FFmpeg/libavfilter/af_afir.h
Paul B Mahol 52bf43eb49 avfilter/af_afir: add support for switching impulse response streams at runtime
Currently, switching is not free of artifacts, to be resolved later.
2020-01-10 13:14:54 +01:00

107 lines
2.4 KiB
C

/*
* Copyright (c) 2017 Paul B Mahol
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVFILTER_AFIR_H
#define AVFILTER_AFIR_H
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "libavutil/opt.h"
#include "libavcodec/avfft.h"
#include "audio.h"
#include "avfilter.h"
#include "formats.h"
#include "internal.h"
typedef struct AudioFIRSegment {
int nb_partitions;
int part_size;
int block_size;
int fft_length;
int coeff_size;
int input_size;
int input_offset;
int *output_offset;
int *part_index;
AVFrame *sum;
AVFrame *block;
AVFrame *buffer;
AVFrame *coeff;
AVFrame *input;
AVFrame *output;
RDFTContext **rdft, **irdft;
} AudioFIRSegment;
typedef struct AudioFIRDSPContext {
void (*fcmul_add)(float *sum, const float *t, const float *c,
ptrdiff_t len);
} AudioFIRDSPContext;
typedef struct AudioFIRContext {
const AVClass *class;
float wet_gain;
float dry_gain;
float length;
int gtype;
float ir_gain;
int ir_format;
float max_ir_len;
int response;
int w, h;
AVRational frame_rate;
int ir_channel;
int minp;
int maxp;
int nb_irs;
int selir;
float gain;
int eof_coeffs[32];
int have_coeffs;
int nb_taps;
int nb_channels;
int nb_coef_channels;
int one2many;
AudioFIRSegment seg[1024];
int nb_segments;
AVFrame *in;
AVFrame *ir[32];
AVFrame *video;
int min_part_size;
int64_t pts;
AudioFIRDSPContext afirdsp;
AVFloatDSPContext *fdsp;
} AudioFIRContext;
void ff_afir_init(AudioFIRDSPContext *s);
void ff_afir_init_x86(AudioFIRDSPContext *s);
#endif /* AVFILTER_AFIR_H */