
562  if(active_lines == 0.0)
563  return 0;
564 
565  thr_avg = exp2f((a - pe) / (4.0f * active_lines));
566  reduction = exp2f((a - desired_pe) / (4.0f * active_lines)) - thr_avg;
567 
568  return FFMAX(reduction, 0.0f);
569 }
570 
571 static float calc_reduced_thr_3gpp(AacPsyBand *band, float min_snr,
572  float reduction)
573 {
574  float thr = band->thr;
575 
576  if (band->energy > thr) {
577  thr = sqrtf(thr);
578  thr = sqrtf(thr) + reduction;
579  thr *= thr;
580  thr *= thr;
581 
582  /* This deviates from the 3GPP spec to match the reference encoder.
583  * It performs min(thr_reduced, max(thr, energy/min_snr)) only for bands
584  * that have hole avoidance on (active or inactive). It always reduces the
585  * threshold of bands with hole avoidance off.
586  */
587  if (thr > band->energy * min_snr && band->avoid_holes != PSY_3GPP_AH_NONE) {
588  thr = FFMAX(band->thr, band->energy * min_snr);
590  }
591  }
592 
593  return thr;
594 }
595 
596 #ifndef calc_thr_3gpp
597 static void calc_thr_3gpp(const FFPsyWindowInfo *wi, const int num_bands, AacPsyChannel *pch,
598  const uint8_t *band_sizes, const float *coefs, const int cutoff)
599 {
600  int i, w, g;
601  int start = 0, wstart = 0;
602  for (w = 0; w < wi->num_windows*16; w += 16) {
603  wstart = 0;
604  for (g = 0; g < num_bands; g++) {
605  AacPsyBand *band = &pch->band[w+g];
606 
607  float form_factor = 0.0f;
608  float Temp;
609  band->energy = 0.0f;
610  if (wstart < cutoff) {
611  for (i = 0; i < band_sizes[g]; i++) {
612  band->energy += coefs[start+i] * coefs[start+i];
613  form_factor += sqrtf(fabs(coefs[start+i]));
614  }
615  }
616  Temp = band->energy > 0 ? sqrtf((float)band_sizes[g] / band->energy) : 0;
617  band->thr = band->energy * 0.001258925f;
618  band->nz_lines = form_factor * sqrtf(Temp);
619 
620  start += band_sizes[g];
621  wstart += band_sizes[g];
622  }
623  }
624 }
625 #endif /* calc_thr_3gpp */
626 
627 #ifndef psy_hp_filter
628 static void psy_hp_filter(const float *firbuf, float *hpfsmpl, const float *psy_fir_coeffs)
629 {
630  int i, j;
631  for (i = 0; i < AAC_BLOCK_SIZE_LONG; i++) {
632  float sum1, sum2;
633  sum1 = firbuf[i + (PSY_LAME_FIR_LEN - 1) / 2];
634  sum2 = 0.0;
635  for (j = 0; j < ((PSY_LAME_FIR_LEN - 1) / 2) - 1; j += 2) {
636  sum1 += psy_fir_coeffs[j] * (firbuf[i + j] + firbuf[i + PSY_LAME_FIR_LEN - j]);
637  sum2 += psy_fir_coeffs[j + 1] * (firbuf[i + j + 1] + firbuf[i + PSY_LAME_FIR_LEN - j - 1]);
638  }
639  /* NOTE: The LAME psymodel expects it's input in the range -32768 to 32768.
