mirror of
https://github.com/FFmpeg/FFmpeg.git
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335 lines
11 KiB
C
335 lines
11 KiB
C
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/*
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* Copyright (c) 2018 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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#include "libavutil/avassert.h"
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#include "libavutil/avstring.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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typedef struct AudioIIRContext {
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const AVClass *class;
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char *a_str, *b_str;
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double dry_gain, wet_gain;
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int *nb_a, *nb_b;
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double **a, **b;
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double **input, **output;
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int clippings;
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int channels;
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void (*iir_frame)(AVFilterContext *ctx, AVFrame *in, AVFrame *out);
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} AudioIIRContext;
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats;
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AVFilterChannelLayouts *layouts;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_NONE
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};
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int ret;
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layouts = ff_all_channel_counts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ret = ff_set_common_channel_layouts(ctx, layouts);
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if (ret < 0)
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return ret;
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formats = ff_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ret = ff_set_common_formats(ctx, formats);
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if (ret < 0)
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return ret;
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formats = ff_all_samplerates();
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if (!formats)
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return AVERROR(ENOMEM);
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return ff_set_common_samplerates(ctx, formats);
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}
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#define IIR_FRAME(name, type, min, max, need_clipping) \
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static void iir_frame_## name(AVFilterContext *ctx, AVFrame *in, AVFrame *out) \
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{ \
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AudioIIRContext *s = ctx->priv; \
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const double ig = s->dry_gain; \
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const double og = s->wet_gain; \
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int ch, n; \
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\
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for (ch = 0; ch < out->channels; ch++) { \
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const type *src = (const type *)in->extended_data[ch]; \
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double *ic = (double *)s->input[ch]; \
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double *oc = (double *)s->output[ch]; \
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const int nb_a = s->nb_a[ch]; \
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const int nb_b = s->nb_b[ch]; \
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const double *a = s->a[ch]; \
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const double *b = s->b[ch]; \
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type *dst = (type *)out->extended_data[ch]; \
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\
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for (n = 0; n < in->nb_samples; n++) { \
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double sample = 0.; \
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int x; \
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\
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memmove(&ic[1], &ic[0], (nb_b - 1) * sizeof(*ic)); \
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memmove(&oc[1], &oc[0], (nb_a - 1) * sizeof(*oc)); \
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ic[0] = src[n] * ig; \
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for (x = 0; x < nb_b; x++) \
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sample += b[x] * ic[x]; \
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\
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for (x = 1; x < nb_a; x++) \
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sample -= a[x] * oc[x]; \
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\
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oc[0] = sample; \
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sample *= og; \
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if (need_clipping && sample < min) { \
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s->clippings++; \
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dst[n] = min; \
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} else if (need_clipping && sample > max) { \
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s->clippings++; \
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dst[n] = max; \
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} else { \
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dst[n] = sample; \
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} \
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} \
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} \
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}
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IIR_FRAME(s16p, int16_t, INT16_MIN, INT16_MAX, 1)
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IIR_FRAME(s32p, int32_t, INT32_MIN, INT32_MAX, 1)
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IIR_FRAME(fltp, float, -1., 1., 0)
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IIR_FRAME(dblp, double, -1., 1., 0)
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static void count_coefficients(char *item_str, int *nb_items)
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{
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char *p;
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*nb_items = 1;
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for (p = item_str; *p && *p != '|'; p++) {
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if (*p == ' ')
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(*nb_items)++;
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}
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}
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static int read_coefficients(AVFilterContext *ctx, char *item_str, int nb_items, double *dst)
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{
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char *p, *arg, *old_str, *saveptr = NULL;
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int i;
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p = old_str = av_strdup(item_str);
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if (!p)
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return AVERROR(ENOMEM);
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for (i = 0; i < nb_items; i++) {
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if (!(arg = av_strtok(p, " ", &saveptr)))
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break;
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p = NULL;
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if (sscanf(arg, "%lf", &dst[i]) != 1) {
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av_log(ctx, AV_LOG_ERROR, "Invalid coefficients supplied: %s\n", arg);
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return AVERROR(EINVAL);
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}
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}
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av_freep(&old_str);
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return 0;
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}
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static int read_channels(AVFilterContext *ctx, int channels, uint8_t *item_str, int *nb, double **c, double **cache)
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{
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char *p, *arg, *old_str, *prev_arg = NULL, *saveptr = NULL;
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int i, ret;
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p = old_str = av_strdup(item_str);
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if (!p)
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return AVERROR(ENOMEM);
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for (i = 0; i < channels; i++) {
