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avfilter: add FIR equalizer coefficients source filter

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This commit is contained in:
Paul B Mahol 2023-01-13 13:32:26 +01:00
parent 595f0cbb4a
commit 19148a5b9f
6 changed files with 360 additions and 5 deletions
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1
Changelog
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@ -5,6 +5,7 @@ version <next>:
- libaribcaption decoder
- Playdate video decoder and demuxer
- Extend VAAPI support for libva-win32 on Windows
- afireqsrc audio source filter
version 6.0:
- Radiance HDR image support

65
doc/filters.texi
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@ -7624,6 +7624,71 @@ Specifies the channel layout, and can be a string representing a channel layout.
The default value of @var{channel_layout} is "stereo".
@end table
@section afireqsrc
Generate a FIR equalizer coefficients.
The resulting stream can be used with @ref{afir} filter for filtering the audio signal.
The filter accepts the following options:
@table @option
@item preset, p
Set equalizer preset.
Default preset is @code{flat}.
Available presets are:
@table @samp
@item custom
@item flat
@item acoustic
@item bass
@item beats
@item classic
@item clear
@item deep bass
@item dubstep
@item electronic
@item hard-style
@item hip-hop
@item jazz
@item metal
@item movie
@item pop
@item r&b
@item rock
@item vocal booster
@end table
@item gains, g
Set custom gains for each band. Only used if the preset option is set to @code{custom}.
Gains are separated by white spaces and each gain is set in dBFS.
Default is @code{0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0}.
@item bands, b
Set the custom bands from where custon equalizer gains are set.
This must be in strictly increasing order. Only used if the preset option is set to @code{custom}.
Bands are separated by white spaces and each band represent frequency in Hz.
Default is @code{25 40 63 100 160 250 400 630 1000 1600 2500 4000 6300 10000 16000 24000}.
@item taps, t
Set number of filter coefficents in output audio stream.
Default value is @code{4096}.
@item sample_rate, r
Set sample rate of output audio stream, default is @code{44100}.
@item nb_samples, n
Set number of samples per each frame in output audio stream. Default is @code{1024}.
@item interp, i
Set interpolation method for FIR equalizer coefficients. Can be @code{linear} or @code{cubic}.
@item phase, h
Set phase type of FIR filter. Can be @code{linear} or @code{min}: minimum-phase.
Default is minimum-phase filter.
@end table
@section afirsrc
Generate a FIR coefficients using frequency sampling method.

1
libavfilter/Makefile
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@ -172,6 +172,7 @@ OBJS-$(CONFIG_VOLUMEDETECT_FILTER) += af_volumedetect.o
OBJS-$(CONFIG_AEVALSRC_FILTER) += aeval.o
OBJS-$(CONFIG_AFDELAYSRC_FILTER) += asrc_afdelaysrc.o
OBJS-$(CONFIG_AFIREQSRC_FILTER) += asrc_afirsrc.o
OBJS-$(CONFIG_AFIRSRC_FILTER) += asrc_afirsrc.o
OBJS-$(CONFIG_ANOISESRC_FILTER) += asrc_anoisesrc.o
OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o

1
libavfilter/allfilters.c
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@ -160,6 +160,7 @@ extern const AVFilter ff_af_volumedetect;
extern const AVFilter ff_asrc_aevalsrc;
extern const AVFilter ff_asrc_afdelaysrc;
