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FFmpeg/libavfilter/af_acrossover.c

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/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Crossover filter
*
* Split an audio stream into several bands.
*/
#include "libavutil/attributes.h"
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/eval.h"
#include "libavutil/internal.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "formats.h"
#include "internal.h"
#define MAX_SPLITS 16
#define MAX_BANDS MAX_SPLITS + 1
typedef struct BiquadContext {
double a0, a1, a2;
double b1, b2;
double i1, i2;
double o1, o2;
} BiquadContext;
typedef struct CrossoverChannel {
BiquadContext lp[MAX_BANDS][4];
} CrossoverChannel;
typedef struct AudioCrossoverContext {
const AVClass *class;
char *splits_str;
int order;
int filter_count;
int nb_splits;
float *splits;
CrossoverChannel *xover;
AVFrame *input_frame;
AVFrame *frames[MAX_BANDS];
} AudioCrossoverContext;
#define OFFSET(x) offsetof(AudioCrossoverContext, x)
#define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
static const AVOption acrossover_options[] = {
{ "split", "set split frequencies", OFFSET(splits_str), AV_OPT_TYPE_STRING, {.str="500"}, 0, 0, AF },
{ "order", "set order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=1}, 0, 2, AF, "m" },
{ "2nd", "2nd order", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "m" },
{ "4th", "4th order", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "m" },
{ "8th", "8th order", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "m" },
{ NULL }
};
AVFILTER_DEFINE_CLASS(acrossover);
static av_cold int init(AVFilterContext *ctx)
{
AudioCrossoverContext *s = ctx->priv;
char *p, *arg, *saveptr = NULL;
int i, ret = 0;
s->splits = av_calloc(MAX_SPLITS, sizeof(*s->splits));
if (!s->splits)
return AVERROR(ENOMEM);
p = s->splits_str;
for (i = 0; i < MAX_SPLITS; i++) {
float freq;
if (!(arg = av_strtok(p, " |", &saveptr)))
break;
p = NULL;
av_sscanf(arg, "%f", &freq);
if (freq <= 0) {
av_log(ctx, AV_LOG_ERROR, "Frequency %f must be positive number.\n", freq);
return AVERROR(EINVAL);
}
if (i > 0 && freq <= s->splits[i-1]) {
av_log(ctx, AV_LOG_ERROR, "Frequency %f must be in increasing order.\n", freq);
return AVERROR(EINVAL);
}
s->splits[i] = freq;
}
s->nb_splits = i;
for (i = 0; i <= s->nb_splits; i++) {
AVFilterPad pad = { 0 };
char *name;
pad.type = AVMEDIA_TYPE_AUDIO;
name = av_asprintf("out%d", ctx->nb_outputs);
if (!name)
return AVERROR(ENOMEM);
pad.name = name;
if ((ret = ff_insert_outpad(ctx, i, &pad)) < 0) {
av_freep(&pad.name);
return ret;
}
}
return ret;
}
static void set_lp(BiquadContext *b, double fc, double q, double sr)
{
double omega = 2.0 * M_PI * fc / sr;
double sn = sin(omega);
double cs = cos(omega);
double alpha = sn / (2. * q);
double inv = 1.0 / (1.0 + alpha);
b->a0 = (1. - cs) * 0.5 * inv;
b->a1 = (1. - cs) * inv;
b->a2 = b->a0;
b->b1 = -2. * cs * inv;
b->b2 = (1. - alpha) * inv;
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioCrossoverContext *s = ctx->priv;
int ch, band, sample_rate = inlink->sample_rate;
double q;
s->xover = av_calloc(inlink->channels, sizeof(*s->xover));
if (!s->xover)
return AVERROR(ENOMEM);
switch (s->order) {
case 0:
q = 0.5;
s->filter_count = 1;
break;
case 1:
q = M_SQRT1_2;
s->filter_count = 2;
break;
case 2:
q = 0.54;
s->filter_count = 4;
break;
}
for (ch = 0; ch < inlink->channels; ch++) {
