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celp_filters.c
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1 /*
2  * various filters 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 "avcodec.h"
26 #include "celp_filters.h"
27 #include "libavutil/avassert.h"
28 #include "libavutil/common.h"
29 
30 void ff_celp_convolve_circ(int16_t* fc_out, const int16_t* fc_in,
31  const int16_t* filter, int len)
32 {
33  int i, k;
34 
35  memset(fc_out, 0, len * sizeof(int16_t));
36 
37  /* Since there are few pulses over an entire subframe (i.e. almost
38  all fc_in[i] are zero) it is faster to loop over fc_in first. */
39  for (i = 0; i < len; i++) {
40  if (fc_in[i]) {
41  for (k = 0; k < i; k++)
42  fc_out[k] += (fc_in[i] * filter[len + k - i]) >> 15;
43 
44  for (k = i; k < len; k++)
45  fc_out[k] += (fc_in[i] * filter[ k - i]) >> 15;
46  }
47  }
48 }
49 
50 void ff_celp_circ_addf(float *out, const float *in,
51  const float *lagged, int lag, float fac, int n)
52 {
53  int k;
54  for (k = 0; k < lag; k++)
55  out[k] = in[k] + fac * lagged[n + k - lag];
56  for (; k < n; k++)
57  out[k] = in[k] + fac * lagged[ k - lag];
58 }
59 
60 int ff_celp_lp_synthesis_filter(int16_t *out, const int16_t *filter_coeffs,
61  const int16_t *in, int buffer_length,
62  int filter_length, int stop_on_overflow,
63  int shift, int rounder)
64 {
65  int i,n;
66 
67  for (n = 0; n < buffer_length; n++) {
68  int sum = -rounder, sum1;
69  for (i = 1; i <= filter_length; i++)
70  sum += filter_coeffs[i-1] * out[n-i];
71 
72  sum1 = ((-sum >> 12) + in[n]) >> shift;
73  sum = av_clip_int16(sum1);
74 
75  if (stop_on_overflow && sum != sum1)
76  return 1;
77 
78  out[n] = sum;
79  }
80 
81  return 0;
82 }
83 
84 void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs,
85  const float* in, int buffer_length,
86  int filter_length)
87 {
88  int i,n;
89 
90 #if 0 // Unoptimized code path for improved readability
91  for (n = 0; n < buffer_length; n++) {
92  out[n] = in[n];
93  for (i = 1; i <= filter_length; i++)
94  out[n] -= filter_coeffs[i-1] * out[n-i];
95  }
96 #else
97  float out0, out1, out2, out3;
98  float old_out0, old_out1, old_out2, old_out3;
99  float a,b,c;
100 
101  a = filter_coeffs[0];
102  b = filter_coeffs[1];
103  c = filter_coeffs[2];
104  b -= filter_coeffs[0] * filter_coeffs[0];
105  c -= filter_coeffs[1] * filter_coeffs[0];
106  c -= filter_coeffs[0] * b;
107 
108  av_assert2((filter_length&1)==0 && filter_length>=4);
109 
110  old_out0 = out[-4];
111  old_out1 = out[-3];
112  old_out2 = out[-2];
113  old_out3 = out[-1];
114  for (n = 0; n <= buffer_length - 4; n+=4) {
115  float tmp0,tmp1,tmp2;
116  float val;
117 
118  out0 = in[0];
119  out1 = in[1];
120  out2 = in[2];
121  out3 = in[3];
122 
123  out0 -= filter_coeffs[2] * old_out1;
124  out1 -= filter_coeffs[2] * old_out2;
125  out2 -= filter_coeffs[2] * old_out3;
126 
127  out0 -= filter_coeffs[1] * old_out2;
128  out1 -= filter_coeffs[1] * old_out3;
129 
130  out0 -= filter_coeffs[0] * old_out3;
131 
132  val = filter_coeffs[3];
133 
134  out0 -= val * old_out0;
135  out1 -= val * old_out1;
136  out2 -= val * old_out2;
137  out3 -= val * old_out3;
138 
139  for (i = 5; i < filter_length; i += 2) {
140  old_out3 = out[-i];
141  val = filter_coeffs[i-1];
142 
143  out0 -= val * old_out3;
