FFmpeg  4.4
acelp_vectors.c
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1 /*
2  * adaptive and fixed codebook vector operations for ACELP-based codecs
3  *
4  * Copyright (c) 2008 Vladimir Voroshilov
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 #include <inttypes.h>
24 
25 #include "libavutil/avassert.h"
26 #include "libavutil/common.h"
27 #include "libavutil/float_dsp.h"
28 #include "avcodec.h"
29 #include "acelp_vectors.h"
30 
32 {
33  1, 3,
34  8, 6,
35  18, 16,
36  11, 13,
37  38, 36,
38  31, 33,
39  21, 23,
40  28, 26,
41 };
42 
44 {
45  0, 2,
46  5, 4,
47  12, 10,
48  7, 9,
49  25, 24,
50  20, 22,
51  14, 15,
52  19, 17,
53  36, 31,
54  21, 26,
55  1, 6,
56  16, 11,
57  27, 29,
58  32, 30,
59  39, 37,
60  34, 35,
61 };
62 
64 {
65  0, 5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60, 65, 70, 75,
66 };
67 
69 {
70  3, 4,
71  8, 9,
72  13, 14,
73  18, 19,
74  23, 24,
75  28, 29,
76  33, 34,
77  38, 39,
78  43, 44,
79  48, 49,
80  53, 54,
81  58, 59,
82  63, 64,
83  68, 69,
84  73, 74,
85  78, 79,
86 };
87 
88 const float ff_pow_0_7[10] = {
89  0.700000, 0.490000, 0.343000, 0.240100, 0.168070,
90  0.117649, 0.082354, 0.057648, 0.040354, 0.028248
91 };
92 
93 const float ff_pow_0_75[10] = {
94  0.750000, 0.562500, 0.421875, 0.316406, 0.237305,
95  0.177979, 0.133484, 0.100113, 0.075085, 0.056314
96 };
97 
98 const float ff_pow_0_55[10] = {
99  0.550000, 0.302500, 0.166375, 0.091506, 0.050328,
100  0.027681, 0.015224, 0.008373, 0.004605, 0.002533
101 };
102 
103 const float ff_b60_sinc[61] = {
104  0.898529 , 0.865051 , 0.769257 , 0.624054 , 0.448639 , 0.265289 ,
105  0.0959167 , -0.0412598 , -0.134338 , -0.178986 , -0.178528 , -0.142609 ,
106 -0.0849304 , -0.0205078 , 0.0369568 , 0.0773926 , 0.0955200 , 0.0912781 ,
107  0.0689392 , 0.0357056 , 0.0 , -0.0305481 , -0.0504150 , -0.0570068 ,
108 -0.0508423 , -0.0350037 , -0.0141602 , 0.00665283, 0.0230713 , 0.0323486 ,
109  0.0335388 , 0.0275879 , 0.0167847 , 0.00411987, -0.00747681, -0.0156860 ,
110 -0.0193481 , -0.0183716 , -0.0137634 , -0.00704956, 0.0 , 0.00582886 ,
111  0.00939941, 0.0103760 , 0.00903320, 0.00604248, 0.00238037, -0.00109863 ,
112 -0.00366211, -0.00497437, -0.00503540, -0.00402832, -0.00241089, -0.000579834,
113  0.00103760, 0.00222778, 0.00277710, 0.00271606, 0.00213623, 0.00115967 ,
114  0.
