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[79.124.17.100]) by mx.google.com with ESMTP id z11si3922711ejo.318.2021.08.26.14.27.01; Thu, 26 Aug 2021 14:27:02 -0700 (PDT) Received-SPF: pass (google.com: domain of ffmpeg-devel-bounces@ffmpeg.org designates 79.124.17.100 as permitted sender) client-ip=79.124.17.100; Authentication-Results: mx.google.com; dkim=neutral (body hash did not verify) header.i=@gmail.com header.s=20161025 header.b=PwOlr8HX; spf=pass (google.com: domain of ffmpeg-devel-bounces@ffmpeg.org designates 79.124.17.100 as permitted sender) smtp.mailfrom=ffmpeg-devel-bounces@ffmpeg.org; dmarc=fail (p=NONE sp=QUARANTINE dis=NONE) header.from=gmail.com Received: from [127.0.1.1] (localhost [127.0.0.1]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTP id 5778F689EC9; Fri, 27 Aug 2021 00:26:56 +0300 (EEST) X-Original-To: ffmpeg-devel@ffmpeg.org Delivered-To: ffmpeg-devel@ffmpeg.org Received: from mail-ej1-f42.google.com (mail-ej1-f42.google.com [209.85.218.42]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTPS id 9A118680749 for ; Fri, 27 Aug 2021 00:26:49 +0300 (EEST) Received: by mail-ej1-f42.google.com with SMTP id h9so9293266ejs.4 for ; Thu, 26 Aug 2021 14:26:49 -0700 (PDT) DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=gmail.com; s=20161025; h=from:to:subject:date:message-id; bh=luQeYSwEhevjhmo2nZWPZuBCy5lzWonmCOp82J65bQw=; b=PwOlr8HXGeeQBz1stl49Icx5lIhhJ5elzBgvRiVjYt2M33CZ52X63Ry7Y7pzxUnslR tXkHoBXLid2olXu8O6J+dFsYTUWGobTgXM3nwOW4r4kR0o7HvoPvcnmp6PBwvMmvWdKy oYUb27cS+Pf9l4ml1eMOf3W9HKi2mJTo/TrSWq4K927C4xuL7uO5br8CeuT/QGcOQa7e lOHoe33enVWDaGocCY+oqmzGI3afW7tsW5c0w3ZfPteTar/+UtxPZ8rYj3OHpqw1+kAL spIlnVj2yBq8iZzJRYqZQSEadp0s6f//zzsR436TEwH426MI+x83+LoQoDKSn7i1+Nnb PtvQ== X-Google-DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/relaxed; d=1e100.net; s=20161025; h=x-gm-message-state:from:to:subject:date:message-id; bh=luQeYSwEhevjhmo2nZWPZuBCy5lzWonmCOp82J65bQw=; b=jlHlnsjm57PPMo1A43iTd/MgM8aRsLSO1bmvQXXKmcs1KydWxkoNykOSbDUaEX5Qnk znXXjAUXcDdyIDD7De9/wh564B2cII1hAibDi3o69NXx1RglrOR/y/jytPl2LL0aqv++ paatGAGu5HYen6ntiGFrXdmIppE7AeGMA/5p4hS/fNzzh9Y3B3FQyXVwKhTI0U5B74UW ge8dQv8YmL6o2R1PH+aSyOcQsTogeEEV1LqbQWfDVzosyl9rc1Er9p6TDeFYuVVfuWtB FUC7oOu1cvqW1uNkY+lQgMSKI9NNjNl52wVqqg998Ty24e7Id0TBtWRpGYzOrhOmkMfB 69rQ== X-Gm-Message-State: AOAM532cHj0vZu5FxgXJYONsvjwGQ54oQ6Lg5GDKYUg4H56TGFfE6NHe bo8syFBbEaPPZ3jH0FlxfvQabaCoWiE= X-Received: by 2002:a17:906:9b87:: with SMTP id dd7mr6066858ejc.99.1630013208990; Thu, 26 Aug 2021 14:26:48 -0700 (PDT) Received: from localhost.localdomain ([95.168.120.12]) by smtp.gmail.com with ESMTPSA id o12sm2397981edv.19.2021.08.26.14.26.47 for (version=TLS1_3 cipher=TLS_AES_256_GCM_SHA384 bits=256/256); Thu, 26 Aug 2021 14:26:48 -0700 (PDT) From: Paul B Mahol To: ffmpeg-devel@ffmpeg.org Date: Thu, 26 Aug 2021 23:26:40 +0200 Message-Id: <20210826212640.26473-1-onemda@gmail.com> X-Mailer: git-send-email 2.17.1 Subject: [FFmpeg-devel] [PATCH] avfilter: add atilt filter X-BeenThere: ffmpeg-devel@ffmpeg.org X-Mailman-Version: 2.1.29 Precedence: list List-Id: FFmpeg development discussions and patches List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Reply-To: FFmpeg development discussions and patches MIME-Version: 1.0 Errors-To: ffmpeg-devel-bounces@ffmpeg.org Sender: "ffmpeg-devel" X-TUID: lCkTd0XIST3Y Signed-off-by: Paul B Mahol --- doc/filters.texi | 29 ++++ libavfilter/Makefile | 1 + libavfilter/af_atilt.c | 287 +++++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + 4 files changed, 318 insertions(+) create mode 100644 libavfilter/af_atilt.c diff --git