for (band = 0; band <= s->nb_splits; band++) {
set_lp(&s->xover[ch].lp[band][0], s->splits[band], q, sample_rate);
if (s->order > 1) {
set_lp(&s->xover[ch].lp[band][1], s->splits[band], 1.34, sample_rate);
set_lp(&s->xover[ch].lp[band][2], s->splits[band], q, sample_rate);
set_lp(&s->xover[ch].lp[band][3], s->splits[band], 1.34, sample_rate);
} else {
set_lp(&s->xover[ch].lp[band][1], s->splits[band], q, sample_rate);
}
}
}
return 0;
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
int ret;
layouts = ff_all_channel_counts();
if (!layouts)
return AVERROR(ENOMEM);
ret = ff_set_common_channel_layouts(ctx, layouts);
if (ret < 0)
return ret;
formats = ff_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
ret = ff_set_common_formats(ctx, formats);
if (ret < 0)
return ret;
formats = ff_all_samplerates();
if (!formats)
return AVERROR(ENOMEM);
return ff_set_common_samplerates(ctx, formats);
}
static double biquad_process(BiquadContext *b, double in)
{
double out = in * b->a0 + b->i1 * b->a1 + b->i2 * b->a2 - b->o1 * b->b1 - b->o2 * b->b2;
b->i2 = b->i1;
b->o2 = b->o1;
b->i1 = in;
b->o1 = out;
return out;
}
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
AudioCrossoverContext *s = ctx->priv;
AVFrame *in = s->input_frame;
AVFrame **frames = s->frames;
const int start = (in->channels * jobnr) / nb_jobs;
const int end = (in->channels * (jobnr+1)) / nb_jobs;
int f, band;
for (int ch = start; ch < end; ch++) {
const double *src = (const double *)in->extended_data[ch];
CrossoverChannel *xover = &s->xover[ch];
for (int i = 0; i < in->nb_samples; i++) {
double sample = src[i], lo, hi;
for (band = 0; band < ctx->nb_outputs; band++) {
double *dst = (double *)frames[band]->extended_data[ch];
lo = sample;
for (f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) {
BiquadContext *lp = &xover->lp[band][f];
lo = biquad_process(lp, lo);
}
hi = sample - lo;
dst[i] = lo;
sample = hi;
}
}
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AudioCrossoverContext *s = ctx->priv;
AVFrame **frames = s->frames;
int i, ret = 0;
for (i = 0; i < ctx->nb_outputs; i++) {
frames[i] = ff_get_audio_buffer(ctx->outputs[i], in->nb_samples);
if (!frames[i]) {
ret = AVERROR(ENOMEM);
break;
}
frames[i]->pts = in->pts;
}
if (ret < 0)
goto fail;
s->input_frame = in;
ctx->internal->execute(ctx, filter_channels, NULL, NULL, FFMIN(inlink->channels,
ff_filter_get_nb_threads(ctx)));
for (i = 0; i < ctx->nb_outputs; i++) {
ret = ff_filter_frame(ctx->outputs[i], frames[i]);
frames[i] = NULL;
if (ret < 0)
break;
}
fail:
for (i = 0; i < ctx->nb_outputs; i++)
av_frame_free(&frames[i]);
av_frame_free(&in);
s->input_frame = NULL;
return ret;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioCrossoverContext *s = ctx->priv;
int i;
av_freep(&s->splits);
av_freep(&s->xover);
for (i = 0; i < ctx->nb_outputs; i++)
av_freep(&ctx->output_pads[i].name);
}
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
},
{ NULL }
};
AVFilter ff_af_acrossover = {
.name = "acrossover",
.description = NULL_IF_CONFIG_SMALL("Split audio into per-bands streams."),
.priv_size = sizeof(AudioCrossoverContext),
.priv_class = &acrossover_class,
.init = init,
.uninit = uninit,
.query_formats = query_formats,
.inputs = inputs,
.outputs = NULL,
.flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
AVFILTER_FLAG_SLICE_THREADS,
};
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