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aacsbr.c
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1 /*
2  * AAC Spectral Band Replication decoding functions
3  * Copyright (c) 2008-2009 Robert Swain ( rob opendot cl )
4  * Copyright (c) 2009-2010 Alex Converse <alex.converse@gmail.com>
5  *
6  * This file is part of FFmpeg.
7  *
8  * FFmpeg is free software; you can redistribute it and/or
9  * modify it under the terms of the GNU Lesser General Public
10  * License as published by the Free Software Foundation; either
11  * version 2.1 of the License, or (at your option) any later version.
12  *
13  * FFmpeg is distributed in the hope that it will be useful,
14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16  * Lesser General Public License for more details.
17  *
18  * You should have received a copy of the GNU Lesser General Public
19  * License along with FFmpeg; if not, write to the Free Software
20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21  */
22 
23 /**
24  * @file
25  * AAC Spectral Band Replication decoding functions
26  * @author Robert Swain ( rob opendot cl )
27  */
28 #define USE_FIXED 0
29 
30 #include "aac.h"
31 #include "sbr.h"
32 #include "aacsbr.h"
33 #include "aacsbrdata.h"
34 #include "aacsbr_tablegen.h"
35 #include "fft.h"
36 #include "aacps.h"
37 #include "sbrdsp.h"
38 #include "libavutil/internal.h"
39 #include "libavutil/libm.h"
40 #include "libavutil/avassert.h"
41 
42 #include <stdint.h>
43 #include <float.h>
44 #include <math.h>
45 
46 #if ARCH_MIPS
47 #include "mips/aacsbr_mips.h"
48 #endif /* ARCH_MIPS */
49 
50 static VLC vlc_sbr[10];
52 
53 static void make_bands(int16_t* bands, int start, int stop, int num_bands)
54 {
55  int k, previous, present;
56  float base, prod;
57 
58  base = powf((float)stop / start, 1.0f / num_bands);
59  prod = start;
60  previous = start;
61 
62  for (k = 0; k < num_bands-1; k++) {
63  prod *= base;
64  present = lrintf(prod);
65  bands[k] = present - previous;
66  previous = present;
67  }
68  bands[num_bands-1] = stop - previous;
69 }
70 
71 /// Dequantization and stereo decoding (14496-3 sp04 p203)
72 static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
73 {
74  int k, e;
75  int ch;
76 
77  if (id_aac == TYPE_CPE && sbr->bs_coupling) {
78  float alpha = sbr->data[0].bs_amp_res ? 1.0f : 0.5f;
79  float pan_offset = sbr->data[0].bs_amp_res ? 12.0f : 24.0f;
80  for (e = 1; e <= sbr->data[0].bs_num_env; e++) {
81  for (k = 0; k < sbr->n[sbr->data[0].bs_freq_res[e]]; k++) {
82  float temp1 = exp2f(sbr->data[0].env_facs[e][k] * alpha + 7.0f);
83  float temp2 = exp2f((pan_offset - sbr->data[1].env_facs[e][k]) * alpha);
84  float fac;
85  if (temp1 > 1E20) {
86  av_log(NULL, AV_LOG_ERROR, "envelope scalefactor overflow in dequant\n");
87  temp1 = 1;
88  }
89  fac = temp1 / (1.0f + temp2);