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if (!(arg = av_strtok(p, "|", &saveptr)))
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arg = prev_arg;
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p = NULL;
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count_coefficients(arg, &nb[i]);
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cache[i] = av_calloc(nb[i], sizeof(cache[i]));
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c[i] = av_calloc(nb[i], sizeof(c[i]));
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if (!c[i] || !cache[i])
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return AVERROR(ENOMEM);
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ret = read_coefficients(ctx, arg, nb[i], c[i]);
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if (ret < 0)
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return ret;
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prev_arg = arg;
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}
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av_freep(&old_str);
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return 0;
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}
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static int config_output(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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AudioIIRContext *s = ctx->priv;
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AVFilterLink *inlink = ctx->inputs[0];
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int ch, ret, i;
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s->channels = inlink->channels;
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s->a = av_calloc(inlink->channels, sizeof(*s->a));
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s->b = av_calloc(inlink->channels, sizeof(*s->b));
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s->nb_a = av_calloc(inlink->channels, sizeof(*s->nb_a));
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s->nb_b = av_calloc(inlink->channels, sizeof(*s->nb_b));
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s->input = av_calloc(inlink->channels, sizeof(*s->input));
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s->output = av_calloc(inlink->channels, sizeof(*s->output));
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if (!s->a || !s->b || !s->nb_a || !s->nb_b || !s->input || !s->output)
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return AVERROR(ENOMEM);
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ret = read_channels(ctx, inlink->channels, s->a_str, s->nb_a, s->a, s->output);
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if (ret < 0)
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return ret;
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ret = read_channels(ctx, inlink->channels, s->b_str, s->nb_b, s->b, s->input);
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if (ret < 0)
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return ret;
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for (ch = 0; ch < inlink->channels; ch++) {
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for (i = 1; i < s->nb_a[ch]; i++) {
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s->a[ch][i] /= s->a[ch][0];
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}
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for (i = 0; i < s->nb_b[ch]; i++) {
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s->b[ch][i] /= s->a[ch][0];
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}
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}
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switch (inlink->format) {
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case AV_SAMPLE_FMT_DBLP: s->iir_frame = iir_frame_dblp; break;
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case AV_SAMPLE_FMT_FLTP: s->iir_frame = iir_frame_fltp; break;
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case AV_SAMPLE_FMT_S32P: s->iir_frame = iir_frame_s32p; break;
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case AV_SAMPLE_FMT_S16P: s->iir_frame = iir_frame_s16p; break;
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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 *in)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioIIRContext *s = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[0];
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AVFrame *out;
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if (av_frame_is_writable(in)) {
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out = in;
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} else {
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out = ff_get_audio_buffer(outlink, in->nb_samples);
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if (!out) {
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av_frame_free(&in);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out, in);
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}
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s->iir_frame(ctx, in, out);
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if (s->clippings > 0)
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av_log(ctx, AV_LOG_WARNING, "clipping %d times. Please reduce gain.\n", s->clippings);
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s->clippings = 0;
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if (in != out)
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av_frame_free(&in);
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return ff_filter_frame(outlink, out);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AudioIIRContext *s = ctx->priv;
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int ch;
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if (s->a) {
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for (ch = 0; ch < s->channels; ch++) {
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av_freep(&s->a[ch]);
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av_freep(&s->output[ch]);
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}
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}
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av_freep(&s->a);
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if (s->b) {
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for (ch = 0; ch < s->channels; ch++) {
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av_freep(&s->b[ch]);
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av_freep(&s->input[ch]);
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}
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}
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av_freep(&s->b);
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av_freep(&s->input);
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av_freep(&s->output);
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av_freep(&s->nb_a);
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av_freep(&s->nb_b);
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}
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static const AVFilterPad 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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{ NULL }
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};
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static const AVFilterPad 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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{ NULL }
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};
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#define OFFSET(x) offsetof(AudioIIRContext, x)
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#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption aiir_options[] = {
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{ "a", "set A/denominator/poles coefficients", OFFSET(a_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, AF },
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{ "b", "set B/numerator/zeros coefficients", OFFSET(b_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, AF },
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{ "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
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{ "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, AF },
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{ NULL },
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};
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AVFILTER_DEFINE_CLASS(aiir);
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AVFilter ff_af_aiir = {
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.name = "aiir",
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.description = NULL_IF_CONFIG_SMALL("Apply Infinite Impulse Response filter with supplied coefficients."),
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.priv_size = sizeof(AudioIIRContext),
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.uninit = uninit,
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.query_formats = query_formats,
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.inputs = inputs,
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.outputs = outputs,
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.priv_class = &aiir_class,
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};
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