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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/float_dsp.h"
#include "libavutil/internal.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
#include "formats.h"
#include "internal.h"
#define MAX_SPLITS 16
#define MAX_BANDS MAX_SPLITS + 1
#define B0 0
#define B1 1
#define B2 2
#define A1 3
#define A2 4
typedef struct BiquadCoeffs {
double cd[5];
float cf[5];
} BiquadCoeffs;
typedef struct AudioCrossoverContext {
const AVClass *class;
char *splits_str;
char *gains_str;
int order_opt;
float level_in;
int precision;
int order;
int filter_count;
int first_order;
int ap_filter_count;
int nb_splits;
float splits[MAX_SPLITS];
float gains[MAX_BANDS];
BiquadCoeffs lp[MAX_BANDS][20];
BiquadCoeffs hp[MAX_BANDS][20];
BiquadCoeffs ap[MAX_BANDS][20];
AVFrame *xover;
AVFrame *frames[MAX_BANDS];
int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
AVFloatDSPContext *fdsp;
} 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 filter order", OFFSET(order_opt), AV_OPT_TYPE_INT, {.i64=1}, 0, 9, AF, .unit = "m" },
{ "2nd", "2nd order (12 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "m" },
{ "4th", "4th order (24 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "m" },
{ "6th", "6th order (36 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "m" },
{ "8th", "8th order (48 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, .unit = "m" },
{ "10th", "10th order (60 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, .unit = "m" },
{ "12th", "12th order (72 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, AF, .unit = "m" },
{ "14th", "14th order (84 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, AF, .unit = "m" },
{ "16th", "16th order (96 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, AF, .unit = "m" },
{ "18th", "18th order (108 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, AF, .unit = "m" },
{ "20th", "20th order (120 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=9}, 0, 0, AF, .unit = "m" },
{ "level", "set input gain", OFFSET(level_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AF },
{ "gain", "set output bands gain", OFFSET(gains_str), AV_OPT_TYPE_STRING, {.str="1.f"}, 0, 0, AF },
{ "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, AF, .unit = "precision" },
{ "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "precision" },
{ "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "precision" },
{ "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "precision" },
{ NULL }
};
AVFILTER_DEFINE_CLASS(acrossover);
static int query_formats(AVFilterContext *ctx)
{
AudioCrossoverContext *s = ctx->priv;
static const enum AVSampleFormat auto_sample_fmts[] = {
AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE
};
const enum AVSampleFormat *sample_fmts_list = sample_fmts;
int ret = ff_set_common_all_channel_counts(ctx);
if (ret < 0)
return ret;
switch (s->precision) {
case 0:
sample_fmts_list = auto_sample_fmts;
break;
case 1:
sample_fmts[0] = AV_SAMPLE_FMT_FLTP;
break;
case 2:
sample_fmts[0] = AV_SAMPLE_FMT_DBLP;
break;
default:
break;
}
ret = ff_set_common_formats_from_list(ctx, sample_fmts_list);
if (ret < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
static int parse_gains(AVFilterContext *ctx)
{
AudioCrossoverContext *s = ctx->priv;
char *p, *arg, *saveptr = NULL;
int i, ret = 0;
saveptr = NULL;
