FFmpeg
flacdsp_template.c
Go to the documentation of this file.
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  int a = in[0][i];
70  int 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  int a = in[0][i];
84  int 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  int 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 }
static int shift(int a, int b)
Definition: sonic.c:82
static void FUNC() flac_decorrelate_indep_c(uint8_t **out, int32_t **in, int channels, int len, int shift)
channels
Definition: aptx.c:30
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:36
Convenience header that includes libavutil&#39;s core.
static void FUNC() flac_decorrelate_ls_c(uint8_t **out, int32_t **in, int channels, int len, int shift)
#define sample
uint8_t
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
#define S(s, c, i)
static void FUNC() flac_decorrelate_rs_c(uint8_t **out, int32_t **in, int channels, int len, int shift)
#define b
Definition: input.c:41
int32_t
static void FUNC() flac_decorrelate_ms_c(uint8_t **out, int32_t **in, int channels, int len, int shift)
#define OUT(n)
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
int
#define FUNC(n)
int len
FILE * out
Definition: movenc.c:54
Filter the word “frame” indicates either a video frame or a group of audio samples