115 };
116 
118  int16_t* fc_v,
119  const uint8_t *tab1,
120  const uint8_t *tab2,
121  int pulse_indexes,
122  int pulse_signs,
123  int pulse_count,
124  int bits)
125 {
126  int mask = (1 << bits) - 1;
127  int i;
128 
129  for(i=0; i<pulse_count; i++)
130  {
131  fc_v[i + tab1[pulse_indexes & mask]] +=
132  (pulse_signs & 1) ? 8191 : -8192; // +/-1 in (2.13)
133 
134  pulse_indexes >>= bits;
135  pulse_signs >>= 1;
136  }
137 
138  fc_v[tab2[pulse_indexes]] += (pulse_signs & 1) ? 8191 : -8192;
139 }
140 
141 void ff_decode_10_pulses_35bits(const int16_t *fixed_index,
142  AMRFixed *fixed_sparse,
143  const uint8_t *gray_decode,
144  int half_pulse_count, int bits)
145 {
146  int i;
147  int mask = (1 << bits) - 1;
148 
149  fixed_sparse->no_repeat_mask = 0;
150  fixed_sparse->n = 2 * half_pulse_count;
151  for (i = 0; i < half_pulse_count; i++) {
152  const int pos1 = gray_decode[fixed_index[2*i+1] & mask] + i;
153  const int pos2 = gray_decode[fixed_index[2*i ] & mask] + i;
154  const float sign = (fixed_index[2*i+1] & (1 << bits)) ? -1.0 : 1.0;
155  fixed_sparse->x[2*i+1] = pos1;
156  fixed_sparse->x[2*i ] = pos2;
157  fixed_sparse->y[2*i+1] = sign;
158  fixed_sparse->y[2*i ] = pos2 < pos1 ? -sign : sign;
159  }
160 }
161 
163  int16_t* out,
164  const int16_t *in_a,
165  const int16_t *in_b,
166  int16_t weight_coeff_a,
167  int16_t weight_coeff_b,
168  int16_t rounder,
169  int shift,
170  int length)
171 {
172  int i;
173 
174  // Clipping required here; breaks OVERFLOW test.
175  for(i=0; i<length; i++)
176  out[i] = av_clip_int16((
177  in_a[i] * weight_coeff_a +
178  in_b[i] * weight_coeff_b +
179  rounder) >> shift);
180 }
181 
182 void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b,
183  float weight_coeff_a, float weight_coeff_b, int length)
184 {
185  int i;
186 
187  for(i=0; i<length; i++)
188  out[i] = weight_coeff_a * in_a[i]
189  + weight_coeff_b * in_b[i];
190 }
191 
192 void ff_adaptive_gain_control(float *out, const float *in, float speech_energ,
193  int size, float alpha, float *gain_mem)
194 {
195  int i;
196  float postfilter_energ = avpriv_scalarproduct_float_c(in, in, size);
197  float gain_scale_factor = 1.0;
198  float mem = *gain_mem;
199 
200  if (postfilter_energ)
201  gain_scale_factor = sqrt(speech_energ / postfilter_energ);
202 
203  gain_scale_factor *= 1.0 - alpha;
204 
205  for (i = 0; i < size; i++) {
206  mem = alpha * mem + gain_scale_factor;
207  out[i] = in[i] * mem;
208  }
209 
210  *gain_mem = mem;
211 }
212 
214  float sum_of_squares, const int n)
215 {
216  int i;
217  float scalefactor = avpriv_scalarproduct_float_c(in, in, n);
218  if (scalefactor)
219  scalefactor = sqrt(sum_of_squares / scalefactor);
220  for (i = 0; i < n; i++)
221  out[i] = in[i] * scalefactor;
222 }
223 
224 void ff_set_fixed_vector(float *out, const AMRFixed *in, float scale, int size)
225 {
226  int i;
227 
228  for (i=0; i < in->n; i++) {
229  int x = in->x[i], repeats = !((in->no_repeat_mask >> i) & 1);
230  float y = in->y[i] * scale;
231 
232  if (in->pitch_lag > 0)
233  av_assert0(x < size);
234  do {
235  out[x] += y;
236  y *= in->pitch_fac;
237  x += in->pitch_lag;
238  } while (x < size && repeats);
239  }
240 }
241 
242 void ff_clear_fixed_vector(float *out, const AMRFixed *in, int size)
243 {
244  int i;
245 
246  for (i=0; i < in->n; i++) {
247  int x = in->x[i], repeats = !((in->no_repeat_mask >> i) & 1);
248 
249  if (in->pitch_lag > 0)
250  do {
251  out[x] = 0.0;
252  x += in->pitch_lag;
253  } while (x < size && repeats);
254  }
255 }
256 
258 {
259  c->weighted_vector_sumf = ff_weighted_vector_sumf;
260 
261  if(HAVE_MIPSFPU)
263 }
const float ff_pow_0_7[10]
Table of pow(0.7,n)
Definition: acelp_vectors.c:88
void ff_decode_10_pulses_35bits(const int16_t *fixed_index, AMRFixed *fixed_sparse, const uint8_t *gray_decode, int half_pulse_count, int bits)
Decode the algebraic codebook index to pulse positions and signs and construct the algebraic codebook...