p = s->gains_str;
for (i = 0; i < MAX_BANDS; i++) {
float gain;
char c[3] = { 0 };
if (!(arg = av_strtok(p, " |", &saveptr)))
break;
p = NULL;
if (av_sscanf(arg, "%f%2s", &gain, c) < 1) {
av_log(ctx, AV_LOG_ERROR, "Invalid syntax for gain[%d].\n", i);
ret = AVERROR(EINVAL);
break;
}
if (c[0] == 'd' && c[1] == 'B')
s->gains[i] = expf(gain * M_LN10 / 20.f);
else
s->gains[i] = gain;
}
for (; i < MAX_BANDS; i++)
s->gains[i] = 1.f;
return ret;
}
static av_cold int init(AVFilterContext *ctx)
{
AudioCrossoverContext *s = ctx->priv;
char *p, *arg, *saveptr = NULL;
int i, ret = 0;
s->fdsp = avpriv_float_dsp_alloc(0);
if (!s->fdsp)
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;
if (av_sscanf(arg, "%f", &freq) != 1) {
av_log(ctx, AV_LOG_ERROR, "Invalid syntax for frequency[%d].\n", i);
return AVERROR(EINVAL);
}
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;
ret = parse_gains(ctx);
if (ret < 0)
return ret;
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_append_outpad_free_name(ctx, &pad)) < 0)
return ret;
}
return ret;
}
static void set_lp(BiquadCoeffs *b, double fc, double q, double sr)
{
double omega = 2. * M_PI * fc / sr;
double cosine = cos(omega);
double alpha = sin(omega) / (2. * q);
double b0 = (1. - cosine) / 2.;
double b1 = 1. - cosine;
double b2 = (1. - cosine) / 2.;
double a0 = 1. + alpha;
double a1 = -2. * cosine;
double a2 = 1. - alpha;
b->cd[B0] = b0 / a0;
b->cd[B1] = b1 / a0;
b->cd[B2] = b2 / a0;
b->cd[A1] = -a1 / a0;
b->cd[A2] = -a2 / a0;
b->cf[B0] = b->cd[B0];
b->cf[B1] = b->cd[B1];
b->cf[B2] = b->cd[B2];
b->cf[A1] = b->cd[A1];
b->cf[A2] = b->cd[A2];
}
static void set_hp(BiquadCoeffs *b, double fc, double q, double sr)
{
double omega = 2. * M_PI * fc / sr;
double cosine = cos(omega);
double alpha = sin(omega) / (2. * q);
double b0 = (1. + cosine) / 2.;
double b1 = -1. - cosine;
double b2 = (1. + cosine) / 2.;
double a0 = 1. + alpha;
double a1 = -2. * cosine;
double a2 = 1. - alpha;
b->cd[B0] = b0 / a0;
b->cd[B1] = b1 / a0;
b->cd[B2] = b2 / a0;
b->cd[A1] = -a1 / a0;
b->cd[A2] = -a2 / a0;
b->cf[B0] = b->cd[B0];
b->cf[B1] = b->cd[B1];
b->cf[B2] = b->cd[B2];
b->cf[A1] = b->cd[A1];
b->cf[A2] = b->cd[A2];
}
static void set_ap(BiquadCoeffs *b, double fc, double q, double sr)
{
double omega = 2. * M_PI * fc / sr;
double cosine = cos(omega);
double alpha = sin(omega) / (2. * q);
double a0 = 1. + alpha;
double a1 = -2. * cosine;
double a2 = 1. - alpha;
double b0 = a2;
double b1 = a1;
double b2 = a0;
b->cd[B0] = b0 / a0;
b->cd[B1] = b1 / a0;
b->cd[B2] = b2 / a0;
b->cd[A1] = -a1 / a0;
b->cd[A2] = -a2 / a0;
b->cf[B0] = b->cd[B0];
b->cf[B1] = b->cd[B1];
b->cf[B2] = b->cd[B2];
b->cf[A1] = b->cd[A1];
b->cf[A2] = b->cd[A2];
}
static void set_ap1(BiquadCoeffs *b, double fc, double sr)
{
double omega = 2. * M_PI * fc / sr;
b->cd[A1] = exp(-omega);
b->cd[A2] = 0.;
b->cd[B0] = -b->cd[A1];
b->cd[B1] = 1.;
b->cd[B2] = 0.;
b->cf[B0] = b->cd[B0];
b->cf[B1] = b->cd[B1];
b->cf[B2] = b->cd[B2];
b->cf[A1] = b->cd[A1];
b->cf[A2] = b->cd[A2];
}
static void calc_q_factors(int order, double *q)
{
double n = order / 2.;
for (int i = 0; i < n / 2; i++)
q[i] = 1. / (-2. * cos(M_PI * (2. * (i + 1) + n - 1.) / (2. * n)));
}
#define BIQUAD_PROCESS(name, type) \
static void biquad_process_## name(const type *const c, \
type *b, \
type *dst, const type *src, \
int nb_samples) \
{ \
const type b0 = c[B0]; \
const type b1 = c[B1]; \
