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author | Paul B Mahol <onemda@gmail.com> | 2021-08-23 19:52:32 +0200 |
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committer | Paul B Mahol <onemda@gmail.com> | 2021-08-28 18:46:39 +0200 |
commit | b53a7d2d4d8d45d94d23ff9e31873b82025b1e15 (patch) | |
tree | 433a4badc07ecfc078eee03410f6ce30383bd59c | |
parent | 0871273a2ff8e7ec7d44d9b4e24e2d78346eb2f0 (diff) | |
download | ffmpeg-b53a7d2d4d8d45d94d23ff9e31873b82025b1e15.tar.gz |
avfilter: add adecorrelate filter
-rw-r--r-- | Changelog | 1 | ||||
-rw-r--r-- | doc/filters.texi | 14 | ||||
-rw-r--r-- | libavfilter/Makefile | 1 | ||||
-rw-r--r-- | libavfilter/af_adecorrelate.c | 268 | ||||
-rw-r--r-- | libavfilter/allfilters.c | 1 | ||||
-rw-r--r-- | libavfilter/version.h | 2 |
6 files changed, 286 insertions, 1 deletions
@@ -12,6 +12,7 @@ version <next>: - audio and video segment filters - Apple Graphics (SMC) encoder - hsvkey and hsvhold video filters +- adecorrelate audio filter version 4.4: diff --git a/doc/filters.texi b/doc/filters.texi index 44a9a47d1b..c85f82616c 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -745,6 +745,20 @@ Select overlap-save method. Not interpolated samples remain unchanged. Default value is @code{a}. @end table +@section adecorrelate +Apply decorrelation to input audio stream. + +The filter accepts the following options: + +@table @option +@item stages +Set decorrelation stages of filtering. Allowed +range is from 1 to 16. Default value is 6. + +@item seed +Set random seed used for setting delay in samples across channels. +@end table + @section adelay Delay one or more audio channels. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 3166e70b58..399a4a5083 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -40,6 +40,7 @@ OBJS-$(CONFIG_ACRUSHER_FILTER) += af_acrusher.o OBJS-$(CONFIG_ACUE_FILTER) += f_cue.o OBJS-$(CONFIG_ADECLICK_FILTER) += af_adeclick.o OBJS-$(CONFIG_ADECLIP_FILTER) += af_adeclick.o +OBJS-$(CONFIG_ADECORRELATE_FILTER) += af_adecorrelate.o OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o OBJS-$(CONFIG_ADENORM_FILTER) += af_adenorm.o OBJS-$(CONFIG_ADERIVATIVE_FILTER) += af_aderivative.o diff --git a/libavfilter/af_adecorrelate.c b/libavfilter/af_adecorrelate.c new file mode 100644 index 0000000000..6113574125 --- /dev/null +++ b/libavfilter/af_adecorrelate.c @@ -0,0 +1,268 @@ +/* + * Copyright (c) 2013-2020 Michael Barbour <barbour.michael.0@gmail.com> + * Copyright (c) 2021 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/channel_layout.h" +#include "libavutil/ffmath.h" +#include "libavutil/lfg.h" +#include "libavutil/random_seed.h" +#include "libavutil/opt.h" +#include "avfilter.h" +#include "audio.h" +#include "formats.h" + +#define MAX_STAGES 16 +#define FILTER_FC 1100.0 +#define RT60_LF 0.1 +#define RT60_HF 0.008 + +typedef struct APContext { + int len, p; + double *mx, *my; + double b0, b1, a0, a1; +} APContext; + +typedef struct ADecorrelateContext { + const AVClass *class; + + int stages; + int64_t seed; + + int nb_channels; + APContext (*ap)[MAX_STAGES]; + + AVLFG c; + + void (*filter_channel)(AVFilterContext *ctx, + int channel, + AVFrame *in, AVFrame *out); +} ADecorrelateContext; + +static int query_formats(AVFilterContext *ctx) +{ + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_DBLP, + AV_SAMPLE_FMT_NONE + }; + int ret = ff_set_common_formats_from_list(ctx, sample_fmts); + if (ret < 0) + return ret; + + ret = ff_set_common_all_channel_counts(ctx); + if (ret < 0) + return ret; + + return ff_set_common_all_samplerates(ctx); +} + +static int ap_init(APContext *ap, int fs, double delay) +{ + const int delay_samples = lrint(round(delay * fs)); + const double gain_lf = -60.0 / (RT60_LF * fs) * delay_samples; + const double gain_hf = -60.0 / (RT60_HF * fs) * delay_samples; + const double w0 = 2.0 * M_PI * FILTER_FC / fs; + const double t = tan(w0 / 2.0); + const double g_hf = ff_exp10(gain_hf / 20.0); + const double gd = ff_exp10((gain_lf-gain_hf) / 20.0); + const double sgd = sqrt(gd); + + ap->len = delay_samples + 1; + ap->p = 0; + ap->mx = av_calloc(ap->len, sizeof(*ap->mx)); + ap->my = av_calloc(ap->len, sizeof(*ap->my)); + if (!ap->mx || !ap->my) + return AVERROR(ENOMEM); + + ap->a0 = t + sgd; + ap->a1 = (t - sgd) / ap->a0; + ap->b0 = (gd*t - sgd) / ap->a0 * g_hf; + ap->b1 = (gd*t + sgd) / ap->a0 * g_hf; + ap->a0 = 1.0; + + return 0; +} + +static void ap_free(APContext *ap) +{ + av_freep(&ap->mx); + av_freep(&ap->my); +} + +static double ap_run(APContext *ap, double x) +{ + const int i0 = ((ap->p < 1) ? ap->len : ap->p)-1, i_n1 = ap->p, i_n2 = (ap->p+1 >= ap->len) ? 