mirror of
https://github.com/jellyfin/jellyfin-ffmpeg.git
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252 lines
8.1 KiB
C
252 lines
8.1 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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#include "libavutil/channel_layout.h"
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#include "libavutil/ffmath.h"
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "audio.h"
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typedef struct ASubBoostContext {
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const AVClass *class;
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double dry_gain;
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double wet_gain;
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double feedback;
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double max_boost;
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double decay;
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double delay;
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double cutoff;
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double slope;
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double a0, a1, a2;
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double b0, b1, b2;
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char *ch_layout_str;
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AVChannelLayout ch_layout;
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int *write_pos;
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int buffer_samples;
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AVFrame *w;
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AVFrame *buffer;
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} ASubBoostContext;
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static int get_coeffs(AVFilterContext *ctx)
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{
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ASubBoostContext *s = ctx->priv;
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AVFilterLink *inlink = ctx->inputs[0];
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double w0 = 2 * M_PI * s->cutoff / inlink->sample_rate;
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double alpha = sin(w0) / 2 * sqrt(2. * (1. / s->slope - 1.) + 2.);
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s->a0 = 1 + alpha;
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s->a1 = -2 * cos(w0);
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s->a2 = 1 - alpha;
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s->b0 = (1 - cos(w0)) / 2;
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s->b1 = 1 - cos(w0);
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s->b2 = (1 - cos(w0)) / 2;
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s->a1 /= s->a0;
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s->a2 /= s->a0;
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s->b0 /= s->a0;
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s->b1 /= s->a0;
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s->b2 /= s->a0;
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s->buffer_samples = inlink->sample_rate * s->delay / 1000;
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return 0;
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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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ASubBoostContext *s = ctx->priv;
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s->buffer = ff_get_audio_buffer(inlink, inlink->sample_rate / 10);
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s->w = ff_get_audio_buffer(inlink, 3);
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s->write_pos = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->write_pos));
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if (!s->buffer || !s->w || !s->write_pos)
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return AVERROR(ENOMEM);
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return get_coeffs(ctx);
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}
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typedef struct ThreadData {
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AVFrame *in, *out;
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} ThreadData;
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static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
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{
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ASubBoostContext *s = ctx->priv;
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ThreadData *td = arg;
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AVFrame *out = td->out;
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AVFrame *in = td->in;
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const double mix = ctx->is_disabled ? 0. : 1.;
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const double wet = ctx->is_disabled ? 1. : s->wet_gain;
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const double dry = ctx->is_disabled ? 1. : s->dry_gain;
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const double feedback = s->feedback, decay = s->decay;
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const double max_boost = s->max_boost;
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const double b0 = s->b0;
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const double b1 = s->b1;
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const double b2 = s->b2;
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const double a1 = -s->a1;
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const double a2 = -s->a2;
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const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
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const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
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const int buffer_samples = s->buffer_samples;
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for (int ch = start; ch < end; ch++) {
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const double *src = (const double *)in->extended_data[ch];
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double *dst = (double *)out->extended_data[ch];
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double *buffer = (double *)s->buffer->extended_data[ch];
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double *w = (double *)s->w->extended_data[ch];
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int write_pos = s->write_pos[ch];
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enum AVChannel channel = av_channel_layout_channel_from_index(&in->ch_layout, ch);
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const int bypass = av_channel_layout_index_from_channel(&s->ch_layout, channel) < 0;
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const double a = 0.00001;
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const double b = 1. - a;
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if (bypass) {
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if (in != out)
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memcpy(out->extended_data[ch], in->extended_data[ch],
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in->nb_samples * sizeof(double));
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continue;
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}
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for (int n = 0; n < in->nb_samples; n++) {
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double out_sample, boost;
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out_sample = src[n] * b0 + w[0];
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w[0] = b1 * src[n] + w[1] + a1 * out_sample;
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w[1] = b2 * src[n] + a2 * out_sample;
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buffer[write_pos] = buffer[write_pos] * decay + out_sample * feedback;
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boost = av_clipd((1. - (fabs(src[n] * dry))) / fabs(buffer[write_pos]), 0., max_boost);
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w[2] = boost > w[2] ? w[2] * b + a * boost : w[2] * a + b * boost;
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w[2] = av_clipd(w[2], 0., max_boost);
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dst[n] = (src[n] * dry + w[2] * buffer[write_pos] * mix) * wet;
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if (++write_pos >= buffer_samples)
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write_pos = 0;
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}
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s->write_pos[ch] = write_pos;
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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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ASubBoostContext *s = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[0];
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ThreadData td;
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AVFrame *out;
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int ret;
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ret = av_channel_layout_copy(&s->ch_layout, &inlink->ch_layout);
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if (ret < 0)
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return ret;
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if (strcmp(s->ch_layout_str, "all"))
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av_channel_layout_from_string(&s->ch_layout,
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s->ch_layout_str);
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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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td.in = in; td.out = out;
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ff_filter_execute(ctx, filter_channels, &td, NULL,
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FFMIN(inlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
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if (out != in)
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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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ASubBoostContext *s = ctx->priv;
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av_channel_layout_uninit(&s->ch_layout);
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av_frame_free(&s->buffer);
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av_frame_free(&s->w);
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av_freep(&s->write_pos);
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}
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static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
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char *res, int res_len, int flags)
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{
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int ret;
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ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
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if (ret < 0)
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return ret;
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return get_coeffs(ctx);
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}
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#define OFFSET(x) offsetof(ASubBoostContext, x)
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
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static const AVOption asubboost_options[] = {
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{ "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1.0}, 0, 1, FLAGS },
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{ "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1.0}, 0, 1, FLAGS },
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{ "boost", "set max boost",OFFSET(max_boost),AV_OPT_TYPE_DOUBLE, {.dbl=2.0}, 1, 12, FLAGS },
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{ "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=0.0}, 0, 1, FLAGS },
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{ "feedback", "set feedback", OFFSET(feedback), AV_OPT_TYPE_DOUBLE, {.dbl=0.9}, 0, 1, FLAGS },
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{ "cutoff", "set cutoff", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, {.dbl=100}, 50, 900, FLAGS },
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{ "slope", "set slope", OFFSET(slope), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.0001, 1, FLAGS },
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{ "delay", "set delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 1, 100, FLAGS },
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{ "channels", "set channels to filter", OFFSET(ch_layout_str), AV_OPT_TYPE_STRING, {.str="all"}, 0, 0, FLAGS },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(asubboost);
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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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};
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const AVFilter ff_af_asubboost = {
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.name = "asubboost",
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.description = NULL_IF_CONFIG_SMALL("Boost subwoofer frequencies."),
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.priv_size = sizeof(ASubBoostContext),
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.priv_class = &asubboost_class,
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.uninit = uninit,
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FILTER_INPUTS(inputs),
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FILTER_OUTPUTS(ff_audio_default_filterpad),
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FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
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.process_command = process_command,
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.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
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AVFILTER_FLAG_SLICE_THREADS,
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};
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