#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"
#define MAX_SPLITS 16
#define MAX_BANDS MAX_SPLITS + 1
-typedef struct BiquadContext {
- double b0, b1, b2;
- double a1, a2;
- double z1, z2;
-} BiquadContext;
+#define B0 0
+#define B1 1
+#define B2 2
+#define A1 3
+#define A2 4
-typedef struct CrossoverChannel {
- BiquadContext lp[MAX_BANDS][20];
- BiquadContext hp[MAX_BANDS][20];
- BiquadContext ap[MAX_BANDS][MAX_BANDS][20];
-} CrossoverChannel;
+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 order;
int filter_count;
int first_order;
int ap_filter_count;
int nb_splits;
- float *splits;
+ float splits[MAX_SPLITS];
- CrossoverChannel *xover;
+ float gains[MAX_BANDS];
+
+ BiquadCoeffs lp[MAX_BANDS][20];
+ BiquadCoeffs hp[MAX_BANDS][20];
+ BiquadCoeffs ap[MAX_BANDS][20];
+
+ AVFrame *xover;
AVFrame *input_frame;
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)
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_opt), AV_OPT_TYPE_INT, {.i64=1}, 0, 9, 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" },
- { "6th", "6th order", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "m" },
- { "8th", "8th order", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "m" },
- { "10th", "10th order", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, "m" },
- { "12th", "12th order", 0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, AF, "m" },
- { "14th", "14th order", 0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, AF, "m" },
- { "16th", "16th order", 0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, AF, "m" },
- { "18th", "18th order", 0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, AF, "m" },
- { "20th", "20th order", 0, AV_OPT_TYPE_CONST, {.i64=9}, 0, 0, AF, "m" },
+ { "order", "set filter order", OFFSET(order_opt), AV_OPT_TYPE_INT, {.i64=1}, 0, 9, AF, "m" },
+ { "2nd", "2nd order (12 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "m" },
+ { "4th", "4th order (24 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "m" },
+ { "6th", "6th order (36 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "m" },
+ { "8th", "8th order (48 dB/8ve)", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "m" },
+ { "10th", "10th order (60 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, AF, "m" },
+ { "12th", "12th order (72 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, AF, "m" },
+ { "14th", "14th order (84 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, AF, "m" },
+ { "16th", "16th order (96 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, AF, "m" },
+ { "18th", "18th order (108 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, AF, "m" },
+ { "20th", "20th order (120 dB/8ve)",0, AV_OPT_TYPE_CONST, {.i64=9}, 0, 0, AF, "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 },
{ NULL }
};
AVFILTER_DEFINE_CLASS(acrossover);
+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->splits = av_calloc(MAX_SPLITS, sizeof(*s->splits));
- if (!s->splits)
+ s->fdsp = avpriv_float_dsp_alloc(0);
+ if (!s->fdsp)
return AVERROR(ENOMEM);
p = s->splits_str;
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;
return ret;
}
-static void set_lp(BiquadContext *b, double fc, double q, double sr)
+static void set_lp(BiquadCoeffs *b, double fc, double q, double sr)
{
- double omega = M_PI * fc / sr;
+ double omega = 2. * M_PI * fc / sr;
double cosine = cos(omega);
double alpha = sin(omega) / (2. * q);
double a1 = -2. * cosine;
double a2 = 1. - alpha;
- b->b0 = b0 / a0;
- b->b1 = b1 / a0;
- b->b2 = b2 / a0;
- b->a1 = -a1 / a0;
- b->a2 = -a2 / 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_hp(BiquadContext *b, double fc, double q, double sr)
+static void set_hp(BiquadCoeffs *b, double fc, double q, double sr)
{
- double omega = M_PI * fc / sr;
+ double omega = 2. * M_PI * fc / sr;
double cosine = cos(omega);
double alpha = sin(omega) / (2. * q);
double a1 = -2. * cosine;
double a2 = 1. - alpha;
- b->b0 = b0 / a0;
- b->b1 = b1 / a0;
- b->b2 = b2 / a0;
- b->a1 = -a1 / a0;
- b->a2 = -a2 / 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_ap(BiquadContext *b, double fc, double q, double sr)
+static void set_ap(BiquadCoeffs *b, double fc, double q, double sr)
{
- double omega = M_PI * fc / sr;
+ double omega = 2. * M_PI * fc / sr;
double cosine = cos(omega);
double alpha = sin(omega) / (2. * q);
double b1 = a1;
double b2 = a0;
- b->b0 = b0 / a0;
- b->b1 = b1 / a0;
- b->b2 = b2 / a0;
- b->a1 = -a1 / a0;
- b->a2 = -a2 / 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(BiquadContext *b, double fc, double sr)
+static void set_ap1(BiquadCoeffs *b, double fc, double sr)
{
- double omega = M_PI * fc / sr;
-
- b->a1 = exp(-omega);
- b->a2 = 0.;
- b->b0 = -b->a1;
- b->b1 = 1.;
- b->b2 = 0.;
+ 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)
}
#define BIQUAD_PROCESS(name, type) \
-static void biquad_process_## name(BiquadContext *b, \
+static void biquad_process_## name(const type *const c, \
+ type *b, \
