mirror of
https://github.com/FFmpeg/FFmpeg.git
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272 lines
12 KiB
C
272 lines
12 KiB
C
/*
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* Copyright (c) 2013 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* phaser audio filter
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*/
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#include "libavutil/avassert.h"
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#include "libavutil/opt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "internal.h"
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#include "generate_wave_table.h"
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typedef struct AudioPhaserContext {
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const AVClass *class;
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double in_gain, out_gain;
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double delay;
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double decay;
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double speed;
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int type;
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int delay_buffer_length;
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double *delay_buffer;
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int modulation_buffer_length;
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int32_t *modulation_buffer;
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int delay_pos, modulation_pos;
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void (*phaser)(struct AudioPhaserContext *s,
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uint8_t * const *src, uint8_t **dst,
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int nb_samples, int channels);
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} AudioPhaserContext;
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#define OFFSET(x) offsetof(AudioPhaserContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption aphaser_options[] = {
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{ "in_gain", "set input gain", OFFSET(in_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, 1, FLAGS },
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{ "out_gain", "set output gain", OFFSET(out_gain), AV_OPT_TYPE_DOUBLE, {.dbl=.74}, 0, 1e9, FLAGS },
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{ "delay", "set delay in milliseconds", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=3.}, 0, 5, FLAGS },
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{ "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=.4}, 0, .99, FLAGS },
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{ "speed", "set modulation speed", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, .1, 2, FLAGS },
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{ "type", "set modulation type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=WAVE_TRI}, 0, WAVE_NB-1, FLAGS, "type" },
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{ "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
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{ "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, "type" },
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{ "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
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{ "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, "type" },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(aphaser);
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static av_cold int init(AVFilterContext *ctx)
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{
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AudioPhaserContext *s = ctx->priv;
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if (s->in_gain > (1 - s->decay * s->decay))
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av_log(ctx, AV_LOG_WARNING, "in_gain may cause clipping\n");
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if (s->in_gain / (1 - s->decay) > 1 / s->out_gain)
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av_log(ctx, AV_LOG_WARNING, "out_gain may cause clipping\n");
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return 0;
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}
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#define MOD(a, b) (((a) >= (b)) ? (a) - (b) : (a))
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#define PHASER_PLANAR(name, type) \
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static void phaser_## name ##p(AudioPhaserContext *s, \
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uint8_t * const *ssrc, uint8_t **ddst, \
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int nb_samples, int channels) \
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{ \
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int i, c, delay_pos, modulation_pos; \
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\
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av_assert0(channels > 0); \
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for (c = 0; c < channels; c++) { \
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type *src = (type *)ssrc[c]; \
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type *dst = (type *)ddst[c]; \
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double *buffer = s->delay_buffer + \
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c * s->delay_buffer_length; \
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\
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delay_pos = s->delay_pos; \
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modulation_pos = s->modulation_pos; \
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\
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for (i = 0; i < nb_samples; i++, src++, dst++) { \
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double v = *src * s->in_gain + buffer[ \
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MOD(delay_pos + s->modulation_buffer[ \
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modulation_pos], \
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s->delay_buffer_length)] * s->decay; \
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\
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modulation_pos = MOD(modulation_pos + 1, \
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s->modulation_buffer_length); \
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delay_pos = MOD(delay_pos + 1, s->delay_buffer_length); \
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buffer[delay_pos] = v; \
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\
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*dst = v * s->out_gain; \
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} \
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} \
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\
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s->delay_pos = delay_pos; \
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s->modulation_pos = modulation_pos; \
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}
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#define PHASER(name, type) \
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static void phaser_## name (AudioPhaserContext *s, \
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uint8_t * const *ssrc, uint8_t **ddst, \
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int nb_samples, int channels) \
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{ \
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int i, c, delay_pos, modulation_pos; \
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type *src = (type *)ssrc[0]; \
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type *dst = (type *)ddst[0]; \
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double *buffer = s->delay_buffer; \
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\
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delay_pos = s->delay_pos; \
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modulation_pos = s->modulation_pos; \
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\
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for (i = 0; i < nb_samples; i++) { \