640  * Tuning this for normalized floats would be difficult. */
641  hpfsmpl[i] = (sum1 + sum2) * 32768.0f;
642  }
643 }
644 #endif /* psy_hp_filter */
645 
646 /**
647  * Calculate band thresholds as suggested in 3GPP TS26.403
648  */
650  const float *coefs, const FFPsyWindowInfo *wi)
651 {
653  AacPsyChannel *pch = &pctx->ch[channel];
654  int i, w, g;
655  float desired_bits, desired_pe, delta_pe, reduction= NAN, spread_en[128] = {0};
656  float a = 0.0f, active_lines = 0.0f, norm_fac = 0.0f;
657  float pe = pctx->chan_bitrate > 32000 ? 0.0f : FFMAX(50.0f, 100.0f - pctx->chan_bitrate * 100.0f / 32000.0f);
658  const int num_bands = ctx->num_bands[wi->num_windows == 8];
659  const uint8_t *band_sizes = ctx->bands[wi->num_windows == 8];
660  AacPsyCoeffs *coeffs = pctx->psy_coef[wi->num_windows == 8];
661  const float avoid_hole_thr = wi->num_windows == 8 ? PSY_3GPP_AH_THR_SHORT : PSY_3GPP_AH_THR_LONG;
662  const int bandwidth = ctx->cutoff ? ctx->cutoff : AAC_CUTOFF(ctx->avctx);
663  const int cutoff = bandwidth * 2048 / wi->num_windows / ctx->avctx->sample_rate;
664 
665  //calculate energies, initial thresholds and related values - 5.4.2 "Threshold Calculation"
666  calc_thr_3gpp(wi, num_bands, pch, band_sizes, coefs, cutoff);
667 
668  //modify thresholds and energies - spread, threshold in quiet, pre-echo control
669  for (w = 0; w < wi->num_windows*16; w += 16) {
670  AacPsyBand *bands = &pch->band[w];
671 
672  /* 5.4.2.3 "Spreading" & 5.4.3 "Spread Energy Calculation" */
673  spread_en[0] = bands[0].energy;
674  for (g = 1; g < num_bands; g++) {
675  bands[g].thr = FFMAX(bands[g].thr, bands[g-1].thr * coeffs[g].spread_hi[0]);
676  spread_en[w+g] = FFMAX(bands[g].energy, spread_en[w+g-1] * coeffs[g].spread_hi[1]);
677  }
678  for (g = num_bands - 2; g >= 0; g--) {
679  bands[g].thr = FFMAX(bands[g].thr, bands[g+1].thr * coeffs[g].spread_low[0]);
680  spread_en[w+g] = FFMAX(spread_en[w+g], spread_en[w+g+1] * coeffs[g].spread_low[1]);
681  }
682  //5.4.2.4 "Threshold in quiet"
683  for (g = 0; g < num_bands; g++) {
684  AacPsyBand *band = &bands[g];
685 
686  band->thr_quiet = band->thr = FFMAX(band->thr, coeffs[g].ath);
687  //5.4.2.5 "Pre-echo control"
688  if (!(wi->window_type[0] == LONG_STOP_SEQUENCE || (!w && wi->window_type[1] == LONG_START_SEQUENCE)))
689  band->thr = FFMAX(PSY_3GPP_RPEMIN*band->thr, FFMIN(band->thr,
690  PSY_3GPP_RPELEV*pch->prev_band[w+g].thr_quiet));
691 
692  /* 5.6.1.3.1 "Preparatory steps of the perceptual entropy calculation" */
693  pe += calc_pe_3gpp(band);
694  a += band->pe_const;
695  active_lines += band->active_lines;
696 
697  /* 5.6.1.3.3 "Selection of the bands for avoidance of holes" */
698  if (spread_en[w+g] * avoid_hole_thr > band->energy || coeffs[g].min_snr > 1.0f)
700  else
702  }
703  }
704 
705  /* 5.6.1.3.2 "Calculation of the desired perceptual entropy" */
706  ctx->ch[channel].entropy = pe;
707  if (ctx->avctx->flags & AV_CODEC_FLAG_QSCALE) {
708  /* (2.5 * 120) achieves almost transparent rate, and we want to give
709  * ample room downwards, so we make that equivalent to QSCALE=2.4
710  */
711  desired_pe = pe * (ctx->avctx->global_quality ? ctx->avctx->global_quality : 120) / (2 * 2.5f * 120.0f);
712  desired_bits = FFMIN(2560, PSY_3GPP_PE_TO_BITS(desired_pe));
713  desired_pe = PSY_3GPP_BITS_TO_PE(desired_bits); // reflect clipping
714 
715  /* PE slope smoothing */
716  if (ctx->bitres.bits > 0) {
717  desired_bits = FFMIN(2560, PSY_3GPP_PE_TO_BITS(desired_pe));
718  desired_pe = PSY_3GPP_BITS_TO_PE(desired_bits); // reflect clipping
719  }
720 
721  pctx->pe.max = FFMAX(pe, pctx->pe.max);
722  pctx->pe.min = FFMIN(pe, pctx->pe.min);
723  } else {
724  desired_bits = calc_bit_demand(pctx, pe, ctx->bitres.bits, ctx->bitres.size, wi->num_windows == 8);
725  desired_pe = PSY_3GPP_BITS_TO_PE(desired_bits);
726 
727  /* NOTE: PE correction is kept simple. During initial testing it had very
728  * little effect on the final bitrate. Probably a good idea to come
729  * back and do more testing later.