90  sbr->data[0].env_facs[e][k] = fac;
91  sbr->data[1].env_facs[e][k] = fac * temp2;
92  }
93  }
94  for (e = 1; e <= sbr->data[0].bs_num_noise; e++) {
95  for (k = 0; k < sbr->n_q; k++) {
96  float temp1 = exp2f(NOISE_FLOOR_OFFSET - sbr->data[0].noise_facs[e][k] + 1);
97  float temp2 = exp2f(12 - sbr->data[1].noise_facs[e][k]);
98  float fac;
99  if (temp1 > 1E20) {
100  av_log(NULL, AV_LOG_ERROR, "envelope scalefactor overflow in dequant\n");
101  temp1 = 1;
102  }
103  fac = temp1 / (1.0f + temp2);
104  sbr->data[0].noise_facs[e][k] = fac;
105  sbr->data[1].noise_facs[e][k] = fac * temp2;
106  }
107  }
108  } else { // SCE or one non-coupled CPE
109  for (ch = 0; ch < (id_aac == TYPE_CPE) + 1; ch++) {
110  float alpha = sbr->data[ch].bs_amp_res ? 1.0f : 0.5f;
111  for (e = 1; e <= sbr->data[ch].bs_num_env; e++)
112  for (k = 0; k < sbr->n[sbr->data[ch].bs_freq_res[e]]; k++){
113  sbr->data[ch].env_facs[e][k] =
114  exp2f(alpha * sbr->data[ch].env_facs[e][k] + 6.0f);
115  if (sbr->data[ch].env_facs[e][k] > 1E20) {
116  av_log(NULL, AV_LOG_ERROR, "envelope scalefactor overflow in dequant\n");
117  sbr->data[ch].env_facs[e][k] = 1;
118  }
119  }
120 
121  for (e = 1; e <= sbr->data[ch].bs_num_noise; e++)
122  for (k = 0; k < sbr->n_q; k++)
123  sbr->data[ch].noise_facs[e][k] =
124  exp2f(NOISE_FLOOR_OFFSET - sbr->data[ch].noise_facs[e][k]);
125  }
126  }
127 }
128 
129 /** High Frequency Generation (14496-3 sp04 p214+) and Inverse Filtering
130  * (14496-3 sp04 p214)
131  * Warning: This routine does not seem numerically stable.
132  */
134  float (*alpha0)[2], float (*alpha1)[2],
135  const float X_low[32][40][2], int k0)
136 {
137  int k;
138  for (k = 0; k < k0; k++) {
139  LOCAL_ALIGNED_16(float, phi, [3], [2][2]);
140  float dk;
141 
142  dsp->autocorrelate(X_low[k], phi);
143 
144  dk = phi[2][1][0] * phi[1][0][0] -
145  (phi[1][1][0] * phi[1][1][0] + phi[1][1][1] * phi[1][1][1]) / 1.000001f;
146 
147  if (!dk) {
148  alpha1[k][0] = 0;
149  alpha1[k][1] = 0;
150  } else {
151  float temp_real, temp_im;
152  temp_real = phi[0][0][0] * phi[1][1][0] -
153  phi[0][0][1] * phi[1][1][1] -
154  phi[0][1][0] * phi[1][0][0];
155  temp_im = phi[0][0][0] * phi[1][1][1] +
156  phi[0][0][1] * phi[1][1][0] -
157  phi[0][1][1] * phi[1][0][0];
158 
159  alpha1[k][0] = temp_real / dk;
160  alpha1[k][1] = temp_im / dk;
161  }
162 
163  if (!phi[1][0][0]) {
164  alpha0[k][0] = 0;
165  alpha0[k][1] = 0;
166  } else {
167  float temp_real, temp_im;
168  temp_real = phi[0][0][0] + alpha1[k][0] * phi[1][1][0] +
169  alpha1[k][1] * phi[1][1][1];
170  temp_im = phi[0][0][1] + alpha1[k][1] * phi[1][1][0] -
171  alpha1[k][0] * phi[1][1][1];
172 
173  alpha0[k][0] = -temp_real / phi[1][0][0];
174  alpha0[k][1] = -temp_im / phi[1][0][0];
175  }
176 
