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aacpsdsp_template.c
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1 /*
2  * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  *
20  * Note: Rounding-to-nearest used unless otherwise stated
21  *
22  */
23 #include <stdint.h>
24 
25 #include "config.h"
26 #include "libavutil/attributes.h"
27 #include "aacpsdsp.h"
28 
29 static void ps_add_squares_c(INTFLOAT *dst, const INTFLOAT (*src)[2], int n)
30 {
31  int i;
32  for (i = 0; i < n; i++)
33  dst[i] += (UINTFLOAT)AAC_MADD28(src[i][0], src[i][0], src[i][1], src[i][1]);
34 }
35 
36 static void ps_mul_pair_single_c(INTFLOAT (*dst)[2], INTFLOAT (*src0)[2], INTFLOAT *src1,
37  int n)
38 {
39  int i;
40  for (i = 0; i < n; i++) {
41  dst[i][0] = AAC_MUL16(src0[i][0], src1[i]);
42  dst[i][1] = AAC_MUL16(src0[i][1], src1[i]);
43  }
44 }
45 
46 static void ps_hybrid_analysis_c(INTFLOAT (*out)[2], INTFLOAT (*in)[2],
47  const INTFLOAT (*filter)[8][2],
48  ptrdiff_t stride, int n)
49 {
50  int i, j;
51 
52  for (i = 0; i < n; i++) {
53  INT64FLOAT sum_re = (INT64FLOAT)filter[i][6][0] * in[6][0];
54  INT64FLOAT sum_im = (INT64FLOAT)filter[i][6][0] * in[6][1];
55 
56  for (j = 0; j < 6; j++) {
57  INTFLOAT in0_re = in[j][0];
58  INTFLOAT in0_im = in[j][1];
59  INTFLOAT in1_re = in[12-j][0];
60  INTFLOAT in1_im = in[12-j][1];
61  sum_re += (INT64FLOAT)filter[i][j][0] * (in0_re + in1_re) -
62  (INT64FLOAT)filter[i][j][1] * (in0_im - in1_im);
63  sum_im += (INT64FLOAT)filter[i][j][0] * (in0_im + in1_im) +
64  (INT64FLOAT)filter[i][j][1] * (in0_re - in1_re);
65  }
66 #if USE_FIXED
67  out[i * stride][0] = (int)((sum_re + 0x40000000) >> 31);
68  out[i * stride][1] = (int)((sum_im + 0x40000000) >> 31);
69 #else
70  out[i * stride][0] = sum_re;
71  out[i * stride][1] = sum_im;
72 #endif /* USE_FIXED */
73  }
74 }
75 
76 static void ps_hybrid_analysis_ileave_c(INTFLOAT (*out)[32][2], INTFLOAT L[2][38][64],
77  int i, int len)
78 {
79  int j;
80 
81  for (; i < 64; i++) {
82  for (j = 0; j < len; j++) {
83  out[i][j][0] = L[0][j][i];
84  out[i][j][1] = L[1][j][i];
85  }
86  }
87 }
88 
89 static void ps_hybrid_synthesis_deint_c(INTFLOAT out[2][38][64],
90  INTFLOAT (*in)[32][2],
91  int i, int len)
92 {
93  int n;
94 
95  for (; i < 64; i++) {
96  for (n = 0; n < len; n++) {
97  out[0][n][i] = in[i][n][0];
98  out[1][n][i] = in[i][n][1];
99  }
100  }
101 }
102 
103 static void ps_decorrelate_c(INTFLOAT (*out)[2], INTFLOAT (*delay)[2],
104  INTFLOAT (*ap_delay)[PS_QMF_TIME_SLOTS + PS_MAX_AP_DELAY][2],
105  const INTFLOAT phi_fract[2], const INTFLOAT (*Q_fract)[2],
106  const INTFLOAT *transient_gain,
107  INTFLOAT g_decay_slope,
108  int len)
109 {
110  static const INTFLOAT a[] = { Q31(0.65143905753106f),
111  Q31(0.56471812200776f),
112  Q31(0.48954165955695f) };
113  INTFLOAT ag[PS_AP_LINKS];
114  int m, n;
115 
116  for (m = 0; m < PS_AP_LINKS; m++)
117  ag[m] = AAC_MUL30(a[m], g_decay_slope);
118 
119  for (n = 0; n < len; n++) {
120  INTFLOAT in_re = AAC_MSUB30(delay[n][0], phi_fract[0], delay[n][1], phi_fract[1]);
121  INTFLOAT in_im = AAC_MADD30(delay[n][0], phi_fract[1], delay[n][1], phi_fract[0]);
122  for (m = 0; m < PS_AP_LINKS; m++) {
123  INTFLOAT a_re = AAC_MUL31(ag[m], in_re);
124  INTFLOAT a_im = AAC_MUL31(ag[m], in_im);
125  INTFLOAT link_delay_re = ap_delay[m][n+2-m][0];
126  INTFLOAT link_delay_im = ap_delay[m][n+2-m][1];
127  INTFLOAT fractional_delay_re = Q_fract[m][0];
128  INTFLOAT fractional_delay_im = Q_fract[m][1];
