FFmpeg  4.3
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 {
150  c->acelp_interpolatef = ff_acelp_interpolatef;
151  c->acelp_apply_order_2_transfer_function = ff_acelp_apply_order_2_transfer_function;
152 
153  if(HAVE_MIPSFPU)
155 }
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
out
FILE * out
Definition: movenc.c:54
ff_acelp_interp_filter
const int16_t ff_acelp_interp_filter[61]
low-pass Finite Impulse Response filter coefficients.
Definition: acelp_filters.c:30
ff_acelp_interpolate
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
tmp
static uint8_t tmp[11]
Definition: aes_ctr.c:26
ff_acelp_high_pass_filter
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
ff_acelp_apply_order_2_transfer_function
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.
Definition: acelp_filters.c:119
HAVE_MIPSFPU
#define HAVE_MIPSFPU
Definition: config.h:74
ff_acelp_filter_init
void ff_acelp_filter_init(ACELPFContext *c)
Initialize ACELPFContext.
Definition: acelp_filters.c:148
samples
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new samples
Definition: fate.txt:139
avassert.h
NULL
#define NULL
Definition: coverity.c:32
c
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
ACELPFContext
Definition: acelp_filters.h:28
size
int size
Definition: twinvq_data.h:11134
ff_tilt_compensation
void ff_tilt_compensation(float *mem, float tilt, float *samples, int size)
Apply tilt compensation filter, 1 - tilt * z-1.
Definition: acelp_filters.c:136
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
Definition: audio_convert.c:326
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
common.h
av_assert1
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
Definition: avassert.h:53
acelp_filters.h
avcodec.h
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
ff_acelp_filter_init_mips
void ff_acelp_filter_init_mips(ACELPFContext *c)
Definition: acelp_filters_mips.c:213
ff_acelp_interpolatef
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