730  */
731  if (ctx->bitres.bits > 0)
732  desired_pe *= av_clipf(pctx->pe.previous / PSY_3GPP_BITS_TO_PE(ctx->bitres.bits),
733  0.85f, 1.15f);
734  }
735  pctx->pe.previous = PSY_3GPP_BITS_TO_PE(desired_bits);
736  ctx->bitres.alloc = desired_bits;
737 
738  if (desired_pe < pe) {
739  /* 5.6.1.3.4 "First Estimation of the reduction value" */
740  for (w = 0; w < wi->num_windows*16; w += 16) {
741  reduction = calc_reduction_3gpp(a, desired_pe, pe, active_lines);
742  pe = 0.0f;
743  a = 0.0f;
744  active_lines = 0.0f;
745  for (g = 0; g < num_bands; g++) {
746  AacPsyBand *band = &pch->band[w+g];
747 
748  band->thr = calc_reduced_thr_3gpp(band, coeffs[g].min_snr, reduction);
749  /* recalculate PE */
750  pe += calc_pe_3gpp(band);
751  a += band->pe_const;
752  active_lines += band->active_lines;
753  }
754  }
755 
756  /* 5.6.1.3.5 "Second Estimation of the reduction value" */
757  for (i = 0; i < 2; i++) {
758  float pe_no_ah = 0.0f, desired_pe_no_ah;
759  active_lines = a = 0.0f;
760  for (w = 0; w < wi->num_windows*16; w += 16) {
761  for (g = 0; g < num_bands; g++) {
762  AacPsyBand *band = &pch->band[w+g];
763 
764  if (band->avoid_holes != PSY_3GPP_AH_ACTIVE) {
765  pe_no_ah += band->pe;
766  a += band->pe_const;
767  active_lines += band->active_lines;
768  }
769  }
770  }
771  desired_pe_no_ah = FFMAX(desired_pe - (pe - pe_no_ah), 0.0f);
772  if (active_lines > 0.0f)
773  reduction = calc_reduction_3gpp(a, desired_pe_no_ah, pe_no_ah, active_lines);
774 
775  pe = 0.0f;
776  for (w = 0; w < wi->num_windows*16; w += 16) {
777  for (g = 0; g < num_bands; g++) {
778  AacPsyBand *band = &pch->band[w+g];
779 
780  if (active_lines > 0.0f)
781  band->thr = calc_reduced_thr_3gpp(band, coeffs[g].min_snr, reduction);
782  pe += calc_pe_3gpp(band);
783  if (band->thr > 0.0f)
784  band->norm_fac = band->active_lines / band->thr;
785  else
786  band->norm_fac = 0.0f;
787  norm_fac += band->norm_fac;
788  }
789  }
790  delta_pe = desired_pe - pe;
791  if (fabs(delta_pe) > 0.05f * desired_pe)
792  break;
793  }
794 
795  if (pe < 1.15f * desired_pe) {
796  /* 6.6.1.3.6 "Final threshold modification by linearization" */
797  norm_fac = 1.0f / norm_fac;
798  for (w = 0; w < wi->num_windows*16; w += 16) {
799  for (g = 0; g < num_bands; g++) {
800  AacPsyBand *band = &pch->band[w+g];
801 
802  if (band->active_lines > 0.5f) {
803  float delta_sfb_pe = band->norm_fac * norm_fac * delta_pe;
804  float thr = band->thr;
805 
806  thr *= exp2f(delta_sfb_pe / band->active_lines);
807  if (thr > coeffs[g].min_snr * band->energy && band->avoid_holes == PSY_3GPP_AH_INACTIVE)
808  thr = FFMAX(band->thr, coeffs[g].min_snr * band->energy);
809  band->thr = thr;
810  }
811  }
812  }
813  } else {
814  /* 5.6.1.3.7 "Further perceptual entropy reduction" */
815  g = num_bands;
816  while (pe > desired_pe && g--) {
817  for (w = 0; w < wi->num_windows*16; w+= 16) {
818  AacPsyBand *band = &pch->band[w+g];
819  if (band->avoid_holes != PSY_3GPP_AH_NONE && coeffs[g].min_snr < PSY_SNR_1DB) {
820  coeffs[g].min_snr = PSY_SNR_1DB;
821  band->thr = band->energy * PSY_SNR_1DB;
822  pe += band->active_lines * 1.5f - band->pe;
823  }
824  }
825  }
826  /* TODO: allow more holes (unused without mid/side) */
827  }
828  }
829 
830  for (w = 0; w < wi->num_windows*16; w += 16) {
