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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  6, 8,
35  11, 13,
36  16, 18,
37  21, 23,
38  26, 28,
39  31, 33,
40  36, 38
41 };
43 {
44  1, 3,
45  8, 6,
46  18, 16,
47  11, 13,
48  38, 36,
49  31, 33,
50  21, 23,
51  28, 26,
52 };
53 
55 {
56  0, 2,
57  5, 4,
58  12, 10,
59  7, 9,
60  25, 24,
61  20, 22,
62  14, 15,
63  19, 17,
64  36, 31,
65  21, 26,
66  1, 6,
67  16, 11,
68  27, 29,
69  32, 30,
70  39, 37,
71  34, 35,
72 };
73 
75 {
76  0, 5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60, 65, 70, 75,
77 };
78 
80 {
81  3, 4,
82  8, 9,
83  13, 14,
84  18, 19,
85  23, 24,
86  28, 29,
87  33, 34,
88  38, 39,
89  43, 44,
90  48, 49,
91  53, 54,
92  58, 59,
93  63, 64,
94  68, 69,
95  73, 74,
96  78, 79,
97 };
98 
99 const float ff_pow_0_7[10] = {
100  0.700000, 0.490000, 0.343000, 0.240100, 0.168070,
101  0.117649, 0.082354, 0.057648, 0.040354, 0.028248
102 };
103 
104 const float ff_pow_0_75[10] = {
105  0.750000, 0.562500, 0.421875, 0.316406, 0.237305,
106  0.177979, 0.133484, 0.100113, 0.075085, 0.056314
107 };
108 
109 const float ff_pow_0_55[10] = {
110  0.550000, 0.302500, 0.166375, 0.091506, 0.050328,
111  0.027681, 0.015224, 0.008373, 0.004605, 0.002533
112 };
113 
114 const float ff_b60_sinc[61] = {
115  0.898529 , 0.865051 , 0.769257 , 0.624054 , 0.448639 , 0.265289 ,
116  0.0959167 , -0.0412598 , -0.134338 , -0.178986 , -0.178528 , -0.142609 ,
117 -0.0849304 , -0.0205078 , 0.0369568 , 0.0773926 , 0.0955200 , 0.0912781 ,
118  0.0689392 , 0.0357056 , 0.0 , -0.0305481 , -0.0504150 , -0.0570068 ,
119 -0.0508423 , -0.0350037 , -0.0141602 , 0.00665283, 0.0230713 , 0.0323486 ,
120  0.0335388 , 0.0275879 , 0.0167847 , 0.00411987, -0.00747681, -0.0156860 ,
121 -0.0193481 , -0.0183716 , -0.0137634 , -0.00704956, 0.0 , 0.00582886 ,
122  0.00939941, 0.0103760 , 0.00903320, 0.00604248, 0.00238037, -0.00109863 ,
123 -0.00366211, -0.00497437, -0.00503540, -0.00402832, -0.00241089, -0.000579834,
124  0.00103760, 0.00222778, 0.00277710, 0.00271606, 0.00213623, 0.00115967 ,
125  0.
126 };
127 
129  int16_t* fc_v,
130  const uint8_t *tab1,
131  const uint8_t *tab2,
132  int pulse_indexes,
133  int pulse_signs,
134  int pulse_count,
135  int bits)
136 {
137  int mask = (1 << bits) - 1;
138  int i;
139 
140  for(i=0; i<pulse_count; i++)
141  {
142  fc_v[i + tab1[pulse_indexes & mask]] +=
143  (pulse_signs & 1) ? 8191 : -8192; // +/-1 in (2.13)
144 
145  pulse_indexes >>= bits;
146  pulse_signs >>= 1;
147  }
148 
149  fc_v[tab2[pulse_indexes]] += (pulse_signs & 1) ? 8191 : -8192;
150 }
151 
152 void ff_decode_10_pulses_35bits(const int16_t *fixed_index,
153  AMRFixed *fixed_sparse,
154  const uint8_t *gray_decode,
155  int half_pulse_count, int bits)
156 {
157  int i;
158  int mask = (1 << bits) - 1;
159 
160  fixed_sparse->no_repeat_mask = 0;
161  fixed_sparse->n = 2 * half_pulse_count;
162  for (i = 0; i < half_pulse_count; i++) {
163  const int pos1 = gray_decode[fixed_index[2*i+1] & mask] + i;
164  const int pos2 = gray_decode[fixed_index[2*i ] & mask] + i;
165  const float sign = (fixed_index[2*i+1] & (1 << bits)) ? -1.0 : 1.0;
166  fixed_sparse->x[2*i+1] = pos1;
167  fixed_sparse->x[2*i ] = pos2;
168  fixed_sparse->y[2*i+1] = sign;
169  fixed_sparse->y[2*i ] = pos2 < pos1 ? -sign : sign;
170  }
171 }
172 
174  int16_t* out,
175  const int16_t *in_a,
176  const int16_t *in_b,
177  int16_t weight_coeff_a,
178  int16_t weight_coeff_b,
179  int16_t rounder,
180  int shift,
181  int length)
182 {
183  int i;
184 
185  // Clipping required here; breaks OVERFLOW test.