extern const AVFilter ff_asrc_afireqsrc;
extern const AVFilter ff_asrc_afirsrc;
extern const AVFilter ff_asrc_anoisesrc;
extern const AVFilter ff_asrc_anullsrc;

295
libavfilter/asrc_afirsrc.c
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@ -18,7 +18,9 @@
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/cpu.h"
#include "libavutil/channel_layout.h"
#include "libavutil/ffmath.h"
#include "libavutil/eval.h"
#include "libavutil/opt.h"
#include "libavutil/tx.h"
@ -38,6 +40,9 @@ typedef struct AudioFIRSourceContext {
int sample_rate;
int nb_samples;
int win_func;
int preset;
int interp;
int phaset;
AVComplexFloat *complexf;
float *freq;
@ -54,8 +59,8 @@ typedef struct AudioFIRSourceContext {
float *win;
int64_t pts;
AVTXContext *tx_ctx;
av_tx_fn tx_fn;
AVTXContext *tx_ctx, *itx_ctx;
av_tx_fn tx_fn, itx_fn;
} AudioFIRSourceContext;
#define OFFSET(x) offsetof(AudioFIRSourceContext, x)
@ -104,6 +109,7 @@ static av_cold void uninit(AVFilterContext *ctx)
av_freep(&s->phase);
av_freep(&s->complexf);
av_tx_uninit(&s->tx_ctx);
av_tx_uninit(&s->itx_ctx);
}
static av_cold int query_formats(AVFilterContext *ctx)
@ -131,7 +137,7 @@ static int parse_string(char *str, float **items, int *nb_items, int *items_size
float *new_items;
char *tail;
new_items = av_fast_realloc(NULL, items_size, 1 * sizeof(float));
new_items = av_fast_realloc(NULL, items_size, sizeof(float));
if (!new_items)
return AVERROR(ENOMEM);
*items = new_items;
@ -142,7 +148,7 @@ static int parse_string(char *str, float **items, int *nb_items, int *items_size
do {
(*items)[(*nb_items)++] = av_strtod(tail, &tail);
new_items = av_fast_realloc(*items, items_size, (*nb_items + 1) * sizeof(float));
new_items = av_fast_realloc(*items, items_size, (*nb_items + 2) * sizeof(float));
if (!new_items)
return AVERROR(ENOMEM);
*items = new_items;
@ -300,3 +306,284 @@ const AVFilter ff_asrc_afirsrc = {
FILTER_QUERY_FUNC(query_formats),
.priv_class = &afirsrc_class,
};
#define DEFAULT_BANDS "25 40 63 100 160 250 400 630 1000 1600 2500 4000 6300 10000 16000 24000"
typedef struct EqPreset {
char name[16];
float gains[16];
} EqPreset;
static const EqPreset eq_presets[] = {
{ "flat", { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 } },
{ "acoustic", { 5.0, 4.5, 4.0, 3.5, 1.5, 1.0, 1.5, 1.5, 2.0, 3.0, 3.5, 4.0, 3.7, 3.0, 3.0 } },
{ "bass", { 10.0, 8.8, 8.5, 6.5, 2.5, 1.5, 0, 0, 0, 0, 0, 0, 0, 0, 0 } },
{ "beats", { -5.5, -5.0, -4.5, -4.2, -3.5, -3.0, -1.9, 0, 0, 0, 0, 0, 0, 0, 0 } },
{ "classic", { -0.3, 0.3, -3.5, -9.0, -1.0, 0.0, 1.8, 2.1, 0.0, 0.0, 0.0, 4.4, 9.0, 9.0, 9.0 } },
{ "clear", { 3.5, 5.5, 6.5, 9.5, 8.0, 6.5, 3.5, 2.5, 1.3, 5.0, 7.0, 9.0, 10.0, 11.0, 9.0 } },
{ "deep bass", { 12.0, 8.0, 0.0, -6.7, -12.0, -9.0, -3.5, -3.5, -6.1, 0.0, -3.0, -5.0, 0.0, 1.2, 3.0 } },
{ "dubstep", { 12.0, 10.0, 0.5, -1.0, -3.0, -5.0, -5.0, -4.8, -4.5, -2.5, -1.0, 0.0, -2.5, -2.5, 0.0 } },
{ "electronic", { 4.0, 4.0, 3.5, 1.0, 0.0, -0.5, -2.0, 0.0, 2.0, 0.0, 0.0, 1.0, 3.0, 4.0, 4.5 } },
{ "hardstyle", { 6.1, 7.0, 12.0, 6.1, -5.0, -12.0, -2.5, 3.0, 6.5, 0.0, -2.2, -4.5, -6.1, -9.2, -10.0 } },
{ "hip-hop", { 4.5, 4.3, 4.0, 2.5, 1.5, 3.0, -1.0, -1.5, -1.5, 1.5, 0.0, -1.0, 0.0, 1.5, 3.0 } },
{ "jazz", { 0.0, 0.0, 0.0, 2.0, 4.0, 5.9, -5.9, -4.5, -2.5, 2.5, 1.0, -0.8, -0.8, -0.8, -0.8 } },
{ "metal", { 10.5, 10.5, 7.5, 0.0, 2.0, 5.5, 0.0, 0.0, 0.0, 6.1, 0.0, 0.0, 6.1, 10.0, 12.0 } },