831  for (g = 0; g < num_bands; g++) {
832  AacPsyBand *band = &pch->band[w+g];
833  FFPsyBand *psy_band = &ctx->ch[channel].psy_bands[w+g];
834 
835  psy_band->threshold = band->thr;
836  psy_band->energy = band->energy;
837  psy_band->spread = band->active_lines * 2.0f / band_sizes[g];
838  psy_band->bits = PSY_3GPP_PE_TO_BITS(band->pe);
839  }
840  }
841 
842  memcpy(pch->prev_band, pch->band, sizeof(pch->band));
843 }
844 
846  const float **coeffs, const FFPsyWindowInfo *wi)
847 {
848  int ch;
849  FFPsyChannelGroup *group = ff_psy_find_group(ctx, channel);
850 
851  for (ch = 0; ch < group->num_ch; ch++)
852  psy_3gpp_analyze_channel(ctx, channel + ch, coeffs[ch], &wi[ch]);
853 }
854 
856 {
858  av_freep(&pctx->ch);
859  av_freep(&apc->model_priv_data);
860 }
861 
862 static void lame_apply_block_type(AacPsyChannel *ctx, FFPsyWindowInfo *wi, int uselongblock)
863 {
864  int blocktype = ONLY_LONG_SEQUENCE;
865  if (uselongblock) {
867  blocktype = LONG_STOP_SEQUENCE;
868  } else {
869  blocktype = EIGHT_SHORT_SEQUENCE;
874  }
875 
876  wi->window_type[0] = ctx->next_window_seq;
877  ctx->next_window_seq = blocktype;
878 }
879 
880 static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio,
881  const float *la, int channel, int prev_type)
882 {
884  AacPsyChannel *pch = &pctx->ch[channel];
885  int grouping = 0;
886  int uselongblock = 1;
887  int attacks[AAC_NUM_BLOCKS_SHORT + 1] = { 0 };
888  int i;
889  FFPsyWindowInfo wi = { { 0 } };
890 
891  if (la) {
892  float hpfsmpl[AAC_BLOCK_SIZE_LONG];
893  const float *pf = hpfsmpl;
894  float attack_intensity[(AAC_NUM_BLOCKS_SHORT + 1) * PSY_LAME_NUM_SUBBLOCKS];
895  float energy_subshort[(AAC_NUM_BLOCKS_SHORT + 1) * PSY_LAME_NUM_SUBBLOCKS];
896  float energy_short[AAC_NUM_BLOCKS_SHORT + 1] = { 0 };
897  const float *firbuf = la + (AAC_BLOCK_SIZE_SHORT/4 - PSY_LAME_FIR_LEN);
898  int att_sum = 0;
899 
900  /* LAME comment: apply high pass filter of fs/4 */
901  psy_hp_filter(firbuf, hpfsmpl, psy_fir_coeffs);
902 
903  /* Calculate the energies of each sub-shortblock */
904  for (i = 0; i < PSY_LAME_NUM_SUBBLOCKS; i++) {
905  energy_subshort[i] = pch->prev_energy_subshort[i + ((AAC_NUM_BLOCKS_SHORT - 1) * PSY_LAME_NUM_SUBBLOCKS)];
906  assert(pch->prev_energy_subshort[i + ((AAC_NUM_BLOCKS_SHORT - 2) * PSY_LAME_NUM_SUBBLOCKS + 1)] > 0);
907  attack_intensity[i] = energy_subshort[i] / pch->prev_energy_subshort[i + ((AAC_NUM_BLOCKS_SHORT - 2) * PSY_LAME_NUM_SUBBLOCKS + 1)];
908  energy_short[0] += energy_subshort[i];
909  }
910 
911  for (i = 0; i < AAC_NUM_BLOCKS_SHORT * PSY_LAME_NUM_SUBBLOCKS; i++) {
912  const float *const pfe = pf + AAC_BLOCK_SIZE_LONG / (AAC_NUM_BLOCKS_SHORT * PSY_LAME_NUM_SUBBLOCKS);
913  float p = 1.0f;
914  for (; pf < pfe; pf++)
915  p = FFMAX(p, fabsf(*pf));
916  pch->prev_energy_subshort[i] = energy_subshort[i + PSY_LAME_NUM_SUBBLOCKS] = p;
917  energy_short[1 + i / PSY_LAME_NUM_SUBBLOCKS] += p;
918  /* NOTE: The indexes below are [i + 3 - 2] in the LAME source.
919  * Obviously the 3 and 2 have some significance, or this would be just [i + 1]
920  * (which is what we use here). What the 3 stands for is ambiguous, as it is both
921  * number of short blocks, and the number of sub-short blocks.
922  * It seems that LAME is comparing each sub-block to sub-block + 1 in the
923  * previous block.
