FFmpeg  4.3
aacenc_utils.h
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1 /*
2  * AAC encoder utilities
3  * Copyright (C) 2015 Rostislav Pehlivanov
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 utilities
25  * @author Rostislav Pehlivanov ( atomnuker gmail com )
26  */
27 
28 #ifndef AVCODEC_AACENC_UTILS_H
29 #define AVCODEC_AACENC_UTILS_H
30 
31 #include "libavutil/ffmath.h"
32 #include "aac.h"
33 #include "aacenctab.h"
34 #include "aactab.h"
35 
36 #define ROUND_STANDARD 0.4054f
37 #define ROUND_TO_ZERO 0.1054f
38 #define C_QUANT 0.4054f
39 
40 static inline void abs_pow34_v(float *out, const float *in, const int size)
41 {
42  int i;
43  for (i = 0; i < size; i++) {
44  float a = fabsf(in[i]);
45  out[i] = sqrtf(a * sqrtf(a));
46  }
47 }
48 
49 static inline float pos_pow34(float a)
50 {
51  return sqrtf(a * sqrtf(a));
52 }
53 
54 /**
55  * Quantize one coefficient.
56  * @return absolute value of the quantized coefficient
57  * @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
58  */
59 static inline int quant(float coef, const float Q, const float rounding)
60 {
61  float a = coef * Q;
62  return sqrtf(a * sqrtf(a)) + rounding;
63 }
64 
65 static inline void quantize_bands(int *out, const float *in, const float *scaled,
66  int size, int is_signed, int maxval, const float Q34,
67  const float rounding)
68 {
69  int i;
70  for (i = 0; i < size; i++) {
71  float qc = scaled[i] * Q34;
72  int tmp = (int)FFMIN(qc + rounding, (float)maxval);
73  if (is_signed && in[i] < 0.0f) {
74  tmp = -tmp;
75  }
76  out[i] = tmp;
77  }
78 }
79 
80 static inline float find_max_val(int group_len, int swb_size, const float *scaled)
81 {
82  float maxval = 0.0f;
83  int w2, i;
84  for (w2 = 0; w2 < group_len; w2++) {
85  for (i = 0; i < swb_size; i++) {
86  maxval = FFMAX(maxval, scaled[w2*128+i]);
87  }
88  }
89  return maxval;
90 }
91 
92 static inline int find_min_book(float maxval, int sf)
93 {
95  int qmaxval, cb;
96  qmaxval = maxval * Q34 + C_QUANT;
97  if (qmaxval >= (FF_ARRAY_ELEMS(aac_maxval_cb)))
98  cb = 11;
99  else
100  cb = aac_maxval_cb[qmaxval];
101  return cb;
102 }
103 
104 static inline float find_form_factor(int group_len, int swb_size, float thresh,
105  const float *scaled, float nzslope) {
106  const float iswb_size = 1.0f / swb_size;
107  const float iswb_sizem1 = 1.0f / (swb_size - 1);
108  const float ethresh = thresh;
109  float form = 0.0f, weight = 0.0f;
110  int w2, i;
111  for (w2 = 0; w2 < group_len; w2++) {
112  float e = 0.0f, e2 = 0.0f, var = 0.0f, maxval = 0.0f;
113  float nzl = 0;
114  for (i = 0; i < swb_size; i++) {
115  float s = fabsf(scaled[w2*128+i]);
116  maxval = FFMAX(maxval, s);
117  e += s;
118  e2 += s *= s;
119  /* We really don't want a hard non-zero-line count, since
120  * even below-threshold lines do add up towards band spectral power.
