mirror of
https://github.com/nyanmisaka/ffmpeg-rockchip.git
synced 2025-10-30 04:02:04 +08:00
344 lines
9.4 KiB
C
344 lines
9.4 KiB
C
/*
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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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* Crossover filter
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*
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* Split an audio stream into several bands.
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*/
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#include "libavutil/attributes.h"
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#include "libavutil/avstring.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/eval.h"
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#include "libavutil/internal.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 "formats.h"
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#include "internal.h"
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#define MAX_SPLITS 16
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#define MAX_BANDS MAX_SPLITS + 1
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typedef struct BiquadContext {
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double a0, a1, a2;
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double b1, b2;
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double i1, i2;
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double o1, o2;
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} BiquadContext;
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typedef struct CrossoverChannel {
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BiquadContext lp[MAX_BANDS][4];
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BiquadContext hp[MAX_BANDS][4];
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} CrossoverChannel;
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typedef struct AudioCrossoverContext {
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const AVClass *class;
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char *splits_str;
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int order;
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int filter_count;
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int nb_splits;
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float *splits;
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CrossoverChannel *xover;
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} AudioCrossoverContext;
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#define OFFSET(x) offsetof(AudioCrossoverContext, x)
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#define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption acrossover_options[] = {
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{ "split", "set split frequencies", OFFSET(splits_str), AV_OPT_TYPE_STRING, {.str="500"}, 0, 0, AF },
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{ "order", "set order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=1}, 0, 2, AF, "m" },
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{ "2nd", "2nd order", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "m" },
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{ "4th", "4th order", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "m" },
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{ "8th", "8th order", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "m" },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(acrossover);
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static av_cold int init(AVFilterContext *ctx)
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{
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AudioCrossoverContext *s = ctx->priv;
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char *p, *arg, *saveptr = NULL;
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int i, ret = 0;
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s->splits = av_calloc(MAX_SPLITS, sizeof(*s->splits));
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if (!s->splits)
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return AVERROR(ENOMEM);
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p = s->splits_str;
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for (i = 0; i < MAX_SPLITS; i++) {
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float freq;
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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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av_sscanf(arg, "%f", &freq);
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if (freq <= 0) {
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av_log(ctx, AV_LOG_ERROR, "Frequency %f must be positive number.\n", freq);
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return AVERROR(EINVAL);
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}
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if (i > 0 && freq <= s->splits[i-1]) {
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av_log(ctx, AV_LOG_ERROR, "Frequency %f must be in increasing order.\n", freq);
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return AVERROR(EINVAL);
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}
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s->splits[i] = freq;
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}
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s->nb_splits = i;
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for (i = 0; i <= s->nb_splits; i++) {
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AVFilterPad pad = { 0 };
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char *name;
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pad.type = AVMEDIA_TYPE_AUDIO;
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name = av_asprintf("out%d", ctx->nb_outputs);
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if (!name)
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return AVERROR(ENOMEM);
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pad.name = name;
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if ((ret = ff_insert_outpad(ctx, i, &pad)) < 0) {
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av_freep(&pad.name);
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return ret;
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}
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}
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return ret;
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}
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static void set_lp(BiquadContext *b, float fc, float q, float sr)
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{
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double omega = (2.0 * M_PI * fc / sr);
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double sn = sin(omega);
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double cs = cos(omega);
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double alpha = (sn / (2 * q));
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double inv = (1.0 / (1.0 + alpha));
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b->a2 = b->a0 = (inv * (1.0 - cs) * 0.5);
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b->a1 = b->a0 + b->a0;
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b->b1 = -2. * cs * inv;
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b->b2 = (1. - alpha) * inv;
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}
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static void set_hp(BiquadContext *b, float fc, float q, float sr)
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{
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double omega = 2 * M_PI * fc / sr;
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double sn = sin(omega);
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double cs = cos(omega);
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double alpha = sn / (2 * q);
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double inv = 1.0 / (1.0 + alpha);
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b->a0 = inv * (1. + cs) / 2.;
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b->a1 = -2. * b->a0;
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b->a2 = b->a0;
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b->b1 = -2. * cs * inv;
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b->b2 = (1. - alpha) * inv;
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}
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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AudioCrossoverContext *s = ctx->priv;
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int ch, band, sample_rate = inlink->sample_rate;