0 : ap->p+1; + const double r = ap->b1*x + ap->b0*ap->mx[i0] + ap->a1*ap->mx[i_n2] + ap->a0*ap->mx[i_n1] - + ap->a1*ap->my[i0] - ap->b0*ap->my[i_n2] - ap->b1*ap->my[i_n1]; + + ap->mx[ap->p] = x; + ap->my[ap->p] = r; + ap->p = (ap->p+1 >= ap->len) ? 0 : ap->p+1; + + return r; +} + +static void filter_channel_dbl(AVFilterContext *ctx, int ch, + AVFrame *in, AVFrame *out) +{ + ADecorrelateContext *s = ctx->priv; + const double *src = (const double *)in->extended_data[ch]; + double *dst = (double *)out->extended_data[ch]; + const int nb_samples = in->nb_samples; + const int stages = s->stages; + APContext *ap0 = &s->ap[ch][0]; + + for (int n = 0; n < nb_samples; n++) { + dst[n] = ap_run(ap0, src[n]); + for (int i = 1; i < stages; i++) { + APContext *ap = &s->ap[ch][i]; + + dst[n] = ap_run(ap, dst[n]); + } + } +} + +static int config_input(AVFilterLink *inlink) +{ + AVFilterContext *ctx = inlink->dst; + ADecorrelateContext *s = ctx->priv; + int ret; + + if (s->seed == -1) + s->seed = av_get_random_seed(); + av_lfg_init(&s->c, s->seed); + + s->nb_channels = inlink->channels; + s->ap = av_calloc(inlink->channels, sizeof(*s->ap)); + if (!s->ap) + return AVERROR(ENOMEM); + + for (int i = 0; i < inlink->channels; i++) { + for (int j = 0; j < s->stages; j++) { + ret = ap_init(&s->ap[i][j], inlink->sample_rate, + (double)av_lfg_get(&s->c) / 0xffffffff * 2.2917e-3 + 0.83333e-3); + if (ret < 0) + return ret; + } + } + + s->filter_channel = filter_channel_dbl; + + return 0; +} + +typedef struct ThreadData { + AVFrame *in, *out; +} ThreadData; + +static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) +{ + ADecorrelateContext *s = ctx->priv; + ThreadData *td = arg; + AVFrame *out = td->out; + AVFrame *in = td->in; + const int start = (in->channels * jobnr) / nb_jobs; + const int end = (in->channels * (jobnr+1)) / nb_jobs; + + for (int ch = start; ch < end; ch++) + s->filter_channel(ctx, ch, in, out); + + return 0; +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + AVFilterContext *ctx = inlink->dst; + AVFilterLink *outlink = ctx->outputs[0]; + AVFrame *out; + ThreadData td; + + if (av_frame_is_writable(in)) { + out = in; + } else { + out = ff_get_audio_buffer(outlink, in->nb_samples); + if (!out) { + av_frame_free(&in); + return AVERROR(ENOMEM); + } + av_frame_copy_props(out, in); + } + + td.in = in; td.out = out; + ff_filter_execute(ctx, filter_channels, &td, NULL, + FFMIN(inlink->channels, ff_filter_get_nb_threads(ctx))); + + if (out != in) + av_frame_free(&in); + return ff_filter_frame(outlink, out); +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + ADecorrelateContext *s = ctx->priv; + + if (s->ap) { + for (int ch = 0; ch < s->nb_channels; ch++) { + for (int stage = 0; stage < s->stages; stage++) + ap_free(&s->ap[ch][stage]); + } + } + + av_freep(&s->ap); +} + +#define OFFSET(x) offsetof(ADecorrelateContext, x) +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption adecorrelate_options[] = { + { "stages", "set filtering stages", OFFSET(stages), AV_OPT_TYPE_INT, {.i64=6}, 1, MAX_STAGES, FLAGS }, + { "seed", "set random seed", OFFSET(seed), AV_OPT_TYPE_INT64, {.i64=-1}, -1, UINT_MAX, FLAGS }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(adecorrelate); + +static const AVFilterPad inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + .config_props = config_input, + }, +}; + +static const AVFilterPad outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + }, +}; + +const AVFilter ff_af_adecorrelate = { + .name = "adecorrelate", + .description = NULL_IF_CONFIG_SMALL("Apply decorrelation to input audio."), + .query_formats = query_formats, + .priv_size = sizeof(ADecorrelateContext), + .priv_class = &adecorrelate_class, + .uninit = uninit, + FILTER_INPUTS(inputs), + FILTER_OUTPUTS(outputs), + .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC | + AVFILTER_FLAG_SLICE_THREADS, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index c49c831203..745fc69e66 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -33,6 +33,7 @@ extern const AVFilter ff_af_acrossover; extern const AVFilter ff_af_acrusher; extern const AVFilter ff_af_adeclick; extern const AVFilter ff_af_adeclip; +extern const AVFilter ff_af_adecorrelate; extern const AVFilter ff_af_adelay; extern const AVFilter ff_af_adenorm; extern const AVFilter ff_af_aderivative; diff --git a/libavfilter/version.h b/libavfilter/version.h index e9a76c5ac3..e3a86d9b01 100644 --- a/libavfilter/version.h +++ b/libavfilter/version.h @@ -30,7 +30,7 @@ #include "libavutil/version.h" #define LIBAVFILTER_VERSION_MAJOR 8 -#define LIBAVFILTER_VERSION_MINOR 4 +#define LIBAVFILTER_VERSION_MINOR 5 #define LIBAVFILTER_VERSION_MICRO 100 |