129  INTFLOAT apd_re = in_re;
130  INTFLOAT apd_im = in_im;
131  in_re = AAC_MSUB30(link_delay_re, fractional_delay_re,
132  link_delay_im, fractional_delay_im);
133  in_re -= (UINTFLOAT)a_re;
134  in_im = AAC_MADD30(link_delay_re, fractional_delay_im,
135  link_delay_im, fractional_delay_re);
136  in_im -= (UINTFLOAT)a_im;
137  ap_delay[m][n+5][0] = apd_re + (UINTFLOAT)AAC_MUL31(ag[m], in_re);
138  ap_delay[m][n+5][1] = apd_im + (UINTFLOAT)AAC_MUL31(ag[m], in_im);
139  }
140  out[n][0] = AAC_MUL16(transient_gain[n], in_re);
141  out[n][1] = AAC_MUL16(transient_gain[n], in_im);
142  }
143 }
144 
145 static void ps_stereo_interpolate_c(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
146  INTFLOAT h[2][4], INTFLOAT h_step[2][4],
147  int len)
148 {
149  INTFLOAT h0 = h[0][0];
150  INTFLOAT h1 = h[0][1];
151  INTFLOAT h2 = h[0][2];
152  INTFLOAT h3 = h[0][3];
153  INTFLOAT hs0 = h_step[0][0];
154  INTFLOAT hs1 = h_step[0][1];
155  INTFLOAT hs2 = h_step[0][2];
156  INTFLOAT hs3 = h_step[0][3];
157  int n;
158 
159  for (n = 0; n < len; n++) {
160  //l is s, r is d
161  INTFLOAT l_re = l[n][0];
162  INTFLOAT l_im = l[n][1];
163  INTFLOAT r_re = r[n][0];
164  INTFLOAT r_im = r[n][1];
165  h0 += hs0;
166  h1 += hs1;
167  h2 += hs2;
168  h3 += hs3;
169  l[n][0] = AAC_MADD30(h0, l_re, h2, r_re);
170  l[n][1] = AAC_MADD30(h0, l_im, h2, r_im);
171  r[n][0] = AAC_MADD30(h1, l_re, h3, r_re);
172  r[n][1] = AAC_MADD30(h1, l_im, h3, r_im);
173  }
174 }
175 
177  INTFLOAT h[2][4], INTFLOAT h_step[2][4],
178  int len)
179 {
180  INTFLOAT h00 = h[0][0], h10 = h[1][0];
181  INTFLOAT h01 = h[0][1], h11 = h[1][1];
182  INTFLOAT h02 = h[0][2], h12 = h[1][2];
183  INTFLOAT h03 = h[0][3], h13 = h[1][3];
184  INTFLOAT hs00 = h_step[0][0], hs10 = h_step[1][0];
185  INTFLOAT hs01 = h_step[0][1], hs11 = h_step[1][1];
186  INTFLOAT hs02 = h_step[0][2], hs12 = h_step[1][2];
187  INTFLOAT hs03 = h_step[0][3], hs13 = h_step[1][3];
188  int n;
189 
190  for (n = 0; n < len; n++) {
191  //l is s, r is d
192  INTFLOAT l_re = l[n][0];
193  INTFLOAT l_im = l[n][1];
194  INTFLOAT r_re = r[n][0];
195  INTFLOAT r_im = r[n][1];
196  h00 += hs00;
197  h01 += hs01;
198  h02 += hs02;
199  h03 += hs03;
200  h10 += hs10;
201  h11 += hs11;
202  h12 += hs12;
203  h13 += hs13;
204 
205  l[n][0] = AAC_MSUB30_V8(h00, l_re, h02, r_re, h10, l_im, h12, r_im);
206  l[n][1] = AAC_MADD30_V8(h00, l_im, h02, r_im, h10, l_re, h12, r_re);
207  r[n][0] = AAC_MSUB30_V8(h01, l_re, h03, r_re, h11, l_im, h13, r_im);
208  r[n][1] = AAC_MADD30_V8(h01, l_im, h03, r_im, h11, l_re, h13, r_re);
209  }
210 }
211 
213 {
214  s->add_squares = ps_add_squares_c;
215  s->mul_pair_single = ps_mul_pair_single_c;
216  s->hybrid_analysis = ps_hybrid_analysis_c;
217  s->hybrid_analysis_ileave = ps_hybrid_analysis_ileave_c;
218  s->hybrid_synthesis_deint = ps_hybrid_synthesis_deint_c;
219  s->decorrelate = ps_decorrelate_c;
220  s->stereo_interpolate[0] = ps_stereo_interpolate_c;
221  s->stereo_interpolate[1] = ps_stereo_interpolate_ipdopd_c;
222 
223 #if !USE_FIXED
224  if (ARCH_ARM)
226  if (ARCH_AARCH64)
228  if (ARCH_MIPS)
230  if (ARCH_X86)
232 #endif /* !USE_FIXED */
233 }
float UINTFLOAT
Definition: aac_defines.h:87
const char * s
Definition: avisynth_c.h:768
void ff_psdsp_init_arm(PSDSPContext *s)
static void ps_add_squares_c(INTFLOAT *dst, const INTFLOAT(*src)[2], int n)
#define src
Definition: vp8dsp.c:254
void ff_psdsp_init_aarch64(PSDSPContext *s)
Macro definitions for various function/variable attributes.