{ "movie", { 3.0, 3.0, 6.1, 8.5, 9.0, 7.0, 6.1, 6.1, 5.0, 8.0, 3.5, 3.5, 8.0, 10.0, 8.0 } },
{ "pop", { 0.0, 0.0, 0.0, 0.0, 0.0, 1.3, 2.0, 2.5, 5.0, -1.5, -2.0, -3.0, -3.0, -3.0, -3.0 } },
{ "r&b", { 3.0, 3.0, 7.0, 6.1, 4.5, 1.5, -1.5, -2.0, -1.5, 2.0, 2.5, 3.0, 3.5, 3.8, 4.0 } },
{ "rock", { 0.0, 0.0, 0.0, 3.0, 3.0, -10.0, -4.0, -1.0, 0.8, 3.0, 3.0, 3.0, 3.0, 3.0, 3.0 } },
{ "vocal booster", { -1.5, -2.0, -3.0, -3.0, -0.5, 1.5, 3.5, 3.5, 3.5, 3.0, 2.0, 1.5, 0.0, 0.0, -1.5 } },
};
static const AVOption afireqsrc_options[] = {
{ "preset","set equalizer preset", OFFSET(preset), AV_OPT_TYPE_INT, {.i64=0}, -1, FF_ARRAY_ELEMS(eq_presets)-1, FLAGS, "preset" },
{ "p", "set equalizer preset", OFFSET(preset), AV_OPT_TYPE_INT, {.i64=0}, -1, FF_ARRAY_ELEMS(eq_presets)-1, FLAGS, "preset" },
{ "custom", NULL, 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, FLAGS, "preset" },
{ eq_presets[ 0].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 0}, 0, 0, FLAGS, "preset" },
{ eq_presets[ 1].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 1}, 0, 0, FLAGS, "preset" },
{ eq_presets[ 2].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 2}, 0, 0, FLAGS, "preset" },
{ eq_presets[ 3].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 3}, 0, 0, FLAGS, "preset" },
{ eq_presets[ 4].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 4}, 0, 0, FLAGS, "preset" },
{ eq_presets[ 5].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 5}, 0, 0, FLAGS, "preset" },
{ eq_presets[ 6].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 6}, 0, 0, FLAGS, "preset" },
{ eq_presets[ 7].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 7}, 0, 0, FLAGS, "preset" },
{ eq_presets[ 8].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 8}, 0, 0, FLAGS, "preset" },
{ eq_presets[ 9].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64= 9}, 0, 0, FLAGS, "preset" },
{ eq_presets[10].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=10}, 0, 0, FLAGS, "preset" },
{ eq_presets[11].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=11}, 0, 0, FLAGS, "preset" },
{ eq_presets[12].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=12}, 0, 0, FLAGS, "preset" },
{ eq_presets[13].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=13}, 0, 0, FLAGS, "preset" },
{ eq_presets[14].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=14}, 0, 0, FLAGS, "preset" },
{ eq_presets[15].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=15}, 0, 0, FLAGS, "preset" },
{ eq_presets[16].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=16}, 0, 0, FLAGS, "preset" },
{ eq_presets[17].name, NULL, 0, AV_OPT_TYPE_CONST, {.i64=17}, 0, 0, FLAGS, "preset" },
{ "gains", "set gain values per band", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0"}, 0, 0, FLAGS },
{ "g", "set gain values per band", OFFSET(magnitude_str), AV_OPT_TYPE_STRING, {.str="0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0"}, 0, 0, FLAGS },
{ "bands", "set central frequency values per band", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str=DEFAULT_BANDS}, 0, 0, FLAGS },
{ "b", "set central frequency values per band", OFFSET(freq_points_str), AV_OPT_TYPE_STRING, {.str=DEFAULT_BANDS}, 0, 0, FLAGS },
{ "taps", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=4096}, 16, UINT16_MAX, FLAGS },
