dct.h
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1 /*
2  * (I)DCT Transforms
3  * Copyright (c) 2009 Peter Ross <pross@xvid.org>
4  * Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
5  * Copyright (c) 2010 Vitor Sessak
6  *
7  * This file is part of FFmpeg.
8  *
9  * FFmpeg is free software; you can redistribute it and/or
10  * modify it under the terms of the GNU Lesser General Public
11  * License as published by the Free Software Foundation; either
12  * version 2.1 of the License, or (at your option) any later version.
13  *
14  * FFmpeg is distributed in the hope that it will be useful,
15  * but WITHOUT ANY WARRANTY; without even the implied warranty of
16  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
17  * Lesser General Public License for more details.
18  *
19  * You should have received a copy of the GNU Lesser General Public
20  * License along with FFmpeg; if not, write to the Free Software
21  * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
22  */
23 
24 #ifndef AVCODEC_DCT_H
25 #define AVCODEC_DCT_H
26 
27 #include <stdint.h>
28 
29 #include "rdft.h"
30 
31 struct DCTContext {
32  int nbits;
33  int inverse;
35  const float *costab;
39 };
40 
41 /**
42  * Set up DCT.
43  * @param nbits size of the input array:
44  * (1 << nbits) for DCT-II, DCT-III and DST-I
45  * (1 << nbits) + 1 for DCT-I
46  *
47  * @note the first element of the input of DST-I is ignored
48  */
50 void ff_dct_end (DCTContext *s);
51 
53 
54 void ff_fdct_ifast(int16_t *data);
55 void ff_fdct_ifast248(int16_t *data);
56 void ff_jpeg_fdct_islow_8(int16_t *data);
57 void ff_jpeg_fdct_islow_10(int16_t *data);
58 void ff_fdct248_islow_8(int16_t *data);
59 void ff_fdct248_islow_10(int16_t *data);
60 
61 void ff_j_rev_dct(int16_t *data);
62 void ff_j_rev_dct4(int16_t *data);
63 void ff_j_rev_dct2(int16_t *data);
64 void ff_j_rev_dct1(int16_t *data);
65 
66 void ff_fdct_mmx(int16_t *block);
67 void ff_fdct_mmxext(int16_t *block);
68 void ff_fdct_sse2(int16_t *block);
69 
70 #endif /* AVCODEC_DCT_H */
void ff_jpeg_fdct_islow_10(int16_t *data)
void ff_fdct248_islow_10(int16_t *data)
const char * s
Definition: avisynth_c.h:668
void ff_fdct_ifast(int16_t *data)
Definition: jfdctfst.c:208
const float * costab
Definition: dct.h:35
void ff_j_rev_dct4(int16_t *data)
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
RDFTContext rdft
Definition: dct.h:34
void ff_dct_init_x86(DCTContext *s)
int nbits
Definition: dct.h:32
void ff_fdct_mmxext(int16_t *block)
DCTTransformType
Definition: avfft.h:93
Spectrum Plot time data
void ff_fdct_mmx(int16_t *block)
float FFTSample
Definition: avfft.h:35
void ff_fdct_sse2(int16_t *block)
void(* dct_calc)(struct DCTContext *s, FFTSample *data)
Definition: dct.h:37
Definition: dct.h:31
void ff_fdct248_islow_8(int16_t *data)
void ff_j_rev_dct1(int16_t *data)
int ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType type)
Set up DCT.
Definition: dct.c:177
void ff_jpeg_fdct_islow_8(int16_t *data)
void ff_j_rev_dct(int16_t *data)
typedef void(RENAME(mix_any_func_type))
int inverse
Definition: dct.h:33
void ff_dct_end(DCTContext *s)
Definition: dct.c:218
#define type
void ff_fdct_ifast248(int16_t *data)
Definition: jfdctfst.c:274
void(* dct32)(FFTSample *out, const FFTSample *in)
Definition: dct.h:38
void ff_j_rev_dct2(int16_t *data)
FFTSample * csc2
Definition: dct.h:36
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