void ff_acelp_vectors_init(ACELPVContext *c)
Initialize ACELPVContext.
void ff_adaptive_gain_control(float *out, const float *in, float speech_energ, int size, float alpha, float *gain_mem)
Adaptive gain control (as used in AMR postfiltering)
const uint8_t ff_fc_2pulses_9bits_track1_gray[16]
Definition: acelp_vectors.c:31
void ff_scale_vector_to_given_sum_of_squares(float *out, const float *in, float sum_of_squares, const int n)
Set the sum of squares of a signal by scaling.
const uint8_t ff_fc_4pulses_8bits_tracks_13[16]
Definition: acelp_vectors.c:63
const uint8_t ff_fc_2pulses_9bits_track2_gray[32]
Definition: acelp_vectors.c:43
const float ff_b60_sinc[61]
b60 hamming windowed sinc function coefficients
const float ff_pow_0_55[10]
Table of pow(0.55,n)
Definition: acelp_vectors.c:98
void ff_acelp_fc_pulse_per_track(int16_t *fc_v, const uint8_t *tab1, const uint8_t *tab2, int pulse_indexes, int pulse_signs, int pulse_count, int bits)
Decode fixed-codebook vector (3.8 and D.5.8 of G.729, 5.7.1 of AMR).
const float ff_pow_0_75[10]
Table of pow(0.75,n)
Definition: acelp_vectors.c:93
const uint8_t ff_fc_4pulses_8bits_track_4[32]
Definition: acelp_vectors.c:68
void ff_set_fixed_vector(float *out, const AMRFixed *in, float scale, int size)
Add fixed vector to an array from a sparse representation.
void ff_acelp_weighted_vector_sum(int16_t *out, const int16_t *in_a, const int16_t *in_b, int16_t weight_coeff_a, int16_t weight_coeff_b, int16_t rounder, int shift, int length)
weighted sum of two vectors with rounding.
void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b, float weight_coeff_a, float weight_coeff_b, int length)
float implementation of weighted sum of two vectors.
void ff_clear_fixed_vector(float *out, const AMRFixed *in, int size)
Clear array values set by set_fixed_vector.
void ff_acelp_vectors_init_mips(ACELPVContext *c)
static const uint8_t gray_decode[8]
3-bit Gray code to binary lookup table
Definition: amrnbdata.h:1433
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
uint8_t
simple assert() macros that are a bit more flexible than ISO C assert().
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
Libavcodec external API header.
common internal and external API header
#define av_clip_int16
Definition: common.h:137
#define HAVE_MIPSFPU
Definition: config.h:75
static const int16_t alpha[]
Definition: ilbcdata.h:55
int i
Definition: input.c:407
float avpriv_scalarproduct_float_c(const float *v1, const float *v2, int len)
Return the scalar product of two vectors.
Definition: float_dsp.c:124
static const uint16_t mask[17]
Definition: lzw.c:38
const int16_t * tab2
Definition: mace.c:144
const int16_t * tab1
Definition: mace.c:144
static int shift(int a, int b)
Definition: sonic.c:82
Sparse representation for the algebraic codebook (fixed) vector.
Definition: acelp_vectors.h:53
int x[10]
Definition: acelp_vectors.h:55
int no_repeat_mask
Definition: acelp_vectors.h:57
float y[10]
Definition: acelp_vectors.h:56
FILE * out
Definition: movenc.c:54
int size
uint8_t bits
Definition: vp3data.h:141
static double c[64]