144  out1 -= val * old_out0;
145  out2 -= val * old_out1;
146  out3 -= val * old_out2;
147 
148  old_out2 = out[-i-1];
149 
150  val = filter_coeffs[i];
151 
152  out0 -= val * old_out2;
153  out1 -= val * old_out3;
154  out2 -= val * old_out0;
155  out3 -= val * old_out1;
156 
157  FFSWAP(float, old_out0, old_out2);
158  old_out1 = old_out3;
159  }
160 
161  tmp0 = out0;
162  tmp1 = out1;
163  tmp2 = out2;
164 
165  out3 -= a * tmp2;
166  out2 -= a * tmp1;
167  out1 -= a * tmp0;
168 
169  out3 -= b * tmp1;
170  out2 -= b * tmp0;
171 
172  out3 -= c * tmp0;
173 
174 
175  out[0] = out0;
176  out[1] = out1;
177  out[2] = out2;
178  out[3] = out3;
179 
180  old_out0 = out0;
181  old_out1 = out1;
182  old_out2 = out2;
183  old_out3 = out3;
184 
185  out += 4;
186  in += 4;
187  }
188 
189  out -= n;
190  in -= n;
191  for (; n < buffer_length; n++) {
192  out[n] = in[n];
193  for (i = 1; i <= filter_length; i++)
194  out[n] -= filter_coeffs[i-1] * out[n-i];
195  }
196 #endif
197 }
198 
199 void ff_celp_lp_zero_synthesis_filterf(float *out, const float *filter_coeffs,
200  const float *in, int buffer_length,
201  int filter_length)
202 {
203  int i,n;
204 
205  for (n = 0; n < buffer_length; n++) {
206  out[n] = in[n];
207  for (i = 1; i <= filter_length; i++)
208  out[n] += filter_coeffs[i-1] * in[n-i];
209  }
210 }
211 
213 {
216 
217  if(HAVE_MIPSFPU)
219 }
const char const char void * val
Definition: avisynth_c.h:634
void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP synthesis filter.
Definition: celp_filters.c:84
static int shift(int a, int b)
Definition: sonic.c:82
int ff_celp_lp_synthesis_filter(int16_t *out, const int16_t *filter_coeffs, const int16_t *in, int buffer_length, int filter_length, int stop_on_overflow, int shift, int rounder)
LP synthesis filter.
Definition: celp_filters.c:60
const char * b
Definition: vf_curves.c:109
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:63
void(* celp_lp_synthesis_filterf)(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP synthesis filter.
Definition: celp_filters.h:45
simple assert() macros that are a bit more flexible than ISO C assert().
void ff_celp_convolve_circ(int16_t *fc_out, const int16_t *fc_in, const int16_t *filter, int len)
Circularly convolve fixed vector with a phase dispersion impulse response filter (D.6.2 of G.729 and 6.1.5 of AMR).
Definition: celp_filters.c:30
Libavcodec external API header.
int n
Definition: avisynth_c.h:547
void ff_celp_circ_addf(float *out, const float *in, const float *lagged, int lag, float fac, int n)
Add an array to a rotated array.
Definition: celp_filters.c:50
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
void ff_celp_filter_init(CELPFContext *c)
Initialize CELPFContext.
Definition: celp_filters.c:212
void ff_celp_lp_zero_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP zero synthesis filter.
Definition: celp_filters.c:199
void(* celp_lp_zero_synthesis_filterf)(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length)
LP zero synthesis filter.
Definition: celp_filters.h:65
static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
common internal and external API header
static double c[64]
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-> out
#define FFSWAP(type, a, b)
Definition: common.h:69
void ff_celp_filter_init_mips(CELPFContext *c)