FFmpeg  4.2.1
af_afir.h
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1 /*
2  * Copyright (c) 2017 Paul B Mahol
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 AVFILTER_AFIR_H
22 #define AVFILTER_AFIR_H
23 
24 #include "libavutil/common.h"
25 #include "libavutil/float_dsp.h"
26 #include "libavutil/opt.h"
27 #include "libavcodec/avfft.h"
28 
29 #include "audio.h"
30 #include "avfilter.h"
31 #include "formats.h"
32 #include "internal.h"
33 
34 typedef struct AudioFIRSegment {
36  int part_size;
42 
44  int *part_index;
45 
52 
55 
56 typedef struct AudioFIRDSPContext {
57  void (*fcmul_add)(float *sum, const float *t, const float *c,
58  ptrdiff_t len);
60 
61 typedef struct AudioFIRContext {
62  const AVClass *class;
63 
64  float wet_gain;
65  float dry_gain;
66  float length;
67  int gtype;
68  float ir_gain;
69  int ir_format;
70  float max_ir_len;
71  int response;
72  int w, h;
75  int minp;
76  int maxp;
77 
78  float gain;
79 
82  int nb_taps;
85  int one2many;
86 
87  AudioFIRSegment seg[1024];
89 
90  AVFrame *in[2];
93  int64_t pts;
94 
97 
99 
102 
103 #endif /* AVFILTER_AFIR_H */
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
int nb_coef_channels
Definition: af_afir.h:84
int * part_index
Definition: af_afir.h:44
int nb_channels
Definition: af_afir.h:83
Main libavfilter public API header.
AVFrame * block
Definition: af_afir.h:47
int input_offset
Definition: af_afir.h:41
int input_size
Definition: af_afir.h:40
int min_part_size
Definition: af_afir.h:92
float ir_gain
Definition: af_afir.h:68
AVFrame * coeff
Definition: af_afir.h:49
int eof_coeffs
Definition: af_afir.h:80
float dry_gain
Definition: af_afir.h:65
AVOptions.
AVFrame * buffer
Definition: af_afir.h:48
void ff_afir_init_x86(AudioFIRDSPContext *s)
Definition: af_afir_init.c:30
AVFrame * output
Definition: af_afir.h:51
AVFrame * sum
Definition: af_afir.h:46
RDFTContext ** irdft
Definition: af_afir.h:53
int coeff_size
Definition: af_afir.h:39
RDFTContext ** rdft
Definition: af_afir.h:53
AudioFIRDSPContext afirdsp
Definition: af_afir.h:95
int ir_channel
Definition: af_afir.h:74
#define s(width, name)
Definition: cbs_vp9.c:257
int64_t pts
Definition: af_afir.h:93
AVFloatDSPContext * fdsp
Definition: af_afir.h:96
AVFrame * input
Definition: af_afir.h:50
typedef void(RENAME(mix_any_func_type))
FFT functions.
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
Describe the class of an AVClass context structure.
Definition: log.h:67
int have_coeffs
Definition: af_afir.h:81
Rational number (pair of numerator and denominator).
Definition: rational.h:58
float max_ir_len
Definition: af_afir.h:70
float length
Definition: af_afir.h:66
int nb_partitions
Definition: af_afir.h:35
void ff_afir_init(AudioFIRDSPContext *s)
Definition: af_afir.c:756
common internal and external API header
static double c[64]
AVFrame * video
Definition: af_afir.h:91
float gain
Definition: af_afir.h:78
int fft_length
Definition: af_afir.h:38
int len
float wet_gain
Definition: af_afir.h:64
int * output_offset
Definition: af_afir.h:43
int block_size
Definition: af_afir.h:37
AVRational frame_rate
Definition: af_afir.h:73
internal API functions
int nb_segments
Definition: af_afir.h:88