121  * So, fall steeply towards zero, but smoothly
122  */
123  if (s >= ethresh) {
124  nzl += 1.0f;
125  } else {
126  if (nzslope == 2.f)
127  nzl += (s / ethresh) * (s / ethresh);
128  else
129  nzl += ff_fast_powf(s / ethresh, nzslope);
130  }
131  }
132  if (e2 > thresh) {
133  float frm;
134  e *= iswb_size;
135 
136  /** compute variance */
137  for (i = 0; i < swb_size; i++) {
138  float d = fabsf(scaled[w2*128+i]) - e;
139  var += d*d;
140  }
141  var = sqrtf(var * iswb_sizem1);
142 
143  e2 *= iswb_size;
144  frm = e / FFMIN(e+4*var,maxval);
145  form += e2 * sqrtf(frm) / FFMAX(0.5f,nzl);
146  weight += e2;
147  }
148  }
149  if (weight > 0) {
150  return form / weight;
151  } else {
152  return 1.0f;
153  }
154 }
155 
156 /** Return the minimum scalefactor where the quantized coef does not clip. */
157 static inline uint8_t coef2minsf(float coef)
158 {
159  return av_clip_uint8(log2f(coef)*4 - 69 + SCALE_ONE_POS - SCALE_DIV_512);
160 }
161 
162 /** Return the maximum scalefactor where the quantized coef is not zero. */
163 static inline uint8_t coef2maxsf(float coef)
164 {
165  return av_clip_uint8(log2f(coef)*4 + 6 + SCALE_ONE_POS - SCALE_DIV_512);
166 }
167 
168 /*
169  * Returns the closest possible index to an array of float values, given a value.
170  */
171 static inline int quant_array_idx(const float val, const float *arr, const int num)
172 {
173  int i, index = 0;
174  float quant_min_err = INFINITY;
175  for (i = 0; i < num; i++) {
176  float error = (val - arr[i])*(val - arr[i]);
177  if (error < quant_min_err) {
178  quant_min_err = error;
179  index = i;
180  }
181  }
182  return index;
183 }
184 
185 /**
186  * approximates exp10f(-3.0f*(0.5f + 0.5f * cosf(FFMIN(b,15.5f) / 15.5f)))
187  */
188 static av_always_inline float bval2bmax(float b)
189 {
190  return 0.001f + 0.0035f * (b*b*b) / (15.5f*15.5f*15.5f);
191 }
192 
193 /*
194  * Compute a nextband map to be used with SF delta constraint utilities.
195  * The nextband array should contain 128 elements, and positions that don't
196  * map to valid, nonzero bands of the form w*16+g (with w being the initial
197  * window of the window group, only) are left indetermined.
198  */
199 static inline void ff_init_nextband_map(const SingleChannelElement *sce, uint8_t *nextband)
200 {
201  unsigned char prevband = 0;
202  int w, g;
203  /** Just a safe default */
204  for (g = 0; g < 128; g++)
205  nextband[g] = g;
206 
207  /** Now really navigate the nonzero band chain */
208  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
209  for (g = 0; g < sce->ics.num_swb; g++) {
210  if (!sce->zeroes[w*16+g] && sce->band_type[w*16+g] < RESERVED_BT)
211  prevband = nextband[prevband] = w*16+g;
212  }
213  }
214  nextband[prevband] = prevband; /* terminate */
215 }
216 
217 /*
218  * Updates nextband to reflect a removed band (equivalent to
219  * calling ff_init_nextband_map after marking a band as zero)
220  */
221 static inline void ff_nextband_remove(uint8_t *nextband, int prevband, int band)
222 {
223  nextband[prevband] = nextband[band];
224 }
225 
226 /*
227  * Checks whether the specified band could be removed without inducing
228  * scalefactor delta that violates SF delta encoding constraints.
229  * prev_sf has to be the scalefactor of the previous nonzero, nonspecial
230  * band, in encoding order, or negative if there was no such band.
231  */
233  const uint8_t *nextband, int prev_sf, int band)
234 {
235  return prev_sf >= 0
236  && sce->sf_idx[nextband[band]] >= (prev_sf - SCALE_MAX_DIFF)
237  && sce->sf_idx[nextband[band]] <= (prev_sf + SCALE_MAX_DIFF);
238 }
239 
240 /*
241  * Checks whether the specified band's scalefactor could be replaced
242  * with another one without violating SF delta encoding constraints.