924  */
925  if (p > energy_subshort[i + 1])
926  p = p / energy_subshort[i + 1];
927  else if (energy_subshort[i + 1] > p * 10.0f)
928  p = energy_subshort[i + 1] / (p * 10.0f);
929  else
930  p = 0.0;
931  attack_intensity[i + PSY_LAME_NUM_SUBBLOCKS] = p;
932  }
933 
934  /* compare energy between sub-short blocks */
935  for (i = 0; i < (AAC_NUM_BLOCKS_SHORT + 1) * PSY_LAME_NUM_SUBBLOCKS; i++)
936  if (!attacks[i / PSY_LAME_NUM_SUBBLOCKS])
937  if (attack_intensity[i] > pch->attack_threshold)
938  attacks[i / PSY_LAME_NUM_SUBBLOCKS] = (i % PSY_LAME_NUM_SUBBLOCKS) + 1;
939 
940  /* should have energy change between short blocks, in order to avoid periodic signals */
941  /* Good samples to show the effect are Trumpet test songs */
942  /* GB: tuned (1) to avoid too many short blocks for test sample TRUMPET */
943  /* RH: tuned (2) to let enough short blocks through for test sample FSOL and SNAPS */
944  for (i = 1; i < AAC_NUM_BLOCKS_SHORT + 1; i++) {
945  const float u = energy_short[i - 1];
946  const float v = energy_short[i];
947  const float m = FFMAX(u, v);
948  if (m < 40000) { /* (2) */
949  if (u < 1.7f * v && v < 1.7f * u) { /* (1) */
950  if (i == 1 && attacks[0] < attacks[i])
951  attacks[0] = 0;
952  attacks[i] = 0;
953  }
954  }
955  att_sum += attacks[i];
956  }
957 
958  if (attacks[0] <= pch->prev_attack)
959  attacks[0] = 0;
960 
961  att_sum += attacks[0];
962  /* 3 below indicates the previous attack happened in the last sub-block of the previous sequence */
963  if (pch->prev_attack == 3 || att_sum) {
964  uselongblock = 0;
965 
966  for (i = 1; i < AAC_NUM_BLOCKS_SHORT + 1; i++)
967  if (attacks[i] && attacks[i-1])
968  attacks[i] = 0;
969  }
970  } else {
971  /* We have no lookahead info, so just use same type as the previous sequence. */
972  uselongblock = !(prev_type == EIGHT_SHORT_SEQUENCE);
973  }
974 
975  lame_apply_block_type(pch, &wi, uselongblock);
976 
977  wi.window_type[1] = prev_type;
978  if (wi.window_type[0] != EIGHT_SHORT_SEQUENCE) {
979 
980  wi.num_windows = 1;
981  wi.grouping[0] = 1;
982  if (wi.window_type[0] == LONG_START_SEQUENCE)
983  wi.window_shape = 0;
984  else
985  wi.window_shape = 1;
986 
987  } else {
988  int lastgrp = 0;
989 
990  wi.num_windows = 8;
991  wi.window_shape = 0;
992  for (i = 0; i < 8; i++) {
993  if (!((pch->next_grouping >> i) & 1))
994  lastgrp = i;
995  wi.grouping[lastgrp]++;
996  }
997  }
998 
999  /* Determine grouping, based on the location of the first attack, and save for
1000  * the next frame.
1001  * FIXME: Move this to analysis.
1002  * TODO: Tune groupings depending on attack location
1003  * TODO: Handle more than one attack in a group
1004  */
1005  for (i = 0; i < 9; i++) {
1006  if (attacks[i]) {
1007  grouping = i;
1008  break;
1009  }
1010  }
1011  pch->next_grouping = window_grouping[grouping];
1012 
1013  pch->prev_attack = attacks[8];
1014 
1015  return wi;
1016 }
1017 
1019 {
1020  .name = "3GPP TS 26.403-inspired model",
1021  .init = psy_3gpp_init,
1022  .window = psy_lame_window,
1023  .analyze = psy_3gpp_analyze,
1024  .end = psy_3gpp_end,
1025 };
int quality
Quality to map the rest of the vaules to.
Definition: aacpsy.c:170
float global_quality
normalized global quality taken from avctx
Definition: aacpsy.c:163
const char * s
Definition: avisynth_c.h:768
int size
static const uint8_t window_grouping[9]
window grouping information stored as bits (0 - new group, 1 - group continues)
Definition: aacpsy.c:397
int grouping[8]
window grouping (for e.g. AAC)
Definition: psymodel.h:81
#define AAC_BLOCK_SIZE_SHORT
short block size
Definition: aacpsy.c:97
static int calc_bit_demand(AacPsyContext *ctx, float pe, int bits, int size, int short_window)
Definition: aacpsy.c:491
uint8_t ** bands
scalefactor band sizes for possible frame sizes
Definition: psymodel.h:98
#define PSY_3GPP_AH_THR_SHORT
Definition: aacpsy.c:81
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1568
const char * g
Definition: vf_curves.c:112
static const PsyLamePreset psy_vbr_map[]
LAME psy model preset table for constant quality.
