FFmpeg
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  INT64FLOAT in0_re = in[j][0];
58  INT64FLOAT in0_im = in[j][1];
59  INT64FLOAT in1_re = in[12-j][0];
60  INT64FLOAT 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  UINTFLOAT hs0 = h_step[0][0];
154  UINTFLOAT hs1 = h_step[0][1];
155  UINTFLOAT hs2 = h_step[0][2];
156  UINTFLOAT 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 
176 static void ps_stereo_interpolate_ipdopd_c(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
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  UINTFLOAT hs00 = h_step[0][0], hs10 = h_step[1][0];
185  UINTFLOAT hs01 = h_step[0][1], hs11 = h_step[1][1];
186  UINTFLOAT hs02 = h_step[0][2], hs12 = h_step[1][2];
187  UINTFLOAT 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 
212 av_cold void AAC_RENAME(ff_psdsp_init)(PSDSPContext *s)
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)
225  ff_psdsp_init_arm(s);
226  if (ARCH_AARCH64)
227  ff_psdsp_init_aarch64(s);
228  if (ARCH_MIPS)
229  ff_psdsp_init_mips(s);
230  if (ARCH_X86)
231  ff_psdsp_init_x86(s);
232 #endif /* !USE_FIXED */
233 }
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Definition: aac_defines.h:104
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void ff_psdsp_init_x86(PSDSPContext *s)
Definition: aacpsdsp_init.c:52
PS_QMF_TIME_SLOTS
#define PS_QMF_TIME_SLOTS
Definition: aacps.h:36
filter
filter_frame For filters that do not use the this method is called when a frame is pushed to the filter s input It can be called at any time except in a reentrant way If the input frame is enough to produce then the filter should push the output frames on the output link immediately As an exception to the previous rule if the input frame is enough to produce several output frames then the filter needs output only at least one per link The additional frames can be left buffered in the filter
Definition: filter_design.txt:228
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void ff_psdsp_init_mips(PSDSPContext *s)
Definition: aacpsdsp_mips.c:450
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Definition: aacpsdsp_template.c:145
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void ff_psdsp_init_arm(PSDSPContext *s)
Definition: aacpsdsp_init_arm.c:46
ps_hybrid_analysis_ileave_c
static void ps_hybrid_analysis_ileave_c(INTFLOAT(*out)[32][2], INTFLOAT L[2][38][64], int i, int len)
Definition: aacpsdsp_template.c:76
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#define src
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Definition: aac_defines.h:102
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Definition: aacpsdsp_init_aarch64.c:37
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Definition: aac_defines.h:101
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Definition: aac_defines.h:107
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Definition: cbs_vp9.c:255
ps_stereo_interpolate_ipdopd_c
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Definition: aacpsdsp_template.c:176
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Definition: aacpsdsp_template.c:29
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Definition: aacps.h:39
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Definition: aacpsdsp_template.c:89
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static void ps_hybrid_analysis_c(INTFLOAT(*out)[2], INTFLOAT(*in)[2], const INTFLOAT(*filter)[8][2], ptrdiff_t stride, int n)
Definition: aacpsdsp_template.c:46
ps_decorrelate_c
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)
Definition: aacpsdsp_template.c:103
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
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float INT64FLOAT
Definition: aac_defines.h:88
attributes.h
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#define src0
Definition: h264pred.c:138
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Definition: h264pred.c:139
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
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Definition: aac_defines.h:105
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Definition: aac_defines.h:99
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Definition: aacpsdsp_template.c:36
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Definition: aacpsdsp_template.c:212
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Definition: vp56_arith.h:36
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static int phi_fract[2][50][2]
Definition: aacps_fixed_tablegen.h:56
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Definition: aacpsdsp.h:32
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Definition: aac_defines.h:96
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#define AAC_MSUB30_V8(x, y, a, b, c, d, e, f)
Definition: aac_defines.h:108
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Definition: vp9dsp_template.c:2038
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Definition: aac_defines.h:103
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Definition: aac_defines.h:86
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