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 cliping 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)
float v
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.
const int16_t ff_acelp_interp_filter[61]
low-pass Finite Impulse Response filter coefficients.
Definition: acelp_filters.c:30
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
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:154
About Git write you should know how to use GIT properly Luckily Git comes with excellent documentation git help man git shows you the available git< command > help man git< command > shows information about the subcommand< command > The most comprehensive manual is the website Git Reference visit they are quite exhaustive You do not need a special username or password All you need is to provide a ssh public key to the Git server admin What follows now is a basic introduction to Git and some FFmpeg specific guidelines Read it at least if you are granted commit privileges to the FFmpeg project you are expected to be familiar with these rules I if not You can get git from etc no matter how small Every one of them has been saved from looking like a fool by this many times It s very easy for stray debug output or cosmetic modifications to slip in
Definition: git-howto.txt:5
int mem
Definition: avisynth_c.h:721
simple assert() macros that are a bit more flexible than ISO C assert().
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:246
external API header
int size
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
Definition: avassert.h:53
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
void ff_tilt_compensation(float *mem, float tilt, float *samples, int size)
Apply tilt compensation filter, 1 - tilt * z-1.
NULL
Definition: eval.c:55
synthesis window for stochastic i
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.
#define HAVE_MIPSFPU
Definition: config.h:59
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
Filter the word “frame” indicates either a video frame or a group of audio samples
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=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;}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_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_dlog(ac->avr,"%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-> out
const char int length
Definition: avisynth_c.h:668