Definition: aacpsy.c:201
psychoacoustic information for an arbitrary group of channels
Definition: psymodel.h:68
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi - 0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(INT64_C(1)<< 63))) #define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={ FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64), };static void cpy1(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, len);} static void cpy2(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 2 *len);} static void cpy4(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 4 *len);} static void cpy8(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 8 *len);} AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags) { AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;if(channels==1){ in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);} ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map) { switch(av_get_bytes_per_sample(in_fmt)){ case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;} } if(HAVE_X86ASM &&HAVE_MMX) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);return ctx;} void swri_audio_convert_free(AudioConvert **ctx) { av_freep(ctx);} int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len) { int ch;int off=0;const int os=(out->planar ? 1 :out->ch_count) *out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask) { int planes=in->planar ? in->ch_count :1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;} if(ctx->out_simd_align_mask) { int planes=out->planar ? out->ch_count :1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;} if(ctx->simd_f &&!ctx->ch_map &&!misaligned){ off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){ if(out->planar==in->planar){ int planes=out->planar ? out->ch_count :1;for(ch=0;ch< planes;ch++){ ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
static float calc_reduction_3gpp(float a, float desired_pe, float pe, float active_lines)
Definition: aacpsy.c:557
float ath
absolute threshold of hearing per bands
Definition: aacpsy.c:141
#define PSY_3GPP_EN_SPREAD_HI_L1
Definition: aacpsy.c:47
static av_cold float ath(float f, float add)
Calculate ATH value for given frequency.
Definition: aacpsy.c:292
float prev_energy_subshort[AAC_NUM_BLOCKS_SHORT *PSY_LAME_NUM_SUBBLOCKS]
Definition: aacpsy.c:133
enum WindowSequence next_window_seq
window sequence to be used in the next frame
Definition: aacpsy.c:130
#define PSY_SNR_25DB
Definition: aacpsy.c:65
#define AAC_BLOCK_SIZE_LONG
long block size
Definition: aacpsy.c:96
int * num_bands
number of scalefactor bands for possible frame sizes
Definition: psymodel.h:99
Macro definitions for various function/variable attributes.
LAME psy model preset struct.
Definition: aacpsy.c:169
float thr
energy threshold
Definition: aacpsy.c:110
float correction
PE correction factor.
Definition: aacpsy.c:159
static av_cold void psy_3gpp_end(FFPsyContext *apc)
Definition: aacpsy.c:855
float attack_threshold
attack threshold for this channel
Definition: aacpsy.c:132
#define PSY_3GPP_EN_SPREAD_LOW_L
Definition: aacpsy.c:53
float nz_lines
number of non-zero spectral lines
Definition: aacpsy.c:112
uint8_t
psychoacoustic model frame type-dependent coefficients
Definition: aacpsy.c:140
#define av_cold
Definition: attributes.h:82
int size
size of the bitresevoir in bits
Definition: psymodel.h:103
static float calc_reduced_thr_3gpp(AacPsyBand *band, float min_snr, float reduction)
Definition: aacpsy.c:571
#define PSY_3GPP_C2
Definition: aacpsy.c:61
#define PSY_LAME_FIR_LEN
LAME psy model FIR order.
Definition: aacpsy.c:95
#define PSY_3GPP_CLIP_LO_L
Definition: aacpsy.c:75
#define PSY_3GPP_SPEND_SLOPE_S
Definition: aacpsy.c:72
#define u(width, name, range_min, range_max)
Definition: cbs_h2645.c:344
#define PSY_3GPP_THR_SPREAD_LOW
Definition: aacpsy.c:45
context used by psychoacoustic model
Definition: psymodel.h:89
#define atanf(x)
Definition: libm.h:40
int flags
Flags modifying the (de)muxer behaviour.
Definition: avformat.h:1473
#define AAC_CUTOFF(s)
Definition: psymodel.h:41
single band psychoacoustic information
Definition: psymodel.h:50
static float lame_calc_attack_threshold(int bitrate)
Calculate the ABR attack threshold from the above LAME psymodel table.