{ "t", "set number of taps", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=4096}, 16, UINT16_MAX, FLAGS },
{ "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
{ "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, FLAGS },
{ "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
{ "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, FLAGS },
{ "interp","set the interpolation", OFFSET(interp), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS, "interp" },
{ "i", "set the interpolation", OFFSET(interp), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS, "interp" },
{ "linear", NULL, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "interp" },
{ "cubic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "interp" },
{ "phase","set the phase", OFFSET(phaset), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS, "phase" },
{ "h", "set the phase", OFFSET(phaset), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, FLAGS, "phase" },
{ "linear", "linear phase", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "phase" },
{ "min", "minimum phase", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "phase" },
{NULL}
};
AVFILTER_DEFINE_CLASS(afireqsrc);
static void eq_interp(AVComplexFloat *complexf,
const float *freq,
const float *magnitude,
int m, int interp, int minterp,
const float factor)
{
for (int i = 0; i < minterp; i++) {
for (int j = 0; j < m; j++) {
const float x = factor * i;
if (x <= freq[j+1]) {
float g;
if (interp == 0) {
const float d = freq[j+1] - freq[j];
const float d0 = x - freq[j];
const float d1 = freq[j+1] - x;
const float g0 = magnitude[j];
const float g1 = magnitude[j+1];
if (d0 && d1) {
g = (d0 * g1 + d1 * g0) / d;
} else if (d0) {
g = g1;
} else {
g = g0;
}
} else {
if (x <= freq[j]) {
g = magnitude[j];
} else {
float x1, x2, x3;
float a, b, c, d;
float m0, m1, m2, msum;
const float unit = freq[j+1] - freq[j];
m0 = j != 0 ? unit * (magnitude[j] - magnitude[j-1]) / (freq[j] - freq[j-1]) : 0;
m1 = magnitude[j+1] - magnitude[j];
m2 = j != minterp - 1 ? unit * (magnitude[j+2] - magnitude[j+1]) / (freq[j+2] - freq[j+1]) : 0;
msum = fabsf(m0) + fabsf(m1);
m0 = msum > 0.f ? (fabsf(m0) * m1 + fabsf(m1) * m0) / msum : 0.f;
msum = fabsf(m1) + fabsf(m2);
m1 = msum > 0.f ? (fabsf(m1) * m2 + fabsf(m2) * m1) / msum : 0.f;
d = magnitude[j];
c = m0;
b = 3.f * magnitude[j+1] - m1 - 2.f * c - 3.f * d;
a = magnitude[j+1] - b - c - d;
x1 = (x - freq[j]) / unit;
x2 = x1 * x1;
x3 = x2 * x1;
g = a * x3 + b * x2 + c * x1 + d;
}
}
complexf[i].re = g;
complexf[i].im = 0;
complexf[minterp * 2 - i - 1].re = g;
complexf[minterp * 2 - i - 1].im = 0;
break;
}
}
}
}
static av_cold int config_eq_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioFIRSourceContext *s = ctx->priv;
int fft_size, middle, asize, ret;
float scale, factor;
s->nb_freq = s->nb_magnitude = 0;
if (s->preset < 0) {
ret = parse_string(s->freq_points_str, &s->freq, &s->nb_freq, &s->freq_size);
if (ret < 0)
return ret;
ret = parse_string(s->magnitude_str, &s->magnitude, &s->nb_magnitude, &s->magnitude_size);
if (ret < 0)
return ret;
} else {
char *freq_str;
s->nb_magnitude = FF_ARRAY_ELEMS(eq_presets[s->preset].gains);
freq_str = av_strdup(DEFAULT_BANDS);
if (!freq_str)
return AVERROR(ENOMEM);
ret = parse_string(freq_str, &s->freq, &s->nb_freq, &s->freq_size);
av_free(freq_str);
