FFmpeg  4.3
proresdsp.c
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1 /*
2  * Apple ProRes compatible decoder
3  *
4  * Copyright (c) 2010-2011 Maxim Poliakovski
5  *
6  * This file is part of FFmpeg.
7  *
8  * FFmpeg is free software; you can redistribute it and/or
9  * modify it under the terms of the GNU Lesser General Public
10  * License as published by the Free Software Foundation; either
11  * version 2.1 of the License, or (at your option) any later version.
12  *
13  * FFmpeg is distributed in the hope that it will be useful,
14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16  * Lesser General Public License for more details.
17  *
18  * You should have received a copy of the GNU Lesser General Public
19  * License along with FFmpeg; if not, write to the Free Software
20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21  */
22 
23 #include "config.h"
24 #include "libavutil/attributes.h"
25 #include "libavutil/common.h"
26 #include "idctdsp.h"
27 #include "proresdsp.h"
28 #include "simple_idct.h"
29 
30 #define CLIP_MIN (1 << 2) ///< minimum value for clipping resulting pixels
31 #define CLIP_MAX_10 (1 << 10) - CLIP_MIN - 1 ///< maximum value for clipping resulting pixels
32 #define CLIP_MAX_12 (1 << 12) - CLIP_MIN - 1 ///< maximum value for clipping resulting pixels
33 
34 #define CLIP_10(x) (av_clip((x), CLIP_MIN, CLIP_MAX_10))
35 #define CLIP_12(x) (av_clip((x), CLIP_MIN, CLIP_MAX_12))
36 
37 /**
38  * Add bias value, clamp and output pixels of a slice
39  */
40 
41 static inline void put_pixel(uint16_t *dst, ptrdiff_t linesize, const int16_t *in, int bits_per_raw_sample) {
42  int x, y, src_offset, dst_offset;
43 
44  for (y = 0, dst_offset = 0; y < 8; y++, dst_offset += linesize) {
45  for (x = 0; x < 8; x++) {
46  src_offset = (y << 3) + x;
47 
48  if (bits_per_raw_sample == 10) {
49  dst[dst_offset + x] = CLIP_10(in[src_offset]);
50  } else {//12b
51  dst[dst_offset + x] = CLIP_12(in[src_offset]);
52  }
53  }
54  }
55 }
56 
57 static void put_pixels_10(uint16_t *dst, ptrdiff_t linesize, const int16_t *in)
58 {
59  put_pixel(dst, linesize, in, 10);
60 }
61 
62 static void put_pixels_12(uint16_t *dst, ptrdiff_t linesize, const int16_t *in)
63 {
64  put_pixel(dst, linesize, in, 12);
65 }
66 
67 static void prores_idct_put_10_c(uint16_t *out, ptrdiff_t linesize, int16_t *block, const int16_t *qmat)
68 {
69  ff_prores_idct_10(block, qmat);
70  put_pixels_10(out, linesize >> 1, block);
71 }
72 
73 static void prores_idct_put_12_c(uint16_t *out, ptrdiff_t linesize, int16_t *block, const int16_t *qmat)
74 {
75  ff_prores_idct_12(block, qmat);
76  put_pixels_12(out, linesize >> 1, block);
77 }
78 
80 {
81  if (avctx->bits_per_raw_sample == 10) {
84  } else if (avctx->bits_per_raw_sample == 12) {
87  } else {
88  return AVERROR_BUG;
89  }
90 
91  if (ARCH_X86)
92  ff_proresdsp_init_x86(dsp, avctx);
93 
96  return 0;
97 }
ARCH_X86
#define ARCH_X86
Definition: config.h:38
out
FILE * out
Definition: movenc.c:54
ff_proresdsp_init
av_cold int ff_proresdsp_init(ProresDSPContext *dsp, AVCodecContext *avctx)
Definition: proresdsp.c:79
ProresDSPContext::idct_permutation_type
int idct_permutation_type
Definition: proresdsp.h:31
put_pixel
static void put_pixel(uint16_t *dst, ptrdiff_t linesize, const int16_t *in, int bits_per_raw_sample)
Add bias value, clamp and output pixels of a slice.
Definition: proresdsp.c:41
put_pixels_12
static void put_pixels_12(uint16_t *dst, ptrdiff_t linesize, const int16_t *in)
Definition: proresdsp.c:62
ProresDSPContext
Definition: proresdsp.h:30
x
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new please make sure they are as small as space on each network bandwidth and so on benefit from smaller test cases Also keep in mind older checkouts use existing sample that means in practice generally do not remove or overwrite files as it likely would break older checkouts or releases Also all needed samples for a commit should be ideally before the push If you need an account for frequently uploading samples or you wish to help others by doing that send a mail to ffmpeg devel rsync vauL Duo x
Definition: fate.txt:150
av_cold
#define av_cold
Definition: attributes.h:90
AVCodecContext::bits_per_raw_sample
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
Definition: avcodec.h:1757
simple_idct.h
CLIP_12
#define CLIP_12(x)
Definition: proresdsp.c:35
ProresDSPContext::idct_permutation
uint8_t idct_permutation[64]
Definition: proresdsp.h:32
FF_IDCT_PERM_NONE
@ FF_IDCT_PERM_NONE
Definition: idctdsp.h:38
CLIP_10
#define CLIP_10(x)
Definition: proresdsp.c:34
attributes.h
proresdsp.h
ff_init_scantable_permutation
av_cold void ff_init_scantable_permutation(uint8_t *idct_permutation, enum idct_permutation_type perm_type)
Definition: idctdsp.c:50
in
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
Definition: audio_convert.c:326
common.h
idctdsp.h
AVCodecContext
main external API structure.
Definition: avcodec.h:526
config.h
ff_prores_idct_12
void ff_prores_idct_12(int16_t *block, const int16_t *qmat)
Definition: simple_idct.c:255
put_pixels_10
static void put_pixels_10(uint16_t *dst, ptrdiff_t linesize, const int16_t *in)
Definition: proresdsp.c:57
ff_prores_idct_10
void ff_prores_idct_10(int16_t *block, const int16_t *qmat)
Special version of ff_simple_idct_int16_10bit() which does dequantization and scales by a factor of 2...
Definition: simple_idct.c:239
prores_idct_put_10_c
static void prores_idct_put_10_c(uint16_t *out, ptrdiff_t linesize, int16_t *block, const int16_t *qmat)
Definition: proresdsp.c:67
ff_proresdsp_init_x86
void ff_proresdsp_init_x86(ProresDSPContext *dsp, AVCodecContext *avctx)
Definition: proresdsp_init.c:33
AVERROR_BUG
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:50
block
The exact code depends on how similar the blocks are and how related they are to the block
Definition: filter_design.txt:207
ProresDSPContext::idct_put
void(* idct_put)(uint16_t *out, ptrdiff_t linesize, int16_t *block, const int16_t *qmat)
Definition: proresdsp.h:33
prores_idct_put_12_c
static void prores_idct_put_12_c(uint16_t *out, ptrdiff_t linesize, int16_t *block, const int16_t *qmat)
Definition: proresdsp.c:73