diff --git a/Changelog b/Changelog index cd91f63cb3..1f5ca70655 100644 --- a/Changelog +++ b/Changelog @@ -20,6 +20,7 @@ version : - sofalizer filter switched to libmysofa - Gremlin Digital Video demuxer and decoder - headphone audio filter +- superequalizer audio filter version 3.3: - CrystalHD decoder moved to new decode API diff --git a/doc/filters.texi b/doc/filters.texi index 41b4b8249c..53e057c774 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -3835,6 +3835,49 @@ channels. Default is 0.3. Set level of input signal of original channel. Default is 0.8. @end table +@section superequalizer +Apply 18th band equalizer. + +The filter accpets the following options: +@table @option +@item 1b +Set 65Hz band gain. +@item 2b +Set 92Hz band gain. +@item 3b +Set 131Hz band gain. +@item 4b +Set 185Hz band gain. +@item 5b +Set 262Hz band gain. +@item 6b +Set 370Hz band gain. +@item 7b +Set 523Hz band gain. +@item 8b +Set 740Hz band gain. +@item 9b +Set 1047Hz band gain. +@item 10b +Set 1480Hz band gain. +@item 11b +Set 2093Hz band gain. +@item 12b +Set 2960Hz band gain. +@item 13b +Set 4186Hz band gain. +@item 14b +Set 5920Hz band gain. +@item 15b +Set 8372Hz band gain. +@item 16b +Set 11840Hz band gain. +@item 17b +Set 16744Hz band gain. +@item 18b +Set 20000Hz band gain. +@end table + @section surround Apply audio surround upmix filter. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 04ec9b8b8f..52c44d266f 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -109,6 +109,7 @@ OBJS-$(CONFIG_SILENCEREMOVE_FILTER) += af_silenceremove.o OBJS-$(CONFIG_SOFALIZER_FILTER) += af_sofalizer.o OBJS-$(CONFIG_STEREOTOOLS_FILTER) += af_stereotools.o OBJS-$(CONFIG_STEREOWIDEN_FILTER) += af_stereowiden.o +OBJS-$(CONFIG_SUPEREQUALIZER_FILTER) += af_superequalizer.o OBJS-$(CONFIG_SURROUND_FILTER) += af_surround.o OBJS-$(CONFIG_TREBLE_FILTER) += af_biquads.o OBJS-$(CONFIG_TREMOLO_FILTER) += af_tremolo.o diff --git a/libavfilter/af_superequalizer.c b/libavfilter/af_superequalizer.c new file mode 100644 index 0000000000..4c9f215f4c --- /dev/null +++ b/libavfilter/af_superequalizer.c @@ -0,0 +1,368 @@ +/* + * Copyright (c) 2002 Naoki Shibata + * Copyright (c) 2017 Paul B Mahol + * + * 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 + */ + +#include "libavutil/opt.h" + +#include "libavcodec/avfft.h" + +#include "audio.h" +#include "avfilter.h" +#include "internal.h" + +#define NBANDS 17 +#define M 15 + +typedef struct EqParameter { + float lower, upper, gain; +} EqParameter; + +typedef struct SuperEqualizerContext { + const AVClass *class; + + EqParameter params[NBANDS + 1]; + + float gains[NBANDS + 1]; + + float fact[M + 1]; + float aa; + float iza; + float *ires, *irest; + float *fsamples; + int winlen, tabsize; + + AVFrame *in, *out; + RDFTContext *rdft, *irdft; +} SuperEqualizerContext; + +static const float bands[] = { + 65.406392, 92.498606, 130.81278, 184.99721, 261.62557, 369.99442, 523.25113, 739.9884, 1046.5023, + 1479.9768, 2093.0045, 2959.9536, 4186.0091, 5919.9072, 8372.0181, 11839.814, 16744.036 +}; + +static float izero(SuperEqualizerContext *s, float x) +{ + float ret = 1; + int m; + + for (m = 1; m <= M; m++) { + float t; + + t = pow(x / 2, m) / s->fact[m]; + ret += t*t; + } + + return ret; +} + +static float hn_lpf(int n, float f, float fs) +{ + float t = 1 / fs; + float omega = 2 * M_PI * f; + + if (n * omega * t == 0) + return 2 * f * t; + return 2 * f * t * sinf(n * omega * t) / (n * omega * t); +} + +static float hn_imp(int n) +{ + return n == 0 ? 1.f : 0.f; +} + +static float hn(int n, EqParameter *param, float fs) +{ + float ret, lhn; + int i; + + lhn = hn_lpf(n, param[0].upper, fs); + ret = param[0].gain*lhn; + + for (i = 1; i < NBANDS + 1 && param[i].upper < fs / 2; i++) { + float lhn2 = hn_lpf(n, param[i].upper, fs); + ret += param[i].gain * (lhn2 - lhn); + lhn = lhn2; + } + + ret += param[i].gain * (hn_imp(n) - lhn); + + return ret; +} + +static float alpha(float a) +{ + if (a <= 21) + return 0; + if (a <= 50) + return .5842f * pow(a - 21, 0.4f) + 0.07886f * (a - 21); + return .1102f * (a - 8.7f); +} + +static float win(SuperEqualizerContext *s, float n, int N) +{ + return izero(s, alpha(s->aa) * sqrtf(1 - 4 * n * n / ((N - 1) * (N - 1)))) / s->iza; +} + +static void process_param(float *bc, EqParameter *param, float fs) +{ + int i; + + for (i = 0; i <= NBANDS; i++) { + param[i].lower = i == 0 ? 