FFmpeg  4.0
flacdsp.c
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1 /*
2  * Copyright (c) 2015 James Almer
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or modify
7  * it under the terms of the GNU General Public License as published by
8  * the Free Software Foundation; either version 2 of the License, or
9  * (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
14  * GNU General Public License for more details.
15  *
16  * You should have received a copy of the GNU General Public License along
17  * with FFmpeg; if not, write to the Free Software Foundation, Inc.,
18  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
19  */
20 
21 #include <string.h>
22 #include "checkasm.h"
23 #include "libavcodec/flacdsp.h"
24 #include "libavutil/common.h"
25 #include "libavutil/internal.h"
26 #include "libavutil/intreadwrite.h"
27 
28 #define BUF_SIZE 256
29 #define MAX_CHANNELS 8
30 
31 #define randomize_buffers() \
32  do { \
33  int i, j; \
34  for (i = 0; i < BUF_SIZE; i += 4) { \
35  for (j = 0; j < channels; j++) { \
36  uint32_t r = rnd() & (1 << (bits - 2)) - 1; \
37  AV_WN32A(ref_src[j] + i, r); \
38  AV_WN32A(new_src[j] + i, r); \
39  } \
40  } \
41  } while (0)
42 
43 static void check_decorrelate(uint8_t **ref_dst, uint8_t **ref_src, uint8_t **new_dst, uint8_t **new_src,
44  int channels, int bits) {
45  declare_func(void, uint8_t **out, int32_t **in, int channels, int len, int shift);
46 
48  call_ref(ref_dst, (int32_t **)ref_src, channels, BUF_SIZE / sizeof(int32_t), 8);
49  call_new(new_dst, (int32_t **)new_src, channels, BUF_SIZE / sizeof(int32_t), 8);
50  if (memcmp(*ref_dst, *new_dst, bits == 16 ? BUF_SIZE * (channels/2) : BUF_SIZE * channels) ||
51  memcmp(*ref_src, *new_src, BUF_SIZE * channels))
52  fail();
53  bench_new(new_dst, (int32_t **)new_src, channels, BUF_SIZE / sizeof(int32_t), 8);
54 }
55 
57 {
59  LOCAL_ALIGNED_16(uint8_t, ref_buf, [BUF_SIZE*MAX_CHANNELS]);
60  LOCAL_ALIGNED_16(uint8_t, new_dst, [BUF_SIZE*MAX_CHANNELS]);
61  LOCAL_ALIGNED_16(uint8_t, new_buf, [BUF_SIZE*MAX_CHANNELS]);
62  uint8_t *ref_src[] = { &ref_buf[BUF_SIZE*0], &ref_buf[BUF_SIZE*1], &ref_buf[BUF_SIZE*2], &ref_buf[BUF_SIZE*3],
63  &ref_buf[BUF_SIZE*4], &ref_buf[BUF_SIZE*5], &ref_buf[BUF_SIZE*6], &ref_buf[BUF_SIZE*7] };
64  uint8_t *new_src[] = { &new_buf[BUF_SIZE*0], &new_buf[BUF_SIZE*1], &new_buf[BUF_SIZE*2], &new_buf[BUF_SIZE*3],
65  &new_buf[BUF_SIZE*4], &new_buf[BUF_SIZE*5], &new_buf[BUF_SIZE*6], &new_buf[BUF_SIZE*7] };
66  static const char * const names[3] = { "ls", "rs", "ms" };
67  static const struct {
68  enum AVSampleFormat fmt;
69  int bits;
70  } fmts[] = {
71  { AV_SAMPLE_FMT_S16, 16 },
72  { AV_SAMPLE_FMT_S32, 32 },
73  };
75  int i, j;
76 
77  for (i = 0; i < 2; i++) {
78  ff_flacdsp_init(&h, fmts[i].fmt, 2, 0);
79  for (j = 0; j < 3; j++)
80  if (check_func(h.decorrelate[j], "flac_decorrelate_%s_%d", names[j], fmts[i].bits))
81  check_decorrelate(&ref_dst, ref_src, &new_dst, new_src, 2, fmts[i].bits);
82  for (j = 2; j <= MAX_CHANNELS; j += 2) {
83  ff_flacdsp_init(&h, fmts[i].fmt, j, 0);
84  if (check_func(h.decorrelate[0], "flac_decorrelate_indep%d_%d", j, fmts[i].bits))
85  check_decorrelate(&ref_dst, ref_src, &new_dst, new_src, j, fmts[i].bits);
86  }
87  }
88 
89  report("decorrelate");
90 }
static int shift(int a, int b)
Definition: sonic.c:82
#define randomize_buffers()
Definition: flacdsp.c:31
const char * fmt
Definition: avisynth_c.h:769
av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps)
Definition: flacdsp.c:88
channels
Definition: aptx.c:30
#define report
Definition: checkasm.h:119
uint8_t
signed 32 bits
Definition: samplefmt.h:62
#define declare_func(ret,...)
Definition: checkasm.h:111
#define fail()
Definition: checkasm.h:116
common internal API header
int32_t
void checkasm_check_flacdsp(void)
Definition: flacdsp.c:56
#define MAX_CHANNELS
Definition: flacdsp.c:29
#define call_ref(...)
Definition: checkasm.h:122
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
#define BUF_SIZE
Definition: flacdsp.c:28
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)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
#define check_func(func,...)
Definition: checkasm.h:107
common internal and external API header
static void check_decorrelate(uint8_t **ref_dst, uint8_t **ref_src, uint8_t **new_dst, uint8_t **new_src, int channels, int bits)
Definition: flacdsp.c:43
signed 16 bits
Definition: samplefmt.h:61
int len
#define bench_new(...)
Definition: checkasm.h:249
FILE * out
Definition: movenc.c:54
#define LOCAL_ALIGNED_16(t, v,...)
Definition: internal.h:131
#define call_new(...)
Definition: checkasm.h:189
void(* decorrelate[4])(uint8_t **out, int32_t **in, int channels, int len, int shift)
Definition: flacdsp.h:27