FFmpeg  4.0
aacpsdsp.h
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1 /*
2  * Copyright (c) 2012 Mans Rullgard
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 #ifndef AVCODEC_AACPSDSP_H
22 #define AVCODEC_AACPSDSP_H
23 
24 #include <stddef.h>
25 
26 #include "aac_defines.h"
27 
28 #define PS_QMF_TIME_SLOTS 32
29 #define PS_AP_LINKS 3
30 #define PS_MAX_AP_DELAY 5
31 
32 typedef struct PSDSPContext {
33  void (*add_squares)(INTFLOAT *dst, const INTFLOAT (*src)[2], int n);
35  int n);
37  const INTFLOAT (*filter)[8][2],
38  ptrdiff_t stride, int n);
39  void (*hybrid_analysis_ileave)(INTFLOAT (*out)[32][2], INTFLOAT L[2][38][64],
40  int i, int len);
41  void (*hybrid_synthesis_deint)(INTFLOAT out[2][38][64], INTFLOAT (*in)[32][2],
42  int i, int len);
43  void (*decorrelate)(INTFLOAT (*out)[2], INTFLOAT (*delay)[2],
45  const INTFLOAT phi_fract[2], const INTFLOAT (*Q_fract)[2],
46  const INTFLOAT *transient_gain,
47  INTFLOAT g_decay_slope,
48  int len);
49  void (*stereo_interpolate[2])(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
50  INTFLOAT h[2][4], INTFLOAT h_step[2][4],
51  int len);
52 } PSDSPContext;
53 
59 
60 #endif /* AVCODEC_AACPSDSP_H */
void AAC_RENAME() ff_psdsp_init(PSDSPContext *s)
void(* mul_pair_single)(INTFLOAT(*dst)[2], INTFLOAT(*src0)[2], INTFLOAT *src1, int n)
Definition: aacpsdsp.h:34
const char * s
Definition: avisynth_c.h:768
void ff_psdsp_init_arm(PSDSPContext *s)
#define PS_MAX_AP_DELAY
Definition: aacpsdsp.h:30
#define src
Definition: vp8dsp.c:254
int stride
Definition: mace.c:144
void(* hybrid_synthesis_deint)(INTFLOAT out[2][38][64], INTFLOAT(*in)[32][2], int i, int len)
Definition: aacpsdsp.h:41
void ff_psdsp_init_aarch64(PSDSPContext *s)
float INTFLOAT
Definition: aac_defines.h:86
#define PS_QMF_TIME_SLOTS
Definition: aacpsdsp.h:28
static void filter(int16_t *output, ptrdiff_t out_stride, int16_t *low, ptrdiff_t low_stride, int16_t *high, ptrdiff_t high_stride, int len, int clip)
Definition: cfhd.c:114
void(* decorrelate)(INTFLOAT(*out)[2], INTFLOAT(*delay)[2], INTFLOAT(*ap_delay)[PS_QMF_TIME_SLOTS+PS_MAX_AP_DELAY][2], const INTFLOAT phi_fract[2], const INTFLOAT(*Q_fract)[2], const INTFLOAT *transient_gain, INTFLOAT g_decay_slope, int len)
Definition: aacpsdsp.h:43
void(* add_squares)(INTFLOAT *dst, const INTFLOAT(*src)[2], int n)
Definition: aacpsdsp.h:33
const char * r
Definition: vf_curves.c:111
void(* hybrid_analysis)(INTFLOAT(*out)[2], INTFLOAT(*in)[2], const INTFLOAT(*filter)[8][2], ptrdiff_t stride, int n)
Definition: aacpsdsp.h:36
static int phi_fract[2][50][2]
#define AAC_RENAME(x)
Definition: aac_defines.h:84
void(* stereo_interpolate[2])(INTFLOAT(*l)[2], INTFLOAT(*r)[2], INTFLOAT h[2][4], INTFLOAT h_step[2][4], int len)
Definition: aacpsdsp.h:49
int n
Definition: avisynth_c.h:684
#define L(x)
Definition: vp56_arith.h:36
#define INTFLOAT
void ff_psdsp_init_x86(PSDSPContext *s)
Definition: aacpsdsp_init.c:52
#define src1
Definition: h264pred.c:139
typedef void(RENAME(mix_any_func_type))
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 src0
Definition: h264pred.c:138
void ff_psdsp_init_mips(PSDSPContext *s)
void(* hybrid_analysis_ileave)(INTFLOAT(*out)[32][2], INTFLOAT L[2][38][64], int i, int len)
Definition: aacpsdsp.h:39
int len
FILE * out
Definition: movenc.c:54