243  * prev_sf has to be the scalefactor of the previous nonzero, nonsepcial
244  * band, in encoding order, or negative if there was no such band.
245  */
246 static inline int ff_sfdelta_can_replace(const SingleChannelElement *sce,
247  const uint8_t *nextband, int prev_sf, int new_sf, int band)
248 {
249  return new_sf >= (prev_sf - SCALE_MAX_DIFF)
250  && new_sf <= (prev_sf + SCALE_MAX_DIFF)
251  && sce->sf_idx[nextband[band]] >= (new_sf - SCALE_MAX_DIFF)
252  && sce->sf_idx[nextband[band]] <= (new_sf + SCALE_MAX_DIFF);
253 }
254 
255 /**
256  * linear congruential pseudorandom number generator
257  *
258  * @param previous_val pointer to the current state of the generator
259  *
260  * @return Returns a 32-bit pseudorandom integer
261  */
262 static av_always_inline int lcg_random(unsigned previous_val)
263 {
264  union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
265  return v.s;
266 }
267 
268 #define ERROR_IF(cond, ...) \
269  if (cond) { \
270  av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
271  return AVERROR(EINVAL); \
272  }
273 
274 #define WARN_IF(cond, ...) \
275  if (cond) { \
276  av_log(avctx, AV_LOG_WARNING, __VA_ARGS__); \
277  }
278 
279 #endif /* AVCODEC_AACENC_UTILS_H */
error
static void error(const char *err)
Definition: target_bsf_fuzzer.c:29
INFINITY
#define INFINITY
Definition: mathematics.h:67
out
FILE * out
Definition: movenc.c:54
cb
static double cb(void *priv, double x, double y)
Definition: vf_geq.c:215
u
#define u(width, name, range_min, range_max)
Definition: cbs_h2645.c:262
aacenctab.h
abs_pow34_v
static void abs_pow34_v(float *out, const float *in, const int size)
Definition: aacenc_utils.h:40
log2f
#define log2f(x)
Definition: libm.h:409
SingleChannelElement::zeroes
uint8_t zeroes[128]
band is not coded (used by encoder)
Definition: aac.h:257
tmp
static uint8_t tmp[11]
Definition: aes_ctr.c:26
bval2bmax
static av_always_inline float bval2bmax(float b)
approximates exp10f(-3.0f*(0.5f + 0.5f * cosf(FFMIN(b,15.5f) / 15.5f)))
Definition: aacenc_utils.h:188
b
#define b
Definition: input.c:41
ff_fast_powf
static av_always_inline float ff_fast_powf(float x, float y)
Compute x^y for floating point x, y.
Definition: ffmath.h:62
ff_sfdelta_can_remove_band
static int ff_sfdelta_can_remove_band(const SingleChannelElement *sce, const uint8_t *nextband, int prev_sf, int band)
Definition: aacenc_utils.h:232
coef2maxsf
static uint8_t coef2maxsf(float coef)
Return the maximum scalefactor where the quantized coef is not zero.
Definition: aacenc_utils.h:163
IndividualChannelStream::num_swb
int num_swb
number of scalefactor window bands
Definition: aac.h:183
SCALE_DIV_512
#define SCALE_DIV_512
scalefactor difference that corresponds to scale difference in 512 times
Definition: aac.h:148
ff_sfdelta_can_replace
static int ff_sfdelta_can_replace(const SingleChannelElement *sce, const uint8_t *nextband, int prev_sf, int new_sf, int band)
Definition: aacenc_utils.h:246
find_form_factor
static float find_form_factor(int group_len, int swb_size, float thresh, const float *scaled, float nzslope)
Definition: aacenc_utils.h:104
POW_SF2_ZERO
#define POW_SF2_ZERO
ff_aac_pow2sf_tab index corresponding to pow(2, 0);
Definition: aac.h:154
val
static double val(void *priv, double ch)
Definition: aeval.c:76
quant
static int quant(float coef, const float Q, const float rounding)
Quantize one coefficient.
