From e35cba23149d2367b20ad85c967a84ec732d39a4 Mon Sep 17 00:00:00 2001 From: Paul B Mahol Date: Wed, 3 Aug 2016 11:02:34 +0200 Subject: [PATCH] avfilter: add simple audio dynamic range compressor --- doc/filters.texi | 18 ++++ libavfilter/Makefile | 1 + libavfilter/af_sadrc.c | 218 +++++++++++++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + 4 files changed, 238 insertions(+) create mode 100644 libavfilter/af_sadrc.c diff --git a/doc/filters.texi b/doc/filters.texi index 923d481..08ea373 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -2994,6 +2994,24 @@ Possible values are: @end table @end table +@section sadrc + +Apply simple audio dynamic range compression. This maximizes the volume by +compressing the audio signal's dynamic range. + +@table @option +@item method +Set the used method. Can be @code{1} or @code{2}. +@code{1} uses a single sample to smooth the variations via the standard weighted mean +over past samples (default). +@code{2} uses several samples to smooth the variations via the standard weighted mean +over past samples. + +@item target +Sets the target amplitude as a fraction of the maximum for the sample type. +Default is 0.25. +@end table + @section sidechaincompress This filter acts like normal compressor but has the ability to compress diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 65a831e..492564d 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -95,6 +95,7 @@ OBJS-$(CONFIG_PAN_FILTER) += af_pan.o OBJS-$(CONFIG_REPLAYGAIN_FILTER) += af_replaygain.o OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o OBJS-$(CONFIG_RUBBERBAND_FILTER) += af_rubberband.o +OBJS-$(CONFIG_SADRC_FILTER) += af_sadrc.o OBJS-$(CONFIG_SIDECHAINCOMPRESS_FILTER) += af_sidechaincompress.o OBJS-$(CONFIG_SIDECHAINGATE_FILTER) += af_agate.o OBJS-$(CONFIG_SILENCEDETECT_FILTER) += af_silencedetect.o diff --git a/libavfilter/af_sadrc.c b/libavfilter/af_sadrc.c new file mode 100644 index 0000000..a68b8c0 --- /dev/null +++ b/libavfilter/af_sadrc.c @@ -0,0 +1,218 @@ +/* + * Copyright (c) 2004 Alex Beregszaszi & Pierre Lombard + * + * This file is part of FFmpeg. + * + * FFmpeg is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public + * License as published by the Free Software Foundation; either + * version 2 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 General Public License for more details. + * + * You should have received a copy of the GNU 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 "avfilter.h" +#include "internal.h" +#include "audio.h" + +typedef struct SADRCMem { + float avg; + int len; +} SADRCMem; + +typedef struct SADRCContext { + const AVClass *class; + int method; + float mul, lastavg; + int idx; + int mem_size; + SADRCMem *mem; + float mid; +} SADRCContext; + +#define OFFSET(x) offsetof(SADRCContext, x) +#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption sadrc_options[] = { + { "m", "set method", OFFSET(method), AV_OPT_TYPE_INT, {.i64 = 1.0}, 1, 2, FLAGS }, + { "t", "set target", OFFSET(mid), AV_OPT_TYPE_FLOAT, {.dbl = 0.25}, 0.0001, 1.0, FLAGS }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(sadrc); + +static int filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + AVFilterContext *ctx = inlink->dst; + AVFilterLink *outlink = ctx->outputs[0]; + SADRCContext *s = ctx->priv; + const float *src = (const float *)in->data[0]; + const int channels = inlink->channels; + const int nb_samples = in->nb_samples; + const int len = channels * nb_samples; + AVFrame *out; + float *dst; + + if (av_frame_is_writable(in)) { + out = in; + } else { + out = ff_get_audio_buffer(inlink, in->nb_samples); + if (!out) { + av_frame_free(&in); + return AVERROR(ENOMEM); + } + av_frame_copy_props(out, in); + } + dst = (float *)out->data[0]; + + if (s->method == 1) { + float curavg = 0.0, newavg, neededmul, tmp; + int i; + + for (i = 0; i < len; i++) { + tmp = src[i]; + curavg += tmp * tmp; + } + curavg = sqrtf(curavg / (float)len); + + if (curavg > 0.01) { + neededmul = s->mid / (curavg * s->mul); + s->mul = (1.0 - 0.06) * s->mul + 0.06 * neededmul; + s->mul = av_clipf(s->mul, 0.1, 5.0); + } + + for (i = 0; i < len; i++) + dst[i] = src[i] * s->mul; + + newavg = s->mul * curavg; + + s->lastavg = (1.0 - 0.06) * s->lastavg + 0.06 * newavg; + } else { + float curavg = 0.0, newavg, avg = 0.0, tmp; + int i, totallen = 0; + + for (i = 0; i < len; i++) { + tmp = src[i]; + curavg += tmp * tmp; + } + curavg = sqrtf(curavg / (float)len); + + for (i = 0; i < s->mem_size; i++) { + avg += s->mem[i].avg * (float)s->mem[i].len; + totallen += s->mem[i].len; + } + + if (totallen > 32000) { + avg /= (float)totallen; + if (avg >= 0.01) { + s->mul = s->mid / avg; + s->mul = av_clipf(s->mul, 0.1, 5.0); + } + } + + for (i = 0; i < len; i++) + dst[i] = src[i] * s->mul; + + newavg = s->mul * curavg; + + s->mem[s->idx].len = len; + s->mem[s->idx].avg = newavg; + s->idx = (s->idx + 1) % s->mem_size; + } + + if (in != out) + av_frame_free(&in); + + return ff_filter_frame(outlink, out); +} + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats; + AVFilterChannelLayouts *layouts; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_FLT, + 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 (!formats) + return AVERROR(ENOMEM); + ret = ff_set_common_formats(ctx, formats); + if (ret < 0) + return ret; + + formats = ff_all_samplerates(); + if (!formats) + return AVERROR(ENOMEM); + return ff_set_common_samplerates(ctx, formats); +} + +static int config_input(AVFilterLink *inlink) +{ + AVFilterContext *ctx = inlink->dst; + SADRCContext *s = ctx->priv; + + s->mul = 1.0; + s->lastavg = .25; + s->idx = 0; + s->mem_size = 128; + + s->mem = av_calloc(s->mem_size, sizeof(*s->mem)); + if (!s->mem) + return AVERROR(ENOMEM); + + return 0; +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + SADRCContext *s = ctx->priv; + + av_freep(&s->mem); +} + +static const AVFilterPad avfilter_af_sadrc_inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_input, + .filter_frame = filter_frame, + }, + { NULL } +}; + +static const AVFilterPad avfilter_af_sadrc_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + }, + { NULL } +}; + +AVFilter ff_af_sadrc = { + .name = "sadrc", + .description = NULL_IF_CONFIG_SMALL("Simple audio dynamic range compressor."), + .priv_size = sizeof(SADRCContext), + .priv_class = &sadrc_class, + .uninit = uninit, + .query_formats = query_formats, + .inputs = avfilter_af_sadrc_inputs, + .outputs = avfilter_af_sadrc_outputs, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index d0d491e..cab44d7 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -114,6 +114,7 @@ void avfilter_register_all(void) REGISTER_FILTER(REPLAYGAIN, replaygain, af); REGISTER_FILTER(RESAMPLE, resample, af); REGISTER_FILTER(RUBBERBAND, rubberband, af); + REGISTER_FILTER(SADRC, sadrc, af); REGISTER_FILTER(SIDECHAINCOMPRESS, sidechaincompress, af); REGISTER_FILTER(SIDECHAINGATE, sidechaingate, af); REGISTER_FILTER(SILENCEDETECT, silencedetect, af); -- 2.8.3