Definition: aacpsy.c:232
uint8_t next_grouping
stored grouping scheme for the next frame (in case of 8 short window sequence)
Definition: aacpsy.c:129
#define PSY_3GPP_SAVE_ADD_L
Definition: aacpsy.c:69
static av_cold float calc_bark(float f)
Calculate Bark value for given line.
Definition: aacpsy.c:282
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
Definition: ffmath.h:42
#define AVERROR(e)
Definition: error.h:43
#define PSY_3GPP_SPEND_ADD_S
Definition: aacpsy.c:74
#define PSY_SNR_1DB
Definition: aacpsy.c:64
3GPP TS26.403-inspired psychoacoustic model specific data
Definition: aacpsy.c:151
single/pair channel context for psychoacoustic model
Definition: aacpsy.c:123
static const float psy_fir_coeffs[]
LAME psy model FIR coefficient table.
Definition: aacpsy.c:219
int bits
Definition: psymodel.h:51
float barks
Bark value for each spectral band in long frame.
Definition: aacpsy.c:142
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:1598
float pe_const
constant part of the PE calculation
Definition: aacpsy.c:115
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:236
int num_windows
number of windows in a frame
Definition: psymodel.h:80
static FFPsyWindowInfo psy_lame_window(FFPsyContext *ctx, const float *audio, const float *la, int channel, int prev_type)
Definition: aacpsy.c:880
#define PSY_3GPP_SPEND_SLOPE_L
Definition: aacpsy.c:71
#define PSY_3GPP_THR_SPREAD_HI
constants for 3GPP AAC psychoacoustic model
Definition: aacpsy.c:44
float energy
Definition: psymodel.h:52
WindowSequence
Definition: aac.h:75
#define FFMAX(a, b)
Definition: common.h:94
codec-specific psychoacoustic model implementation
Definition: psymodel.h:114
#define PSY_3GPP_RPELEV
Definition: aacpsy.c:58
int8_t exp
Definition: eval.c:72
struct AacPsyContext::@31 pe
float thr_quiet
threshold in quiet
Definition: aacpsy.c:111
static void psy_3gpp_analyze(FFPsyContext *ctx, int channel, const float **coeffs, const FFPsyWindowInfo *wi)
Definition: aacpsy.c:845
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:833
#define NAN
Definition: mathematics.h:64
#define FFMIN(a, b)
Definition: common.h:96
int prev_attack
attack value for the last short block in the previous sequence
Definition: aacpsy.c:134
#define PSY_3GPP_SAVE_SLOPE_S
Definition: aacpsy.c:68
#define PSY_3GPP_C3
Definition: aacpsy.c:62
uint8_t w
Definition: llviddspenc.c:38
uint8_t num_ch
number of channels in this group
Definition: psymodel.h:70
int frame_bits
average bits per frame
Definition: aacpsy.c:153
int fill_level
bit reservoir fill level
Definition: aacpsy.c:154
AVFormatContext * ctx
Definition: movenc.c:48
static void lame_apply_block_type(AacPsyChannel *ctx, FFPsyWindowInfo *wi, int uselongblock)
Definition: aacpsy.c:862
#define PSY_3GPP_SAVE_SLOPE_L
Definition: aacpsy.c:67
Reference: libavcodec/aacpsy.c.
#define PSY_LAME_NUM_SUBBLOCKS
Number of sub-blocks in each short block.
Definition: aacpsy.c:99
#define ATH_ADD
Definition: aacpsy.c:287
float energy
band energy
Definition: aacpsy.c:109
static struct @271 state
const FFPsyModel ff_aac_psy_model
Definition: aacpsy.c:1018
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.
Definition: aacpsy.c:649
float st_lrm
short threshold for L, R, and M channels
Definition: aacpsy.c:174
#define PSY_3GPP_EN_SPREAD_LOW_S
Definition: aacpsy.c:55
#define exp2f(x)
Definition: libm.h:293
Libavcodec external API header.
struct FFPsyContext::@106 bitres
int sample_rate
samples per second
Definition: avcodec.h:2173
FFPsyChannelGroup * ff_psy_find_group(FFPsyContext *ctx, int channel)
Determine what group a channel belongs to.
Definition: psymodel.c:73
main external API structure.
Definition: avcodec.h:1518
float win_energy
sliding average of channel energy
Definition: aacpsy.c:127
void * model_priv_data
psychoacoustic model implementation private data
Definition: psymodel.h:108
float active_lines
number of active spectral lines
Definition: aacpsy.c:113
static const float bands[]
static float iir_filter(int in, float state[2])
IIR filter used in block switching decision.
