FFmpeg  4.3
flacdsp_template.c
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1 /*
2  * Copyright (c) 2012 Mans Rullgard <mans@mansr.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 
21 #include <stdint.h>
22 #include "libavutil/avutil.h"
23 
24 #undef FUNC
25 #undef FSUF
26 #undef sample
27 #undef sample_type
28 #undef OUT
29 #undef S
30 
31 #if SAMPLE_SIZE == 32
32 # define sample_type int32_t
33 #else
34 # define sample_type int16_t
35 #endif
36 
37 #if PLANAR
38 # define FSUF AV_JOIN(SAMPLE_SIZE, p)
39 # define sample sample_type *
40 # define OUT(n) n
41 # define S(s, c, i) (s[c][i])
42 #else
43 # define FSUF SAMPLE_SIZE
44 # define sample sample_type
45 # define OUT(n) n[0]
46 # define S(s, c, i) (*s++)
47 #endif
48 
49 #define FUNC(n) AV_JOIN(n ## _, FSUF)
50 
52  int channels, int len, int shift)
53 {
54  sample *samples = (sample *) OUT(out);
55  int i, j;
56 
57  for (j = 0; j < len; j++)
58  for (i = 0; i < channels; i++)
59  S(samples, i, j) = (int)((unsigned)in[i][j] << shift);
60 }
61 
63  int channels, int len, int shift)
64 {
65  sample *samples = (sample *) OUT(out);
66  int i;
67 
68  for (i = 0; i < len; i++) {
69  unsigned a = in[0][i];
70  unsigned b = in[1][i];
71  S(samples, 0, i) = a << shift;
72  S(samples, 1, i) = (a - b) << shift;
73  }
74 }
75 
77  int channels, int len, int shift)
78 {
79  sample *samples = (sample *) OUT(out);
80  int i;
81 
82  for (i = 0; i < len; i++) {
83  unsigned a = in[0][i];
84  unsigned b = in[1][i];
85  S(samples, 0, i) = (a + b) << shift;
86  S(samples, 1, i) = b << shift;
87  }
88 }
89 
91  int channels, int len, int shift)
92 {
93  sample *samples = (sample *) OUT(out);
94  int i;
95 
96  for (i = 0; i < len; i++) {
97  unsigned a = in[0][i];
98  int b = in[1][i];
99  a -= b >> 1;
100  S(samples, 0, i) = (a + b) << shift;
101  S(samples, 1, i) = a << shift;
102  }
103 }
flac_decorrelate_rs_c
static void FUNC() flac_decorrelate_rs_c(uint8_t **out, int32_t **in, int channels, int len, int shift)
Definition: flacdsp_template.c:76
out
FILE * out
Definition: movenc.c:54
b
#define b
Definition: input.c:41
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
FUNC
#define FUNC(n)
Definition: flacdsp_template.c:49
S
#define S(s, c, i)
Definition: flacdsp_template.c:46
flac_decorrelate_ms_c
static void FUNC() flac_decorrelate_ms_c(uint8_t **out, int32_t **in, int channels, int len, int shift)
Definition: flacdsp_template.c:90
channels
channels
Definition: aptx.h:33
OUT
#define OUT(n)
Definition: flacdsp_template.c:45
int32_t
int32_t
Definition: audio_convert.c:194
flac_decorrelate_ls_c
static void FUNC() flac_decorrelate_ls_c(uint8_t **out, int32_t **in, int channels, int len, int shift)
Definition: flacdsp_template.c:62
sample
#define sample
Definition: flacdsp_template.c:44
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
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
uint8_t
uint8_t
Definition: audio_convert.c:194
len
int len
Definition: vorbis_enc_data.h:452
shift
static int shift(int a, int b)
Definition: sonic.c:82
avutil.h
flac_decorrelate_indep_c
static void FUNC() flac_decorrelate_indep_c(uint8_t **out, int32_t **in, int channels, int len, int shift)
Definition: flacdsp_template.c:51
int
int
Definition: ffmpeg_filter.c:192