aacpsdsp.h
Go to the documentation of this file.
1 /*
2  * Copyright (c) 2012 Mans Rullgard
3  *
4  * This file is part of Libav.
5  *
6  * Libav 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  * Libav 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 Libav; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #ifndef LIBAVCODEC_AACPSDSP_H
22 #define LIBAVCODEC_AACPSDSP_H
23 
24 #define PS_QMF_TIME_SLOTS 32
25 #define PS_AP_LINKS 3
26 #define PS_MAX_AP_DELAY 5
27 
28 typedef struct PSDSPContext {
29  void (*add_squares)(float *dst, const float (*src)[2], int n);
30  void (*mul_pair_single)(float (*dst)[2], float (*src0)[2], float *src1,
31  int n);
32  void (*hybrid_analysis)(float (*out)[2], float (*in)[2],
33  const float (*filter)[8][2],
34  int stride, int n);
35  void (*hybrid_analysis_ileave)(float (*out)[32][2], float L[2][38][64],
36  int i, int len);
37  void (*hybrid_synthesis_deint)(float out[2][38][64], float (*in)[32][2],
38  int i, int len);
39  void (*decorrelate)(float (*out)[2], float (*delay)[2],
40  float (*ap_delay)[PS_QMF_TIME_SLOTS+PS_MAX_AP_DELAY][2],
41  const float phi_fract[2], float (*Q_fract)[2],
42  const float *transient_gain,
43  float g_decay_slope,
44  int len);
45  void (*stereo_interpolate[2])(float (*l)[2], float (*r)[2],
46  float h[2][4], float h_step[2][4],
47  int len);
48 } PSDSPContext;
49 
50 void ff_psdsp_init(PSDSPContext *s);
51 void ff_psdsp_init_arm(PSDSPContext *s);
52 void ff_psdsp_init_mips(PSDSPContext *s);
53 
54 #endif /* LIBAVCODEC_AACPSDSP_H */
s
const char * s
Definition: avisynth_c.h:668
ff_psdsp_init_arm
void ff_psdsp_init_arm(PSDSPContext *s)
Definition: aacpsdsp_init_arm.c:46
PS_MAX_AP_DELAY
#define PS_MAX_AP_DELAY
Definition: aacpsdsp.h:26
in
About Git write you should know how to use GIT properly Luckily Git comes with excellent documentation git help man git shows you the available git< command > help man git< command > shows information about the subcommand< command > The most comprehensive manual is the website Git Reference visit they are quite exhaustive You do not need a special username or password All you need is to provide a ssh public key to the Git server admin What follows now is a basic introduction to Git and some FFmpeg specific guidelines Read it at least if you are granted commit privileges to the FFmpeg project you are expected to be familiar with these rules I if not You can get git from etc no matter how small Every one of them has been saved from looking like a fool by this many times It s very easy for stray debug output or cosmetic modifications to slip in
Definition: git-howto.txt:5
ff_psdsp_init
void ff_psdsp_init(PSDSPContext *s)
Definition: aacpsdsp.c:201
stride
int stride
Definition: mace.c:144
PS_QMF_TIME_SLOTS
#define PS_QMF_TIME_SLOTS
Definition: aacpsdsp.h:24
PSDSPContext::stereo_interpolate
void(* stereo_interpolate[2])(float(*l)[2], float(*r)[2], float h[2][4], float h_step[2][4], int len)
Definition: aacpsdsp.h:45
PSDSPContext::mul_pair_single
void(* mul_pair_single)(float(*dst)[2], float(*src0)[2], float *src1, int n)
Definition: aacpsdsp.h:30
filter
the mask is usually to keep the same permissions Filters should remove permissions on reference they give to output whenever necessary It can be automatically done by setting the rej_perms field on the output pad Here are a few guidelines corresponding to common 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
Definition: filter_design.txt:201
PSDSPContext::hybrid_synthesis_deint
void(* hybrid_synthesis_deint)(float out[2][38][64], float(*in)[32][2], int i, int len)
Definition: aacpsdsp.h:37
r
const char * r
Definition: vf_curves.c:94
L
#define L(x)
Definition: arm/vp56_arith.h:36
PSDSPContext::hybrid_analysis
void(* hybrid_analysis)(float(*out)[2], float(*in)[2], const float(*filter)[8][2], int stride, int n)
Definition: aacpsdsp.h:32
PSDSPContext::hybrid_analysis_ileave
void(* hybrid_analysis_ileave)(float(*out)[32][2], float L[2][38][64], int i, int len)
Definition: aacpsdsp.h:35
src
AVS_Value src
Definition: avisynth_c.h:523
void
typedef void(RENAME(mix_any_func_type))
Definition: rematrix_template.c:41
PSDSPContext
struct PSDSPContext PSDSPContext
i
synthesis window for stochastic i
Definition: code Kat/hpsmodel.m:34
ff_psdsp_init_mips
void ff_psdsp_init_mips(PSDSPContext *s)
Definition: aacpsdsp_mips.c:447
n
n
Definition: genspecsines_sin.m:10
PSDSPContext::decorrelate
void(* decorrelate)(float(*out)[2], float(*delay)[2], float(*ap_delay)[PS_QMF_TIME_SLOTS+PS_MAX_AP_DELAY][2], const float phi_fract[2], float(*Q_fract)[2], const float *transient_gain, float g_decay_slope, int len)
Definition: aacpsdsp.h:39
PSDSPContext::add_squares
void(* add_squares)(float *dst, const float(*src)[2], int n)
Definition: aacpsdsp.h:29
len
int len
Definition: vorbis_enc_data.h:452
dst
else dst[i][x+y *dst_stride[i]]
Definition: vf_mcdeint.c:160
out
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=av_sample_fmt_is_planar(in_fmt);out_planar=av_sample_fmt_is_planar(out_fmt);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_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_dlog(ac->avr,"%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-> out
Definition: audio_convert.c:194
Generated on Fri Dec 20 2024 06:55:57 for FFmpeg by   doxygen 1.8.11