Definition: aacpsy.c:384
int avoid_holes
hole avoidance flag
Definition: aacpsy.c:117
AacPsyBand band[128]
bands information
Definition: aacpsy.c:124
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)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
#define PSY_3GPP_CLIP_HI_S
Definition: aacpsy.c:78
#define PSY_3GPP_RPEMIN
Definition: aacpsy.c:57
static const PsyLamePreset psy_abr_map[]
LAME psy model preset table for ABR.
Definition: aacpsy.c:180
int window_shape
window shape (sine/KBD/whatever)
Definition: psymodel.h:79
#define PSY_PE_FORGET_SLOPE
Definition: aacpsy.c:83
#define PSY_3GPP_PE_TO_BITS(bits)
Definition: aacpsy.c:92
int cutoff
lowpass frequency cutoff for analysis
Definition: psymodel.h:96
float min_snr
minimal SNR
Definition: aacpsy.c:145
float max
maximum allowed PE for bit factor calculation
Definition: aacpsy.c:157
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)
Definition: aacpsy.c:597
float previous
allowed PE of the previous frame
Definition: aacpsy.c:158
AacPsyCoeffs psy_coef[2][64]
Definition: aacpsy.c:161
float min
minimum allowed PE for bit factor calculation
Definition: aacpsy.c:156
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:1584
static av_cold int psy_3gpp_init(FFPsyContext *ctx)
Definition: aacpsy.c:301
static void psy_hp_filter(const float *firbuf, float *hpfsmpl, const float *psy_fir_coeffs)
Definition: aacpsy.c:628
float spread_hi[2]
spreading factor for high-to-low threshold spreading in long frame
Definition: aacpsy.c:144
const char * name
Definition: psymodel.h:115
internal math functions header
static av_unused FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx, const int16_t *audio, const int16_t *la, int channel, int prev_type)
Tell encoder which window types to use.
Definition: aacpsy.c:405
int
static float calc_pe_3gpp(AacPsyBand *band)
Definition: aacpsy.c:534
#define exp2(x)
Definition: libm.h:288
windowing related information
Definition: psymodel.h:77
#define log2f(x)
Definition: libm.h:409
channel
Use these values when setting the channel map with ebur128_set_channel().
Definition: ebur128.h:39
#define PSY_3GPP_BITS_TO_PE(bits)
Definition: aacpsy.c:91
#define PSY_3GPP_C1
Definition: aacpsy.c:60
float norm_fac
normalization factor for linearization
Definition: aacpsy.c:116
int chan_bitrate
bitrate per channel
Definition: aacpsy.c:152
#define PSY_3GPP_CLIP_LO_S
Definition: aacpsy.c:76
#define PSY_3GPP_AH_THR_LONG
Definition: aacpsy.c:80
static const int16_t coeffs[]
int channels
number of audio channels
Definition: avcodec.h:2174
float pe
perceptual entropy
Definition: aacpsy.c:114
#define PSY_3GPP_EN_SPREAD_HI_S
Definition: aacpsy.c:51
static const double coeff[2][5]
Definition: vf_owdenoise.c:72
#define FF_QP2LAMBDA
factor to convert from H.263 QP to lambda
Definition: avutil.h:227
AacPsyChannel * ch
Definition: aacpsy.c:162
#define PSY_3GPP_SAVE_ADD_S
Definition: aacpsy.c:70
#define av_freep(p)
void INT64 start
Definition: avisynth_c.h:690
information for single band used by 3GPP TS26.403-inspired psychoacoustic model
Definition: aacpsy.c:108
AVCodecContext * avctx
encoder context
Definition: psymodel.h:90
float threshold
Definition: psymodel.h:53
AAC data declarations.
float spread_low[2]
spreading factor for low-to-high threshold spreading in long frame
Definition: aacpsy.c:143
#define PSY_3GPP_CLIP_HI_L
Definition: aacpsy.c:77
float spread
Definition: psymodel.h:54
int window_type[3]
window type (short/long/transitional, etc.) - current, previous and next
Definition: psymodel.h:78
#define AAC_NUM_BLOCKS_SHORT
number of blocks in a short sequence
Definition: aacpsy.c:98
#define av_unused
Definition: attributes.h:125
#define PSY_3GPP_SPEND_ADD_L
Definition: aacpsy.c:73
void * av_mallocz_array(size_t nmemb, size_t size)
Allocate a memory block for an array with av_mallocz().
Definition: mem.c:191
static av_cold void lame_window_init(AacPsyContext *ctx, AVCodecContext *avctx)
LAME psy model specific initialization.
Definition: aacpsy.c:262