mirror of https://github.com/FFmpeg/FFmpeg.git
Signed-off-by: Paul B Mahol <onemda@gmail.com>pull/303/head
Paul B Mahol
7 years ago
parent
b2be76c0a4
commit
7bb1be9af0
7 changed files with 375 additions and 1 deletions
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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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