177  if (alpha1[k][0] * alpha1[k][0] + alpha1[k][1] * alpha1[k][1] >= 16.0f ||
178  alpha0[k][0] * alpha0[k][0] + alpha0[k][1] * alpha0[k][1] >= 16.0f) {
179  alpha1[k][0] = 0;
180  alpha1[k][1] = 0;
181  alpha0[k][0] = 0;
182  alpha0[k][1] = 0;
183  }
184  }
185 }
186 
187 /// Chirp Factors (14496-3 sp04 p214)
188 static void sbr_chirp(SpectralBandReplication *sbr, SBRData *ch_data)
189 {
190  int i;
191  float new_bw;
192  static const float bw_tab[] = { 0.0f, 0.75f, 0.9f, 0.98f };
193 
194  for (i = 0; i < sbr->n_q; i++) {
195  if (ch_data->bs_invf_mode[0][i] + ch_data->bs_invf_mode[1][i] == 1) {
196  new_bw = 0.6f;
197  } else
198  new_bw = bw_tab[ch_data->bs_invf_mode[0][i]];
199 
200  if (new_bw < ch_data->bw_array[i]) {
201  new_bw = 0.75f * new_bw + 0.25f * ch_data->bw_array[i];
202  } else
203  new_bw = 0.90625f * new_bw + 0.09375f * ch_data->bw_array[i];
204  ch_data->bw_array[i] = new_bw < 0.015625f ? 0.0f : new_bw;
205  }
206 }
207 
208 /**
209  * Calculation of levels of additional HF signal components (14496-3 sp04 p219)
210  * and Calculation of gain (14496-3 sp04 p219)
211  */
213  SBRData *ch_data, const int e_a[2])
214 {
215  int e, k, m;
216  // max gain limits : -3dB, 0dB, 3dB, inf dB (limiter off)
217  static const float limgain[4] = { 0.70795, 1.0, 1.41254, 10000000000 };
218 
219  for (e = 0; e < ch_data->bs_num_env; e++) {
220  int delta = !((e == e_a[1]) || (e == e_a[0]));
221  for (k = 0; k < sbr->n_lim; k++) {
222  float gain_boost, gain_max;
223  float sum[2] = { 0.0f, 0.0f };
224  for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
225  const float temp = sbr->e_origmapped[e][m] / (1.0f + sbr->q_mapped[e][m]);
226  sbr->q_m[e][m] = sqrtf(temp * sbr->q_mapped[e][m]);
227  sbr->s_m[e][m] = sqrtf(temp * ch_data->s_indexmapped[e + 1][m]);
228  if (!sbr->s_mapped[e][m]) {
229  sbr->gain[e][m] = sqrtf(sbr->e_origmapped[e][m] /
230  ((1.0f + sbr->e_curr[e][m]) *
231  (1.0f + sbr->q_mapped[e][m] * delta)));
232  } else {
233  sbr->gain[e][m] = sqrtf(sbr->e_origmapped[e][m] * sbr->q_mapped[e][m] /
234  ((1.0f + sbr->e_curr[e][m]) *
235  (1.0f + sbr->q_mapped[e][m])));
236  }
237  }
238  for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
239  sum[0] += sbr->e_origmapped[e][m];
240  sum[1] += sbr->e_curr[e][m];
241  }
242  gain_max = limgain[sbr->bs_limiter_gains] * sqrtf((FLT_EPSILON + sum[0]) / (FLT_EPSILON + sum[1]));
243  gain_max = FFMIN(100000.f, gain_max);
244  for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
245  float q_m_max = sbr->q_m[e][m] * gain_max / sbr->gain[e][m];
246  sbr->q_m[e][m] = FFMIN(sbr->q_m[e][m], q_m_max);
247  sbr->gain[e][m] = FFMIN(sbr->gain[e][m], gain_max);
248  }
249  sum[0] = sum[1] = 0.0f;
250  for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
251  sum[0] += sbr->e_origmapped[e][m];
252  sum[1] += sbr->e_curr[e][m] * sbr->gain[e][m] * sbr->gain[e][m]