186  for(i=0; i<length; i++)
187  out[i] = av_clip_int16((
188  in_a[i] * weight_coeff_a +
189  in_b[i] * weight_coeff_b +
190  rounder) >> shift);
191 }
192 
193 void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b,
194  float weight_coeff_a, float weight_coeff_b, int length)
195 {
196  int i;
197 
198  for(i=0; i<length; i++)
199  out[i] = weight_coeff_a * in_a[i]
200  + weight_coeff_b * in_b[i];
201 }
202 
203 void ff_adaptive_gain_control(float *out, const float *in, float speech_energ,
204  int size, float alpha, float *gain_mem)
205 {
206  int i;
207  float postfilter_energ = avpriv_scalarproduct_float_c(in, in, size);
208  float gain_scale_factor = 1.0;
209  float mem = *gain_mem;
210 
211  if (postfilter_energ)
212  gain_scale_factor = sqrt(speech_energ / postfilter_energ);
213 
214  gain_scale_factor *= 1.0 - alpha;
215 
216  for (i = 0; i < size; i++) {
217  mem = alpha * mem + gain_scale_factor;
218  out[i] = in[i] * mem;
219  }
220 
221  *gain_mem = mem;
222 }
223 
225  float sum_of_squares, const int n)
226 {
227  int i;
228  float scalefactor = avpriv_scalarproduct_float_c(in, in, n);
229  if (scalefactor)
230  scalefactor = sqrt(sum_of_squares / scalefactor);
231  for (i = 0; i < n; i++)
232  out[i] = in[i] * scalefactor;
233 }
234 
235 void ff_set_fixed_vector(float *out, const AMRFixed *in, float scale, int size)
236 {
237  int i;
238 
239  for (i=0; i < in->n; i++) {
240  int x = in->x[i], repeats = !((in->no_repeat_mask >> i) & 1);
241  float y = in->y[i] * scale;
242 
243  if (in->pitch_lag > 0)
244  av_assert0(x < size);
245  do {
246  out[x] += y;
247  y *= in->pitch_fac;
248  x += in->pitch_lag;
249  } while (x < size && repeats);
250  }
251 }
252 
253 void ff_clear_fixed_vector(float *out, const AMRFixed *in, int size)
254 {
255  int i;
256 
257  for (i=0; i < in->n; i++) {
258  int x = in->x[i], repeats = !((in->no_repeat_mask >> i) & 1);
259 
260  if (in->pitch_lag > 0)
261  do {
262  out[x] = 0.0;
263  x += in->pitch_lag;
264  } while (x < size && repeats);
265  }
266 }
267 
269 {
271 
272  if(HAVE_MIPSFPU)
274 }
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).
static int shift(int a, int b)
Definition: sonic.c:82
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_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.
int x[10]
Definition: acelp_vectors.h:55
const uint8_t ff_fc_2pulses_9bits_track1_gray[16]
Definition: acelp_vectors.c:42
void ff_set_fixed_vector(float *out, const AMRFixed *in, float scale, int size)
Add fixed vector to an array from a sparse representation.
float pitch_fac
Definition: acelp_vectors.h:59
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
float avpriv_scalarproduct_float_c(const float *v1, const float *v2, int len)
Return the scalar product of two vectors.
Definition: float_dsp.c:108
void ff_acelp_vectors_init_mips(ACELPVContext *c)
uint8_t bits
Definition: crc.c:296
int mem
Definition: avisynth_c.h:684
const uint8_t ff_fc_4pulses_8bits_track_4[32]
Track|Pulse| Positions 4 | 3 | 3, 8, 13, 18, 23, 28, 33, 38, 43, 48, 53, 58, 63, 68, 73, 78 | | 4, 9, 14, 19, 24, 29, 34, 39, 44, 49, 54, 59, 64, 69, 74, 79
Definition: acelp_vectors.c:79
uint8_t
Sparse representation for the algebraic codebook (fixed) vector.
Definition: acelp_vectors.h:53
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)
ptrdiff_t size
Definition: opengl_enc.c:101
static double alpha(void *priv, double x, double y)
Definition: vf_geq.c:99
static const uint16_t mask[17]
Definition: lzw.c:38
const float ff_pow_0_7[10]
Table of pow(0.7,n)
Definition: acelp_vectors.c:99
simple assert() macros that are a bit more flexible than ISO C assert().
GLsizei GLsizei * length
Definition: opengl_enc.c:115
int no_repeat_mask
Definition: acelp_vectors.h:57
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 float ff_pow_0_75[10]
Table of pow(0.75,n)
void(* 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.
Definition: acelp_vectors.h:40
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.
const uint8_t ff_fc_2pulses_9bits_track1[16]
Track|Pulse| Positions 1 | 0 | 1, 6, 11, 16, 21, 26, 31, 36 | | 3, 8, 13, 18, 23, 28...
Definition: acelp_vectors.c:31
float y[10]
Definition: acelp_vectors.h:56
int n
Definition: avisynth_c.h:547
FILE * out
Definition: movenc-test.c:54
Libavcodec external API header.
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
const int16_t * tab1
Definition: mace.c:144
const uint8_t ff_fc_4pulses_8bits_tracks_13[16]
Track|Pulse| Positions 1 | 0 | 0, 5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60, 65, 70, 75 2 | 1 | 1, 6, 11, 16, 21, 26, 31, 36, 41, 46, 51, 56, 61, 66, 71, 76 3 | 2 | 2, 7, 12, 17, 22, 27, 32, 37, 42, 47, 52, 57, 62, 67, 72, 77
Definition: acelp_vectors.c:74
void ff_acelp_vectors_init(ACELPVContext *c)
Initialize ACELPVContext.
const uint8_t ff_fc_2pulses_9bits_track2_gray[32]
Track|Pulse| Positions 2 | 1 | 0, 7, 14, 20, 27, 34, 1, 21 | | 2, 9, 15, 22, 29, 35, 6, 26 | | 4,10, 17, 24, 30, 37, 11, 31 | | 5,12, 19, 25, 32, 39, 16, 36
Definition: acelp_vectors.c:54
static const uint8_t gray_decode[8]
3-bit Gray code to binary lookup table
Definition: amrnbdata.h:1438
common internal and external API header
static double c[64]
int pitch_lag
Definition: acelp_vectors.h:58
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)
const int16_t * tab2
Definition: mace.c:144