float INTFLOAT
Definition: aac_defines.h:86
static void filter(int16_t *output, ptrdiff_t out_stride, int16_t *low, ptrdiff_t low_stride, int16_t *high, ptrdiff_t high_stride, int len, int clip)
Definition: cfhd.c:97
#define av_cold
Definition: attributes.h:82
#define PS_MAX_AP_DELAY
Definition: aacps.h:39
#define AAC_MADD28(x, y, a, b)
Definition: aac_defines.h:103
#define AAC_MUL31(x, y)
Definition: aac_defines.h:102
static void ps_hybrid_synthesis_deint_c(INTFLOAT out[2][38][64], INTFLOAT(*in)[32][2], int i, int len)
av_cold void AAC_RENAME() ff_psdsp_init(PSDSPContext *s)
#define AAC_MSUB30(x, y, a, b)
Definition: aac_defines.h:107
const char * r
Definition: vf_curves.c:111
static void ps_stereo_interpolate_c(INTFLOAT(*l)[2], INTFLOAT(*r)[2], INTFLOAT h[2][4], INTFLOAT h_step[2][4], int len)
static int phi_fract[2][50][2]
#define AAC_MUL16(x, y)
Definition: aac_defines.h:99
#define AAC_RENAME(x)
Definition: aac_defines.h:84
#define Q31(x)
Definition: aac_defines.h:96
#define AAC_MSUB30_V8(x, y, a, b, c, d, e, f)
Definition: aac_defines.h:108
static void ps_hybrid_analysis_c(INTFLOAT(*out)[2], INTFLOAT(*in)[2], const INTFLOAT(*filter)[8][2], ptrdiff_t stride, int n)
int n
Definition: avisynth_c.h:684
#define L(x)
Definition: vp56_arith.h:36
void ff_psdsp_init_x86(PSDSPContext *s)
Definition: aacpsdsp_init.c:52
#define src1
Definition: h264pred.c:139
static void ps_mul_pair_single_c(INTFLOAT(*dst)[2], INTFLOAT(*src0)[2], INTFLOAT *src1, int n)
static void ps_stereo_interpolate_ipdopd_c(INTFLOAT(*l)[2], INTFLOAT(*r)[2], INTFLOAT h[2][4], INTFLOAT h_step[2][4], int len)
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
#define AAC_MUL30(x, y)
Definition: aac_defines.h:101
#define src0
Definition: h264pred.c:138
void ff_psdsp_init_mips(PSDSPContext *s)
#define AAC_MADD30(x, y, a, b)
Definition: aac_defines.h:104
#define PS_AP_LINKS
Definition: aacps.h:38
int
GLint GLenum GLboolean GLsizei stride
Definition: opengl_enc.c:105
#define PS_QMF_TIME_SLOTS
Definition: aacps.h:36
int len
FILE * out
Definition: movenc.c:54
static void ps_hybrid_analysis_ileave_c(INTFLOAT(*out)[32][2], INTFLOAT L[2][38][64], int i, int len)
#define stride
static void ps_decorrelate_c(INTFLOAT(*out)[2], INTFLOAT(*delay)[2], INTFLOAT(*ap_delay)[PS_QMF_TIME_SLOTS+PS_MAX_AP_DELAY][2], const INTFLOAT phi_fract[2], const INTFLOAT(*Q_fract)[2], const INTFLOAT *transient_gain, INTFLOAT g_decay_slope, int len)
#define AAC_MADD30_V8(x, y, a, b, c, d, e, f)
Definition: aac_defines.h:105
float INT64FLOAT
Definition: aac_defines.h:88