253  + sbr->s_m[e][m] * sbr->s_m[e][m]
254  + (delta && !sbr->s_m[e][m]) * sbr->q_m[e][m] * sbr->q_m[e][m];
255  }
256  gain_boost = sqrtf((FLT_EPSILON + sum[0]) / (FLT_EPSILON + sum[1]));
257  gain_boost = FFMIN(1.584893192f, gain_boost);
258  for (m = sbr->f_tablelim[k] - sbr->kx[1]; m < sbr->f_tablelim[k + 1] - sbr->kx[1]; m++) {
259  sbr->gain[e][m] *= gain_boost;
260  sbr->q_m[e][m] *= gain_boost;
261  sbr->s_m[e][m] *= gain_boost;
262  }
263  }
264  }
265 }
266 
267 /// Assembling HF Signals (14496-3 sp04 p220)
268 static void sbr_hf_assemble(float Y1[38][64][2],
269  const float X_high[64][40][2],
270  SpectralBandReplication *sbr, SBRData *ch_data,
271  const int e_a[2])
272 {
273  int e, i, j, m;
274  const int h_SL = 4 * !sbr->bs_smoothing_mode;
275  const int kx = sbr->kx[1];
276  const int m_max = sbr->m[1];
277  static const float h_smooth[5] = {
278  0.33333333333333,
279  0.30150283239582,
280  0.21816949906249,
281  0.11516383427084,
282  0.03183050093751,
283  };
284  float (*g_temp)[48] = ch_data->g_temp, (*q_temp)[48] = ch_data->q_temp;
285  int indexnoise = ch_data->f_indexnoise;
286  int indexsine = ch_data->f_indexsine;
287 
288  if (sbr->reset) {
289  for (i = 0; i < h_SL; i++) {
290  memcpy(g_temp[i + 2*ch_data->t_env[0]], sbr->gain[0], m_max * sizeof(sbr->gain[0][0]));
291  memcpy(q_temp[i + 2*ch_data->t_env[0]], sbr->q_m[0], m_max * sizeof(sbr->q_m[0][0]));
292  }
293  } else if (h_SL) {
294  for (i = 0; i < 4; i++) {
295  memcpy(g_temp[i + 2 * ch_data->t_env[0]],
296  g_temp[i + 2 * ch_data->t_env_num_env_old],
297  sizeof(g_temp[0]));
298  memcpy(q_temp[i + 2 * ch_data->t_env[0]],
299  q_temp[i + 2 * ch_data->t_env_num_env_old],
300  sizeof(q_temp[0]));
301  }
302  }
303 
304  for (e = 0; e < ch_data->bs_num_env; e++) {
305  for (i = 2 * ch_data->t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
306  memcpy(g_temp[h_SL + i], sbr->gain[e], m_max * sizeof(sbr->gain[0][0]));
307  memcpy(q_temp[h_SL + i], sbr->q_m[e], m_max * sizeof(sbr->q_m[0][0]));
308  }
309  }
310 
311  for (e = 0; e < ch_data->bs_num_env; e++) {
312  for (i = 2 * ch_data->t_env[e]; i < 2 * ch_data->t_env[e + 1]; i++) {
313  LOCAL_ALIGNED_16(float, g_filt_tab, [48]);
314  LOCAL_ALIGNED_16(float, q_filt_tab, [48]);
315  float *g_filt, *q_filt;
316 
317  if (h_SL && e != e_a[0] && e != e_a[1]) {
318  g_filt = g_filt_tab;
319  q_filt = q_filt_tab;
320  for (m = 0; m < m_max; m++) {
321  const int idx1 = i + h_SL;
322  g_filt[m] = 0.0f;
323  q_filt[m] = 0.0f;
324  for (j = 0; j <= h_SL; j++) {
325  g_filt[m] += g_temp[idx1 - j][m] * h_smooth[j];
326  q_filt[m] += q_temp[idx1 - j][m] * h_smooth[j];
327  }
328  }
329  } else {
330  g_filt = g_temp[i + h_SL];
331  q_filt = q_temp[i];
332  }
333 
334  sbr->dsp.hf_g_filt(Y1[i] + kx, X_high + kx, g_filt, m_max,