if (ret < 0)
return ret;
s->magnitude = av_calloc(s->nb_magnitude, sizeof(*s->magnitude));
if (!s->magnitude)
return AVERROR(ENOMEM);
memcpy(s->magnitude, eq_presets[s->preset].gains, sizeof(*s->magnitude) * s->nb_magnitude);
}
if (s->nb_freq != s->nb_magnitude || s->nb_freq < 2) {
av_log(ctx, AV_LOG_ERROR, "Number of bands and gains must be same and >= 2.\n");
return AVERROR(EINVAL);
}
s->freq[s->nb_freq] = outlink->sample_rate * 0.5f;
s->magnitude[s->nb_freq] = s->magnitude[s->nb_freq-1];
fft_size = s->nb_taps * 2;
factor = FFMIN(outlink->sample_rate * 0.5f, s->freq[s->nb_freq - 1]) / (float)fft_size;
asize = FFALIGN(fft_size, av_cpu_max_align());
s->complexf = av_calloc(asize * 2, sizeof(*s->complexf));
if (!s->complexf)
return AVERROR(ENOMEM);
scale = 1.f;
ret = av_tx_init(&s->itx_ctx, &s->itx_fn, AV_TX_FLOAT_FFT, 1, fft_size, &scale, 0);
if (ret < 0)
return ret;
s->taps = av_calloc(s->nb_taps, sizeof(*s->taps));
if (!s->taps)
return AVERROR(ENOMEM);
eq_interp(s->complexf, s->freq, s->magnitude, s->nb_freq, s->interp, s->nb_taps, factor);
for (int i = 0; i < fft_size; i++)
s->complexf[i].re = ff_exp10f(s->complexf[i].re / 20.f);
if (s->phaset) {
const float threshold = powf(10.f, -100.f / 20.f);
const float logt = logf(threshold);
scale = 1.f;
ret = av_tx_init(&s->tx_ctx, &s->tx_fn, AV_TX_FLOAT_FFT, 0, fft_size, &scale, 0);
if (ret < 0)
return ret;
for (int i = 0; i < fft_size; i++)
s->complexf[i].re = s->complexf[i].re < threshold ? logt : logf(s->complexf[i].re);
s->itx_fn(s->itx_ctx, s->complexf + asize, s->complexf, sizeof(float));
for (int i = 0; i < fft_size; i++) {
s->complexf[i + asize].re /= fft_size;
s->complexf[i + asize].im /= fft_size;
}
for (int i = 1; i < s->nb_taps; i++) {
s->complexf[asize + i].re += s->complexf[asize + fft_size - i].re;
s->complexf[asize + i].im -= s->complexf[asize + fft_size - i].im;
s->complexf[asize + fft_size - i].re = 0.f;
s->complexf[asize + fft_size - i].im = 0.f;
}
s->complexf[asize + s->nb_taps - 1].im *= -1.f;
s->tx_fn(s->tx_ctx, s->complexf, s->complexf + asize, sizeof(float));
for (int i = 0; i < fft_size; i++) {
float eR = expf(s->complexf[i].re);
s->complexf[i].re = eR * cosf(s->complexf[i].im);
s->complexf[i].im = eR * sinf(s->complexf[i].im);
}
s->itx_fn(s->itx_ctx, s->complexf + asize, s->complexf, sizeof(float));
for (int i = 0; i < s->nb_taps; i++)
s->taps[i] = s->complexf[i + asize].re / fft_size;
} else {
s->itx_fn(s->itx_ctx, s->complexf + asize, s->complexf, sizeof(float));
middle = s->nb_taps / 2;
for (int i = 0; i < middle; i++) {
s->taps[middle - i] = s->complexf[i + asize].re / fft_size;
s->taps[middle + i] = s->complexf[i + asize].re / fft_size;
}
}
s->pts = 0;
return 0;
}
static const AVFilterPad afireqsrc_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_eq_output,
},
};
const AVFilter ff_asrc_afireqsrc = {
.name = "afireqsrc",
.description = NULL_IF_CONFIG_SMALL("Generate a FIR equalizer coefficients audio stream."),
.uninit = uninit,
.activate = activate,
.priv_size = sizeof(AudioFIRSourceContext),
.inputs = NULL,
FILTER_OUTPUTS(afireqsrc_outputs),
FILTER_QUERY_FUNC(query_formats),
.priv_class = &afireqsrc_class,
};

2
libavfilter/version.h
View File

@ -31,7 +31,7 @@
#include "version_major.h"
#define LIBAVFILTER_VERSION_MINOR 5
#define LIBAVFILTER_VERSION_MINOR 6
#define LIBAVFILTER_VERSION_MICRO 100