Definition: aacenc_utils.h:59
SingleChannelElement::ics
IndividualChannelStream ics
Definition: aac.h:249
s
#define s(width, name)
Definition: cbs_vp9.c:257
g
const char * g
Definition: vf_curves.c:115
IndividualChannelStream::group_len
uint8_t group_len[8]
Definition: aac.h:179
form
This is the more generic form
Definition: tablegen.txt:34
f
#define f(width, name)
Definition: cbs_vp9.c:255
aac.h
aactab.h
ff_init_nextband_map
static void ff_init_nextband_map(const SingleChannelElement *sce, uint8_t *nextband)
Definition: aacenc_utils.h:199
index
int index
Definition: gxfenc.c:89
SingleChannelElement::sf_idx
int sf_idx[128]
scalefactor indices (used by encoder)
Definition: aac.h:256
weight
static int weight(int i, int blen, int offset)
Definition: diracdec.c:1560
coef2minsf
static uint8_t coef2minsf(float coef)
Return the minimum scalefactor where the quantized coef does not clip.
Definition: aacenc_utils.h:157
quantize_bands
static void quantize_bands(int *out, const float *in, const float *scaled, int size, int is_signed, int maxval, const float Q34, const float rounding)
Definition: aacenc_utils.h:65
FFMAX
#define FFMAX(a, b)
Definition: common.h:94
C_QUANT
#define C_QUANT
Definition: aacenc_utils.h:38
size
int size
Definition: twinvq_data.h:11134
quant_array_idx
static int quant_array_idx(const float val, const float *arr, const int num)
Definition: aacenc_utils.h:171
FFMIN
#define FFMIN(a, b)
Definition: common.h:96
a
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
Definition: undefined.txt:41
SCALE_MAX_DIFF
#define SCALE_MAX_DIFF
maximum scalefactor difference allowed by standard
Definition: aac.h:151
pos_pow34
static float pos_pow34(float a)
Definition: aacenc_utils.h:49
in
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)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
Definition: audio_convert.c:326
SingleChannelElement
Single Channel Element - used for both SCE and LFE elements.
Definition: aac.h:248
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
IndividualChannelStream::num_windows
int num_windows
Definition: aac.h:184
SCALE_ONE_POS
#define SCALE_ONE_POS
scalefactor index that corresponds to scale=1.0
Definition: aac.h:149
find_min_book
static int find_min_book(float maxval, int sf)
Definition: aacenc_utils.h:92
av_always_inline
#define av_always_inline
Definition: attributes.h:49
uint8_t
uint8_t
Definition: audio_convert.c:194
lcg_random
static av_always_inline int lcg_random(unsigned previous_val)
linear congruential pseudorandom number generator
Definition: aacenc_utils.h:262
w
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new please make sure they are as small as space on each network bandwidth and so on benefit from smaller test cases Also keep in mind older checkouts use existing sample that means in practice generally do not remove or overwrite files as it likely would break older checkouts or releases Also all needed samples for a commit should be ideally before the push If you need an account for frequently uploading samples or you wish to help others by doing that send a mail to ffmpeg devel rsync vauL Duo ug o o w
Definition: fate.txt:150
RESERVED_BT
@ RESERVED_BT
Band types following are encoded differently from others.
Definition: aac.h:86
FF_ARRAY_ELEMS
#define FF_ARRAY_ELEMS(a)
Definition: sinewin_tablegen_template.c:38
aac_maxval_cb
static const unsigned char aac_maxval_cb[]
Definition: aacenctab.h:128
ff_aac_pow34sf_tab
float ff_aac_pow34sf_tab[428]
Definition: aactab.c:36
ffmath.h
find_max_val
static float find_max_val(int group_len, int swb_size, const float *scaled)
Definition: aacenc_utils.h:80
int
int
Definition: ffmpeg_filter.c:192
SingleChannelElement::band_type
enum BandType band_type[128]
band types
Definition: aac.h:252
ff_nextband_remove
static void ff_nextband_remove(uint8_t *nextband, int prevband, int band)
Definition: aacenc_utils.h:221