Libav
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 Libav.
7  *
8  * Libav 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  * Libav 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 Libav; 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/common.h"
26 #include "libavutil/float_dsp.h"
27 #include "avcodec.h"
28 #include "acelp_vectors.h"
29 
31 {
32  1, 3,
33  6, 8,
34  11, 13,
35  16, 18,
36  21, 23,
37  26, 28,
38  31, 33,
39  36, 38
40 };
42 {
43  1, 3,
44  8, 6,
45  18, 16,
46  11, 13,
47  38, 36,
48  31, 33,
49  21, 23,
50  28, 26,
51 };
52 
54 {
55  0, 5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60, 65, 70, 75,
56 };
57 
59 {
60  3, 4,
61  8, 9,
62  13, 14,
63  18, 19,
64  23, 24,
65  28, 29,
66  33, 34,
67  38, 39,
68  43, 44,
69  48, 49,
70  53, 54,
71  58, 59,
72  63, 64,
73  68, 69,
74  73, 74,
75  78, 79,
76 };
77 
78 const float ff_pow_0_7[10] = {
79  0.700000, 0.490000, 0.343000, 0.240100, 0.168070,
80  0.117649, 0.082354, 0.057648, 0.040354, 0.028248
81 };
82 
83 const float ff_pow_0_75[10] = {
84  0.750000, 0.562500, 0.421875, 0.316406, 0.237305,
85  0.177979, 0.133484, 0.100113, 0.075085, 0.056314
86 };
87 
88 const float ff_pow_0_55[10] = {
89  0.550000, 0.302500, 0.166375, 0.091506, 0.050328,
90  0.027681, 0.015224, 0.008373, 0.004605, 0.002533
91 };
92 
93 const float ff_b60_sinc[61] = {
94  0.898529 , 0.865051 , 0.769257 , 0.624054 , 0.448639 , 0.265289 ,
95  0.0959167 , -0.0412598 , -0.134338 , -0.178986 , -0.178528 , -0.142609 ,
96 -0.0849304 , -0.0205078 , 0.0369568 , 0.0773926 , 0.0955200 , 0.0912781 ,
97  0.0689392 , 0.0357056 , 0.0 , -0.0305481 , -0.0504150 , -0.0570068 ,
98 -0.0508423 , -0.0350037 , -0.0141602 , 0.00665283, 0.0230713 , 0.0323486 ,
99  0.0335388 , 0.0275879 , 0.0167847 , 0.00411987, -0.00747681, -0.0156860 ,
100 -0.0193481 , -0.0183716 , -0.0137634 , -0.00704956, 0.0 , 0.00582886 ,
101  0.00939941, 0.0103760 , 0.00903320, 0.00604248, 0.00238037, -0.00109863 ,
102 -0.00366211, -0.00497437, -0.00503540, -0.00402832, -0.00241089, -0.000579834,
103  0.00103760, 0.00222778, 0.00277710, 0.00271606, 0.00213623, 0.00115967 ,
104  0.
105 };
106 
108  int16_t* fc_v,
109  const uint8_t *tab1,
110  const uint8_t *tab2,
111  int pulse_indexes,
112  int pulse_signs,
113  int pulse_count,
114  int bits)
115 {
116  int mask = (1 << bits) - 1;
117  int i;
118 
119  for(i=0; i<pulse_count; i++)
120  {
121  fc_v[i + tab1[pulse_indexes & mask]] +=
122  (pulse_signs & 1) ? 8191 : -8192; // +/-1 in (2.13)
123 
124  pulse_indexes >>= bits;
125  pulse_signs >>= 1;
126  }
127 
128  fc_v[tab2[pulse_indexes]] += (pulse_signs & 1) ? 8191 : -8192;
129 }
130 
131 void ff_decode_10_pulses_35bits(const int16_t *fixed_index,
132  AMRFixed *fixed_sparse,
133  const uint8_t *gray_decode,
134  int half_pulse_count, int bits)
135 {
136  int i;
137  int mask = (1 << bits) - 1;
138 
139  fixed_sparse->no_repeat_mask = 0;
140  fixed_sparse->n = 2 * half_pulse_count;
141  for (i = 0; i < half_pulse_count; i++) {
142  const int pos1 = gray_decode[fixed_index[2*i+1] & mask] + i;
143  const int pos2 = gray_decode[fixed_index[2*i ] & mask] + i;
144  const float sign = (fixed_index[2*i+1] & (1 << bits)) ? -1.0 : 1.0;
145  fixed_sparse->x[2*i+1] = pos1;
146  fixed_sparse->x[2*i ] = pos2;
147  fixed_sparse->y[2*i+1] = sign;
148  fixed_sparse->y[2*i ] = pos2 < pos1 ? -sign : sign;
149  }
150 }
151 
153  int16_t* out,
154  const int16_t *in_a,
155  const int16_t *in_b,
156  int16_t weight_coeff_a,
157  int16_t weight_coeff_b,
158  int16_t rounder,
159  int shift,
160  int length)
161 {
162  int i;
163 
164  // Clipping required here; breaks OVERFLOW test.