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int pos = MOD(delay_pos + s->modulation_buffer[modulation_pos], \
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s->delay_buffer_length) * channels; \
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int npos; \
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\
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delay_pos = MOD(delay_pos + 1, s->delay_buffer_length); \
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npos = delay_pos * channels; \
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for (c = 0; c < channels; c++, src++, dst++) { \
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double v = *src * s->in_gain + buffer[pos + c] * s->decay; \
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\
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buffer[npos + c] = v; \
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\
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*dst = v * s->out_gain; \
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} \
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\
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modulation_pos = MOD(modulation_pos + 1, \
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s->modulation_buffer_length); \
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} \
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\
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s->delay_pos = delay_pos; \
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s->modulation_pos = modulation_pos; \
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}
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PHASER_PLANAR(dbl, double)
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PHASER_PLANAR(flt, float)
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PHASER_PLANAR(s16, int16_t)
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PHASER_PLANAR(s32, int32_t)
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PHASER(dbl, double)
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PHASER(flt, float)
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PHASER(s16, int16_t)
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PHASER(s32, int32_t)
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static int config_output(AVFilterLink *outlink)
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{
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AudioPhaserContext *s = outlink->src->priv;
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AVFilterLink *inlink = outlink->src->inputs[0];
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s->delay_buffer_length = s->delay * 0.001 * inlink->sample_rate + 0.5;
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if (s->delay_buffer_length <= 0) {
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av_log(outlink->src, AV_LOG_ERROR, "delay is too small\n");
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return AVERROR(EINVAL);
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}
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s->delay_buffer = av_calloc(s->delay_buffer_length, sizeof(*s->delay_buffer) * inlink->channels);
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s->modulation_buffer_length = inlink->sample_rate / s->speed + 0.5;
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s->modulation_buffer = av_malloc_array(s->modulation_buffer_length, sizeof(*s->modulation_buffer));
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if (!s->modulation_buffer || !s->delay_buffer)
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return AVERROR(ENOMEM);
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ff_generate_wave_table(s->type, AV_SAMPLE_FMT_S32,
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s->modulation_buffer, s->modulation_buffer_length,
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1., s->delay_buffer_length, M_PI / 2.0);
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s->delay_pos = s->modulation_pos = 0;
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switch (inlink->format) {
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case AV_SAMPLE_FMT_DBL: s->phaser = phaser_dbl; break;
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case AV_SAMPLE_FMT_DBLP: s->phaser = phaser_dblp; break;
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case AV_SAMPLE_FMT_FLT: s->phaser = phaser_flt; break;
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case AV_SAMPLE_FMT_FLTP: s->phaser = phaser_fltp; break;
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case AV_SAMPLE_FMT_S16: s->phaser = phaser_s16; break;
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case AV_SAMPLE_FMT_S16P: s->phaser = phaser_s16p; break;
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case AV_SAMPLE_FMT_S32: s->phaser = phaser_s32; break;
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case AV_SAMPLE_FMT_S32P: s->phaser = phaser_s32p; break;
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default: av_assert0(0);
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}
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return 0;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *inbuf)
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{
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AudioPhaserContext *s = inlink->dst->priv;
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AVFilterLink *outlink = inlink->dst->outputs[0];
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AVFrame *outbuf;
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if (av_frame_is_writable(inbuf)) {
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outbuf = inbuf;
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} else {
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outbuf = ff_get_audio_buffer(outlink, inbuf->nb_samples);
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if (!outbuf) {
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av_frame_free(&inbuf);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(outbuf, inbuf);
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}
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s->phaser(s, inbuf->extended_data, outbuf->extended_data,
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outbuf->nb_samples, outbuf->channels);
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if (inbuf != outbuf)
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av_frame_free(&inbuf);
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return ff_filter_frame(outlink, outbuf);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AudioPhaserContext *s = ctx->priv;
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av_freep(&s->delay_buffer);
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av_freep(&s->modulation_buffer);
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}
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static const AVFilterPad aphaser_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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},
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};
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static const AVFilterPad aphaser_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_output,
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},
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};
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const AVFilter ff_af_aphaser = {
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.name = "aphaser",
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.description = NULL_IF_CONFIG_SMALL("Add a phasing effect to the audio."),
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.priv_size = sizeof(AudioPhaserContext),
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.init = init,
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.uninit = uninit,
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FILTER_INPUTS(aphaser_inputs),
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FILTER_OUTPUTS(aphaser_outputs),
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FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P),
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.priv_class = &aphaser_class,
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};
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