diff --git a/Changelog b/Changelog index 8b2e17378cfa6..ebb1727875f1f 100644 --- a/Changelog +++ b/Changelog @@ -47,6 +47,7 @@ version : - DXVA2/D3D11VA hardware accelerated AV1 decoding - speechnorm filter - SpeedHQ encoder +- asupercut filter version 4.3: diff --git a/doc/filters.texi b/doc/filters.texi index 109cacc23d2e8..488d3c4654253 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -1838,7 +1838,7 @@ Set central frequency for band. If input doesn't have that frequency the entry is ignored. @item w -Set band width in hertz. +Set band width in Hertz. @item g Set band gain in dB. @@ -1903,7 +1903,7 @@ Syntax for the commands is : "@var{fN}|f=@var{freq}|w=@var{width}|g=@var{gain}" @var{fN} is existing filter number, starting from 0, if no such filter is available error is returned. @var{freq} set new frequency parameter. -@var{width} set new width parameter in herz. +@var{width} set new width parameter in Hertz. @var{gain} set new gain parameter in dB. Full filter invocation with asendcmd may look like this: @@ -2584,7 +2584,7 @@ Set delay line feedback gain value. Allowed range is from 0 to 1. Default value is 0.5. @item cutoff -Set cutoff frequency in herz. Allowed range is 50 to 900. +Set cutoff frequency in Hertz. Allowed range is 50 to 900. Default value is 100. @item slope @@ -2600,6 +2600,21 @@ Default value is 20. This filter supports the all above options as @ref{commands}. +@section asupercut +Cut super frequencies. + +The filter accepts the following options: + +@table @option +@item cutoff +Set cutoff frequency in Hertz. Allowed range is 20000 to 192000. +Default value is 20000. +@end table + +@subsection Commands + +This filter supports the all above options as @ref{commands}. + @section atempo Adjust audio tempo. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 794a55ac3dabc..cff94029899fb 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -90,6 +90,7 @@ OBJS-$(CONFIG_ASR_FILTER) += af_asr.o OBJS-$(CONFIG_ASTATS_FILTER) += af_astats.o OBJS-$(CONFIG_ASTREAMSELECT_FILTER) += f_streamselect.o framesync.o OBJS-$(CONFIG_ASUBBOOST_FILTER) += af_asubboost.o +OBJS-$(CONFIG_ASUPERCUT_FILTER) += af_asupercut.o OBJS-$(CONFIG_ATEMPO_FILTER) += af_atempo.o OBJS-$(CONFIG_ATRIM_FILTER) += trim.o OBJS-$(CONFIG_AXCORRELATE_FILTER) += af_axcorrelate.o diff --git a/libavfilter/af_asupercut.c b/libavfilter/af_asupercut.c new file mode 100644 index 0000000000000..a22830c2e80dc --- /dev/null +++ b/libavfilter/af_asupercut.c @@ -0,0 +1,247 @@ +/* + * 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/opt.h" +#include "avfilter.h" +#include "audio.h" +#include "formats.h" + +typedef struct BiquadCoeffs { + double a1, a2; + double b0, b1, b2; +} BiquadCoeffs; + +typedef struct ASuperCutContext { + const AVClass *class; + + double cutoff; + + int bypass; + + BiquadCoeffs coeffs[5]; + + AVFrame *w; +} ASuperCutContext; + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats = NULL; + AVFilterChannelLayouts *layouts = NULL; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_DBLP, + AV_SAMPLE_FMT_NONE + }; + int ret; + + formats = ff_make_format_list(sample_fmts); + if (!formats) + return AVERROR(ENOMEM); + ret = ff_set_common_formats(ctx, formats); + if (ret < 0) + return 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_all_samplerates(); + return ff_set_common_samplerates(ctx, formats); +} + +static int get_coeffs(AVFilterContext *ctx) +{ + ASuperCutContext *s = ctx->priv; + AVFilterLink *inlink = ctx->inputs[0]; + double w0 = s->cutoff / inlink->sample_rate; + double K = tan(M_PI * w0); + double q[5]; + + s->bypass = w0 >= 0.5; + if (s->bypass) + return 0; + + q[0] = 0.50623256; + q[1] = 0.56116312; + q[2] = 0.70710678; + q[3] = 1.10134463; + q[4] = 3.19622661; + + for (int b = 0; b < 5; b++) { + BiquadCoeffs *coeffs = &s->coeffs[b]; + double norm = 1.0 / (1.0 + K / q[b] + K * K); + + coeffs->b0 = K * K * norm; + coeffs->b1 = 2.0 * coeffs->b0; + coeffs->b2 = coeffs->b0; + coeffs->a1 = -2.0 * (K * K - 1.0) * norm; + coeffs->a2 = -(1.0 - K / q[b] + K * K) * norm; + } + + return 0; +} + +static int config_input(AVFilterLink *inlink) +{ + AVFilterContext *ctx = inlink->dst; + ASuperCutContext *s = ctx->priv; + + s->w = ff_get_audio_buffer(inlink, 2 * 5); + if (!s->w) + return AVERROR(ENOMEM); + + return get_coeffs(ctx); +} + +typedef struct ThreadData { + AVFrame *in, *out; +} ThreadData; + +static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) +{ + ASuperCutContext *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++) { + const double *src = (const double *)in->extended_data[ch]; + double *dst = (double *)out->extended_data[ch]; + + for (int b = 0; b < 5; b++) { + BiquadCoeffs *coeffs = &s->coeffs[b]; + const double a1 = coeffs->a1; + const double a2 = coeffs->a2; + const double b0 = coeffs->b0; + const double b1 = coeffs->b1; + const double b2 = coeffs->b2; + double *w = ((double *)s->w->extended_data[ch]) + b * 2; + + for (int n = 0; n < in->nb_samples; n++) { + double sin = b ? dst[n] : src[n]; + double sout = sin * b0 + w[0]; + + w[0] = b1 * sin + w[1] + a1 * sout; + w[1] = b2 * sin + a2 * sout; + + dst[n] = sout; + } + } + } + + return 0; +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + AVFilterContext *ctx = inlink->dst; + ASuperCutContext *s = ctx->priv; + AVFilterLink *outlink = ctx->outputs[0]; + ThreadData td; + AVFrame *out; + + if (s->bypass) + return ff_filter_frame(outlink, in); + + 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; + ctx->internal->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 int process_command(AVFilterContext *ctx, const char *cmd, const char *args, + char *res, int res_len, int flags) +{ + int ret; + + ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags); + if (ret < 0) + return ret; + + return get_coeffs(ctx); +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + ASuperCutContext *s = ctx->priv; + + av_frame_free(&s->w); +} + +#define OFFSET(x) offsetof(ASuperCutContext, x) +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM + +static const AVOption asupercut_options[] = { + { "cutoff", "set cutoff frequency", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, {.dbl=20000}, 20000, 192000, FLAGS }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(asupercut); + +static const AVFilterPad inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + .config_props = config_input, + }, + { NULL } +}; + +static const AVFilterPad outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + }, + { NULL } +}; + +AVFilter ff_af_asupercut = { + .name = "asupercut", + .description = NULL_IF_CONFIG_SMALL("Cut super frequencies."), + .query_formats = query_formats, + .priv_size = sizeof(ASuperCutContext), + .priv_class = &asupercut_class, + .uninit = uninit, + .inputs = inputs, + .outputs = outputs, + .process_command = process_command, + .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC | + AVFILTER_FLAG_SLICE_THREADS, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index fbfd8989c6f4e..83f434bc2709d 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -83,6 +83,7 @@ extern AVFilter ff_af_asr; extern AVFilter ff_af_astats; extern AVFilter ff_af_astreamselect; extern AVFilter ff_af_asubboost; +extern AVFilter ff_af_asupercut; extern AVFilter ff_af_atempo; extern AVFilter ff_af_atrim; extern AVFilter ff_af_axcorrelate; diff --git a/libavfilter/version.h b/libavfilter/version.h index 23c9d374ad854..bd20eaee73935 100644 --- a/libavfilter/version.h +++ b/libavfilter/version.h @@ -30,7 +30,7 @@ #include "libavutil/version.h" #define LIBAVFILTER_VERSION_MAJOR 7 -#define LIBAVFILTER_VERSION_MINOR 90 +#define LIBAVFILTER_VERSION_MINOR 91 #define LIBAVFILTER_VERSION_MICRO 100