type *dst, const type *src, \
int nb_samples) \
{ \
- const type b0 = b->b0; \
- const type b1 = b->b1; \
- const type b2 = b->b2; \
- const type a1 = b->a1; \
- const type a2 = b->a2; \
- type z1 = b->z1; \
- type z2 = b->z2; \
+ 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; \
\
- for (int n = 0; n < nb_samples; n++) { \
+ 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; \
\
dst[n] = out; \
} \
\
- b->z1 = z1; \
- b->z2 = z2; \
+ b[0] = z1; \
+ b[1] = z2; \
}
BIQUAD_PROCESS(fltp, float)
BIQUAD_PROCESS(dblp, double)
-#define XOVER_PROCESS(name, type, one) \
+#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; \
const int start = (in->channels * jobnr) / nb_jobs; \
const int end = (in->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++) { \
- CrossoverChannel *xover = &s->xover[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 < ctx->nb_outputs; band++) { \
- for (int f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) { \
- const type *src = (const type *)in->extended_data[ch]; \
+ 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 = (band == 0 && f == 0) ? src : f == 0 ? prv : dst; \
- BiquadContext *hp = &xover->hp[band][f]; \
+ 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(hp, dst, hsrc, nb_samples); \
+ biquad_process_## name(hpc, hp, dst, hsrc, nb_samples); \
} \
\
- for (int f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) { \
- const type *src = (const type *)in->extended_data[ch]; \
+ for (int f = 0; band + 1 < nb_outs && f < s->filter_count; f++) { \
type *dst = (type *)frames[band]->extended_data[ch]; \
- const type *lsrc = (band == 0 && f == 0) ? src : dst; \
- BiquadContext *lp = &xover->lp[band][f]; \
+ 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(lp, dst, lsrc, nb_samples); \
+ biquad_process_## name(lpc, lp, dst, lsrc, nb_samples); \
} \
\
- for (int aband = band + 1; aband + 1 < ctx->nb_outputs; aband++) { \
- if (s->first_order) { \
+ 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]; \
- BiquadContext *ap = &xover->ap[band][aband][0]; \
+ 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(ap, dst, asrc, nb_samples); \
+ biquad_process_## name(apc, ap, dst, asrc, nb_samples); \
} \
\
- for (int f = s->first_order; f < s->ap_filter_count; f++) { \
+ 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]; \
- BiquadContext *ap = &xover->ap[band][aband][f]; \
+ 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(ap, dst, asrc, nb_samples); \
+ biquad_process_## name(apc, ap, dst, asrc, nb_samples); \
} \
} \
} \
\
- for (int band = 0; band < ctx->nb_outputs && s->first_order; band++) { \
- if (band & 1) { \
- type *dst = (type *)frames[band]->extended_data[ch]; \
+ 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]; \
\
- for (int n = 0; n < nb_samples; n++) \
- dst[n] *= -one; \
- } \
+ s->fdsp->vector_## ff ##mul_scalar(dst, dst, gain, \
+ FFALIGN(nb_samples, sizeof(type))); \
} \
} \
\
return 0; \
}
-XOVER_PROCESS(fltp, float, 1.f)
-XOVER_PROCESS(dblp, double, 1.0)
+XOVER_PROCESS(fltp, float, 1.f, f)
+XOVER_PROCESS(dblp, double, 1.0, d)
static int config_input(AVFilterLink *inlink)
{
int sample_rate = inlink->sample_rate;
double q[16];
- s->xover = av_calloc(inlink->channels, sizeof(*s->xover));
- if (!s->xover)
- return AVERROR(ENOMEM);
-
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 ch = 0; ch < inlink->channels; ch++) {
- for (int band = 0; band <= s->nb_splits; band++) {
- if (s->first_order) {
- set_lp(&s->xover[ch].lp[band][0], s->splits[band], 0.5, sample_rate);
- set_hp(&s->xover[ch].hp[band][0], s->splits[band], 0.5, sample_rate);
- }
+ 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;
+ 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->xover[ch].lp[band][n], s->splits[band], q[idx], sample_rate);
- set_hp(&s->xover[ch].hp[band][n], s->splits[band], q[idx], sample_rate);
- }
+ 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);
+ }
- for (int x = 0; x <= s->nb_splits && s->first_order; x++)
- set_ap1(&s->xover[ch].ap[x][band][0], s->splits[band], 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);
+ 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);
- for (int x = 0; x <= s->nb_splits; x++)
- set_ap(&s->xover[ch].ap[x][band][n], s->splits[band], q[idx], sample_rate);
- }
+ set_ap(&s->ap[band][n], s->splits[band], q[idx], sample_rate);
}
}
case AV_SAMPLE_FMT_DBLP: s->filter_channels = filter_channels_dblp; break;
}
+ 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;
}
AudioCrossoverContext *s = ctx->priv;
int i;
- av_freep(&s->splits);
- av_freep(&s->xover);
+ av_freep(&s->fdsp);
+ av_frame_free(&s->xover);
for (i = 0; i < ctx->nb_outputs; i++)
av_freep(&ctx->output_pads[i].name);
{ NULL }
};
-AVFilter ff_af_acrossover = {
+const AVFilter ff_af_acrossover = {
.name = "acrossover",
.description = NULL_IF_CONFIG_SMALL("Split audio into per-bands streams."),
.priv_size = sizeof(AudioCrossoverContext),
projects / ffmpeg / blobdiff