0 : bands[i - 1]; + param[i].upper = i == NBANDS - 1 ? fs : bands[i]; + param[i].gain = bc[i]; + } +} + +static int equ_init(SuperEqualizerContext *s, int wb) +{ + int i,j; + + s->rdft = av_rdft_init(wb, DFT_R2C); + s->irdft = av_rdft_init(wb, IDFT_C2R); + if (!s->rdft || !s->irdft) + return AVERROR(ENOMEM); + + s->aa = 96; + s->winlen = (1 << (wb-1))-1; + s->tabsize = 1 << wb; + + s->ires = av_calloc(s->tabsize, sizeof(float)); + s->irest = av_calloc(s->tabsize, sizeof(float)); + s->fsamples = av_calloc(s->tabsize, sizeof(float)); + + for (i = 0; i <= M; i++) { + s->fact[i] = 1; + for (j = 1; j <= i; j++) + s->fact[i] *= j; + } + + s->iza = izero(s, alpha(s->aa)); + + return 0; +} + +static void make_fir(SuperEqualizerContext *s, float *lbc, float *rbc, EqParameter *param, float fs) +{ + const int winlen = s->winlen; + const int tabsize = s->tabsize; + float *nires; + int i; + + if (fs <= 0) + return; + + process_param(lbc, param, fs); + for (i = 0; i < winlen; i++) + s->irest[i] = hn(i - winlen / 2, param, fs) * win(s, i - winlen / 2, winlen); + for (; i < tabsize; i++) + s->irest[i] = 0; + + av_rdft_calc(s->rdft, s->irest); + nires = s->ires; + for (i = 0; i < tabsize; i++) + nires[i] = s->irest[i]; +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + AVFilterContext *ctx = inlink->dst; + SuperEqualizerContext *s = ctx->priv; + AVFilterLink *outlink = ctx->outputs[0]; + const float *ires = s->ires; + float *fsamples = s->fsamples; + int ch, i; + + AVFrame *out = ff_get_audio_buffer(outlink, s->winlen); + float *src, *dst, *ptr; + + if (!out) { + av_frame_free(&in); + return AVERROR(ENOMEM); + } + + for (ch = 0; ch < in->channels; ch++) { + ptr = (float *)out->extended_data[ch]; + dst = (float *)s->out->extended_data[ch]; + src = (float *)in->extended_data[ch]; + + for (i = 0; i < s->winlen; i++) + fsamples[i] = src[i]; + for (; i < s->tabsize; i++) + fsamples[i] = 0; + + av_rdft_calc(s->rdft, fsamples); + + fsamples[0] = ires[0] * fsamples[0]; + fsamples[1] = ires[1] * fsamples[1]; + for (i = 1; i < s->tabsize / 2; i++) { + float re, im; + + re = ires[i*2 ] * fsamples[i*2] - ires[i*2+1] * fsamples[i*2+1]; + im = ires[i*2+1] * fsamples[i*2] + ires[i*2 ] * fsamples[i*2+1]; + + fsamples[i*2 ] = re; + fsamples[i*2+1] = im; + } + + av_rdft_calc(s->irdft, fsamples); + + for (i = 0; i < s->winlen; i++) + dst[i] += fsamples[i] / s->tabsize * 2; + for (i = s->winlen; i < s->tabsize; i++) + dst[i] = fsamples[i] / s->tabsize * 2; + for (i = 0; i < s->winlen; i++) + ptr[i] = dst[i]; + for (i = 0; i < s->winlen; i++) + dst[i] = dst[i+s->winlen]; + } + + out->pts = in->pts; + av_frame_free(&in); + + return ff_filter_frame(outlink, out); +} + +static av_cold int init(AVFilterContext *ctx) +{ + SuperEqualizerContext *s = ctx->priv; + + return equ_init(s, 14); +} + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats; + AVFilterChannelLayouts *layouts; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_FLTP, + 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 ((ret = ff_set_common_formats(ctx, formats)) < 0) + return ret; + + formats = ff_all_samplerates(); + return ff_set_common_samplerates(ctx, formats); +} + +static int config_input(AVFilterLink *inlink) +{ + AVFilterContext *ctx = inlink->dst; + SuperEqualizerContext *s = ctx->priv; + + inlink->partial_buf_size = + inlink->min_samples = + inlink->max_samples = s->winlen; + + s->out = ff_get_audio_buffer(inlink, s->tabsize); + if (!s->out) + return AVERROR(ENOMEM); + + return 0; +} + +static int config_output(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + SuperEqualizerContext *s = ctx->priv; + + make_fir(s, s->gains, s->gains, s->params, outlink->sample_rate); + + return 0; +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + SuperEqualizerContext *s = ctx->priv; + + av_freep(&s->irest); + av_freep(&s->ires); + av_freep(&s->fsamples); + av_rdft_end(s->rdft); + av_rdft_end(s->irdft); +} + +static const AVFilterPad superequalizer_inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + .config_props = config_input, + }, + { NULL } +}; + +static const AVFilterPad superequalizer_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_output, + }, + { NULL } +}; + +#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM +#define OFFSET(x) offsetof(SuperEqualizerContext, x) + +static const AVOption superequalizer_options[] = { + { "1b", "set 65Hz band gain", OFFSET(gains [0]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF }, + { "2b", "set 92Hz band gain", OFFSET(gains [1]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF }, + { "3b", "set 131Hz band gain", OFFSET(gains [2]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF }, + { "4b", "set 185Hz band gain", OFFSET(gains [3]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF }, + { "5b", "set 262Hz band gain", OFFSET(gains [4]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF }, + { "6b", "set 370Hz band gain", OFFSET(gains [5]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF }, + { "7b", "set 523Hz band gain", OFFSET(gains [6]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF }, + { "8b", "set 740Hz band gain", OFFSET(gains [7]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF }, + { "9b", "set 1047Hz band gain", OFFSET(gains [8]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF }, + { "10b", "set 1480Hz band gain", OFFSET(gains [9]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF }, + { "11b", "set 2093Hz band gain", OFFSET(gains[10]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF }, + { "12b", "set 2960Hz band gain", OFFSET(gains[11]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF }, + { "13b", "set 4186Hz band gain", OFFSET(gains[12]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF }, + { "14b", "set 5920Hz band gain", OFFSET(gains[13]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF }, + { "15b", "set 8372Hz band gain", OFFSET(gains[14]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF }, + { "16b", "set 11840Hz band gain", OFFSET(gains[15]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF }, + { "17b", "set 16744Hz band gain", OFFSET(gains[16]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF }, + { "18b", "set 20000Hz band gain", OFFSET(gains[17]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(superequalizer); + +AVFilter ff_af_superequalizer = { + .name = "superequalizer", + .description = NULL_IF_CONFIG_SMALL("Apply 18-th band equalization filter."), + .priv_size = sizeof(SuperEqualizerContext), + .priv_class = &superequalizer_class, + .query_formats = query_formats, + .init = init, + .uninit = uninit, + .inputs = superequalizer_inputs, + .outputs = superequalizer_outputs, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 94f7cf31a6..bd81091000 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -122,6 +122,7 @@ static void register_all(void) REGISTER_FILTER(SOFALIZER, sofalizer, af); REGISTER_FILTER(STEREOTOOLS, stereotools, af); REGISTER_FILTER(STEREOWIDEN, stereowiden, af); + REGISTER_FILTER(SUPEREQUALIZER, superequalizer, af); REGISTER_FILTER(SURROUND, surround, af); REGISTER_FILTER(TREBLE, treble, af); REGISTER_FILTER(TREMOLO, tremolo, af); diff --git a/libavfilter/version.h b/libavfilter/version.h index 1fa3cf7535..c37a34242f 100644 --- a/libavfilter/version.h +++ b/libavfilter/version.h @@ -30,7 +30,7 @@ #include "libavutil/version.h" #define LIBAVFILTER_VERSION_MAJOR 6 -#define LIBAVFILTER_VERSION_MINOR 92 +#define LIBAVFILTER_VERSION_MINOR 93 #define LIBAVFILTER_VERSION_MICRO 100 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \