FFmpeg
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
void ff_psdsp_init_arm(PSDSPContext *s)
#define PS_MAX_AP_DELAY
Definition: aacpsdsp.h:30
#define src
Definition: vp8dsp.c:254
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
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
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
filter_frame For filters that do not use the this method is called when a frame is pushed to the filter s input It can be called at any time except in a reentrant way If the input frame is enough to produce then the filter should push the output frames on the output link immediately As an exception to the previous rule if the input frame is enough to produce several output frames then the filter needs output only at least one per link The additional frames can be left buffered in the filter
const char * r
Definition: vf_curves.c:114
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
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
#define s(width, name)
Definition: cbs_vp9.c:257
int n
Definition: avisynth_c.h:760
#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
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
#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
#define stride