a/doc/filters.texi b/doc/filters.texi index fb3a683027..f160d70419 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -2989,6 +2989,35 @@ Change filter tempo scale factor. Syntax for the command is : "@var{tempo}" @end table +@section atilt +Apply spectral tilt filter to audio stream. + +This filter apply any spectral roll-off slope over any specified frequency band. + +The filter accepts the following options: + +@table @option +@item freq +Set central frequency of tilt in Hz. Default is 10000 Hz. + +@item slope +Set slope direction of tilt. Default is 0. Allowed range is from -1 to 1. + +@item width +Set width of tilt. Default is 1000. Allowed range is from 100 to 10000. + +@item order +Set order of tilt filter. + +@item level +Set input volume level. Allowed range is from 0 to 4. +Defalt is 1. +@end table + +@subsection Commands + +This filter supports the all above options as @ref{commands}. + @section atrim Trim the input so that the output contains one continuous subpart of the input. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 399a4a5083..2b58325fee 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -99,6 +99,7 @@ OBJS-$(CONFIG_ASUPERCUT_FILTER) += af_asupercut.o OBJS-$(CONFIG_ASUPERPASS_FILTER) += af_asupercut.o OBJS-$(CONFIG_ASUPERSTOP_FILTER) += af_asupercut.o OBJS-$(CONFIG_ATEMPO_FILTER) += af_atempo.o +OBJS-$(CONFIG_ATILT_FILTER) += af_atilt.o OBJS-$(CONFIG_ATRIM_FILTER) += trim.o OBJS-$(CONFIG_AXCORRELATE_FILTER) += af_axcorrelate.o OBJS-$(CONFIG_AZMQ_FILTER) += f_zmq.o diff --git a/libavfilter/af_atilt.c b/libavfilter/af_atilt.c new file mode 100644 index 0000000000..833e0c571b --- /dev/null +++ b/libavfilter/af_atilt.c @@ -0,0 +1,287 @@ +/* + * 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" + +#define MAX_ORDER 30 + +typedef struct Coeffs { + double g; + double a1; + double b0, b1; +} Coeffs; + +typedef struct ATiltContext { + const AVClass *class; + + double freq; + double level; + double slope; + double width; + int order; + + Coeffs coeffs[MAX_ORDER]; + + AVFrame *w; + + int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs); +} ATiltContext; + +static int query_formats(AVFilterContext *ctx) +{ + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, + AV_SAMPLE_FMT_NONE + }; + int ret; + + 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 double prewarp(double w, double T, double wp) +{ + return wp * tan(w * T * 0.5) / tan(wp * T * 0.5); +} + +static double mz(int i, double w0, double r, double alpha) +{ + return w0 * pow(r, -alpha + i); +} + +static double mp(int i, double w0, double r) +{ + return w0 * pow(r, i); +} + +static double mzh(int i, double T, double w0, double r, double alpha) +{ + return prewarp(mz(i, w0, r, alpha), T, w0); +} + +static double mph(int i, double T, double w0, double r) +{ + return prewarp(mp(i, w0, r), T, w0); +} + +static void set_tf1s(Coeffs *coeffs, double b1, double b0, double a0, + double w1, double sr, double alpha) +{ + double c = 1.0 / tan(w1 * 0.5 / sr); + double d = a0 + c; + + coeffs->b1 = (b0 - b1 * c) / d; + coeffs->b0 = (b0 + b1 * c) / d; + coeffs->a1 = (a0 - c) / d; + coeffs->g = a0 / b0; +} + +static void set_filter(AVFilterContext *ctx, + int order, double sr, double f0, + double bw, double alpha) +{ + ATiltContext *s = ctx->priv; + const double w0 = 2. * M_PI * f0; + const double f1 = f0 + bw; + const double w1 = 1.; + const double r = pow(f1 / f0, 1.0 / (order - 1.0)); + const double T = 1. / sr; + + for (int i = 0; i < order; i++) { + Coeffs *coeffs = &s->coeffs[i]; + + set_tf1s(coeffs, 1.0, mzh(i, T, w0, r, alpha), mph(i, T, w0, r), + w1, sr, alpha); + } +} + +static