const type b2 = c[B2]; \
const type a1 = c[A1]; \
const type a2 = c[A2]; \
type z1 = b[0]; \
type z2 = b[1]; \
\
for (int n = 0; n + 1 < nb_samples; n++) { \
type in = src[n]; \
type out; \
\
out = in * b0 + z1; \
z1 = b1 * in + z2 + a1 * out; \
z2 = b2 * in + a2 * out; \
dst[n] = out; \
\
n++; \
in = src[n]; \
out = in * b0 + z1; \
z1 = b1 * in + z2 + a1 * out; \
z2 = b2 * in + a2 * out; \
dst[n] = out; \
} \
\
if (nb_samples & 1) { \
const int n = nb_samples - 1; \
const type in = src[n]; \
type out; \
\
out = in * b0 + z1; \
z1 = b1 * in + z2 + a1 * out; \
z2 = b2 * in + a2 * out; \
dst[n] = out; \
} \
\
b[0] = z1; \
b[1] = z2; \
}
BIQUAD_PROCESS(fltp, float)
BIQUAD_PROCESS(dblp, double)
#define XOVER_PROCESS(name, type, one, ff) \
static int filter_channels_## name(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) \
{ \
AudioCrossoverContext *s = ctx->priv; \
AVFrame *in = arg; \
AVFrame **frames = s->frames; \
const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs; \
const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs; \
const int nb_samples = in->nb_samples; \
const int nb_outs = ctx->nb_outputs; \
const int first_order = s->first_order; \
\
for (int ch = start; ch < end; ch++) { \
const type *src = (const type *)in->extended_data[ch]; \
type *xover = (type *)s->xover->extended_data[ch]; \
\
s->fdsp->vector_## ff ##mul_scalar((type *)frames[0]->extended_data[ch], src, \
s->level_in, FFALIGN(nb_samples, sizeof(type))); \
\
for (int band = 0; band < nb_outs; band++) { \
for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) { \
const type *prv = (const type *)frames[band]->extended_data[ch]; \
type *dst = (type *)frames[band + 1]->extended_data[ch]; \
const type *hsrc = f == 0 ? prv : dst; \
type *hp = xover + nb_outs * 20 + band * 20 + f * 2; \
const type *const hpc = (type *)&s->hp[band][f].c ## ff; \
\
biquad_process_## name(hpc, hp, dst, hsrc, nb_samples); \
} \
\
for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) { \
type *dst = (type *)frames[band]->extended_data[ch]; \
const type *lsrc = dst; \
type *lp = xover + band * 20 + f * 2; \
const type *const lpc = (type *)&s->lp[band][f].c ## ff; \
\
biquad_process_## name(lpc, lp, dst, lsrc, nb_samples); \
} \
\
for (int aband = band + 1; aband + 1 < nb_outs; aband++) { \
if (first_order) { \
const type *asrc = (const type *)frames[band]->extended_data[ch]; \
type *dst = (type *)frames[band]->extended_data[ch]; \
type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20; \
const type *const apc = (type *)&s->ap[aband][0].c ## ff; \
\
biquad_process_## name(apc, ap, dst, asrc, nb_samples); \
} \
\
for (int f = first_order; f < s->ap_filter_count; f++) { \
const type *asrc = (const type *)frames[band]->extended_data[ch]; \
type *dst = (type *)frames[band]->extended_data[ch]; \
type *ap = xover + nb_outs * 40 + (aband * nb_outs + band) * 20 + f * 2;\
const type *const apc = (type *)&s->ap[aband][f].c ## ff; \
\
biquad_process_## name(apc, ap, dst, asrc, nb_samples); \
} \
} \
} \
\
for (int band = 0; band < nb_outs; band++) { \
const type gain = s->gains[band] * ((band & 1 && first_order) ? -one : one); \
type *dst = (type *)frames[band]->extended_data[ch]; \
\
s->fdsp->vector_## ff ##mul_scalar(dst, dst, gain, \
FFALIGN(nb_samples, sizeof(type))); \
} \
} \
\
return 0; \
}
XOVER_PROCESS(fltp, float, 1.f, f)
XOVER_PROCESS(dblp, double, 1.0, d)
