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-rw-r--r--Changelog1
-rw-r--r--doc/filters.texi65
-rw-r--r--libavfilter/Makefile1
-rw-r--r--libavfilter/allfilters.c1
-rw-r--r--libavfilter/asrc_afirsrc.c295
-rw-r--r--libavfilter/version.h2
6 files changed, 360 insertions, 5 deletions
diff --git a/Changelog b/Changelog
index 8268e42cbc..b6f6682904 100644
--- a/Changelog
+++ b/Changelog
@@ -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
diff --git a/doc/filters.texi b/doc/filters.texi
index 5dde79919a..5022f96e46 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -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.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 71e198bbf9..3347e283d9 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -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
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index d7db46c2af..fd353ff2cc 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -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;
diff --git a/libavfilter/asrc_afirsrc.c b/libavfilter/asrc_afirsrc.c
index d2ea92c41c..2f09579c9c 100644
--- a/libavfilter/asrc_afirsrc.c
+++ b/libavfilter/asrc_afirsrc.c
@@ -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,
+};
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 523a7fe0a6..f191d98883 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -31,7 +31,7 @@
#include "version_major.h"
-#define LIBAVFILTER_VERSION_MINOR 5
+#define LIBAVFILTER_VERSION_MINOR 6
#define LIBAVFILTER_VERSION_MICRO 100
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