int get_coeffs(AVFilterContext *ctx) +{ + ATiltContext *s = ctx->priv; + AVFilterLink *inlink = ctx->inputs[0]; + + set_filter(ctx, s->order, inlink->sample_rate, s->freq, s->width, s->slope); + + return 0; +} + +typedef struct ThreadData { + AVFrame *in, *out; +} ThreadData; + +#define FILTER(name, type) \ +static int filter_channels_## name(AVFilterContext *ctx, void *arg, \ + int jobnr, int nb_jobs) \ +{ \ + ATiltContext *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; \ + const type level = s->level; \ + \ + for (int ch = start; ch < end; ch++) { \ + const type *src = (const type *)in->extended_data[ch]; \ + type *dst = (type *)out->extended_data[ch]; \ + \ + for (int b = 0; b < s->order; b++) { \ + Coeffs *coeffs = &s->coeffs[b]; \ + const type g = coeffs->g; \ + const type a1 = coeffs->a1; \ + const type b0 = coeffs->b0; \ + const type b1 = coeffs->b1; \ + type *w = ((type *)s->w->extended_data[ch]) + b * 2; \ + \ + for (int n = 0; n < in->nb_samples; n++) { \ + type sain = b ? dst[n] : src[n] * level; \ + type saout = sain * b0 + w[0] * b1 - w[1] * a1; \ + \ + w[0] = sain; \ + w[1] = saout; \ + \ + dst[n] = saout * g; \ + } \ + } \ + } \ + \ + return 0; \ +} + +FILTER(fltp, float) +FILTER(dblp, double) + +static int config_input(AVFilterLink *inlink) +{ + AVFilterContext *ctx = inlink->dst; + ATiltContext *s = ctx->priv; + + switch (inlink->format) { + case AV_SAMPLE_FMT_FLTP: s->filter_channels = filter_channels_fltp; break; + case AV_SAMPLE_FMT_DBLP: s->filter_channels = filter_channels_dblp; break; + } + + s->w = ff_get_audio_buffer(inlink, 2 * MAX_ORDER); + if (!s->w) + return AVERROR(ENOMEM); + + return get_coeffs(ctx); +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + AVFilterContext *ctx = inlink->dst; + ATiltContext *s = ctx->priv; + AVFilterLink *outlink = ctx->outputs[0]; + ThreadData td; + AVFrame *out; + + 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, s->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) +{ + ATiltContext *s = ctx->priv; + + av_frame_free(&s->w); +} + +#define OFFSET(x) offsetof(ATiltContext, x) +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM + +static const AVOption atilt_options[] = { + { "freq", "set central frequency",OFFSET(freq), AV_OPT_TYPE_DOUBLE, {.dbl=10000}, 20, 192000, FLAGS }, + { "slope", "set filter slope", OFFSET(slope), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, FLAGS }, + { "width", "set filter width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=1000}, 100, 10000, FLAGS }, + { "order", "set filter order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=5}, 2,MAX_ORDER, FLAGS }, + { "level", "set input level", OFFSET(level), AV_OPT_TYPE_DOUBLE, {.dbl=1.}, 0., 4., FLAGS }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(atilt); + +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, + }, +}; + +AVFilter ff_af_atilt = { + .name = "atilt", + .description = NULL_IF_CONFIG_SMALL("Apply spectral tilt to audio."), + .query_formats = query_formats, + .priv_size = sizeof(ATiltContext), + .priv_class = &atilt_class, + .uninit = uninit, + FILTER_INPUTS(inputs), + FILTER_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 745fc69e66..6f6677546d 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -92,6 +92,7 @@ extern const AVFilter ff_af_asupercut; extern const AVFilter ff_af_asuperpass; extern const AVFilter ff_af_asuperstop; extern const AVFilter ff_af_atempo; +extern const AVFilter ff_af_atilt; extern const AVFilter ff_af_atrim; extern const AVFilter ff_af_axcorrelate; extern const AVFilter ff_af_azmq;