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acelp_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 "libavutil/avassert.h"
26 #include "libavutil/common.h"
27 #include "avcodec.h"
28 #include "acelp_filters.h"
29 
30 const int16_t ff_acelp_interp_filter[61] = { /* (0.15) */
31  29443, 28346, 25207, 20449, 14701, 8693,
32  3143, -1352, -4402, -5865, -5850, -4673,
33  -2783, -672, 1211, 2536, 3130, 2991,
34  2259, 1170, 0, -1001, -1652, -1868,
35  -1666, -1147, -464, 218, 756, 1060,
36  1099, 904, 550, 135, -245, -514,
37  -634, -602, -451, -231, 0, 191,
38  308, 340, 296, 198, 78, -36,
39  -120, -163, -165, -132, -79, -19,
40  34, 73, 91, 89, 70, 38,
41  0,
42 };
43 
44 void ff_acelp_interpolate(int16_t* out, const int16_t* in,
45  const int16_t* filter_coeffs, int precision,
46  int frac_pos, int filter_length, int length)
47 {
48  int n, i;
49 
50  av_assert1(frac_pos >= 0 && frac_pos < precision);
51 
52  for (n = 0; n < length; n++) {
53  int idx = 0;
54  int v = 0x4000;
55 
56  for (i = 0; i < filter_length;) {
57 
58  /* The reference G.729 and AMR fixed point code performs clipping after
59  each of the two following accumulations.
60  Since clipping affects only the synthetic OVERFLOW test without
61  causing an int type overflow, it was moved outside the loop. */
62 
63  /* R(x):=ac_v[-k+x]
64  v += R(n-i)*ff_acelp_interp_filter(t+6i)
65  v += R(n+i+1)*ff_acelp_interp_filter(6-t+6i) */
66 
67  v += in[n + i] * filter_coeffs[idx + frac_pos];
68  idx += precision;
69  i++;
70  v += in[n - i] * filter_coeffs[idx - frac_pos];
71  }
72  if (av_clip_int16(v >> 15) != (v >> 15))
73  av_log(NULL, AV_LOG_WARNING, "overflow that would need clipping in ff_acelp_interpolate()\n");
74  out[n] = v >> 15;
75  }
76 }
77 
78 void ff_acelp_interpolatef(float *out, const float *in,
79  const float *filter_coeffs, int precision,
80  int frac_pos, int filter_length, int length)
81 {
82  int n, i;
83 
84  for (n = 0; n < length; n++) {
85  int idx = 0;
86  float v = 0;
87 
88  for (i = 0; i < filter_length;) {
89  v += in[n + i] * filter_coeffs[idx + frac_pos];
90  idx += precision;
91  i++;
92  v += in[n - i] * filter_coeffs[idx - frac_pos];
93  }
94  out[n] = v;
95  }
96 }
97 
98 
99 void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
100  const int16_t* in, int length)
101 {
102  int i;
103  int tmp;
104 
105  for (i = 0; i < length; i++) {
106  tmp = (hpf_f[0]* 15836LL) >> 13;
107  tmp += (hpf_f[1]* -7667LL) >> 13;
108  tmp += 7699 * (in[i] - 2*in[i-1] + in[i-2]);
109 
110  /* With "+0x800" rounding, clipping is needed
111  for ALGTHM and SPEECH tests. */
112  out[i] = av_clip_int16((tmp + 0x800) >> 12);
113 
114  hpf_f[1] = hpf_f[0];
115  hpf_f[0] = tmp;
116  }
117 }
118 
120  const float zero_coeffs[2],
121  const float pole_coeffs[2],
122  float gain, float mem[2], int n)
123 {
124  int i;
125  float tmp;
126 
127  for (i = 0; i < n; i++) {
128  tmp = gain * in[i] - pole_coeffs[0] * mem[0] - pole_coeffs[1] * mem[1];
129  out[i] = tmp + zero_coeffs[0] * mem[0] + zero_coeffs[1] * mem[1];
130 
131  mem[1] = mem[0];
132  mem[0] = tmp;
133  }
134 }
135 
136 void ff_tilt_compensation(float *mem, float tilt, float *samples, int size)
137 {
138  float new_tilt_mem = samples[size - 1];
139  int i;
140 
141  for (i = size - 1; i > 0; i--)
142  samples[i] -= tilt * samples[i - 1];
143 
144  samples[0] -= tilt * *mem;
145  *mem = new_tilt_mem;
146 }
147 
149 {
152 
153  if(HAVE_MIPSFPU)
155 }
void ff_acelp_high_pass_filter(int16_t *out, int hpf_f[2], const int16_t *in, int length)
high-pass filtering and upscaling (4.2.5 of G.729).
Definition: acelp_filters.c:99
void ff_acelp_filter_init_mips(ACELPFContext *c)
#define NULL
Definition: coverity.c:32
void ff_acelp_apply_order_2_transfer_function(float *out, const float *in, const float zero_coeffs[2], const float pole_coeffs[2], float gain, float mem[2], int n)
Apply an order 2 rational transfer function in-place.
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
const int16_t ff_acelp_interp_filter[61]
low-pass Finite Impulse Response filter coefficients.
Definition: acelp_filters.c:30
void(* acelp_apply_order_2_transfer_function)(float *out, const float *in, const float zero_coeffs[2], const float pole_coeffs[2], float gain, float mem[2], int n)
Apply an order 2 rational transfer function in-place.
Definition: acelp_filters.h:47
int mem
Definition: avisynth_c.h:684
ptrdiff_t size
Definition: opengl_enc.c:101
#define av_log(a,...)
simple assert() macros that are a bit more flexible than ISO C assert().
GLsizei GLsizei * length
Definition: opengl_enc.c:115
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
Definition: avassert.h:53
static uint8_t tmp[8]
Definition: des-test.c:38
int n
Definition: avisynth_c.h:547
void ff_tilt_compensation(float *mem, float tilt, float *samples, int size)
Apply tilt compensation filter, 1 - tilt * z-1.
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
void ff_acelp_interpolate(int16_t *out, const int16_t *in, const int16_t *filter_coeffs, int precision, int frac_pos, int filter_length, int length)
Generic FIR interpolation routine.
Definition: acelp_filters.c:44
common internal and external API header
static double c[64]
void ff_acelp_filter_init(ACELPFContext *c)
Initialize ACELPFContext.
void ff_acelp_interpolatef(float *out, const float *in, const float *filter_coeffs, int precision, int frac_pos, int filter_length, int length)
Floating point version of ff_acelp_interpolate()
Definition: acelp_filters.c:78
void(* acelp_interpolatef)(float *out, const float *in, const float *filter_coeffs, int precision, int frac_pos, int filter_length, int length)
Floating point version of ff_acelp_interpolate()
Definition: acelp_filters.h:32