336 
337  if (e != e_a[0] && e != e_a[1]) {
338  sbr->dsp.hf_apply_noise[indexsine](Y1[i] + kx, sbr->s_m[e],
339  q_filt, indexnoise,
340  kx, m_max);
341  } else {
342  int idx = indexsine&1;
343  int A = (1-((indexsine+(kx & 1))&2));
344  int B = (A^(-idx)) + idx;
345  float *out = &Y1[i][kx][idx];
346  float *in = sbr->s_m[e];
347  for (m = 0; m+1 < m_max; m+=2) {
348  out[2*m ] += in[m ] * A;
349  out[2*m+2] += in[m+1] * B;
350  }
351  if(m_max&1)
352  out[2*m ] += in[m ] * A;
353  }
354  indexnoise = (indexnoise + m_max) & 0x1ff;
355  indexsine = (indexsine + 1) & 3;
356  }
357  }
358  ch_data->f_indexnoise = indexnoise;
359  ch_data->f_indexsine = indexsine;
360 }
361 
362 #include "aacsbr_template.c"
uint8_t s_indexmapped[8][48]
Definition: sbr.h:97
#define NULL
Definition: coverity.c:32
static void sbr_hf_assemble(float Y1[38][64][2], const float X_high[64][40][2], SpectralBandReplication *sbr, SBRData *ch_data, const int e_a[2])
Assembling HF Signals (14496-3 sp04 p220)
Definition: aacsbr.c:268
unsigned bs_smoothing_mode
Definition: sbr.h:151
INTFLOAT bw_array[5]
Chirp factors.
Definition: sbr.h:89
else temp
Definition: vf_mcdeint.c:257
Definition: aac.h:57
static void aacsbr_func_ptr_init(AACSBRContext *c)
AAC_SIGNE kx[2]
kx', and kx respectively, kx is the first QMF subband where SBR is used.
Definition: sbr.h:157
AAC_FLOAT gain[7][48]
Definition: sbr.h:206
AAC_FLOAT noise_facs[3][5]
Noise scalefactors.
Definition: sbr.h:101
float delta
AAC_SIGNE n_lim
Number of limiter bands.
Definition: sbr.h:170
#define ENVELOPE_ADJUSTMENT_OFFSET
Definition: aacsbr.h:36
void(* hf_g_filt)(INTFLOAT(*Y)[2], const INTFLOAT(*X_high)[40][2], const AAC_FLOAT *g_filt, int m_max, intptr_t ixh)
Definition: sbrdsp.h:40
AAC Spectral Band Replication decoding data.
AAC_SIGNE bs_num_noise
Definition: sbr.h:71
#define lrintf(x)
Definition: libm_mips.h:70
SBRData data[2]
Definition: sbr.h:163
#define A(x)
Definition: vp56_arith.h:28
#define av_log(a,...)
unsigned m
Definition: audioconvert.c:187
static double alpha(void *priv, double x, double y)
Definition: vf_geq.c:99
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
static void sbr_hf_inverse_filter(SBRDSPContext *dsp, float(*alpha0)[2], float(*alpha1)[2], const float X_low[32][40][2], int k0)
High Frequency Generation (14496-3 sp04 p214+) and Inverse Filtering (14496-3 sp04 p214) Warning: Thi...
Definition: aacsbr.c:133
AAC_SIGNE m[2]
M' and M respectively, M is the number of QMF subbands that use SBR.
Definition: sbr.h:159
static void sbr_dequant(SpectralBandReplication *sbr, int id_aac)
Dequantization and stereo decoding (14496-3 sp04 p203)
Definition: aacsbr.c:72
Spectral Band Replication definitions and structures.
simple assert() macros that are a bit more flexible than ISO C assert().
Reference: libavcodec/aacsbr.c.
