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790f793844
There are lots of files that don't need it: The number of object files that actually need it went down from 2011 to 884 here. Keep it for external users in order to not cause breakages. Also improve the other headers a bit while just at it. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
273 lines
12 KiB
C
273 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/mem.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, .unit = "type" },
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{ "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, .unit = "type" },
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{ "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, FLAGS, .unit = "type" },
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{ "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, .unit = "type" },
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{ "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, FLAGS, .unit = "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->ch_layout.nb_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->ch_layout.nb_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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