165  for(i=0; i<length; i++)
166  out[i] = av_clip_int16((
167  in_a[i] * weight_coeff_a +
168  in_b[i] * weight_coeff_b +
169  rounder) >> shift);
170 }
171 
172 void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b,
173  float weight_coeff_a, float weight_coeff_b, int length)
174 {
175  int i;
176 
177  for(i=0; i<length; i++)
178  out[i] = weight_coeff_a * in_a[i]
179  + weight_coeff_b * in_b[i];
180 }
181 
182 void ff_adaptive_gain_control(float *out, const float *in, float speech_energ,
183  int size, float alpha, float *gain_mem)
184 {
185  int i;
186  float postfilter_energ = avpriv_scalarproduct_float_c(in, in, size);
187  float gain_scale_factor = 1.0;
188  float mem = *gain_mem;
189 
190  if (postfilter_energ)
191  gain_scale_factor = sqrt(speech_energ / postfilter_energ);
192 
193  gain_scale_factor *= 1.0 - alpha;
194 
195  for (i = 0; i < size; i++) {
196  mem = alpha * mem + gain_scale_factor;
197  out[i] = in[i] * mem;
198  }
199 
200  *gain_mem = mem;
201 }
202 
204  float sum_of_squares, const int n)
205 {
206  int i;
207  float scalefactor = avpriv_scalarproduct_float_c(in, in, n);
208  if (scalefactor)
209  scalefactor = sqrt(sum_of_squares / scalefactor);
210  for (i = 0; i < n; i++)
211  out[i] = in[i] * scalefactor;
212 }
213 
214 void ff_set_fixed_vector(float *out, const AMRFixed *in, float scale, int size)
215 {
216  int i;
217 
218  for (i=0; i < in->n; i++) {
219  int x = in->x[i], repeats = !((in->no_repeat_mask >> i) & 1);
220  float y = in->y[i] * scale;
221 
222  do {
223  out[x] += y;
224  y *= in->pitch_fac;
225  x += in->pitch_lag;
226  } while (x < size && repeats);
227  }
228 }
229 
230 void ff_clear_fixed_vector(float *out, const AMRFixed *in, int size)
231 {
232  int i;
233 
234  for (i=0; i < in->n; i++) {
235  int x = in->x[i], repeats = !((in->no_repeat_mask >> i) & 1);
236 
237  do {
238  out[x] = 0.0;
239  x += in->pitch_lag;
240  } while (x < size && repeats);
241  }
242 }
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).
int size
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:31
const uint8_t ff_fc_2pulses_9bits_track1_gray[16]
Definition: acelp_vectors.c:41
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:35
float avpriv_scalarproduct_float_c(const float *v1, const float *v2, int len)
Return the scalar product of two vectors.
Definition: float_dsp.c:104
uint8_t bits
Definition: crc.c:216
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:58
uint8_t
Sparse representation for the algebraic codebook (fixed) vector.
Definition: acelp_vectors.h:29
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=av_sample_fmt_is_planar(in_fmt);out_planar=av_sample_fmt_is_planar(out_fmt);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_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_dlog(ac->avr,"%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
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)
static const uint16_t mask[17]
Definition: lzw.c:38
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=av_sample_fmt_is_planar(in_fmt);out_planar=av_sample_fmt_is_planar(out_fmt);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_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_dlog(ac->avr,"%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 float ff_pow_0_7[10]
Table of pow(0.7,n)
Definition: acelp_vectors.c:78
int no_repeat_mask
Definition: acelp_vectors.h:33
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)
Definition: acelp_vectors.c:83
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:30
float y[10]
Definition: acelp_vectors.h:32
Libavcodec external API header.
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:53
static const uint16_t scale[4]
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
int pitch_lag
Definition: acelp_vectors.h:34
const float ff_b60_sinc[61]
b60 hamming windowed sinc function coefficients
Definition: acelp_vectors.c:93
static const uint16_t rounder[4]
const float ff_pow_0_55[10]
Table of pow(0.55,n)
Definition: acelp_vectors.c:88
const int16_t * tab2
Definition: mace.c:144