Definition: get_bits.h:63
#define powf(x, y)
Definition: libm.h:48
AAC Spectral Band Replication decoding functions.
unsigned f_indexnoise
Definition: sbr.h:108
common internal API header
uint8_t t_env_num_env_old
Envelope time border of the last envelope of the previous frame.
Definition: sbr.h:105
AAC Spectral Band Replication function declarations.
unsigned bs_amp_res
Definition: sbr.h:76
#define FFMIN(a, b)
Definition: common.h:81
unsigned bs_limiter_gains
Definition: sbr.h:149
AAC_FLOAT e_origmapped[7][48]
Dequantized envelope scalefactors, remapped.
Definition: sbr.h:195
uint8_t s_mapped[7][48]
Sinusoidal presence, remapped.
Definition: sbr.h:199
AAC definitions and structures.
uint8_t bs_freq_res[7]
Definition: sbr.h:70
static void sbr_gain_calc(AACContext *ac, SpectralBandReplication *sbr, SBRData *ch_data, const int e_a[2])
Calculation of levels of additional HF signal components (14496-3 sp04 p219) and Calculation of gain ...
Definition: aacsbr.c:212
AAC_FLOAT q_temp[42][48]
Definition: sbr.h:96
AAC_SIGNE bs_num_env
Definition: sbr.h:69
#define exp2f(x)
Definition: libm.h:82
AAC_FLOAT q_mapped[7][48]
Dequantized noise scalefactors, remapped.
Definition: sbr.h:197
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
Replacements for frequently missing libm functions.
static void sbr_chirp(SpectralBandReplication *sbr, SBRData *ch_data)
Chirp Factors (14496-3 sp04 p214)
Definition: aacsbr.c:188
AAC_FLOAT q_m[7][48]
Amplitude adjusted noise scalefactors.
Definition: sbr.h:203
AAC_FLOAT env_facs[6][48]
Envelope scalefactors.
Definition: sbr.h:99
main AAC context
Definition: aac.h:288
#define NOISE_FLOOR_OFFSET
Definition: aacsbr.h:37
AAC_FLOAT e_curr[7][48]
Estimated envelope.
Definition: sbr.h:201
uint8_t bs_invf_mode[2][5]
Definition: sbr.h:74
unsigned f_indexsine
Definition: sbr.h:109
static double c[64]
uint8_t t_env[8]
Envelope time borders.
Definition: sbr.h:103
aacsbr functions pointers
Definition: sbr.h:118
AAC_FLOAT s_m[7][48]
Sinusoidal levels.
Definition: sbr.h:205
uint16_t f_tablelim[30]
Frequency borders for the limiter.
Definition: sbr.h:180
Spectral Band Replication per channel data.
Definition: sbr.h:62
static void make_bands(int16_t *bands, int start, int stop, int num_bands)
Definition: aacsbr.c:53
Definition: vf_geq.c:46
void(* hf_apply_noise[4])(INTFLOAT(*Y)[2], const AAC_FLOAT *s_m, const AAC_FLOAT *q_filt, int noise, int kx, int m_max)
Definition: sbrdsp.h:42
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> out
SBRDSPContext dsp
Definition: sbr.h:210
void(* autocorrelate)(const INTFLOAT x[40][2], AAC_FLOAT phi[3][2][2])
Definition: sbrdsp.h:36
#define LOCAL_ALIGNED_16(t, v,...)
Definition: internal.h:120
void INT64 start
Definition: avisynth_c.h:553
float g_temp[42][48]
Definition: sbr.h:95
static VLC vlc_sbr[10]
Definition: aacsbr.c:50
AAC_SIGNE n_q
Number of noise floor bands.
Definition: sbr.h:168
unsigned bs_coupling
Definition: sbr.h:153
Spectral Band Replication.
Definition: sbr.h:137
AAC_SIGNE n[2]
N_Low and N_High respectively, the number of frequency bands for low and high resolution.
Definition: sbr.h:166