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double q;
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s->xover = av_calloc(inlink->channels, sizeof(*s->xover));
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if (!s->xover)
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return AVERROR(ENOMEM);
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switch (s->order) {
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case 0:
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q = 0.5;
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s->filter_count = 1;
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break;
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case 1:
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q = M_SQRT1_2;
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s->filter_count = 2;
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break;
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case 2:
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q = 0.54;
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s->filter_count = 4;
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break;
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}
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for (ch = 0; ch < inlink->channels; ch++) {
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for (band = 0; band <= s->nb_splits; band++) {
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set_lp(&s->xover[ch].lp[band][0], s->splits[band], q, sample_rate);
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set_hp(&s->xover[ch].hp[band][0], s->splits[band], q, sample_rate);
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if (s->order > 1) {
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set_lp(&s->xover[ch].lp[band][1], s->splits[band], 1.34, sample_rate);
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set_hp(&s->xover[ch].hp[band][1], s->splits[band], 1.34, sample_rate);
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set_lp(&s->xover[ch].lp[band][2], s->splits[band], q, sample_rate);
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set_hp(&s->xover[ch].hp[band][2], s->splits[band], q, sample_rate);
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set_lp(&s->xover[ch].lp[band][3], s->splits[band], 1.34, sample_rate);
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set_hp(&s->xover[ch].hp[band][3], s->splits[band], 1.34, sample_rate);
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} else {
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set_lp(&s->xover[ch].lp[band][1], s->splits[band], q, sample_rate);
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set_hp(&s->xover[ch].hp[band][1], s->splits[band], q, sample_rate);
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}
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}
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}
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return 0;
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}
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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_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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static double biquad_process(BiquadContext *b, double in)
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{
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double out = in * b->a0 + b->i1 * b->a1 + b->i2 * b->a2 - b->o1 * b->b1 - b->o2 * b->b2;
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b->i2 = b->i1;
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b->o2 = b->o1;
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b->i1 = in;
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b->o1 = out;
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return out;
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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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AudioCrossoverContext *s = ctx->priv;
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AVFrame *frames[MAX_BANDS] = { NULL };
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int i, f, ch, band, ret = 0;
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for (i = 0; i < ctx->nb_outputs; i++) {
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frames[i] = ff_get_audio_buffer(ctx->outputs[i], in->nb_samples);
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if (!frames[i]) {
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ret = AVERROR(ENOMEM);
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break;
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}
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frames[i]->pts = in->pts;
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}
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if (ret < 0)
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goto fail;
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for (ch = 0; ch < inlink->channels; ch++) {
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const double *src = (const double *)in->extended_data[ch];
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CrossoverChannel *xover = &s->xover[ch];
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for (band = 0; band < ctx->nb_outputs; band++) {
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double *dst = (double *)frames[band]->extended_data[ch];
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for (i = 0; i < in->nb_samples; i++) {
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dst[i] = src[i];
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for (f = 0; f < s->filter_count; f++) {
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if (band + 1 < ctx->nb_outputs) {
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BiquadContext *lp = &xover->lp[band][f];
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dst[i] = biquad_process(lp, dst[i]);
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}
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if (band - 1 >= 0) {
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BiquadContext *hp = &xover->hp[band - 1][f];
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dst[i] = biquad_process(hp, dst[i]);
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}
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}
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}
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}
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}
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for (i = 0; i < ctx->nb_outputs; i++) {
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ret = ff_filter_frame(ctx->outputs[i], frames[i]);
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if (ret < 0)
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break;
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}
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fail:
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av_frame_free(&in);
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return ret;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AudioCrossoverContext *s = ctx->priv;
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int i;
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av_freep(&s->splits);
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for (i = 0; i < ctx->nb_outputs; i++)
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av_freep(&ctx->output_pads[i].name);
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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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.config_props = config_input,
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},
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{ NULL }
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};
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AVFilter ff_af_acrossover = {
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.name = "acrossover",
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.description = NULL_IF_CONFIG_SMALL("Split audio into per-bands streams."),
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.priv_size = sizeof(AudioCrossoverContext),
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.priv_class = &acrossover_class,
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.init = init,
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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 = NULL,
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.flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS,
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};
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