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioCrossoverContext *s = ctx->priv;
int sample_rate = inlink->sample_rate;
double q[16];
s->order = (s->order_opt + 1) * 2;
s->filter_count = s->order / 2;
s->first_order = s->filter_count & 1;
s->ap_filter_count = s->filter_count / 2 + s->first_order;
calc_q_factors(s->order, q);
for (int band = 0; band <= s->nb_splits; band++) {
if (s->first_order) {
set_lp(&s->lp[band][0], s->splits[band], 0.5, sample_rate);
set_hp(&s->hp[band][0], s->splits[band], 0.5, sample_rate);
}
for (int n = s->first_order; n < s->filter_count; n++) {
const int idx = s->filter_count / 2 - ((n + s->first_order) / 2 - s->first_order) - 1;
set_lp(&s->lp[band][n], s->splits[band], q[idx], sample_rate);
set_hp(&s->hp[band][n], s->splits[band], q[idx], sample_rate);
}
if (s->first_order)
set_ap1(&s->ap[band][0], s->splits[band], sample_rate);
for (int n = s->first_order; n < s->ap_filter_count; n++) {
const int idx = (s->filter_count / 2 - ((n * 2 + s->first_order) / 2 - s->first_order) - 1);
set_ap(&s->ap[band][n], s->splits[band], q[idx], sample_rate);
}
}
switch (inlink->format) {
case AV_SAMPLE_FMT_FLTP: s->filter_channels = filter_channels_fltp; break;
case AV_SAMPLE_FMT_DBLP: s->filter_channels = filter_channels_dblp; break;
default: return AVERROR_BUG;
}
s->xover = ff_get_audio_buffer(inlink, 2 * (ctx->nb_outputs * 10 + ctx->nb_outputs * 10 +
ctx->nb_outputs * ctx->nb_outputs * 10));
if (!s->xover)
return AVERROR(ENOMEM);
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AudioCrossoverContext *s = ctx->priv;
AVFrame **frames = s->frames;
int ret = 0;
for (int 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;
ff_filter_execute(ctx, s->filter_channels, in, NULL,
FFMIN(inlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
for (int i = 0; i < ctx->nb_outputs; i++) {
if (ff_outlink_get_status(ctx->outputs[i])) {
av_frame_free(&frames[i]);
continue;
}
ret = ff_filter_frame(ctx->outputs[i], frames[i]);
frames[i] = NULL;
if (ret < 0)
break;
}
fail:
for (int i = 0; i < ctx->nb_outputs; i++)
av_frame_free(&frames[i]);
return ret;
}
static int activate(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
int status, ret;
AVFrame *in;
int64_t pts;
for (int i = 0; i < ctx->nb_outputs; i++) {
FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[i], ctx);
}
ret = ff_inlink_consume_frame(inlink, &in);
if (ret < 0)
return ret;
if (ret > 0) {
ret = filter_frame(inlink, in);
av_frame_free(&in);
if (ret < 0)
return ret;
}
if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
for (int i = 0; i < ctx->nb_outputs; i++) {
if (ff_outlink_get_status(ctx->outputs[i]))
continue;
ff_outlink_set_status(ctx->outputs[i], status, pts);
}
return 0;
}
for (int i = 0; i < ctx->nb_outputs; i++) {
if (ff_outlink_get_status(ctx->outputs[i]))
continue;
if (ff_outlink_frame_wanted(ctx->outputs[i])) {
ff_inlink_request_frame(inlink);
return 0;
}
}
return FFERROR_NOT_READY;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioCrossoverContext *s = ctx->priv;
av_freep(&s->fdsp);
av_frame_free(&s->xover);
}
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
},
};
const 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,
.activate = activate,
.uninit = uninit,
FILTER_INPUTS(inputs),
.outputs = NULL,
FILTER_QUERY_FUNC(query_formats),
.flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
AVFILTER_FLAG_SLICE_THREADS,
};